AudioFlinger.h revision e10393e72454bfd8298017dc193faf424f4e9a8f
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 <cutils/compiler.h> 27 28#include <media/IAudioFlinger.h> 29#include <media/IAudioFlingerClient.h> 30#include <media/IAudioTrack.h> 31#include <media/IAudioRecord.h> 32#include <media/AudioSystem.h> 33#include <media/AudioTrack.h> 34 35#include <utils/Atomic.h> 36#include <utils/Errors.h> 37#include <utils/threads.h> 38#include <utils/SortedVector.h> 39#include <utils/TypeHelpers.h> 40#include <utils/Vector.h> 41 42#include <binder/BinderService.h> 43#include <binder/MemoryDealer.h> 44 45#include <system/audio.h> 46#include <hardware/audio.h> 47#include <hardware/audio_policy.h> 48 49#include <media/AudioBufferProvider.h> 50#include <media/ExtendedAudioBufferProvider.h> 51 52#include "FastCapture.h" 53#include "FastMixer.h" 54#include <media/nbaio/NBAIO.h> 55#include "AudioWatchdog.h" 56#include "AudioMixer.h" 57#include "AudioStreamOut.h" 58#include "SpdifStreamOut.h" 59#include "AudioHwDevice.h" 60#include "LinearMap.h" 61 62#include <powermanager/IPowerManager.h> 63 64#include <media/nbaio/NBLog.h> 65#include <private/media/AudioTrackShared.h> 66 67namespace android { 68 69struct audio_track_cblk_t; 70struct effect_param_cblk_t; 71class AudioMixer; 72class AudioBuffer; 73class AudioResampler; 74class FastMixer; 75class PassthruBufferProvider; 76class ServerProxy; 77 78// ---------------------------------------------------------------------------- 79 80static const nsecs_t kDefaultStandbyTimeInNsecs = seconds(3); 81 82 83// Max shared memory size for audio tracks and audio records per client process 84static const size_t kClientSharedHeapSizeBytes = 1024*1024; 85// Shared memory size multiplier for non low ram devices 86static const size_t kClientSharedHeapSizeMultiplier = 4; 87 88#define INCLUDING_FROM_AUDIOFLINGER_H 89 90class AudioFlinger : 91 public BinderService<AudioFlinger>, 92 public BnAudioFlinger 93{ 94 friend class BinderService<AudioFlinger>; // for AudioFlinger() 95public: 96 static const char* getServiceName() ANDROID_API { return "media.audio_flinger"; } 97 98 virtual status_t dump(int fd, const Vector<String16>& args); 99 100 // IAudioFlinger interface, in binder opcode order 101 virtual sp<IAudioTrack> createTrack( 102 audio_stream_type_t streamType, 103 uint32_t sampleRate, 104 audio_format_t format, 105 audio_channel_mask_t channelMask, 106 size_t *pFrameCount, 107 IAudioFlinger::track_flags_t *flags, 108 const sp<IMemory>& sharedBuffer, 109 audio_io_handle_t output, 110 pid_t tid, 111 int *sessionId, 112 int clientUid, 113 status_t *status /*non-NULL*/); 114 115 virtual sp<IAudioRecord> openRecord( 116 audio_io_handle_t input, 117 uint32_t sampleRate, 118 audio_format_t format, 119 audio_channel_mask_t channelMask, 120 const String16& opPackageName, 121 size_t *pFrameCount, 122 IAudioFlinger::track_flags_t *flags, 123 pid_t tid, 124 int clientUid, 125 int *sessionId, 126 size_t *notificationFrames, 127 sp<IMemory>& cblk, 128 sp<IMemory>& buffers, 129 status_t *status /*non-NULL*/); 130 131 virtual uint32_t sampleRate(audio_io_handle_t output) const; 132 virtual audio_format_t format(audio_io_handle_t output) const; 133 virtual size_t frameCount(audio_io_handle_t output) const; 134 virtual uint32_t latency(audio_io_handle_t output) const; 135 136 virtual status_t setMasterVolume(float value); 137 virtual status_t setMasterMute(bool muted); 138 139 virtual float masterVolume() const; 140 virtual bool masterMute() const; 141 142 virtual status_t setStreamVolume(audio_stream_type_t stream, float value, 143 audio_io_handle_t output); 144 virtual status_t setStreamMute(audio_stream_type_t stream, bool muted); 145 146 virtual float streamVolume(audio_stream_type_t stream, 147 audio_io_handle_t output) const; 148 virtual bool streamMute(audio_stream_type_t stream) const; 149 150 virtual status_t setMode(audio_mode_t mode); 151 152 virtual status_t setMicMute(bool state); 153 virtual bool getMicMute() const; 154 155 virtual status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs); 156 virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) const; 157 158 virtual void registerClient(const sp<IAudioFlingerClient>& client); 159 160 virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, 161 audio_channel_mask_t channelMask) const; 162 163 virtual status_t openOutput(audio_module_handle_t module, 164 audio_io_handle_t *output, 165 audio_config_t *config, 166 audio_devices_t *devices, 167 const String8& address, 168 uint32_t *latencyMs, 169 audio_output_flags_t flags); 170 171 virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1, 172 audio_io_handle_t output2); 173 174 virtual status_t closeOutput(audio_io_handle_t output); 175 176 virtual status_t suspendOutput(audio_io_handle_t output); 177 178 virtual status_t restoreOutput(audio_io_handle_t output); 179 180 virtual status_t openInput(audio_module_handle_t module, 181 audio_io_handle_t *input, 182 audio_config_t *config, 183 audio_devices_t *device, 184 const String8& address, 185 audio_source_t source, 186 audio_input_flags_t flags); 187 188 virtual status_t closeInput(audio_io_handle_t input); 189 190 virtual status_t invalidateStream(audio_stream_type_t stream); 191 192 virtual status_t setVoiceVolume(float volume); 193 194 virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 195 audio_io_handle_t output) const; 196 197 virtual uint32_t getInputFramesLost(audio_io_handle_t ioHandle) const; 198 199 virtual audio_unique_id_t newAudioUniqueId(); 200 201 virtual void acquireAudioSessionId(int audioSession, pid_t pid); 202 203 virtual void releaseAudioSessionId(int audioSession, pid_t pid); 204 205 virtual status_t queryNumberEffects(uint32_t *numEffects) const; 206 207 virtual status_t queryEffect(uint32_t index, effect_descriptor_t *descriptor) const; 208 209 virtual status_t getEffectDescriptor(const effect_uuid_t *pUuid, 210 effect_descriptor_t *descriptor) const; 211 212 virtual sp<IEffect> createEffect( 213 effect_descriptor_t *pDesc, 214 const sp<IEffectClient>& effectClient, 215 int32_t priority, 216 audio_io_handle_t io, 217 int sessionId, 218 const String16& opPackageName, 219 status_t *status /*non-NULL*/, 220 int *id, 221 int *enabled); 222 223 virtual status_t moveEffects(int sessionId, audio_io_handle_t srcOutput, 224 audio_io_handle_t dstOutput); 225 226 virtual audio_module_handle_t loadHwModule(const char *name); 227 228 virtual uint32_t getPrimaryOutputSamplingRate(); 229 virtual size_t getPrimaryOutputFrameCount(); 230 231 virtual status_t setLowRamDevice(bool isLowRamDevice); 232 233 /* List available audio ports and their attributes */ 234 virtual status_t listAudioPorts(unsigned int *num_ports, 235 struct audio_port *ports); 236 237 /* Get attributes for a given audio port */ 238 virtual status_t getAudioPort(struct audio_port *port); 239 240 /* Create an audio patch between several source and sink ports */ 241 virtual status_t createAudioPatch(const struct audio_patch *patch, 242 audio_patch_handle_t *handle); 243 244 /* Release an audio patch */ 245 virtual status_t releaseAudioPatch(audio_patch_handle_t handle); 246 247 /* List existing audio patches */ 248 virtual status_t listAudioPatches(unsigned int *num_patches, 249 struct audio_patch *patches); 250 251 /* Set audio port configuration */ 252 virtual status_t setAudioPortConfig(const struct audio_port_config *config); 253 254 /* Get the HW synchronization source used for an audio session */ 255 virtual audio_hw_sync_t getAudioHwSyncForSession(audio_session_t sessionId); 256 257 /* Indicate JAVA services are ready (scheduling, power management ...) */ 258 virtual status_t systemReady(); 259 260 virtual status_t onTransact( 261 uint32_t code, 262 const Parcel& data, 263 Parcel* reply, 264 uint32_t flags); 265 266 // end of IAudioFlinger interface 267 268 sp<NBLog::Writer> newWriter_l(size_t size, const char *name); 269 void unregisterWriter(const sp<NBLog::Writer>& writer); 270private: 271 static const size_t kLogMemorySize = 40 * 1024; 272 sp<MemoryDealer> mLogMemoryDealer; // == 0 when NBLog is disabled 273 // When a log writer is unregistered, it is done lazily so that media.log can continue to see it 274 // for as long as possible. The memory is only freed when it is needed for another log writer. 275 Vector< sp<NBLog::Writer> > mUnregisteredWriters; 276 Mutex mUnregisteredWritersLock; 277public: 278 279 class SyncEvent; 280 281 typedef void (*sync_event_callback_t)(const wp<SyncEvent>& event) ; 282 283 class SyncEvent : public RefBase { 284 public: 285 SyncEvent(AudioSystem::sync_event_t type, 286 int triggerSession, 287 int listenerSession, 288 sync_event_callback_t callBack, 289 wp<RefBase> cookie) 290 : mType(type), mTriggerSession(triggerSession), mListenerSession(listenerSession), 291 mCallback(callBack), mCookie(cookie) 292 {} 293 294 virtual ~SyncEvent() {} 295 296 void trigger() { Mutex::Autolock _l(mLock); if (mCallback) mCallback(this); } 297 bool isCancelled() const { Mutex::Autolock _l(mLock); return (mCallback == NULL); } 298 void cancel() { Mutex::Autolock _l(mLock); mCallback = NULL; } 299 AudioSystem::sync_event_t type() const { return mType; } 300 int triggerSession() const { return mTriggerSession; } 301 int listenerSession() const { return mListenerSession; } 302 wp<RefBase> cookie() const { return mCookie; } 303 304 private: 305 const AudioSystem::sync_event_t mType; 306 const int mTriggerSession; 307 const int mListenerSession; 308 sync_event_callback_t mCallback; 309 const wp<RefBase> mCookie; 310 mutable Mutex mLock; 311 }; 312 313 sp<SyncEvent> createSyncEvent(AudioSystem::sync_event_t type, 314 int triggerSession, 315 int listenerSession, 316 sync_event_callback_t callBack, 317 wp<RefBase> cookie); 318 319private: 320 321 audio_mode_t getMode() const { return mMode; } 322 323 bool btNrecIsOff() const { return mBtNrecIsOff; } 324 325 AudioFlinger() ANDROID_API; 326 virtual ~AudioFlinger(); 327 328 // call in any IAudioFlinger method that accesses mPrimaryHardwareDev 329 status_t initCheck() const { return mPrimaryHardwareDev == NULL ? 330 NO_INIT : NO_ERROR; } 331 332 // RefBase 333 virtual void onFirstRef(); 334 335 AudioHwDevice* findSuitableHwDev_l(audio_module_handle_t module, 336 audio_devices_t devices); 337 void purgeStaleEffects_l(); 338 339 // Set kEnableExtendedChannels to true to enable greater than stereo output 340 // for the MixerThread and device sink. Number of channels allowed is 341 // FCC_2 <= channels <= AudioMixer::MAX_NUM_CHANNELS. 342 static const bool kEnableExtendedChannels = true; 343 344 // Returns true if channel mask is permitted for the PCM sink in the MixerThread 345 static inline bool isValidPcmSinkChannelMask(audio_channel_mask_t channelMask) { 346 switch (audio_channel_mask_get_representation(channelMask)) { 347 case AUDIO_CHANNEL_REPRESENTATION_POSITION: { 348 uint32_t channelCount = FCC_2; // stereo is default 349 if (kEnableExtendedChannels) { 350 channelCount = audio_channel_count_from_out_mask(channelMask); 351 if (channelCount < FCC_2 // mono is not supported at this time 352 || channelCount > AudioMixer::MAX_NUM_CHANNELS) { 353 return false; 354 } 355 } 356 // check that channelMask is the "canonical" one we expect for the channelCount. 357 return channelMask == audio_channel_out_mask_from_count(channelCount); 358 } 359 case AUDIO_CHANNEL_REPRESENTATION_INDEX: 360 if (kEnableExtendedChannels) { 361 const uint32_t channelCount = audio_channel_count_from_out_mask(channelMask); 362 if (channelCount >= FCC_2 // mono is not supported at this time 363 && channelCount <= AudioMixer::MAX_NUM_CHANNELS) { 364 return true; 365 } 366 } 367 return false; 368 default: 369 return false; 370 } 371 } 372 373 // Set kEnableExtendedPrecision to true to use extended precision in MixerThread 374 static const bool kEnableExtendedPrecision = true; 375 376 // Returns true if format is permitted for the PCM sink in the MixerThread 377 static inline bool isValidPcmSinkFormat(audio_format_t format) { 378 switch (format) { 379 case AUDIO_FORMAT_PCM_16_BIT: 380 return true; 381 case AUDIO_FORMAT_PCM_FLOAT: 382 case AUDIO_FORMAT_PCM_24_BIT_PACKED: 383 case AUDIO_FORMAT_PCM_32_BIT: 384 case AUDIO_FORMAT_PCM_8_24_BIT: 385 return kEnableExtendedPrecision; 386 default: 387 return false; 388 } 389 } 390 391 // standby delay for MIXER and DUPLICATING playback threads is read from property 392 // ro.audio.flinger_standbytime_ms or defaults to kDefaultStandbyTimeInNsecs 393 static nsecs_t mStandbyTimeInNsecs; 394 395 // incremented by 2 when screen state changes, bit 0 == 1 means "off" 396 // AudioFlinger::setParameters() updates, other threads read w/o lock 397 static uint32_t mScreenState; 398 399 // Internal dump utilities. 400 static const int kDumpLockRetries = 50; 401 static const int kDumpLockSleepUs = 20000; 402 static bool dumpTryLock(Mutex& mutex); 403 void dumpPermissionDenial(int fd, const Vector<String16>& args); 404 void dumpClients(int fd, const Vector<String16>& args); 405 void dumpInternals(int fd, const Vector<String16>& args); 406 407 // --- Client --- 408 class Client : public RefBase { 409 public: 410 Client(const sp<AudioFlinger>& audioFlinger, pid_t pid); 411 virtual ~Client(); 412 sp<MemoryDealer> heap() const; 413 pid_t pid() const { return mPid; } 414 sp<AudioFlinger> audioFlinger() const { return mAudioFlinger; } 415 416 private: 417 Client(const Client&); 418 Client& operator = (const Client&); 419 const sp<AudioFlinger> mAudioFlinger; 420 sp<MemoryDealer> mMemoryDealer; 421 const pid_t mPid; 422 }; 423 424 // --- Notification Client --- 425 class NotificationClient : public IBinder::DeathRecipient { 426 public: 427 NotificationClient(const sp<AudioFlinger>& audioFlinger, 428 const sp<IAudioFlingerClient>& client, 429 pid_t pid); 430 virtual ~NotificationClient(); 431 432 sp<IAudioFlingerClient> audioFlingerClient() const { return mAudioFlingerClient; } 433 434 // IBinder::DeathRecipient 435 virtual void binderDied(const wp<IBinder>& who); 436 437 private: 438 NotificationClient(const NotificationClient&); 439 NotificationClient& operator = (const NotificationClient&); 440 441 const sp<AudioFlinger> mAudioFlinger; 442 const pid_t mPid; 443 const sp<IAudioFlingerClient> mAudioFlingerClient; 444 }; 445 446 class TrackHandle; 447 class RecordHandle; 448 class RecordThread; 449 class PlaybackThread; 450 class MixerThread; 451 class DirectOutputThread; 452 class OffloadThread; 453 class DuplicatingThread; 454 class AsyncCallbackThread; 455 class Track; 456 class RecordTrack; 457 class EffectModule; 458 class EffectHandle; 459 class EffectChain; 460 461 struct AudioStreamIn; 462 463 struct stream_type_t { 464 stream_type_t() 465 : volume(1.0f), 466 mute(false) 467 { 468 } 469 float volume; 470 bool mute; 471 }; 472 473 // --- PlaybackThread --- 474 475#include "Threads.h" 476 477#include "Effects.h" 478 479#include "PatchPanel.h" 480 481 // server side of the client's IAudioTrack 482 class TrackHandle : public android::BnAudioTrack { 483 public: 484 TrackHandle(const sp<PlaybackThread::Track>& track); 485 virtual ~TrackHandle(); 486 virtual sp<IMemory> getCblk() const; 487 virtual status_t start(); 488 virtual void stop(); 489 virtual void flush(); 490 virtual void pause(); 491 virtual status_t attachAuxEffect(int effectId); 492 virtual status_t setParameters(const String8& keyValuePairs); 493 virtual status_t getTimestamp(AudioTimestamp& timestamp); 494 virtual void signal(); // signal playback thread for a change in control block 495 496 virtual status_t onTransact( 497 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 498 499 private: 500 const sp<PlaybackThread::Track> mTrack; 501 }; 502 503 // server side of the client's IAudioRecord 504 class RecordHandle : public android::BnAudioRecord { 505 public: 506 RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack); 507 virtual ~RecordHandle(); 508 virtual status_t start(int /*AudioSystem::sync_event_t*/ event, int triggerSession); 509 virtual void stop(); 510 virtual status_t onTransact( 511 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 512 private: 513 const sp<RecordThread::RecordTrack> mRecordTrack; 514 515 // for use from destructor 516 void stop_nonvirtual(); 517 }; 518 519 520 PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const; 521 MixerThread *checkMixerThread_l(audio_io_handle_t output) const; 522 RecordThread *checkRecordThread_l(audio_io_handle_t input) const; 523 sp<RecordThread> openInput_l(audio_module_handle_t module, 524 audio_io_handle_t *input, 525 audio_config_t *config, 526 audio_devices_t device, 527 const String8& address, 528 audio_source_t source, 529 audio_input_flags_t flags); 530 sp<PlaybackThread> openOutput_l(audio_module_handle_t module, 531 audio_io_handle_t *output, 532 audio_config_t *config, 533 audio_devices_t devices, 534 const String8& address, 535 audio_output_flags_t flags); 536 537 void closeOutputFinish(sp<PlaybackThread> thread); 538 void closeInputFinish(sp<RecordThread> thread); 539 540 // no range check, AudioFlinger::mLock held 541 bool streamMute_l(audio_stream_type_t stream) const 542 { return mStreamTypes[stream].mute; } 543 // no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held 544 float streamVolume_l(audio_stream_type_t stream) const 545 { return mStreamTypes[stream].volume; } 546 void ioConfigChanged(audio_io_config_event event, 547 const sp<AudioIoDescriptor>& ioDesc, 548 pid_t pid = 0); 549 550 // Allocate an audio_io_handle_t, session ID, effect ID, or audio_module_handle_t. 551 // They all share the same ID space, but the namespaces are actually independent 552 // because there are separate KeyedVectors for each kind of ID. 553 // The return value is uint32_t, but is cast to signed for some IDs. 554 // FIXME This API does not handle rollover to zero (for unsigned IDs), 555 // or from positive to negative (for signed IDs). 556 // Thus it may fail by returning an ID of the wrong sign, 557 // or by returning a non-unique ID. 558 uint32_t nextUniqueId(); 559 560 status_t moveEffectChain_l(int sessionId, 561 PlaybackThread *srcThread, 562 PlaybackThread *dstThread, 563 bool reRegister); 564 // return thread associated with primary hardware device, or NULL 565 PlaybackThread *primaryPlaybackThread_l() const; 566 audio_devices_t primaryOutputDevice_l() const; 567 568 sp<PlaybackThread> getEffectThread_l(int sessionId, int EffectId); 569 570 571 void removeClient_l(pid_t pid); 572 void removeNotificationClient(pid_t pid); 573 bool isNonOffloadableGlobalEffectEnabled_l(); 574 void onNonOffloadableGlobalEffectEnable(); 575 576 // Store an effect chain to mOrphanEffectChains keyed vector. 577 // Called when a thread exits and effects are still attached to it. 578 // If effects are later created on the same session, they will reuse the same 579 // effect chain and same instances in the effect library. 580 // return ALREADY_EXISTS if a chain with the same session already exists in 581 // mOrphanEffectChains. Note that this should never happen as there is only one 582 // chain for a given session and it is attached to only one thread at a time. 583 status_t putOrphanEffectChain_l(const sp<EffectChain>& chain); 584 // Get an effect chain for the specified session in mOrphanEffectChains and remove 585 // it if found. Returns 0 if not found (this is the most common case). 586 sp<EffectChain> getOrphanEffectChain_l(audio_session_t session); 587 // Called when the last effect handle on an effect instance is removed. If this 588 // effect belongs to an effect chain in mOrphanEffectChains, the chain is updated 589 // and removed from mOrphanEffectChains if it does not contain any effect. 590 // Return true if the effect was found in mOrphanEffectChains, false otherwise. 591 bool updateOrphanEffectChains(const sp<EffectModule>& effect); 592 593 void broacastParametersToRecordThreads_l(const String8& keyValuePairs); 594 595 // AudioStreamIn is immutable, so their fields are const. 596 // For emphasis, we could also make all pointers to them be "const *", 597 // but that would clutter the code unnecessarily. 598 599 struct AudioStreamIn { 600 AudioHwDevice* const audioHwDev; 601 audio_stream_in_t* const stream; 602 603 audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); } 604 605 AudioStreamIn(AudioHwDevice *dev, audio_stream_in_t *in) : 606 audioHwDev(dev), stream(in) {} 607 }; 608 609 // for mAudioSessionRefs only 610 struct AudioSessionRef { 611 AudioSessionRef(int sessionid, pid_t pid) : 612 mSessionid(sessionid), mPid(pid), mCnt(1) {} 613 const int mSessionid; 614 const pid_t mPid; 615 int mCnt; 616 }; 617 618 mutable Mutex mLock; 619 // protects mClients and mNotificationClients. 620 // must be locked after mLock and ThreadBase::mLock if both must be locked 621 // avoids acquiring AudioFlinger::mLock from inside thread loop. 622 mutable Mutex mClientLock; 623 // protected by mClientLock 624 DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client() 625 626 mutable Mutex mHardwareLock; 627 // NOTE: If both mLock and mHardwareLock mutexes must be held, 628 // always take mLock before mHardwareLock 629 630 // These two fields are immutable after onFirstRef(), so no lock needed to access 631 AudioHwDevice* mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL 632 DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*> mAudioHwDevs; 633 634 // for dump, indicates which hardware operation is currently in progress (but not stream ops) 635 enum hardware_call_state { 636 AUDIO_HW_IDLE = 0, // no operation in progress 637 AUDIO_HW_INIT, // init_check 638 AUDIO_HW_OUTPUT_OPEN, // open_output_stream 639 AUDIO_HW_OUTPUT_CLOSE, // unused 640 AUDIO_HW_INPUT_OPEN, // unused 641 AUDIO_HW_INPUT_CLOSE, // unused 642 AUDIO_HW_STANDBY, // unused 643 AUDIO_HW_SET_MASTER_VOLUME, // set_master_volume 644 AUDIO_HW_GET_ROUTING, // unused 645 AUDIO_HW_SET_ROUTING, // unused 646 AUDIO_HW_GET_MODE, // unused 647 AUDIO_HW_SET_MODE, // set_mode 648 AUDIO_HW_GET_MIC_MUTE, // get_mic_mute 649 AUDIO_HW_SET_MIC_MUTE, // set_mic_mute 650 AUDIO_HW_SET_VOICE_VOLUME, // set_voice_volume 651 AUDIO_HW_SET_PARAMETER, // set_parameters 652 AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size 653 AUDIO_HW_GET_MASTER_VOLUME, // get_master_volume 654 AUDIO_HW_GET_PARAMETER, // get_parameters 655 AUDIO_HW_SET_MASTER_MUTE, // set_master_mute 656 AUDIO_HW_GET_MASTER_MUTE, // get_master_mute 657 }; 658 659 mutable hardware_call_state mHardwareStatus; // for dump only 660 661 662 DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads; 663 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 664 665 // member variables below are protected by mLock 666 float mMasterVolume; 667 bool mMasterMute; 668 // end of variables protected by mLock 669 670 DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads; 671 672 // protected by mClientLock 673 DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients; 674 675 volatile int32_t mNextUniqueId; // updated by android_atomic_inc 676 // nextUniqueId() returns uint32_t, but this is declared int32_t 677 // because the atomic operations require an int32_t 678 679 audio_mode_t mMode; 680 bool mBtNrecIsOff; 681 682 // protected by mLock 683 Vector<AudioSessionRef*> mAudioSessionRefs; 684 685 float masterVolume_l() const; 686 bool masterMute_l() const; 687 audio_module_handle_t loadHwModule_l(const char *name); 688 689 Vector < sp<SyncEvent> > mPendingSyncEvents; // sync events awaiting for a session 690 // to be created 691 692 // Effect chains without a valid thread 693 DefaultKeyedVector< audio_session_t , sp<EffectChain> > mOrphanEffectChains; 694 695 // list of sessions for which a valid HW A/V sync ID was retrieved from the HAL 696 DefaultKeyedVector< audio_session_t , audio_hw_sync_t >mHwAvSyncIds; 697private: 698 sp<Client> registerPid(pid_t pid); // always returns non-0 699 700 // for use from destructor 701 status_t closeOutput_nonvirtual(audio_io_handle_t output); 702 void closeOutputInternal_l(sp<PlaybackThread> thread); 703 status_t closeInput_nonvirtual(audio_io_handle_t input); 704 void closeInputInternal_l(sp<RecordThread> thread); 705 void setAudioHwSyncForSession_l(PlaybackThread *thread, audio_session_t sessionId); 706 707 status_t checkStreamType(audio_stream_type_t stream) const; 708 709#ifdef TEE_SINK 710 // all record threads serially share a common tee sink, which is re-created on format change 711 sp<NBAIO_Sink> mRecordTeeSink; 712 sp<NBAIO_Source> mRecordTeeSource; 713#endif 714 715public: 716 717#ifdef TEE_SINK 718 // tee sink, if enabled by property, allows dumpsys to write most recent audio to .wav file 719 static void dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id = 0); 720 721 // whether tee sink is enabled by property 722 static bool mTeeSinkInputEnabled; 723 static bool mTeeSinkOutputEnabled; 724 static bool mTeeSinkTrackEnabled; 725 726 // runtime configured size of each tee sink pipe, in frames 727 static size_t mTeeSinkInputFrames; 728 static size_t mTeeSinkOutputFrames; 729 static size_t mTeeSinkTrackFrames; 730 731 // compile-time default size of tee sink pipes, in frames 732 // 0x200000 stereo 16-bit PCM frames = 47.5 seconds at 44.1 kHz, 8 megabytes 733 static const size_t kTeeSinkInputFramesDefault = 0x200000; 734 static const size_t kTeeSinkOutputFramesDefault = 0x200000; 735 static const size_t kTeeSinkTrackFramesDefault = 0x200000; 736#endif 737 738 // This method reads from a variable without mLock, but the variable is updated under mLock. So 739 // we might read a stale value, or a value that's inconsistent with respect to other variables. 740 // In this case, it's safe because the return value isn't used for making an important decision. 741 // The reason we don't want to take mLock is because it could block the caller for a long time. 742 bool isLowRamDevice() const { return mIsLowRamDevice; } 743 744private: 745 bool mIsLowRamDevice; 746 bool mIsDeviceTypeKnown; 747 nsecs_t mGlobalEffectEnableTime; // when a global effect was last enabled 748 749 sp<PatchPanel> mPatchPanel; 750 751 bool mSystemReady; 752}; 753 754#undef INCLUDING_FROM_AUDIOFLINGER_H 755 756const char *formatToString(audio_format_t format); 757String8 inputFlagsToString(audio_input_flags_t flags); 758String8 outputFlagsToString(audio_output_flags_t flags); 759String8 devicesToString(audio_devices_t devices); 760const char *sourceToString(audio_source_t source); 761 762// ---------------------------------------------------------------------------- 763 764} // namespace android 765 766#endif // ANDROID_AUDIO_FLINGER_H 767