AudioFlinger.h revision d2e67e1ef59921101fd7b047e2acf84e5d16d66e
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 audio_session_t *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 audio_session_t *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 ioHandle) const; 132 virtual audio_format_t format(audio_io_handle_t output) const; 133 virtual size_t frameCount(audio_io_handle_t ioHandle) 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 // This is the binder API. For the internal API see nextUniqueId(). 200 virtual audio_unique_id_t newAudioUniqueId(audio_unique_id_use_t use); 201 202 virtual void acquireAudioSessionId(audio_session_t audioSession, pid_t pid); 203 204 virtual void releaseAudioSessionId(audio_session_t audioSession, pid_t pid); 205 206 virtual status_t queryNumberEffects(uint32_t *numEffects) const; 207 208 virtual status_t queryEffect(uint32_t index, effect_descriptor_t *descriptor) const; 209 210 virtual status_t getEffectDescriptor(const effect_uuid_t *pUuid, 211 effect_descriptor_t *descriptor) const; 212 213 virtual sp<IEffect> createEffect( 214 effect_descriptor_t *pDesc, 215 const sp<IEffectClient>& effectClient, 216 int32_t priority, 217 audio_io_handle_t io, 218 audio_session_t sessionId, 219 const String16& opPackageName, 220 status_t *status /*non-NULL*/, 221 int *id, 222 int *enabled); 223 224 virtual status_t moveEffects(audio_session_t sessionId, audio_io_handle_t srcOutput, 225 audio_io_handle_t dstOutput); 226 227 virtual audio_module_handle_t loadHwModule(const char *name); 228 229 virtual uint32_t getPrimaryOutputSamplingRate(); 230 virtual size_t getPrimaryOutputFrameCount(); 231 232 virtual status_t setLowRamDevice(bool isLowRamDevice); 233 234 /* List available audio ports and their attributes */ 235 virtual status_t listAudioPorts(unsigned int *num_ports, 236 struct audio_port *ports); 237 238 /* Get attributes for a given audio port */ 239 virtual status_t getAudioPort(struct audio_port *port); 240 241 /* Create an audio patch between several source and sink ports */ 242 virtual status_t createAudioPatch(const struct audio_patch *patch, 243 audio_patch_handle_t *handle); 244 245 /* Release an audio patch */ 246 virtual status_t releaseAudioPatch(audio_patch_handle_t handle); 247 248 /* List existing audio patches */ 249 virtual status_t listAudioPatches(unsigned int *num_patches, 250 struct audio_patch *patches); 251 252 /* Set audio port configuration */ 253 virtual status_t setAudioPortConfig(const struct audio_port_config *config); 254 255 /* Get the HW synchronization source used for an audio session */ 256 virtual audio_hw_sync_t getAudioHwSyncForSession(audio_session_t sessionId); 257 258 /* Indicate JAVA services are ready (scheduling, power management ...) */ 259 virtual status_t systemReady(); 260 261 virtual status_t onTransact( 262 uint32_t code, 263 const Parcel& data, 264 Parcel* reply, 265 uint32_t flags); 266 267 // end of IAudioFlinger interface 268 269 sp<NBLog::Writer> newWriter_l(size_t size, const char *name); 270 void unregisterWriter(const sp<NBLog::Writer>& writer); 271private: 272 static const size_t kLogMemorySize = 40 * 1024; 273 sp<MemoryDealer> mLogMemoryDealer; // == 0 when NBLog is disabled 274 // When a log writer is unregistered, it is done lazily so that media.log can continue to see it 275 // for as long as possible. The memory is only freed when it is needed for another log writer. 276 Vector< sp<NBLog::Writer> > mUnregisteredWriters; 277 Mutex mUnregisteredWritersLock; 278public: 279 280 class SyncEvent; 281 282 typedef void (*sync_event_callback_t)(const wp<SyncEvent>& event) ; 283 284 class SyncEvent : public RefBase { 285 public: 286 SyncEvent(AudioSystem::sync_event_t type, 287 audio_session_t triggerSession, 288 audio_session_t listenerSession, 289 sync_event_callback_t callBack, 290 wp<RefBase> cookie) 291 : mType(type), mTriggerSession(triggerSession), mListenerSession(listenerSession), 292 mCallback(callBack), mCookie(cookie) 293 {} 294 295 virtual ~SyncEvent() {} 296 297 void trigger() { Mutex::Autolock _l(mLock); if (mCallback) mCallback(this); } 298 bool isCancelled() const { Mutex::Autolock _l(mLock); return (mCallback == NULL); } 299 void cancel() { Mutex::Autolock _l(mLock); mCallback = NULL; } 300 AudioSystem::sync_event_t type() const { return mType; } 301 audio_session_t triggerSession() const { return mTriggerSession; } 302 audio_session_t listenerSession() const { return mListenerSession; } 303 wp<RefBase> cookie() const { return mCookie; } 304 305 private: 306 const AudioSystem::sync_event_t mType; 307 const audio_session_t mTriggerSession; 308 const audio_session_t mListenerSession; 309 sync_event_callback_t mCallback; 310 const wp<RefBase> mCookie; 311 mutable Mutex mLock; 312 }; 313 314 sp<SyncEvent> createSyncEvent(AudioSystem::sync_event_t type, 315 audio_session_t triggerSession, 316 audio_session_t listenerSession, 317 sync_event_callback_t callBack, 318 wp<RefBase> cookie); 319 320private: 321 322 audio_mode_t getMode() const { return mMode; } 323 324 bool btNrecIsOff() const { return mBtNrecIsOff; } 325 326 AudioFlinger() ANDROID_API; 327 virtual ~AudioFlinger(); 328 329 // call in any IAudioFlinger method that accesses mPrimaryHardwareDev 330 status_t initCheck() const { return mPrimaryHardwareDev == NULL ? 331 NO_INIT : NO_ERROR; } 332 333 // RefBase 334 virtual void onFirstRef(); 335 336 AudioHwDevice* findSuitableHwDev_l(audio_module_handle_t module, 337 audio_devices_t devices); 338 void purgeStaleEffects_l(); 339 340 // Set kEnableExtendedChannels to true to enable greater than stereo output 341 // for the MixerThread and device sink. Number of channels allowed is 342 // FCC_2 <= channels <= AudioMixer::MAX_NUM_CHANNELS. 343 static const bool kEnableExtendedChannels = true; 344 345 // Returns true if channel mask is permitted for the PCM sink in the MixerThread 346 static inline bool isValidPcmSinkChannelMask(audio_channel_mask_t channelMask) { 347 switch (audio_channel_mask_get_representation(channelMask)) { 348 case AUDIO_CHANNEL_REPRESENTATION_POSITION: { 349 uint32_t channelCount = FCC_2; // stereo is default 350 if (kEnableExtendedChannels) { 351 channelCount = audio_channel_count_from_out_mask(channelMask); 352 if (channelCount < FCC_2 // mono is not supported at this time 353 || channelCount > AudioMixer::MAX_NUM_CHANNELS) { 354 return false; 355 } 356 } 357 // check that channelMask is the "canonical" one we expect for the channelCount. 358 return channelMask == audio_channel_out_mask_from_count(channelCount); 359 } 360 case AUDIO_CHANNEL_REPRESENTATION_INDEX: 361 if (kEnableExtendedChannels) { 362 const uint32_t channelCount = audio_channel_count_from_out_mask(channelMask); 363 if (channelCount >= FCC_2 // mono is not supported at this time 364 && channelCount <= AudioMixer::MAX_NUM_CHANNELS) { 365 return true; 366 } 367 } 368 return false; 369 default: 370 return false; 371 } 372 } 373 374 // Set kEnableExtendedPrecision to true to use extended precision in MixerThread 375 static const bool kEnableExtendedPrecision = true; 376 377 // Returns true if format is permitted for the PCM sink in the MixerThread 378 static inline bool isValidPcmSinkFormat(audio_format_t format) { 379 switch (format) { 380 case AUDIO_FORMAT_PCM_16_BIT: 381 return true; 382 case AUDIO_FORMAT_PCM_FLOAT: 383 case AUDIO_FORMAT_PCM_24_BIT_PACKED: 384 case AUDIO_FORMAT_PCM_32_BIT: 385 case AUDIO_FORMAT_PCM_8_24_BIT: 386 return kEnableExtendedPrecision; 387 default: 388 return false; 389 } 390 } 391 392 // standby delay for MIXER and DUPLICATING playback threads is read from property 393 // ro.audio.flinger_standbytime_ms or defaults to kDefaultStandbyTimeInNsecs 394 static nsecs_t mStandbyTimeInNsecs; 395 396 // incremented by 2 when screen state changes, bit 0 == 1 means "off" 397 // AudioFlinger::setParameters() updates, other threads read w/o lock 398 static uint32_t mScreenState; 399 400 // Internal dump utilities. 401 static const int kDumpLockRetries = 50; 402 static const int kDumpLockSleepUs = 20000; 403 static bool dumpTryLock(Mutex& mutex); 404 void dumpPermissionDenial(int fd, const Vector<String16>& args); 405 void dumpClients(int fd, const Vector<String16>& args); 406 void dumpInternals(int fd, const Vector<String16>& args); 407 408 // --- Client --- 409 class Client : public RefBase { 410 public: 411 Client(const sp<AudioFlinger>& audioFlinger, pid_t pid); 412 virtual ~Client(); 413 sp<MemoryDealer> heap() const; 414 pid_t pid() const { return mPid; } 415 sp<AudioFlinger> audioFlinger() const { return mAudioFlinger; } 416 417 private: 418 Client(const Client&); 419 Client& operator = (const Client&); 420 const sp<AudioFlinger> mAudioFlinger; 421 sp<MemoryDealer> mMemoryDealer; 422 const pid_t mPid; 423 }; 424 425 // --- Notification Client --- 426 class NotificationClient : public IBinder::DeathRecipient { 427 public: 428 NotificationClient(const sp<AudioFlinger>& audioFlinger, 429 const sp<IAudioFlingerClient>& client, 430 pid_t pid); 431 virtual ~NotificationClient(); 432 433 sp<IAudioFlingerClient> audioFlingerClient() const { return mAudioFlingerClient; } 434 435 // IBinder::DeathRecipient 436 virtual void binderDied(const wp<IBinder>& who); 437 438 private: 439 NotificationClient(const NotificationClient&); 440 NotificationClient& operator = (const NotificationClient&); 441 442 const sp<AudioFlinger> mAudioFlinger; 443 const pid_t mPid; 444 const sp<IAudioFlingerClient> mAudioFlingerClient; 445 }; 446 447 class TrackHandle; 448 class RecordHandle; 449 class RecordThread; 450 class PlaybackThread; 451 class MixerThread; 452 class DirectOutputThread; 453 class OffloadThread; 454 class DuplicatingThread; 455 class AsyncCallbackThread; 456 class Track; 457 class RecordTrack; 458 class EffectModule; 459 class EffectHandle; 460 class EffectChain; 461 462 struct AudioStreamIn; 463 464 struct stream_type_t { 465 stream_type_t() 466 : volume(1.0f), 467 mute(false) 468 { 469 } 470 float volume; 471 bool mute; 472 }; 473 474 // --- PlaybackThread --- 475 476#include "Threads.h" 477 478#include "Effects.h" 479 480#include "PatchPanel.h" 481 482 // server side of the client's IAudioTrack 483 class TrackHandle : public android::BnAudioTrack { 484 public: 485 TrackHandle(const sp<PlaybackThread::Track>& track); 486 virtual ~TrackHandle(); 487 virtual sp<IMemory> getCblk() const; 488 virtual status_t start(); 489 virtual void stop(); 490 virtual void flush(); 491 virtual void pause(); 492 virtual status_t attachAuxEffect(int effectId); 493 virtual status_t setParameters(const String8& keyValuePairs); 494 virtual status_t getTimestamp(AudioTimestamp& timestamp); 495 virtual void signal(); // signal playback thread for a change in control block 496 497 virtual status_t onTransact( 498 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 499 500 private: 501 const sp<PlaybackThread::Track> mTrack; 502 }; 503 504 // server side of the client's IAudioRecord 505 class RecordHandle : public android::BnAudioRecord { 506 public: 507 RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack); 508 virtual ~RecordHandle(); 509 virtual status_t start(int /*AudioSystem::sync_event_t*/ event, 510 audio_session_t triggerSession); 511 virtual void stop(); 512 virtual status_t onTransact( 513 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 514 private: 515 const sp<RecordThread::RecordTrack> mRecordTrack; 516 517 // for use from destructor 518 void stop_nonvirtual(); 519 }; 520 521 522 ThreadBase *checkThread_l(audio_io_handle_t ioHandle) const; 523 PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const; 524 MixerThread *checkMixerThread_l(audio_io_handle_t output) const; 525 RecordThread *checkRecordThread_l(audio_io_handle_t input) const; 526 sp<RecordThread> openInput_l(audio_module_handle_t module, 527 audio_io_handle_t *input, 528 audio_config_t *config, 529 audio_devices_t device, 530 const String8& address, 531 audio_source_t source, 532 audio_input_flags_t flags); 533 sp<PlaybackThread> openOutput_l(audio_module_handle_t module, 534 audio_io_handle_t *output, 535 audio_config_t *config, 536 audio_devices_t devices, 537 const String8& address, 538 audio_output_flags_t flags); 539 540 void closeOutputFinish(sp<PlaybackThread> thread); 541 void closeInputFinish(sp<RecordThread> thread); 542 543 // no range check, AudioFlinger::mLock held 544 bool streamMute_l(audio_stream_type_t stream) const 545 { return mStreamTypes[stream].mute; } 546 // no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held 547 float streamVolume_l(audio_stream_type_t stream) const 548 { return mStreamTypes[stream].volume; } 549 void ioConfigChanged(audio_io_config_event event, 550 const sp<AudioIoDescriptor>& ioDesc, 551 pid_t pid = 0); 552 553 // Allocate an audio_unique_id_t. 554 // Specific types are audio_io_handle_t, audio_session_t, effect ID (int), 555 // audio_module_handle_t, and audio_patch_handle_t. 556 // They all share the same ID space, but the namespaces are actually independent 557 // because there are separate KeyedVectors for each kind of ID. 558 // The return value is cast to the specific type depending on how the ID will be used. 559 // FIXME This API does not handle rollover to zero (for unsigned IDs), 560 // or from positive to negative (for signed IDs). 561 // Thus it may fail by returning an ID of the wrong sign, 562 // or by returning a non-unique ID. 563 // This is the internal API. For the binder API see newAudioUniqueId(). 564 audio_unique_id_t nextUniqueId(audio_unique_id_use_t use); 565 566 status_t moveEffectChain_l(audio_session_t sessionId, 567 PlaybackThread *srcThread, 568 PlaybackThread *dstThread, 569 bool reRegister); 570 571 // return thread associated with primary hardware device, or NULL 572 PlaybackThread *primaryPlaybackThread_l() const; 573 audio_devices_t primaryOutputDevice_l() const; 574 575 sp<PlaybackThread> getEffectThread_l(audio_session_t sessionId, int EffectId); 576 577 578 void removeClient_l(pid_t pid); 579 void removeNotificationClient(pid_t pid); 580 bool isNonOffloadableGlobalEffectEnabled_l(); 581 void onNonOffloadableGlobalEffectEnable(); 582 583 // Store an effect chain to mOrphanEffectChains keyed vector. 584 // Called when a thread exits and effects are still attached to it. 585 // If effects are later created on the same session, they will reuse the same 586 // effect chain and same instances in the effect library. 587 // return ALREADY_EXISTS if a chain with the same session already exists in 588 // mOrphanEffectChains. Note that this should never happen as there is only one 589 // chain for a given session and it is attached to only one thread at a time. 590 status_t putOrphanEffectChain_l(const sp<EffectChain>& chain); 591 // Get an effect chain for the specified session in mOrphanEffectChains and remove 592 // it if found. Returns 0 if not found (this is the most common case). 593 sp<EffectChain> getOrphanEffectChain_l(audio_session_t session); 594 // Called when the last effect handle on an effect instance is removed. If this 595 // effect belongs to an effect chain in mOrphanEffectChains, the chain is updated 596 // and removed from mOrphanEffectChains if it does not contain any effect. 597 // Return true if the effect was found in mOrphanEffectChains, false otherwise. 598 bool updateOrphanEffectChains(const sp<EffectModule>& effect); 599 600 void broacastParametersToRecordThreads_l(const String8& keyValuePairs); 601 602 // AudioStreamIn is immutable, so their fields are const. 603 // For emphasis, we could also make all pointers to them be "const *", 604 // but that would clutter the code unnecessarily. 605 606 struct AudioStreamIn { 607 AudioHwDevice* const audioHwDev; 608 audio_stream_in_t* const stream; 609 610 audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); } 611 612 AudioStreamIn(AudioHwDevice *dev, audio_stream_in_t *in) : 613 audioHwDev(dev), stream(in) {} 614 }; 615 616 // for mAudioSessionRefs only 617 struct AudioSessionRef { 618 AudioSessionRef(audio_session_t sessionid, pid_t pid) : 619 mSessionid(sessionid), mPid(pid), mCnt(1) {} 620 const audio_session_t mSessionid; 621 const pid_t mPid; 622 int mCnt; 623 }; 624 625 mutable Mutex mLock; 626 // protects mClients and mNotificationClients. 627 // must be locked after mLock and ThreadBase::mLock if both must be locked 628 // avoids acquiring AudioFlinger::mLock from inside thread loop. 629 mutable Mutex mClientLock; 630 // protected by mClientLock 631 DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client() 632 633 mutable Mutex mHardwareLock; 634 // NOTE: If both mLock and mHardwareLock mutexes must be held, 635 // always take mLock before mHardwareLock 636 637 // These two fields are immutable after onFirstRef(), so no lock needed to access 638 AudioHwDevice* mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL 639 DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*> mAudioHwDevs; 640 641 // for dump, indicates which hardware operation is currently in progress (but not stream ops) 642 enum hardware_call_state { 643 AUDIO_HW_IDLE = 0, // no operation in progress 644 AUDIO_HW_INIT, // init_check 645 AUDIO_HW_OUTPUT_OPEN, // open_output_stream 646 AUDIO_HW_OUTPUT_CLOSE, // unused 647 AUDIO_HW_INPUT_OPEN, // unused 648 AUDIO_HW_INPUT_CLOSE, // unused 649 AUDIO_HW_STANDBY, // unused 650 AUDIO_HW_SET_MASTER_VOLUME, // set_master_volume 651 AUDIO_HW_GET_ROUTING, // unused 652 AUDIO_HW_SET_ROUTING, // unused 653 AUDIO_HW_GET_MODE, // unused 654 AUDIO_HW_SET_MODE, // set_mode 655 AUDIO_HW_GET_MIC_MUTE, // get_mic_mute 656 AUDIO_HW_SET_MIC_MUTE, // set_mic_mute 657 AUDIO_HW_SET_VOICE_VOLUME, // set_voice_volume 658 AUDIO_HW_SET_PARAMETER, // set_parameters 659 AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size 660 AUDIO_HW_GET_MASTER_VOLUME, // get_master_volume 661 AUDIO_HW_GET_PARAMETER, // get_parameters 662 AUDIO_HW_SET_MASTER_MUTE, // set_master_mute 663 AUDIO_HW_GET_MASTER_MUTE, // get_master_mute 664 }; 665 666 mutable hardware_call_state mHardwareStatus; // for dump only 667 668 669 DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads; 670 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 671 672 // member variables below are protected by mLock 673 float mMasterVolume; 674 bool mMasterMute; 675 // end of variables protected by mLock 676 677 DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads; 678 679 // protected by mClientLock 680 DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients; 681 682 // updated by atomic_fetch_add_explicit 683 volatile atomic_uint_fast32_t mNextUniqueIds[AUDIO_UNIQUE_ID_USE_MAX]; 684 685 audio_mode_t mMode; 686 bool mBtNrecIsOff; 687 688 // protected by mLock 689 Vector<AudioSessionRef*> mAudioSessionRefs; 690 691 float masterVolume_l() const; 692 bool masterMute_l() const; 693 audio_module_handle_t loadHwModule_l(const char *name); 694 695 Vector < sp<SyncEvent> > mPendingSyncEvents; // sync events awaiting for a session 696 // to be created 697 698 // Effect chains without a valid thread 699 DefaultKeyedVector< audio_session_t , sp<EffectChain> > mOrphanEffectChains; 700 701 // list of sessions for which a valid HW A/V sync ID was retrieved from the HAL 702 DefaultKeyedVector< audio_session_t , audio_hw_sync_t >mHwAvSyncIds; 703private: 704 sp<Client> registerPid(pid_t pid); // always returns non-0 705 706 // for use from destructor 707 status_t closeOutput_nonvirtual(audio_io_handle_t output); 708 void closeOutputInternal_l(sp<PlaybackThread> thread); 709 status_t closeInput_nonvirtual(audio_io_handle_t input); 710 void closeInputInternal_l(sp<RecordThread> thread); 711 void setAudioHwSyncForSession_l(PlaybackThread *thread, audio_session_t sessionId); 712 713 status_t checkStreamType(audio_stream_type_t stream) const; 714 715#ifdef TEE_SINK 716 // all record threads serially share a common tee sink, which is re-created on format change 717 sp<NBAIO_Sink> mRecordTeeSink; 718 sp<NBAIO_Source> mRecordTeeSource; 719#endif 720 721public: 722 723#ifdef TEE_SINK 724 // tee sink, if enabled by property, allows dumpsys to write most recent audio to .wav file 725 static void dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id = 0); 726 727 // whether tee sink is enabled by property 728 static bool mTeeSinkInputEnabled; 729 static bool mTeeSinkOutputEnabled; 730 static bool mTeeSinkTrackEnabled; 731 732 // runtime configured size of each tee sink pipe, in frames 733 static size_t mTeeSinkInputFrames; 734 static size_t mTeeSinkOutputFrames; 735 static size_t mTeeSinkTrackFrames; 736 737 // compile-time default size of tee sink pipes, in frames 738 // 0x200000 stereo 16-bit PCM frames = 47.5 seconds at 44.1 kHz, 8 megabytes 739 static const size_t kTeeSinkInputFramesDefault = 0x200000; 740 static const size_t kTeeSinkOutputFramesDefault = 0x200000; 741 static const size_t kTeeSinkTrackFramesDefault = 0x200000; 742#endif 743 744 // This method reads from a variable without mLock, but the variable is updated under mLock. So 745 // we might read a stale value, or a value that's inconsistent with respect to other variables. 746 // In this case, it's safe because the return value isn't used for making an important decision. 747 // The reason we don't want to take mLock is because it could block the caller for a long time. 748 bool isLowRamDevice() const { return mIsLowRamDevice; } 749 750private: 751 bool mIsLowRamDevice; 752 bool mIsDeviceTypeKnown; 753 nsecs_t mGlobalEffectEnableTime; // when a global effect was last enabled 754 755 sp<PatchPanel> mPatchPanel; 756 757 bool mSystemReady; 758}; 759 760#undef INCLUDING_FROM_AUDIOFLINGER_H 761 762const char *formatToString(audio_format_t format); 763String8 inputFlagsToString(audio_input_flags_t flags); 764String8 outputFlagsToString(audio_output_flags_t flags); 765String8 devicesToString(audio_devices_t devices); 766const char *sourceToString(audio_source_t source); 767 768// ---------------------------------------------------------------------------- 769 770} // namespace android 771 772#endif // ANDROID_AUDIO_FLINGER_H 773