AudioFlinger.h revision 83b8808faad1e91690c64d7007348be8d9ebde73
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 audio_input_flags_t flags); 184 185 virtual status_t closeInput(audio_io_handle_t input); 186 187 virtual status_t invalidateStream(audio_stream_type_t stream); 188 189 virtual status_t setVoiceVolume(float volume); 190 191 virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 192 audio_io_handle_t output) const; 193 194 virtual uint32_t getInputFramesLost(audio_io_handle_t ioHandle) const; 195 196 virtual int newAudioSessionId(); 197 198 virtual void acquireAudioSessionId(int audioSession, pid_t pid); 199 200 virtual void releaseAudioSessionId(int audioSession, pid_t pid); 201 202 virtual status_t queryNumberEffects(uint32_t *numEffects) const; 203 204 virtual status_t queryEffect(uint32_t index, effect_descriptor_t *descriptor) const; 205 206 virtual status_t getEffectDescriptor(const effect_uuid_t *pUuid, 207 effect_descriptor_t *descriptor) const; 208 209 virtual sp<IEffect> createEffect( 210 effect_descriptor_t *pDesc, 211 const sp<IEffectClient>& effectClient, 212 int32_t priority, 213 audio_io_handle_t io, 214 int sessionId, 215 status_t *status /*non-NULL*/, 216 int *id, 217 int *enabled); 218 219 virtual status_t moveEffects(int sessionId, audio_io_handle_t srcOutput, 220 audio_io_handle_t dstOutput); 221 222 virtual audio_module_handle_t loadHwModule(const char *name); 223 224 virtual uint32_t getPrimaryOutputSamplingRate(); 225 virtual size_t getPrimaryOutputFrameCount(); 226 227 virtual status_t setLowRamDevice(bool isLowRamDevice); 228 229 /* List available audio ports and their attributes */ 230 virtual status_t listAudioPorts(unsigned int *num_ports, 231 struct audio_port *ports); 232 233 /* Get attributes for a given audio port */ 234 virtual status_t getAudioPort(struct audio_port *port); 235 236 /* Create an audio patch between several source and sink ports */ 237 virtual status_t createAudioPatch(const struct audio_patch *patch, 238 audio_patch_handle_t *handle); 239 240 /* Release an audio patch */ 241 virtual status_t releaseAudioPatch(audio_patch_handle_t handle); 242 243 /* List existing audio patches */ 244 virtual status_t listAudioPatches(unsigned int *num_patches, 245 struct audio_patch *patches); 246 247 /* Set audio port configuration */ 248 virtual status_t setAudioPortConfig(const struct audio_port_config *config); 249 250 virtual status_t onTransact( 251 uint32_t code, 252 const Parcel& data, 253 Parcel* reply, 254 uint32_t flags); 255 256 // end of IAudioFlinger interface 257 258 sp<NBLog::Writer> newWriter_l(size_t size, const char *name); 259 void unregisterWriter(const sp<NBLog::Writer>& writer); 260private: 261 static const size_t kLogMemorySize = 40 * 1024; 262 sp<MemoryDealer> mLogMemoryDealer; // == 0 when NBLog is disabled 263 // When a log writer is unregistered, it is done lazily so that media.log can continue to see it 264 // for as long as possible. The memory is only freed when it is needed for another log writer. 265 Vector< sp<NBLog::Writer> > mUnregisteredWriters; 266 Mutex mUnregisteredWritersLock; 267public: 268 269 class SyncEvent; 270 271 typedef void (*sync_event_callback_t)(const wp<SyncEvent>& event) ; 272 273 class SyncEvent : public RefBase { 274 public: 275 SyncEvent(AudioSystem::sync_event_t type, 276 int triggerSession, 277 int listenerSession, 278 sync_event_callback_t callBack, 279 wp<RefBase> cookie) 280 : mType(type), mTriggerSession(triggerSession), mListenerSession(listenerSession), 281 mCallback(callBack), mCookie(cookie) 282 {} 283 284 virtual ~SyncEvent() {} 285 286 void trigger() { Mutex::Autolock _l(mLock); if (mCallback) mCallback(this); } 287 bool isCancelled() const { Mutex::Autolock _l(mLock); return (mCallback == NULL); } 288 void cancel() { Mutex::Autolock _l(mLock); mCallback = NULL; } 289 AudioSystem::sync_event_t type() const { return mType; } 290 int triggerSession() const { return mTriggerSession; } 291 int listenerSession() const { return mListenerSession; } 292 wp<RefBase> cookie() const { return mCookie; } 293 294 private: 295 const AudioSystem::sync_event_t mType; 296 const int mTriggerSession; 297 const int mListenerSession; 298 sync_event_callback_t mCallback; 299 const wp<RefBase> mCookie; 300 mutable Mutex mLock; 301 }; 302 303 sp<SyncEvent> createSyncEvent(AudioSystem::sync_event_t type, 304 int triggerSession, 305 int listenerSession, 306 sync_event_callback_t callBack, 307 wp<RefBase> cookie); 308 309private: 310 class AudioHwDevice; // fwd declaration for findSuitableHwDev_l 311 312 audio_mode_t getMode() const { return mMode; } 313 314 bool btNrecIsOff() const { return mBtNrecIsOff; } 315 316 AudioFlinger() ANDROID_API; 317 virtual ~AudioFlinger(); 318 319 // call in any IAudioFlinger method that accesses mPrimaryHardwareDev 320 status_t initCheck() const { return mPrimaryHardwareDev == NULL ? 321 NO_INIT : NO_ERROR; } 322 323 // RefBase 324 virtual void onFirstRef(); 325 326 AudioHwDevice* findSuitableHwDev_l(audio_module_handle_t module, 327 audio_devices_t devices); 328 void purgeStaleEffects_l(); 329 330 // Set kEnableExtendedPrecision to true to use extended precision in MixerThread 331 static const bool kEnableExtendedPrecision = true; 332 333 // Returns true if format is permitted for the PCM sink in the MixerThread 334 static inline bool isValidPcmSinkFormat(audio_format_t format) { 335 switch (format) { 336 case AUDIO_FORMAT_PCM_16_BIT: 337 return true; 338 case AUDIO_FORMAT_PCM_FLOAT: 339 case AUDIO_FORMAT_PCM_24_BIT_PACKED: 340 case AUDIO_FORMAT_PCM_32_BIT: 341 case AUDIO_FORMAT_PCM_8_24_BIT: 342 return kEnableExtendedPrecision; 343 default: 344 return false; 345 } 346 } 347 348 // standby delay for MIXER and DUPLICATING playback threads is read from property 349 // ro.audio.flinger_standbytime_ms or defaults to kDefaultStandbyTimeInNsecs 350 static nsecs_t mStandbyTimeInNsecs; 351 352 // incremented by 2 when screen state changes, bit 0 == 1 means "off" 353 // AudioFlinger::setParameters() updates, other threads read w/o lock 354 static uint32_t mScreenState; 355 356 // Internal dump utilities. 357 static const int kDumpLockRetries = 50; 358 static const int kDumpLockSleepUs = 20000; 359 static bool dumpTryLock(Mutex& mutex); 360 void dumpPermissionDenial(int fd, const Vector<String16>& args); 361 void dumpClients(int fd, const Vector<String16>& args); 362 void dumpInternals(int fd, const Vector<String16>& args); 363 364 // --- Client --- 365 class Client : public RefBase { 366 public: 367 Client(const sp<AudioFlinger>& audioFlinger, pid_t pid); 368 virtual ~Client(); 369 sp<MemoryDealer> heap() const; 370 pid_t pid() const { return mPid; } 371 sp<AudioFlinger> audioFlinger() const { return mAudioFlinger; } 372 373 bool reserveTimedTrack(); 374 void releaseTimedTrack(); 375 376 private: 377 Client(const Client&); 378 Client& operator = (const Client&); 379 const sp<AudioFlinger> mAudioFlinger; 380 const sp<MemoryDealer> mMemoryDealer; 381 const pid_t mPid; 382 383 Mutex mTimedTrackLock; 384 int mTimedTrackCount; 385 }; 386 387 // --- Notification Client --- 388 class NotificationClient : public IBinder::DeathRecipient { 389 public: 390 NotificationClient(const sp<AudioFlinger>& audioFlinger, 391 const sp<IAudioFlingerClient>& client, 392 pid_t pid); 393 virtual ~NotificationClient(); 394 395 sp<IAudioFlingerClient> audioFlingerClient() const { return mAudioFlingerClient; } 396 397 // IBinder::DeathRecipient 398 virtual void binderDied(const wp<IBinder>& who); 399 400 private: 401 NotificationClient(const NotificationClient&); 402 NotificationClient& operator = (const NotificationClient&); 403 404 const sp<AudioFlinger> mAudioFlinger; 405 const pid_t mPid; 406 const sp<IAudioFlingerClient> mAudioFlingerClient; 407 }; 408 409 class TrackHandle; 410 class RecordHandle; 411 class RecordThread; 412 class PlaybackThread; 413 class MixerThread; 414 class DirectOutputThread; 415 class OffloadThread; 416 class DuplicatingThread; 417 class AsyncCallbackThread; 418 class Track; 419 class RecordTrack; 420 class EffectModule; 421 class EffectHandle; 422 class EffectChain; 423 struct AudioStreamOut; 424 struct AudioStreamIn; 425 426 struct stream_type_t { 427 stream_type_t() 428 : volume(1.0f), 429 mute(false) 430 { 431 } 432 float volume; 433 bool mute; 434 }; 435 436 // --- PlaybackThread --- 437 438#include "Threads.h" 439 440#include "Effects.h" 441 442#include "PatchPanel.h" 443 444 // server side of the client's IAudioTrack 445 class TrackHandle : public android::BnAudioTrack { 446 public: 447 TrackHandle(const sp<PlaybackThread::Track>& track); 448 virtual ~TrackHandle(); 449 virtual sp<IMemory> getCblk() const; 450 virtual status_t start(); 451 virtual void stop(); 452 virtual void flush(); 453 virtual void pause(); 454 virtual status_t attachAuxEffect(int effectId); 455 virtual status_t allocateTimedBuffer(size_t size, 456 sp<IMemory>* buffer); 457 virtual status_t queueTimedBuffer(const sp<IMemory>& buffer, 458 int64_t pts); 459 virtual status_t setMediaTimeTransform(const LinearTransform& xform, 460 int target); 461 virtual status_t setParameters(const String8& keyValuePairs); 462 virtual status_t getTimestamp(AudioTimestamp& timestamp); 463 virtual void signal(); // signal playback thread for a change in control block 464 465 virtual status_t onTransact( 466 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 467 468 private: 469 const sp<PlaybackThread::Track> mTrack; 470 }; 471 472 // server side of the client's IAudioRecord 473 class RecordHandle : public android::BnAudioRecord { 474 public: 475 RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack); 476 virtual ~RecordHandle(); 477 virtual status_t start(int /*AudioSystem::sync_event_t*/ event, int triggerSession); 478 virtual void stop(); 479 virtual status_t onTransact( 480 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 481 private: 482 const sp<RecordThread::RecordTrack> mRecordTrack; 483 484 // for use from destructor 485 void stop_nonvirtual(); 486 }; 487 488 489 PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const; 490 MixerThread *checkMixerThread_l(audio_io_handle_t output) const; 491 RecordThread *checkRecordThread_l(audio_io_handle_t input) const; 492 sp<RecordThread> openInput_l(audio_module_handle_t module, 493 audio_devices_t device, 494 struct audio_config *config, 495 audio_input_flags_t flags); 496 sp<PlaybackThread> openOutput_l(audio_module_handle_t module, 497 audio_devices_t device, 498 struct audio_config *config, 499 audio_output_flags_t flags); 500 501 void closeOutputFinish(sp<PlaybackThread> thread); 502 void closeInputFinish(sp<RecordThread> thread); 503 504 // no range check, AudioFlinger::mLock held 505 bool streamMute_l(audio_stream_type_t stream) const 506 { return mStreamTypes[stream].mute; } 507 // no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held 508 float streamVolume_l(audio_stream_type_t stream) const 509 { return mStreamTypes[stream].volume; } 510 void audioConfigChanged(int event, audio_io_handle_t ioHandle, const void *param2); 511 512 // Allocate an audio_io_handle_t, session ID, effect ID, or audio_module_handle_t. 513 // They all share the same ID space, but the namespaces are actually independent 514 // because there are separate KeyedVectors for each kind of ID. 515 // The return value is uint32_t, but is cast to signed for some IDs. 516 // FIXME This API does not handle rollover to zero (for unsigned IDs), 517 // or from positive to negative (for signed IDs). 518 // Thus it may fail by returning an ID of the wrong sign, 519 // or by returning a non-unique ID. 520 uint32_t nextUniqueId(); 521 522 status_t moveEffectChain_l(int sessionId, 523 PlaybackThread *srcThread, 524 PlaybackThread *dstThread, 525 bool reRegister); 526 // return thread associated with primary hardware device, or NULL 527 PlaybackThread *primaryPlaybackThread_l() const; 528 audio_devices_t primaryOutputDevice_l() const; 529 530 sp<PlaybackThread> getEffectThread_l(int sessionId, int EffectId); 531 532 533 void removeClient_l(pid_t pid); 534 void removeNotificationClient(pid_t pid); 535 bool isNonOffloadableGlobalEffectEnabled_l(); 536 void onNonOffloadableGlobalEffectEnable(); 537 538 class AudioHwDevice { 539 public: 540 enum Flags { 541 AHWD_CAN_SET_MASTER_VOLUME = 0x1, 542 AHWD_CAN_SET_MASTER_MUTE = 0x2, 543 }; 544 545 AudioHwDevice(audio_module_handle_t handle, 546 const char *moduleName, 547 audio_hw_device_t *hwDevice, 548 Flags flags) 549 : mHandle(handle), mModuleName(strdup(moduleName)) 550 , mHwDevice(hwDevice) 551 , mFlags(flags) { } 552 /*virtual*/ ~AudioHwDevice() { free((void *)mModuleName); } 553 554 bool canSetMasterVolume() const { 555 return (0 != (mFlags & AHWD_CAN_SET_MASTER_VOLUME)); 556 } 557 558 bool canSetMasterMute() const { 559 return (0 != (mFlags & AHWD_CAN_SET_MASTER_MUTE)); 560 } 561 562 audio_module_handle_t handle() const { return mHandle; } 563 const char *moduleName() const { return mModuleName; } 564 audio_hw_device_t *hwDevice() const { return mHwDevice; } 565 uint32_t version() const { return mHwDevice->common.version; } 566 567 private: 568 audio_module_handle_t mHandle; 569 const char * const mModuleName; 570 audio_hw_device_t * const mHwDevice; 571 const Flags mFlags; 572 }; 573 574 // AudioStreamOut and AudioStreamIn are immutable, so their fields are const. 575 // For emphasis, we could also make all pointers to them be "const *", 576 // but that would clutter the code unnecessarily. 577 578 struct AudioStreamOut { 579 AudioHwDevice* const audioHwDev; 580 audio_stream_out_t* const stream; 581 const audio_output_flags_t flags; 582 583 audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); } 584 585 AudioStreamOut(AudioHwDevice *dev, audio_stream_out_t *out, audio_output_flags_t flags) : 586 audioHwDev(dev), stream(out), flags(flags) {} 587 }; 588 589 struct AudioStreamIn { 590 AudioHwDevice* const audioHwDev; 591 audio_stream_in_t* const stream; 592 593 audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); } 594 595 AudioStreamIn(AudioHwDevice *dev, audio_stream_in_t *in) : 596 audioHwDev(dev), stream(in) {} 597 }; 598 599 // for mAudioSessionRefs only 600 struct AudioSessionRef { 601 AudioSessionRef(int sessionid, pid_t pid) : 602 mSessionid(sessionid), mPid(pid), mCnt(1) {} 603 const int mSessionid; 604 const pid_t mPid; 605 int mCnt; 606 }; 607 608 mutable Mutex mLock; 609 // protects mClients and mNotificationClients. 610 // must be locked after mLock and ThreadBase::mLock if both must be locked 611 // avoids acquiring AudioFlinger::mLock from inside thread loop. 612 mutable Mutex mClientLock; 613 // protected by mClientLock 614 DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client() 615 616 mutable Mutex mHardwareLock; 617 // NOTE: If both mLock and mHardwareLock mutexes must be held, 618 // always take mLock before mHardwareLock 619 620 // These two fields are immutable after onFirstRef(), so no lock needed to access 621 AudioHwDevice* mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL 622 DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*> mAudioHwDevs; 623 624 // for dump, indicates which hardware operation is currently in progress (but not stream ops) 625 enum hardware_call_state { 626 AUDIO_HW_IDLE = 0, // no operation in progress 627 AUDIO_HW_INIT, // init_check 628 AUDIO_HW_OUTPUT_OPEN, // open_output_stream 629 AUDIO_HW_OUTPUT_CLOSE, // unused 630 AUDIO_HW_INPUT_OPEN, // unused 631 AUDIO_HW_INPUT_CLOSE, // unused 632 AUDIO_HW_STANDBY, // unused 633 AUDIO_HW_SET_MASTER_VOLUME, // set_master_volume 634 AUDIO_HW_GET_ROUTING, // unused 635 AUDIO_HW_SET_ROUTING, // unused 636 AUDIO_HW_GET_MODE, // unused 637 AUDIO_HW_SET_MODE, // set_mode 638 AUDIO_HW_GET_MIC_MUTE, // get_mic_mute 639 AUDIO_HW_SET_MIC_MUTE, // set_mic_mute 640 AUDIO_HW_SET_VOICE_VOLUME, // set_voice_volume 641 AUDIO_HW_SET_PARAMETER, // set_parameters 642 AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size 643 AUDIO_HW_GET_MASTER_VOLUME, // get_master_volume 644 AUDIO_HW_GET_PARAMETER, // get_parameters 645 AUDIO_HW_SET_MASTER_MUTE, // set_master_mute 646 AUDIO_HW_GET_MASTER_MUTE, // get_master_mute 647 }; 648 649 mutable hardware_call_state mHardwareStatus; // for dump only 650 651 652 DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads; 653 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 654 655 // member variables below are protected by mLock 656 float mMasterVolume; 657 bool mMasterMute; 658 // end of variables protected by mLock 659 660 DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads; 661 662 // protected by mClientLock 663 DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients; 664 665 volatile int32_t mNextUniqueId; // updated by android_atomic_inc 666 // nextUniqueId() returns uint32_t, but this is declared int32_t 667 // because the atomic operations require an int32_t 668 669 audio_mode_t mMode; 670 bool mBtNrecIsOff; 671 672 // protected by mLock 673 Vector<AudioSessionRef*> mAudioSessionRefs; 674 675 float masterVolume_l() const; 676 bool masterMute_l() const; 677 audio_module_handle_t loadHwModule_l(const char *name); 678 679 Vector < sp<SyncEvent> > mPendingSyncEvents; // sync events awaiting for a session 680 // to be created 681 682private: 683 sp<Client> registerPid(pid_t pid); // always returns non-0 684 685 // for use from destructor 686 status_t closeOutput_nonvirtual(audio_io_handle_t output); 687 void closeOutputInternal_l(sp<PlaybackThread> thread); 688 status_t closeInput_nonvirtual(audio_io_handle_t input); 689 void closeInputInternal_l(sp<RecordThread> thread); 690 691#ifdef TEE_SINK 692 // all record threads serially share a common tee sink, which is re-created on format change 693 sp<NBAIO_Sink> mRecordTeeSink; 694 sp<NBAIO_Source> mRecordTeeSource; 695#endif 696 697public: 698 699#ifdef TEE_SINK 700 // tee sink, if enabled by property, allows dumpsys to write most recent audio to .wav file 701 static void dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id = 0); 702 703 // whether tee sink is enabled by property 704 static bool mTeeSinkInputEnabled; 705 static bool mTeeSinkOutputEnabled; 706 static bool mTeeSinkTrackEnabled; 707 708 // runtime configured size of each tee sink pipe, in frames 709 static size_t mTeeSinkInputFrames; 710 static size_t mTeeSinkOutputFrames; 711 static size_t mTeeSinkTrackFrames; 712 713 // compile-time default size of tee sink pipes, in frames 714 // 0x200000 stereo 16-bit PCM frames = 47.5 seconds at 44.1 kHz, 8 megabytes 715 static const size_t kTeeSinkInputFramesDefault = 0x200000; 716 static const size_t kTeeSinkOutputFramesDefault = 0x200000; 717 static const size_t kTeeSinkTrackFramesDefault = 0x200000; 718#endif 719 720 // This method reads from a variable without mLock, but the variable is updated under mLock. So 721 // we might read a stale value, or a value that's inconsistent with respect to other variables. 722 // In this case, it's safe because the return value isn't used for making an important decision. 723 // The reason we don't want to take mLock is because it could block the caller for a long time. 724 bool isLowRamDevice() const { return mIsLowRamDevice; } 725 726private: 727 bool mIsLowRamDevice; 728 bool mIsDeviceTypeKnown; 729 nsecs_t mGlobalEffectEnableTime; // when a global effect was last enabled 730 731 sp<PatchPanel> mPatchPanel; 732 733 uint32_t mPrimaryOutputSampleRate; // sample rate of the primary output, or zero if none 734 // protected by mHardwareLock 735}; 736 737#undef INCLUDING_FROM_AUDIOFLINGER_H 738 739const char *formatToString(audio_format_t format); 740 741// ---------------------------------------------------------------------------- 742 743}; // namespace android 744 745#endif // ANDROID_AUDIO_FLINGER_H 746