AudioFlinger.h revision d79072e9dff59f767cce2cda1caab80ce5a0815b
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 61#include <powermanager/IPowerManager.h> 62 63#include <media/nbaio/NBLog.h> 64#include <private/media/AudioTrackShared.h> 65 66namespace android { 67 68struct audio_track_cblk_t; 69struct effect_param_cblk_t; 70class AudioMixer; 71class AudioBuffer; 72class AudioResampler; 73class FastMixer; 74class PassthruBufferProvider; 75class ServerProxy; 76 77// ---------------------------------------------------------------------------- 78 79static const nsecs_t kDefaultStandbyTimeInNsecs = seconds(3); 80 81 82// Max shared memory size for audio tracks and audio records per client process 83static const size_t kClientSharedHeapSizeBytes = 1024*1024; 84// Shared memory size multiplier for non low ram devices 85static const size_t kClientSharedHeapSizeMultiplier = 4; 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 const String16& opPackageName, 120 size_t *pFrameCount, 121 IAudioFlinger::track_flags_t *flags, 122 pid_t tid, 123 int clientUid, 124 int *sessionId, 125 size_t *notificationFrames, 126 sp<IMemory>& cblk, 127 sp<IMemory>& buffers, 128 status_t *status /*non-NULL*/); 129 130 virtual uint32_t sampleRate(audio_io_handle_t output) const; 131 virtual audio_format_t format(audio_io_handle_t output) const; 132 virtual size_t frameCount(audio_io_handle_t output) const; 133 virtual uint32_t latency(audio_io_handle_t output) const; 134 135 virtual status_t setMasterVolume(float value); 136 virtual status_t setMasterMute(bool muted); 137 138 virtual float masterVolume() const; 139 virtual bool masterMute() const; 140 141 virtual status_t setStreamVolume(audio_stream_type_t stream, float value, 142 audio_io_handle_t output); 143 virtual status_t setStreamMute(audio_stream_type_t stream, bool muted); 144 145 virtual float streamVolume(audio_stream_type_t stream, 146 audio_io_handle_t output) const; 147 virtual bool streamMute(audio_stream_type_t stream) const; 148 149 virtual status_t setMode(audio_mode_t mode); 150 151 virtual status_t setMicMute(bool state); 152 virtual bool getMicMute() const; 153 154 virtual status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs); 155 virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) const; 156 157 virtual void registerClient(const sp<IAudioFlingerClient>& client); 158 159 virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, 160 audio_channel_mask_t channelMask) const; 161 162 virtual status_t openOutput(audio_module_handle_t module, 163 audio_io_handle_t *output, 164 audio_config_t *config, 165 audio_devices_t *devices, 166 const String8& address, 167 uint32_t *latencyMs, 168 audio_output_flags_t flags); 169 170 virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1, 171 audio_io_handle_t output2); 172 173 virtual status_t closeOutput(audio_io_handle_t output); 174 175 virtual status_t suspendOutput(audio_io_handle_t output); 176 177 virtual status_t restoreOutput(audio_io_handle_t output); 178 179 virtual status_t openInput(audio_module_handle_t module, 180 audio_io_handle_t *input, 181 audio_config_t *config, 182 audio_devices_t *device, 183 const String8& address, 184 audio_source_t source, 185 audio_input_flags_t flags); 186 187 virtual status_t closeInput(audio_io_handle_t input); 188 189 virtual status_t invalidateStream(audio_stream_type_t stream); 190 191 virtual status_t setVoiceVolume(float volume); 192 193 virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 194 audio_io_handle_t output) const; 195 196 virtual uint32_t getInputFramesLost(audio_io_handle_t ioHandle) const; 197 198 virtual audio_unique_id_t newAudioUniqueId(); 199 200 virtual void acquireAudioSessionId(int audioSession, pid_t pid); 201 202 virtual void releaseAudioSessionId(int audioSession, pid_t pid); 203 204 virtual status_t queryNumberEffects(uint32_t *numEffects) const; 205 206 virtual status_t queryEffect(uint32_t index, effect_descriptor_t *descriptor) const; 207 208 virtual status_t getEffectDescriptor(const effect_uuid_t *pUuid, 209 effect_descriptor_t *descriptor) const; 210 211 virtual sp<IEffect> createEffect( 212 effect_descriptor_t *pDesc, 213 const sp<IEffectClient>& effectClient, 214 int32_t priority, 215 audio_io_handle_t io, 216 int sessionId, 217 const String16& opPackageName, 218 status_t *status /*non-NULL*/, 219 int *id, 220 int *enabled); 221 222 virtual status_t moveEffects(int sessionId, audio_io_handle_t srcOutput, 223 audio_io_handle_t dstOutput); 224 225 virtual audio_module_handle_t loadHwModule(const char *name); 226 227 virtual uint32_t getPrimaryOutputSamplingRate(); 228 virtual size_t getPrimaryOutputFrameCount(); 229 230 virtual status_t setLowRamDevice(bool isLowRamDevice); 231 232 /* List available audio ports and their attributes */ 233 virtual status_t listAudioPorts(unsigned int *num_ports, 234 struct audio_port *ports); 235 236 /* Get attributes for a given audio port */ 237 virtual status_t getAudioPort(struct audio_port *port); 238 239 /* Create an audio patch between several source and sink ports */ 240 virtual status_t createAudioPatch(const struct audio_patch *patch, 241 audio_patch_handle_t *handle); 242 243 /* Release an audio patch */ 244 virtual status_t releaseAudioPatch(audio_patch_handle_t handle); 245 246 /* List existing audio patches */ 247 virtual status_t listAudioPatches(unsigned int *num_patches, 248 struct audio_patch *patches); 249 250 /* Set audio port configuration */ 251 virtual status_t setAudioPortConfig(const struct audio_port_config *config); 252 253 /* Get the HW synchronization source used for an audio session */ 254 virtual audio_hw_sync_t getAudioHwSyncForSession(audio_session_t sessionId); 255 256 /* Indicate JAVA services are ready (scheduling, power management ...) */ 257 virtual status_t systemReady(); 258 259 virtual status_t onTransact( 260 uint32_t code, 261 const Parcel& data, 262 Parcel* reply, 263 uint32_t flags); 264 265 // end of IAudioFlinger interface 266 267 sp<NBLog::Writer> newWriter_l(size_t size, const char *name); 268 void unregisterWriter(const sp<NBLog::Writer>& writer); 269private: 270 static const size_t kLogMemorySize = 40 * 1024; 271 sp<MemoryDealer> mLogMemoryDealer; // == 0 when NBLog is disabled 272 // When a log writer is unregistered, it is done lazily so that media.log can continue to see it 273 // for as long as possible. The memory is only freed when it is needed for another log writer. 274 Vector< sp<NBLog::Writer> > mUnregisteredWriters; 275 Mutex mUnregisteredWritersLock; 276public: 277 278 class SyncEvent; 279 280 typedef void (*sync_event_callback_t)(const wp<SyncEvent>& event) ; 281 282 class SyncEvent : public RefBase { 283 public: 284 SyncEvent(AudioSystem::sync_event_t type, 285 int triggerSession, 286 int listenerSession, 287 sync_event_callback_t callBack, 288 wp<RefBase> cookie) 289 : mType(type), mTriggerSession(triggerSession), mListenerSession(listenerSession), 290 mCallback(callBack), mCookie(cookie) 291 {} 292 293 virtual ~SyncEvent() {} 294 295 void trigger() { Mutex::Autolock _l(mLock); if (mCallback) mCallback(this); } 296 bool isCancelled() const { Mutex::Autolock _l(mLock); return (mCallback == NULL); } 297 void cancel() { Mutex::Autolock _l(mLock); mCallback = NULL; } 298 AudioSystem::sync_event_t type() const { return mType; } 299 int triggerSession() const { return mTriggerSession; } 300 int listenerSession() const { return mListenerSession; } 301 wp<RefBase> cookie() const { return mCookie; } 302 303 private: 304 const AudioSystem::sync_event_t mType; 305 const int mTriggerSession; 306 const int mListenerSession; 307 sync_event_callback_t mCallback; 308 const wp<RefBase> mCookie; 309 mutable Mutex mLock; 310 }; 311 312 sp<SyncEvent> createSyncEvent(AudioSystem::sync_event_t type, 313 int triggerSession, 314 int listenerSession, 315 sync_event_callback_t callBack, 316 wp<RefBase> cookie); 317 318private: 319 320 audio_mode_t getMode() const { return mMode; } 321 322 bool btNrecIsOff() const { return mBtNrecIsOff; } 323 324 AudioFlinger() ANDROID_API; 325 virtual ~AudioFlinger(); 326 327 // call in any IAudioFlinger method that accesses mPrimaryHardwareDev 328 status_t initCheck() const { return mPrimaryHardwareDev == NULL ? 329 NO_INIT : NO_ERROR; } 330 331 // RefBase 332 virtual void onFirstRef(); 333 334 AudioHwDevice* findSuitableHwDev_l(audio_module_handle_t module, 335 audio_devices_t devices); 336 void purgeStaleEffects_l(); 337 338 // Set kEnableExtendedChannels to true to enable greater than stereo output 339 // for the MixerThread and device sink. Number of channels allowed is 340 // FCC_2 <= channels <= AudioMixer::MAX_NUM_CHANNELS. 341 static const bool kEnableExtendedChannels = true; 342 343 // Returns true if channel mask is permitted for the PCM sink in the MixerThread 344 static inline bool isValidPcmSinkChannelMask(audio_channel_mask_t channelMask) { 345 switch (audio_channel_mask_get_representation(channelMask)) { 346 case AUDIO_CHANNEL_REPRESENTATION_POSITION: { 347 uint32_t channelCount = FCC_2; // stereo is default 348 if (kEnableExtendedChannels) { 349 channelCount = audio_channel_count_from_out_mask(channelMask); 350 if (channelCount < FCC_2 // mono is not supported at this time 351 || channelCount > AudioMixer::MAX_NUM_CHANNELS) { 352 return false; 353 } 354 } 355 // check that channelMask is the "canonical" one we expect for the channelCount. 356 return channelMask == audio_channel_out_mask_from_count(channelCount); 357 } 358 case AUDIO_CHANNEL_REPRESENTATION_INDEX: 359 if (kEnableExtendedChannels) { 360 const uint32_t channelCount = audio_channel_count_from_out_mask(channelMask); 361 if (channelCount >= FCC_2 // mono is not supported at this time 362 && channelCount <= AudioMixer::MAX_NUM_CHANNELS) { 363 return true; 364 } 365 } 366 return false; 367 default: 368 return false; 369 } 370 } 371 372 // Set kEnableExtendedPrecision to true to use extended precision in MixerThread 373 static const bool kEnableExtendedPrecision = true; 374 375 // Returns true if format is permitted for the PCM sink in the MixerThread 376 static inline bool isValidPcmSinkFormat(audio_format_t format) { 377 switch (format) { 378 case AUDIO_FORMAT_PCM_16_BIT: 379 return true; 380 case AUDIO_FORMAT_PCM_FLOAT: 381 case AUDIO_FORMAT_PCM_24_BIT_PACKED: 382 case AUDIO_FORMAT_PCM_32_BIT: 383 case AUDIO_FORMAT_PCM_8_24_BIT: 384 return kEnableExtendedPrecision; 385 default: 386 return false; 387 } 388 } 389 390 // standby delay for MIXER and DUPLICATING playback threads is read from property 391 // ro.audio.flinger_standbytime_ms or defaults to kDefaultStandbyTimeInNsecs 392 static nsecs_t mStandbyTimeInNsecs; 393 394 // incremented by 2 when screen state changes, bit 0 == 1 means "off" 395 // AudioFlinger::setParameters() updates, other threads read w/o lock 396 static uint32_t mScreenState; 397 398 // Internal dump utilities. 399 static const int kDumpLockRetries = 50; 400 static const int kDumpLockSleepUs = 20000; 401 static bool dumpTryLock(Mutex& mutex); 402 void dumpPermissionDenial(int fd, const Vector<String16>& args); 403 void dumpClients(int fd, const Vector<String16>& args); 404 void dumpInternals(int fd, const Vector<String16>& args); 405 406 // --- Client --- 407 class Client : public RefBase { 408 public: 409 Client(const sp<AudioFlinger>& audioFlinger, pid_t pid); 410 virtual ~Client(); 411 sp<MemoryDealer> heap() const; 412 pid_t pid() const { return mPid; } 413 sp<AudioFlinger> audioFlinger() const { return mAudioFlinger; } 414 415 private: 416 Client(const Client&); 417 Client& operator = (const Client&); 418 const sp<AudioFlinger> mAudioFlinger; 419 sp<MemoryDealer> mMemoryDealer; 420 const pid_t mPid; 421 }; 422 423 // --- Notification Client --- 424 class NotificationClient : public IBinder::DeathRecipient { 425 public: 426 NotificationClient(const sp<AudioFlinger>& audioFlinger, 427 const sp<IAudioFlingerClient>& client, 428 pid_t pid); 429 virtual ~NotificationClient(); 430 431 sp<IAudioFlingerClient> audioFlingerClient() const { return mAudioFlingerClient; } 432 433 // IBinder::DeathRecipient 434 virtual void binderDied(const wp<IBinder>& who); 435 436 private: 437 NotificationClient(const NotificationClient&); 438 NotificationClient& operator = (const NotificationClient&); 439 440 const sp<AudioFlinger> mAudioFlinger; 441 const pid_t mPid; 442 const sp<IAudioFlingerClient> mAudioFlingerClient; 443 }; 444 445 class TrackHandle; 446 class RecordHandle; 447 class RecordThread; 448 class PlaybackThread; 449 class MixerThread; 450 class DirectOutputThread; 451 class OffloadThread; 452 class DuplicatingThread; 453 class AsyncCallbackThread; 454 class Track; 455 class RecordTrack; 456 class EffectModule; 457 class EffectHandle; 458 class EffectChain; 459 460 struct AudioStreamIn; 461 462 struct stream_type_t { 463 stream_type_t() 464 : volume(1.0f), 465 mute(false) 466 { 467 } 468 float volume; 469 bool mute; 470 }; 471 472 // --- PlaybackThread --- 473 474#include "Threads.h" 475 476#include "Effects.h" 477 478#include "PatchPanel.h" 479 480 // server side of the client's IAudioTrack 481 class TrackHandle : public android::BnAudioTrack { 482 public: 483 TrackHandle(const sp<PlaybackThread::Track>& track); 484 virtual ~TrackHandle(); 485 virtual sp<IMemory> getCblk() const; 486 virtual status_t start(); 487 virtual void stop(); 488 virtual void flush(); 489 virtual void pause(); 490 virtual status_t attachAuxEffect(int effectId); 491 virtual status_t setParameters(const String8& keyValuePairs); 492 virtual status_t getTimestamp(AudioTimestamp& timestamp); 493 virtual void signal(); // signal playback thread for a change in control block 494 495 virtual status_t onTransact( 496 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 497 498 private: 499 const sp<PlaybackThread::Track> mTrack; 500 }; 501 502 // server side of the client's IAudioRecord 503 class RecordHandle : public android::BnAudioRecord { 504 public: 505 RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack); 506 virtual ~RecordHandle(); 507 virtual status_t start(int /*AudioSystem::sync_event_t*/ event, int triggerSession); 508 virtual void stop(); 509 virtual status_t onTransact( 510 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 511 private: 512 const sp<RecordThread::RecordTrack> mRecordTrack; 513 514 // for use from destructor 515 void stop_nonvirtual(); 516 }; 517 518 519 PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const; 520 MixerThread *checkMixerThread_l(audio_io_handle_t output) const; 521 RecordThread *checkRecordThread_l(audio_io_handle_t input) const; 522 sp<RecordThread> openInput_l(audio_module_handle_t module, 523 audio_io_handle_t *input, 524 audio_config_t *config, 525 audio_devices_t device, 526 const String8& address, 527 audio_source_t source, 528 audio_input_flags_t flags); 529 sp<PlaybackThread> openOutput_l(audio_module_handle_t module, 530 audio_io_handle_t *output, 531 audio_config_t *config, 532 audio_devices_t devices, 533 const String8& address, 534 audio_output_flags_t flags); 535 536 void closeOutputFinish(sp<PlaybackThread> thread); 537 void closeInputFinish(sp<RecordThread> thread); 538 539 // no range check, AudioFlinger::mLock held 540 bool streamMute_l(audio_stream_type_t stream) const 541 { return mStreamTypes[stream].mute; } 542 // no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held 543 float streamVolume_l(audio_stream_type_t stream) const 544 { return mStreamTypes[stream].volume; } 545 void ioConfigChanged(audio_io_config_event event, 546 const sp<AudioIoDescriptor>& ioDesc, 547 pid_t pid = 0); 548 549 // Allocate an audio_io_handle_t, session ID, effect ID, or audio_module_handle_t. 550 // They all share the same ID space, but the namespaces are actually independent 551 // because there are separate KeyedVectors for each kind of ID. 552 // The return value is uint32_t, but is cast to signed for some IDs. 553 // FIXME This API does not handle rollover to zero (for unsigned IDs), 554 // or from positive to negative (for signed IDs). 555 // Thus it may fail by returning an ID of the wrong sign, 556 // or by returning a non-unique ID. 557 uint32_t nextUniqueId(); 558 559 status_t moveEffectChain_l(int sessionId, 560 PlaybackThread *srcThread, 561 PlaybackThread *dstThread, 562 bool reRegister); 563 // return thread associated with primary hardware device, or NULL 564 PlaybackThread *primaryPlaybackThread_l() const; 565 audio_devices_t primaryOutputDevice_l() const; 566 567 sp<PlaybackThread> getEffectThread_l(int sessionId, int EffectId); 568 569 570 void removeClient_l(pid_t pid); 571 void removeNotificationClient(pid_t pid); 572 bool isNonOffloadableGlobalEffectEnabled_l(); 573 void onNonOffloadableGlobalEffectEnable(); 574 575 // Store an effect chain to mOrphanEffectChains keyed vector. 576 // Called when a thread exits and effects are still attached to it. 577 // If effects are later created on the same session, they will reuse the same 578 // effect chain and same instances in the effect library. 579 // return ALREADY_EXISTS if a chain with the same session already exists in 580 // mOrphanEffectChains. Note that this should never happen as there is only one 581 // chain for a given session and it is attached to only one thread at a time. 582 status_t putOrphanEffectChain_l(const sp<EffectChain>& chain); 583 // Get an effect chain for the specified session in mOrphanEffectChains and remove 584 // it if found. Returns 0 if not found (this is the most common case). 585 sp<EffectChain> getOrphanEffectChain_l(audio_session_t session); 586 // Called when the last effect handle on an effect instance is removed. If this 587 // effect belongs to an effect chain in mOrphanEffectChains, the chain is updated 588 // and removed from mOrphanEffectChains if it does not contain any effect. 589 // Return true if the effect was found in mOrphanEffectChains, false otherwise. 590 bool updateOrphanEffectChains(const sp<EffectModule>& effect); 591 592 void broacastParametersToRecordThreads_l(const String8& keyValuePairs); 593 594 // AudioStreamIn is immutable, so their fields are const. 595 // For emphasis, we could also make all pointers to them be "const *", 596 // but that would clutter the code unnecessarily. 597 598 struct AudioStreamIn { 599 AudioHwDevice* const audioHwDev; 600 audio_stream_in_t* const stream; 601 602 audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); } 603 604 AudioStreamIn(AudioHwDevice *dev, audio_stream_in_t *in) : 605 audioHwDev(dev), stream(in) {} 606 }; 607 608 // for mAudioSessionRefs only 609 struct AudioSessionRef { 610 AudioSessionRef(int sessionid, pid_t pid) : 611 mSessionid(sessionid), mPid(pid), mCnt(1) {} 612 const int mSessionid; 613 const pid_t mPid; 614 int mCnt; 615 }; 616 617 mutable Mutex mLock; 618 // protects mClients and mNotificationClients. 619 // must be locked after mLock and ThreadBase::mLock if both must be locked 620 // avoids acquiring AudioFlinger::mLock from inside thread loop. 621 mutable Mutex mClientLock; 622 // protected by mClientLock 623 DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client() 624 625 mutable Mutex mHardwareLock; 626 // NOTE: If both mLock and mHardwareLock mutexes must be held, 627 // always take mLock before mHardwareLock 628 629 // These two fields are immutable after onFirstRef(), so no lock needed to access 630 AudioHwDevice* mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL 631 DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*> mAudioHwDevs; 632 633 // for dump, indicates which hardware operation is currently in progress (but not stream ops) 634 enum hardware_call_state { 635 AUDIO_HW_IDLE = 0, // no operation in progress 636 AUDIO_HW_INIT, // init_check 637 AUDIO_HW_OUTPUT_OPEN, // open_output_stream 638 AUDIO_HW_OUTPUT_CLOSE, // unused 639 AUDIO_HW_INPUT_OPEN, // unused 640 AUDIO_HW_INPUT_CLOSE, // unused 641 AUDIO_HW_STANDBY, // unused 642 AUDIO_HW_SET_MASTER_VOLUME, // set_master_volume 643 AUDIO_HW_GET_ROUTING, // unused 644 AUDIO_HW_SET_ROUTING, // unused 645 AUDIO_HW_GET_MODE, // unused 646 AUDIO_HW_SET_MODE, // set_mode 647 AUDIO_HW_GET_MIC_MUTE, // get_mic_mute 648 AUDIO_HW_SET_MIC_MUTE, // set_mic_mute 649 AUDIO_HW_SET_VOICE_VOLUME, // set_voice_volume 650 AUDIO_HW_SET_PARAMETER, // set_parameters 651 AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size 652 AUDIO_HW_GET_MASTER_VOLUME, // get_master_volume 653 AUDIO_HW_GET_PARAMETER, // get_parameters 654 AUDIO_HW_SET_MASTER_MUTE, // set_master_mute 655 AUDIO_HW_GET_MASTER_MUTE, // get_master_mute 656 }; 657 658 mutable hardware_call_state mHardwareStatus; // for dump only 659 660 661 DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads; 662 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 663 664 // member variables below are protected by mLock 665 float mMasterVolume; 666 bool mMasterMute; 667 // end of variables protected by mLock 668 669 DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads; 670 671 // protected by mClientLock 672 DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients; 673 674 volatile int32_t mNextUniqueId; // updated by android_atomic_inc 675 // nextUniqueId() returns uint32_t, but this is declared int32_t 676 // because the atomic operations require an int32_t 677 678 audio_mode_t mMode; 679 bool mBtNrecIsOff; 680 681 // protected by mLock 682 Vector<AudioSessionRef*> mAudioSessionRefs; 683 684 float masterVolume_l() const; 685 bool masterMute_l() const; 686 audio_module_handle_t loadHwModule_l(const char *name); 687 688 Vector < sp<SyncEvent> > mPendingSyncEvents; // sync events awaiting for a session 689 // to be created 690 691 // Effect chains without a valid thread 692 DefaultKeyedVector< audio_session_t , sp<EffectChain> > mOrphanEffectChains; 693 694 // list of sessions for which a valid HW A/V sync ID was retrieved from the HAL 695 DefaultKeyedVector< audio_session_t , audio_hw_sync_t >mHwAvSyncIds; 696private: 697 sp<Client> registerPid(pid_t pid); // always returns non-0 698 699 // for use from destructor 700 status_t closeOutput_nonvirtual(audio_io_handle_t output); 701 void closeOutputInternal_l(sp<PlaybackThread> thread); 702 status_t closeInput_nonvirtual(audio_io_handle_t input); 703 void closeInputInternal_l(sp<RecordThread> thread); 704 void setAudioHwSyncForSession_l(PlaybackThread *thread, audio_session_t sessionId); 705 706 status_t checkStreamType(audio_stream_type_t stream) const; 707 708#ifdef TEE_SINK 709 // all record threads serially share a common tee sink, which is re-created on format change 710 sp<NBAIO_Sink> mRecordTeeSink; 711 sp<NBAIO_Source> mRecordTeeSource; 712#endif 713 714public: 715 716#ifdef TEE_SINK 717 // tee sink, if enabled by property, allows dumpsys to write most recent audio to .wav file 718 static void dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id = 0); 719 720 // whether tee sink is enabled by property 721 static bool mTeeSinkInputEnabled; 722 static bool mTeeSinkOutputEnabled; 723 static bool mTeeSinkTrackEnabled; 724 725 // runtime configured size of each tee sink pipe, in frames 726 static size_t mTeeSinkInputFrames; 727 static size_t mTeeSinkOutputFrames; 728 static size_t mTeeSinkTrackFrames; 729 730 // compile-time default size of tee sink pipes, in frames 731 // 0x200000 stereo 16-bit PCM frames = 47.5 seconds at 44.1 kHz, 8 megabytes 732 static const size_t kTeeSinkInputFramesDefault = 0x200000; 733 static const size_t kTeeSinkOutputFramesDefault = 0x200000; 734 static const size_t kTeeSinkTrackFramesDefault = 0x200000; 735#endif 736 737 // This method reads from a variable without mLock, but the variable is updated under mLock. So 738 // we might read a stale value, or a value that's inconsistent with respect to other variables. 739 // In this case, it's safe because the return value isn't used for making an important decision. 740 // The reason we don't want to take mLock is because it could block the caller for a long time. 741 bool isLowRamDevice() const { return mIsLowRamDevice; } 742 743private: 744 bool mIsLowRamDevice; 745 bool mIsDeviceTypeKnown; 746 nsecs_t mGlobalEffectEnableTime; // when a global effect was last enabled 747 748 sp<PatchPanel> mPatchPanel; 749 750 bool mSystemReady; 751}; 752 753#undef INCLUDING_FROM_AUDIOFLINGER_H 754 755const char *formatToString(audio_format_t format); 756String8 inputFlagsToString(audio_input_flags_t flags); 757String8 outputFlagsToString(audio_output_flags_t flags); 758String8 devicesToString(audio_devices_t devices); 759const char *sourceToString(audio_source_t source); 760 761// ---------------------------------------------------------------------------- 762 763} // namespace android 764 765#endif // ANDROID_AUDIO_FLINGER_H 766