AudioFlinger.h revision a494e82c3c73508b4d3cfe89e9134de94e12fd31
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 size_t *notificationFrames, 124 sp<IMemory>& cblk, 125 sp<IMemory>& buffers, 126 status_t *status /*non-NULL*/); 127 128 virtual uint32_t sampleRate(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 // Set kEnableExtendedPrecision to true to use extended precision in MixerThread 330 static const bool kEnableExtendedPrecision = true; 331 332 // Returns true if format is permitted for the PCM sink in the MixerThread 333 static inline bool isValidPcmSinkFormat(audio_format_t format) { 334 switch (format) { 335 case AUDIO_FORMAT_PCM_16_BIT: 336 return true; 337 case AUDIO_FORMAT_PCM_FLOAT: 338 case AUDIO_FORMAT_PCM_24_BIT_PACKED: 339 case AUDIO_FORMAT_PCM_32_BIT: 340 case AUDIO_FORMAT_PCM_8_24_BIT: 341 return kEnableExtendedPrecision; 342 default: 343 return false; 344 } 345 } 346 347 // standby delay for MIXER and DUPLICATING playback threads is read from property 348 // ro.audio.flinger_standbytime_ms or defaults to kDefaultStandbyTimeInNsecs 349 static nsecs_t mStandbyTimeInNsecs; 350 351 // incremented by 2 when screen state changes, bit 0 == 1 means "off" 352 // AudioFlinger::setParameters() updates, other threads read w/o lock 353 static uint32_t mScreenState; 354 355 // Internal dump utilities. 356 static const int kDumpLockRetries = 50; 357 static const int kDumpLockSleepUs = 20000; 358 static bool dumpTryLock(Mutex& mutex); 359 void dumpPermissionDenial(int fd, const Vector<String16>& args); 360 void dumpClients(int fd, const Vector<String16>& args); 361 void dumpInternals(int fd, const Vector<String16>& args); 362 363 // --- Client --- 364 class Client : public RefBase { 365 public: 366 Client(const sp<AudioFlinger>& audioFlinger, pid_t pid); 367 virtual ~Client(); 368 sp<MemoryDealer> heap() const; 369 pid_t pid() const { return mPid; } 370 sp<AudioFlinger> audioFlinger() const { return mAudioFlinger; } 371 372 bool reserveTimedTrack(); 373 void releaseTimedTrack(); 374 375 private: 376 Client(const Client&); 377 Client& operator = (const Client&); 378 const sp<AudioFlinger> mAudioFlinger; 379 const sp<MemoryDealer> mMemoryDealer; 380 const pid_t mPid; 381 382 Mutex mTimedTrackLock; 383 int mTimedTrackCount; 384 }; 385 386 // --- Notification Client --- 387 class NotificationClient : public IBinder::DeathRecipient { 388 public: 389 NotificationClient(const sp<AudioFlinger>& audioFlinger, 390 const sp<IAudioFlingerClient>& client, 391 pid_t pid); 392 virtual ~NotificationClient(); 393 394 sp<IAudioFlingerClient> audioFlingerClient() const { return mAudioFlingerClient; } 395 396 // IBinder::DeathRecipient 397 virtual void binderDied(const wp<IBinder>& who); 398 399 private: 400 NotificationClient(const NotificationClient&); 401 NotificationClient& operator = (const NotificationClient&); 402 403 const sp<AudioFlinger> mAudioFlinger; 404 const pid_t mPid; 405 const sp<IAudioFlingerClient> mAudioFlingerClient; 406 }; 407 408 class TrackHandle; 409 class RecordHandle; 410 class RecordThread; 411 class PlaybackThread; 412 class MixerThread; 413 class DirectOutputThread; 414 class OffloadThread; 415 class DuplicatingThread; 416 class AsyncCallbackThread; 417 class Track; 418 class RecordTrack; 419 class EffectModule; 420 class EffectHandle; 421 class EffectChain; 422 struct AudioStreamOut; 423 struct AudioStreamIn; 424 425 struct stream_type_t { 426 stream_type_t() 427 : volume(1.0f), 428 mute(false) 429 { 430 } 431 float volume; 432 bool mute; 433 }; 434 435 // --- PlaybackThread --- 436 437#include "Threads.h" 438 439#include "Effects.h" 440 441#include "PatchPanel.h" 442 443 // server side of the client's IAudioTrack 444 class TrackHandle : public android::BnAudioTrack { 445 public: 446 TrackHandle(const sp<PlaybackThread::Track>& track); 447 virtual ~TrackHandle(); 448 virtual sp<IMemory> getCblk() const; 449 virtual status_t start(); 450 virtual void stop(); 451 virtual void flush(); 452 virtual void pause(); 453 virtual status_t attachAuxEffect(int effectId); 454 virtual status_t allocateTimedBuffer(size_t size, 455 sp<IMemory>* buffer); 456 virtual status_t queueTimedBuffer(const sp<IMemory>& buffer, 457 int64_t pts); 458 virtual status_t setMediaTimeTransform(const LinearTransform& xform, 459 int target); 460 virtual status_t setParameters(const String8& keyValuePairs); 461 virtual status_t getTimestamp(AudioTimestamp& timestamp); 462 virtual void signal(); // signal playback thread for a change in control block 463 464 virtual status_t onTransact( 465 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 466 467 private: 468 const sp<PlaybackThread::Track> mTrack; 469 }; 470 471 // server side of the client's IAudioRecord 472 class RecordHandle : public android::BnAudioRecord { 473 public: 474 RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack); 475 virtual ~RecordHandle(); 476 virtual status_t start(int /*AudioSystem::sync_event_t*/ event, int triggerSession); 477 virtual void stop(); 478 virtual status_t onTransact( 479 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 480 private: 481 const sp<RecordThread::RecordTrack> mRecordTrack; 482 483 // for use from destructor 484 void stop_nonvirtual(); 485 }; 486 487 488 PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const; 489 MixerThread *checkMixerThread_l(audio_io_handle_t output) const; 490 RecordThread *checkRecordThread_l(audio_io_handle_t input) const; 491 // no range check, AudioFlinger::mLock held 492 bool streamMute_l(audio_stream_type_t stream) const 493 { return mStreamTypes[stream].mute; } 494 // no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held 495 float streamVolume_l(audio_stream_type_t stream) const 496 { return mStreamTypes[stream].volume; } 497 void audioConfigChanged(int event, audio_io_handle_t ioHandle, const void *param2); 498 499 // Allocate an audio_io_handle_t, session ID, effect ID, or audio_module_handle_t. 500 // They all share the same ID space, but the namespaces are actually independent 501 // because there are separate KeyedVectors for each kind of ID. 502 // The return value is uint32_t, but is cast to signed for some IDs. 503 // FIXME This API does not handle rollover to zero (for unsigned IDs), 504 // or from positive to negative (for signed IDs). 505 // Thus it may fail by returning an ID of the wrong sign, 506 // or by returning a non-unique ID. 507 uint32_t nextUniqueId(); 508 509 status_t moveEffectChain_l(int sessionId, 510 PlaybackThread *srcThread, 511 PlaybackThread *dstThread, 512 bool reRegister); 513 // return thread associated with primary hardware device, or NULL 514 PlaybackThread *primaryPlaybackThread_l() const; 515 audio_devices_t primaryOutputDevice_l() const; 516 517 sp<PlaybackThread> getEffectThread_l(int sessionId, int EffectId); 518 519 520 void removeClient_l(pid_t pid); 521 void removeNotificationClient(pid_t pid); 522 bool isNonOffloadableGlobalEffectEnabled_l(); 523 void onNonOffloadableGlobalEffectEnable(); 524 525 class AudioHwDevice { 526 public: 527 enum Flags { 528 AHWD_CAN_SET_MASTER_VOLUME = 0x1, 529 AHWD_CAN_SET_MASTER_MUTE = 0x2, 530 }; 531 532 AudioHwDevice(const char *moduleName, 533 audio_hw_device_t *hwDevice, 534 Flags flags) 535 : mModuleName(strdup(moduleName)) 536 , mHwDevice(hwDevice) 537 , mFlags(flags) { } 538 /*virtual*/ ~AudioHwDevice() { free((void *)mModuleName); } 539 540 bool canSetMasterVolume() const { 541 return (0 != (mFlags & AHWD_CAN_SET_MASTER_VOLUME)); 542 } 543 544 bool canSetMasterMute() const { 545 return (0 != (mFlags & AHWD_CAN_SET_MASTER_MUTE)); 546 } 547 548 const char *moduleName() const { return mModuleName; } 549 audio_hw_device_t *hwDevice() const { return mHwDevice; } 550 uint32_t version() const { return mHwDevice->common.version; } 551 552 private: 553 const char * const mModuleName; 554 audio_hw_device_t * const mHwDevice; 555 const Flags mFlags; 556 }; 557 558 // AudioStreamOut and AudioStreamIn are immutable, so their fields are const. 559 // For emphasis, we could also make all pointers to them be "const *", 560 // but that would clutter the code unnecessarily. 561 562 struct AudioStreamOut { 563 AudioHwDevice* const audioHwDev; 564 audio_stream_out_t* const stream; 565 const audio_output_flags_t flags; 566 567 audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); } 568 569 AudioStreamOut(AudioHwDevice *dev, audio_stream_out_t *out, audio_output_flags_t flags) : 570 audioHwDev(dev), stream(out), flags(flags) {} 571 }; 572 573 struct AudioStreamIn { 574 AudioHwDevice* const audioHwDev; 575 audio_stream_in_t* const stream; 576 577 audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); } 578 579 AudioStreamIn(AudioHwDevice *dev, audio_stream_in_t *in) : 580 audioHwDev(dev), stream(in) {} 581 }; 582 583 // for mAudioSessionRefs only 584 struct AudioSessionRef { 585 AudioSessionRef(int sessionid, pid_t pid) : 586 mSessionid(sessionid), mPid(pid), mCnt(1) {} 587 const int mSessionid; 588 const pid_t mPid; 589 int mCnt; 590 }; 591 592 mutable Mutex mLock; 593 // protects mClients and mNotificationClients. 594 // must be locked after mLock and ThreadBase::mLock if both must be locked 595 // avoids acquiring AudioFlinger::mLock from inside thread loop. 596 mutable Mutex mClientLock; 597 // protected by mClientLock 598 DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client() 599 600 mutable Mutex mHardwareLock; 601 // NOTE: If both mLock and mHardwareLock mutexes must be held, 602 // always take mLock before mHardwareLock 603 604 // These two fields are immutable after onFirstRef(), so no lock needed to access 605 AudioHwDevice* mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL 606 DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*> mAudioHwDevs; 607 608 // for dump, indicates which hardware operation is currently in progress (but not stream ops) 609 enum hardware_call_state { 610 AUDIO_HW_IDLE = 0, // no operation in progress 611 AUDIO_HW_INIT, // init_check 612 AUDIO_HW_OUTPUT_OPEN, // open_output_stream 613 AUDIO_HW_OUTPUT_CLOSE, // unused 614 AUDIO_HW_INPUT_OPEN, // unused 615 AUDIO_HW_INPUT_CLOSE, // unused 616 AUDIO_HW_STANDBY, // unused 617 AUDIO_HW_SET_MASTER_VOLUME, // set_master_volume 618 AUDIO_HW_GET_ROUTING, // unused 619 AUDIO_HW_SET_ROUTING, // unused 620 AUDIO_HW_GET_MODE, // unused 621 AUDIO_HW_SET_MODE, // set_mode 622 AUDIO_HW_GET_MIC_MUTE, // get_mic_mute 623 AUDIO_HW_SET_MIC_MUTE, // set_mic_mute 624 AUDIO_HW_SET_VOICE_VOLUME, // set_voice_volume 625 AUDIO_HW_SET_PARAMETER, // set_parameters 626 AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size 627 AUDIO_HW_GET_MASTER_VOLUME, // get_master_volume 628 AUDIO_HW_GET_PARAMETER, // get_parameters 629 AUDIO_HW_SET_MASTER_MUTE, // set_master_mute 630 AUDIO_HW_GET_MASTER_MUTE, // get_master_mute 631 }; 632 633 mutable hardware_call_state mHardwareStatus; // for dump only 634 635 636 DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads; 637 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 638 639 // member variables below are protected by mLock 640 float mMasterVolume; 641 bool mMasterMute; 642 // end of variables protected by mLock 643 644 DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads; 645 646 // protected by mClientLock 647 DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients; 648 649 volatile int32_t mNextUniqueId; // updated by android_atomic_inc 650 // nextUniqueId() returns uint32_t, but this is declared int32_t 651 // because the atomic operations require an int32_t 652 653 audio_mode_t mMode; 654 bool mBtNrecIsOff; 655 656 // protected by mLock 657 Vector<AudioSessionRef*> mAudioSessionRefs; 658 659 float masterVolume_l() const; 660 bool masterMute_l() const; 661 audio_module_handle_t loadHwModule_l(const char *name); 662 663 Vector < sp<SyncEvent> > mPendingSyncEvents; // sync events awaiting for a session 664 // to be created 665 666private: 667 sp<Client> registerPid(pid_t pid); // always returns non-0 668 669 // for use from destructor 670 status_t closeOutput_nonvirtual(audio_io_handle_t output); 671 status_t closeInput_nonvirtual(audio_io_handle_t input); 672 673#ifdef TEE_SINK 674 // all record threads serially share a common tee sink, which is re-created on format change 675 sp<NBAIO_Sink> mRecordTeeSink; 676 sp<NBAIO_Source> mRecordTeeSource; 677#endif 678 679public: 680 681#ifdef TEE_SINK 682 // tee sink, if enabled by property, allows dumpsys to write most recent audio to .wav file 683 static void dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id = 0); 684 685 // whether tee sink is enabled by property 686 static bool mTeeSinkInputEnabled; 687 static bool mTeeSinkOutputEnabled; 688 static bool mTeeSinkTrackEnabled; 689 690 // runtime configured size of each tee sink pipe, in frames 691 static size_t mTeeSinkInputFrames; 692 static size_t mTeeSinkOutputFrames; 693 static size_t mTeeSinkTrackFrames; 694 695 // compile-time default size of tee sink pipes, in frames 696 // 0x200000 stereo 16-bit PCM frames = 47.5 seconds at 44.1 kHz, 8 megabytes 697 static const size_t kTeeSinkInputFramesDefault = 0x200000; 698 static const size_t kTeeSinkOutputFramesDefault = 0x200000; 699 static const size_t kTeeSinkTrackFramesDefault = 0x200000; 700#endif 701 702 // This method reads from a variable without mLock, but the variable is updated under mLock. So 703 // we might read a stale value, or a value that's inconsistent with respect to other variables. 704 // In this case, it's safe because the return value isn't used for making an important decision. 705 // The reason we don't want to take mLock is because it could block the caller for a long time. 706 bool isLowRamDevice() const { return mIsLowRamDevice; } 707 708private: 709 bool mIsLowRamDevice; 710 bool mIsDeviceTypeKnown; 711 nsecs_t mGlobalEffectEnableTime; // when a global effect was last enabled 712 713 sp<PatchPanel> mPatchPanel; 714 715 uint32_t mPrimaryOutputSampleRate; // sample rate of the primary output, or zero if none 716 // protected by mHardwareLock 717}; 718 719#undef INCLUDING_FROM_AUDIOFLINGER_H 720 721const char *formatToString(audio_format_t format); 722 723// ---------------------------------------------------------------------------- 724 725}; // namespace android 726 727#endif // ANDROID_AUDIO_FLINGER_H 728