AudioFlinger.h revision 1c333e252cbca3337c1bedbc57a005f3b7d23fdb
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 /* Get attributes for a given audio port */ 231 virtual status_t getAudioPort(struct audio_port *port); 232 233 /* Create an audio patch between several source and sink ports */ 234 virtual status_t createAudioPatch(const struct audio_patch *patch, 235 audio_patch_handle_t *handle); 236 237 /* Release an audio patch */ 238 virtual status_t releaseAudioPatch(audio_patch_handle_t handle); 239 240 /* List existing audio patches */ 241 virtual status_t listAudioPatches(unsigned int *num_patches, 242 struct audio_patch *patches); 243 244 /* Set audio port configuration */ 245 virtual status_t setAudioPortConfig(const struct audio_port_config *config); 246 247 virtual status_t onTransact( 248 uint32_t code, 249 const Parcel& data, 250 Parcel* reply, 251 uint32_t flags); 252 253 // end of IAudioFlinger interface 254 255 sp<NBLog::Writer> newWriter_l(size_t size, const char *name); 256 void unregisterWriter(const sp<NBLog::Writer>& writer); 257private: 258 static const size_t kLogMemorySize = 40 * 1024; 259 sp<MemoryDealer> mLogMemoryDealer; // == 0 when NBLog is disabled 260 // When a log writer is unregistered, it is done lazily so that media.log can continue to see it 261 // for as long as possible. The memory is only freed when it is needed for another log writer. 262 Vector< sp<NBLog::Writer> > mUnregisteredWriters; 263 Mutex mUnregisteredWritersLock; 264public: 265 266 class SyncEvent; 267 268 typedef void (*sync_event_callback_t)(const wp<SyncEvent>& event) ; 269 270 class SyncEvent : public RefBase { 271 public: 272 SyncEvent(AudioSystem::sync_event_t type, 273 int triggerSession, 274 int listenerSession, 275 sync_event_callback_t callBack, 276 wp<RefBase> cookie) 277 : mType(type), mTriggerSession(triggerSession), mListenerSession(listenerSession), 278 mCallback(callBack), mCookie(cookie) 279 {} 280 281 virtual ~SyncEvent() {} 282 283 void trigger() { Mutex::Autolock _l(mLock); if (mCallback) mCallback(this); } 284 bool isCancelled() const { Mutex::Autolock _l(mLock); return (mCallback == NULL); } 285 void cancel() { Mutex::Autolock _l(mLock); mCallback = NULL; } 286 AudioSystem::sync_event_t type() const { return mType; } 287 int triggerSession() const { return mTriggerSession; } 288 int listenerSession() const { return mListenerSession; } 289 wp<RefBase> cookie() const { return mCookie; } 290 291 private: 292 const AudioSystem::sync_event_t mType; 293 const int mTriggerSession; 294 const int mListenerSession; 295 sync_event_callback_t mCallback; 296 const wp<RefBase> mCookie; 297 mutable Mutex mLock; 298 }; 299 300 sp<SyncEvent> createSyncEvent(AudioSystem::sync_event_t type, 301 int triggerSession, 302 int listenerSession, 303 sync_event_callback_t callBack, 304 wp<RefBase> cookie); 305 306private: 307 class AudioHwDevice; // fwd declaration for findSuitableHwDev_l 308 309 audio_mode_t getMode() const { return mMode; } 310 311 bool btNrecIsOff() const { return mBtNrecIsOff; } 312 313 AudioFlinger() ANDROID_API; 314 virtual ~AudioFlinger(); 315 316 // call in any IAudioFlinger method that accesses mPrimaryHardwareDev 317 status_t initCheck() const { return mPrimaryHardwareDev == NULL ? 318 NO_INIT : NO_ERROR; } 319 320 // RefBase 321 virtual void onFirstRef(); 322 323 AudioHwDevice* findSuitableHwDev_l(audio_module_handle_t module, 324 audio_devices_t devices); 325 void purgeStaleEffects_l(); 326 327 // standby delay for MIXER and DUPLICATING playback threads is read from property 328 // ro.audio.flinger_standbytime_ms or defaults to kDefaultStandbyTimeInNsecs 329 static nsecs_t mStandbyTimeInNsecs; 330 331 // incremented by 2 when screen state changes, bit 0 == 1 means "off" 332 // AudioFlinger::setParameters() updates, other threads read w/o lock 333 static uint32_t mScreenState; 334 335 // Internal dump utilities. 336 static const int kDumpLockRetries = 50; 337 static const int kDumpLockSleepUs = 20000; 338 static bool dumpTryLock(Mutex& mutex); 339 void dumpPermissionDenial(int fd, const Vector<String16>& args); 340 void dumpClients(int fd, const Vector<String16>& args); 341 void dumpInternals(int fd, const Vector<String16>& args); 342 343 // --- Client --- 344 class Client : public RefBase { 345 public: 346 Client(const sp<AudioFlinger>& audioFlinger, pid_t pid); 347 virtual ~Client(); 348 sp<MemoryDealer> heap() const; 349 pid_t pid() const { return mPid; } 350 sp<AudioFlinger> audioFlinger() const { return mAudioFlinger; } 351 352 bool reserveTimedTrack(); 353 void releaseTimedTrack(); 354 355 private: 356 Client(const Client&); 357 Client& operator = (const Client&); 358 const sp<AudioFlinger> mAudioFlinger; 359 const sp<MemoryDealer> mMemoryDealer; 360 const pid_t mPid; 361 362 Mutex mTimedTrackLock; 363 int mTimedTrackCount; 364 }; 365 366 // --- Notification Client --- 367 class NotificationClient : public IBinder::DeathRecipient { 368 public: 369 NotificationClient(const sp<AudioFlinger>& audioFlinger, 370 const sp<IAudioFlingerClient>& client, 371 pid_t pid); 372 virtual ~NotificationClient(); 373 374 sp<IAudioFlingerClient> audioFlingerClient() const { return mAudioFlingerClient; } 375 376 // IBinder::DeathRecipient 377 virtual void binderDied(const wp<IBinder>& who); 378 379 private: 380 NotificationClient(const NotificationClient&); 381 NotificationClient& operator = (const NotificationClient&); 382 383 const sp<AudioFlinger> mAudioFlinger; 384 const pid_t mPid; 385 const sp<IAudioFlingerClient> mAudioFlingerClient; 386 }; 387 388 class TrackHandle; 389 class RecordHandle; 390 class RecordThread; 391 class PlaybackThread; 392 class MixerThread; 393 class DirectOutputThread; 394 class OffloadThread; 395 class DuplicatingThread; 396 class AsyncCallbackThread; 397 class Track; 398 class RecordTrack; 399 class EffectModule; 400 class EffectHandle; 401 class EffectChain; 402 struct AudioStreamOut; 403 struct AudioStreamIn; 404 405 struct stream_type_t { 406 stream_type_t() 407 : volume(1.0f), 408 mute(false) 409 { 410 } 411 float volume; 412 bool mute; 413 }; 414 415 // --- PlaybackThread --- 416 417#include "Threads.h" 418 419#include "Effects.h" 420 421#include "PatchPanel.h" 422 423 // server side of the client's IAudioTrack 424 class TrackHandle : public android::BnAudioTrack { 425 public: 426 TrackHandle(const sp<PlaybackThread::Track>& track); 427 virtual ~TrackHandle(); 428 virtual sp<IMemory> getCblk() const; 429 virtual status_t start(); 430 virtual void stop(); 431 virtual void flush(); 432 virtual void pause(); 433 virtual status_t attachAuxEffect(int effectId); 434 virtual status_t allocateTimedBuffer(size_t size, 435 sp<IMemory>* buffer); 436 virtual status_t queueTimedBuffer(const sp<IMemory>& buffer, 437 int64_t pts); 438 virtual status_t setMediaTimeTransform(const LinearTransform& xform, 439 int target); 440 virtual status_t setParameters(const String8& keyValuePairs); 441 virtual status_t getTimestamp(AudioTimestamp& timestamp); 442 virtual void signal(); // signal playback thread for a change in control block 443 444 virtual status_t onTransact( 445 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 446 447 private: 448 const sp<PlaybackThread::Track> mTrack; 449 }; 450 451 // server side of the client's IAudioRecord 452 class RecordHandle : public android::BnAudioRecord { 453 public: 454 RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack); 455 virtual ~RecordHandle(); 456 virtual status_t start(int /*AudioSystem::sync_event_t*/ event, int triggerSession); 457 virtual void stop(); 458 virtual status_t onTransact( 459 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 460 private: 461 const sp<RecordThread::RecordTrack> mRecordTrack; 462 463 // for use from destructor 464 void stop_nonvirtual(); 465 }; 466 467 468 PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const; 469 MixerThread *checkMixerThread_l(audio_io_handle_t output) const; 470 RecordThread *checkRecordThread_l(audio_io_handle_t input) const; 471 // no range check, AudioFlinger::mLock held 472 bool streamMute_l(audio_stream_type_t stream) const 473 { return mStreamTypes[stream].mute; } 474 // no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held 475 float streamVolume_l(audio_stream_type_t stream) const 476 { return mStreamTypes[stream].volume; } 477 void audioConfigChanged(int event, audio_io_handle_t ioHandle, const void *param2); 478 479 // Allocate an audio_io_handle_t, session ID, effect ID, or audio_module_handle_t. 480 // They all share the same ID space, but the namespaces are actually independent 481 // because there are separate KeyedVectors for each kind of ID. 482 // The return value is uint32_t, but is cast to signed for some IDs. 483 // FIXME This API does not handle rollover to zero (for unsigned IDs), 484 // or from positive to negative (for signed IDs). 485 // Thus it may fail by returning an ID of the wrong sign, 486 // or by returning a non-unique ID. 487 uint32_t nextUniqueId(); 488 489 status_t moveEffectChain_l(int sessionId, 490 PlaybackThread *srcThread, 491 PlaybackThread *dstThread, 492 bool reRegister); 493 // return thread associated with primary hardware device, or NULL 494 PlaybackThread *primaryPlaybackThread_l() const; 495 audio_devices_t primaryOutputDevice_l() const; 496 497 sp<PlaybackThread> getEffectThread_l(int sessionId, int EffectId); 498 499 500 void removeClient_l(pid_t pid); 501 void removeNotificationClient(pid_t pid); 502 bool isNonOffloadableGlobalEffectEnabled_l(); 503 void onNonOffloadableGlobalEffectEnable(); 504 505 class AudioHwDevice { 506 public: 507 enum Flags { 508 AHWD_CAN_SET_MASTER_VOLUME = 0x1, 509 AHWD_CAN_SET_MASTER_MUTE = 0x2, 510 }; 511 512 AudioHwDevice(const char *moduleName, 513 audio_hw_device_t *hwDevice, 514 Flags flags) 515 : mModuleName(strdup(moduleName)) 516 , mHwDevice(hwDevice) 517 , mFlags(flags) { } 518 /*virtual*/ ~AudioHwDevice() { free((void *)mModuleName); } 519 520 bool canSetMasterVolume() const { 521 return (0 != (mFlags & AHWD_CAN_SET_MASTER_VOLUME)); 522 } 523 524 bool canSetMasterMute() const { 525 return (0 != (mFlags & AHWD_CAN_SET_MASTER_MUTE)); 526 } 527 528 const char *moduleName() const { return mModuleName; } 529 audio_hw_device_t *hwDevice() const { return mHwDevice; } 530 uint32_t version() const { return mHwDevice->common.version; } 531 532 private: 533 const char * const mModuleName; 534 audio_hw_device_t * const mHwDevice; 535 const Flags mFlags; 536 }; 537 538 // AudioStreamOut and AudioStreamIn are immutable, so their fields are const. 539 // For emphasis, we could also make all pointers to them be "const *", 540 // but that would clutter the code unnecessarily. 541 542 struct AudioStreamOut { 543 AudioHwDevice* const audioHwDev; 544 audio_stream_out_t* const stream; 545 const audio_output_flags_t flags; 546 547 audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); } 548 549 AudioStreamOut(AudioHwDevice *dev, audio_stream_out_t *out, audio_output_flags_t flags) : 550 audioHwDev(dev), stream(out), flags(flags) {} 551 }; 552 553 struct AudioStreamIn { 554 AudioHwDevice* const audioHwDev; 555 audio_stream_in_t* const stream; 556 557 audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); } 558 559 AudioStreamIn(AudioHwDevice *dev, audio_stream_in_t *in) : 560 audioHwDev(dev), stream(in) {} 561 }; 562 563 // for mAudioSessionRefs only 564 struct AudioSessionRef { 565 AudioSessionRef(int sessionid, pid_t pid) : 566 mSessionid(sessionid), mPid(pid), mCnt(1) {} 567 const int mSessionid; 568 const pid_t mPid; 569 int mCnt; 570 }; 571 572 mutable Mutex mLock; 573 // protects mClients and mNotificationClients. 574 // must be locked after mLock and ThreadBase::mLock if both must be locked 575 // avoids acquiring AudioFlinger::mLock from inside thread loop. 576 mutable Mutex mClientLock; 577 // protected by mClientLock 578 DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client() 579 580 mutable Mutex mHardwareLock; 581 // NOTE: If both mLock and mHardwareLock mutexes must be held, 582 // always take mLock before mHardwareLock 583 584 // These two fields are immutable after onFirstRef(), so no lock needed to access 585 AudioHwDevice* mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL 586 DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*> mAudioHwDevs; 587 588 // for dump, indicates which hardware operation is currently in progress (but not stream ops) 589 enum hardware_call_state { 590 AUDIO_HW_IDLE = 0, // no operation in progress 591 AUDIO_HW_INIT, // init_check 592 AUDIO_HW_OUTPUT_OPEN, // open_output_stream 593 AUDIO_HW_OUTPUT_CLOSE, // unused 594 AUDIO_HW_INPUT_OPEN, // unused 595 AUDIO_HW_INPUT_CLOSE, // unused 596 AUDIO_HW_STANDBY, // unused 597 AUDIO_HW_SET_MASTER_VOLUME, // set_master_volume 598 AUDIO_HW_GET_ROUTING, // unused 599 AUDIO_HW_SET_ROUTING, // unused 600 AUDIO_HW_GET_MODE, // unused 601 AUDIO_HW_SET_MODE, // set_mode 602 AUDIO_HW_GET_MIC_MUTE, // get_mic_mute 603 AUDIO_HW_SET_MIC_MUTE, // set_mic_mute 604 AUDIO_HW_SET_VOICE_VOLUME, // set_voice_volume 605 AUDIO_HW_SET_PARAMETER, // set_parameters 606 AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size 607 AUDIO_HW_GET_MASTER_VOLUME, // get_master_volume 608 AUDIO_HW_GET_PARAMETER, // get_parameters 609 AUDIO_HW_SET_MASTER_MUTE, // set_master_mute 610 AUDIO_HW_GET_MASTER_MUTE, // get_master_mute 611 }; 612 613 mutable hardware_call_state mHardwareStatus; // for dump only 614 615 616 DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads; 617 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 618 619 // member variables below are protected by mLock 620 float mMasterVolume; 621 bool mMasterMute; 622 // end of variables protected by mLock 623 624 DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads; 625 626 // protected by mClientLock 627 DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients; 628 629 volatile int32_t mNextUniqueId; // updated by android_atomic_inc 630 // nextUniqueId() returns uint32_t, but this is declared int32_t 631 // because the atomic operations require an int32_t 632 633 audio_mode_t mMode; 634 bool mBtNrecIsOff; 635 636 // protected by mLock 637 Vector<AudioSessionRef*> mAudioSessionRefs; 638 639 float masterVolume_l() const; 640 bool masterMute_l() const; 641 audio_module_handle_t loadHwModule_l(const char *name); 642 643 Vector < sp<SyncEvent> > mPendingSyncEvents; // sync events awaiting for a session 644 // to be created 645 646private: 647 sp<Client> registerPid(pid_t pid); // always returns non-0 648 649 // for use from destructor 650 status_t closeOutput_nonvirtual(audio_io_handle_t output); 651 status_t closeInput_nonvirtual(audio_io_handle_t input); 652 653#ifdef TEE_SINK 654 // all record threads serially share a common tee sink, which is re-created on format change 655 sp<NBAIO_Sink> mRecordTeeSink; 656 sp<NBAIO_Source> mRecordTeeSource; 657#endif 658 659public: 660 661#ifdef TEE_SINK 662 // tee sink, if enabled by property, allows dumpsys to write most recent audio to .wav file 663 static void dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id = 0); 664 665 // whether tee sink is enabled by property 666 static bool mTeeSinkInputEnabled; 667 static bool mTeeSinkOutputEnabled; 668 static bool mTeeSinkTrackEnabled; 669 670 // runtime configured size of each tee sink pipe, in frames 671 static size_t mTeeSinkInputFrames; 672 static size_t mTeeSinkOutputFrames; 673 static size_t mTeeSinkTrackFrames; 674 675 // compile-time default size of tee sink pipes, in frames 676 // 0x200000 stereo 16-bit PCM frames = 47.5 seconds at 44.1 kHz, 8 megabytes 677 static const size_t kTeeSinkInputFramesDefault = 0x200000; 678 static const size_t kTeeSinkOutputFramesDefault = 0x200000; 679 static const size_t kTeeSinkTrackFramesDefault = 0x200000; 680#endif 681 682 // This method reads from a variable without mLock, but the variable is updated under mLock. So 683 // we might read a stale value, or a value that's inconsistent with respect to other variables. 684 // In this case, it's safe because the return value isn't used for making an important decision. 685 // The reason we don't want to take mLock is because it could block the caller for a long time. 686 bool isLowRamDevice() const { return mIsLowRamDevice; } 687 688private: 689 bool mIsLowRamDevice; 690 bool mIsDeviceTypeKnown; 691 nsecs_t mGlobalEffectEnableTime; // when a global effect was last enabled 692 693 sp<PatchPanel> mPatchPanel; 694}; 695 696#undef INCLUDING_FROM_AUDIOFLINGER_H 697 698const char *formatToString(audio_format_t format); 699 700// ---------------------------------------------------------------------------- 701 702}; // namespace android 703 704#endif // ANDROID_AUDIO_FLINGER_H 705