AudioFlinger.h revision 83faee053cfd4251dbb591b62039f563ffdac399
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 <stdint.h> 22#include <sys/types.h> 23#include <limits.h> 24 25#include <common_time/cc_helper.h> 26 27#include <media/IAudioFlinger.h> 28#include <media/IAudioFlingerClient.h> 29#include <media/IAudioTrack.h> 30#include <media/IAudioRecord.h> 31#include <media/AudioSystem.h> 32#include <media/AudioTrack.h> 33 34#include <utils/Atomic.h> 35#include <utils/Errors.h> 36#include <utils/threads.h> 37#include <utils/SortedVector.h> 38#include <utils/TypeHelpers.h> 39#include <utils/Vector.h> 40 41#include <binder/BinderService.h> 42#include <binder/MemoryDealer.h> 43 44#include <system/audio.h> 45#include <hardware/audio.h> 46#include <hardware/audio_policy.h> 47 48#include "AudioBufferProvider.h" 49#include "FastMixer.h" 50#include "NBAIO.h" 51 52#include <powermanager/IPowerManager.h> 53 54namespace android { 55 56class audio_track_cblk_t; 57class effect_param_cblk_t; 58class AudioMixer; 59class AudioBuffer; 60class AudioResampler; 61class FastMixer; 62 63// ---------------------------------------------------------------------------- 64 65// AudioFlinger has a hard-coded upper limit of 2 channels for capture and playback. 66// There is support for > 2 channel tracks down-mixed to 2 channel output via a down-mix effect. 67// Adding full support for > 2 channel capture or playback would require more than simply changing 68// this #define. There is an independent hard-coded upper limit in AudioMixer; 69// removing that AudioMixer limit would be necessary but insufficient to support > 2 channels. 70// The macro FCC_2 highlights some (but not all) places where there is are 2-channel assumptions. 71// Search also for "2", "left", "right", "[0]", "[1]", ">> 16", "<< 16", etc. 72#define FCC_2 2 // FCC_2 = Fixed Channel Count 2 73 74static const nsecs_t kDefaultStandbyTimeInNsecs = seconds(3); 75 76class AudioFlinger : 77 public BinderService<AudioFlinger>, 78 public BnAudioFlinger 79{ 80 friend class BinderService<AudioFlinger>; // for AudioFlinger() 81public: 82 static const char* getServiceName() { return "media.audio_flinger"; } 83 84 virtual status_t dump(int fd, const Vector<String16>& args); 85 86 // IAudioFlinger interface, in binder opcode order 87 virtual sp<IAudioTrack> createTrack( 88 pid_t pid, 89 audio_stream_type_t streamType, 90 uint32_t sampleRate, 91 audio_format_t format, 92 uint32_t channelMask, 93 int frameCount, 94 IAudioFlinger::track_flags_t flags, 95 const sp<IMemory>& sharedBuffer, 96 audio_io_handle_t output, 97 pid_t tid, 98 int *sessionId, 99 status_t *status); 100 101 virtual sp<IAudioRecord> openRecord( 102 pid_t pid, 103 audio_io_handle_t input, 104 uint32_t sampleRate, 105 audio_format_t format, 106 uint32_t channelMask, 107 int frameCount, 108 IAudioFlinger::track_flags_t flags, 109 int *sessionId, 110 status_t *status); 111 112 virtual uint32_t sampleRate(audio_io_handle_t output) const; 113 virtual int channelCount(audio_io_handle_t output) const; 114 virtual audio_format_t format(audio_io_handle_t output) const; 115 virtual size_t frameCount(audio_io_handle_t output) const; 116 virtual uint32_t latency(audio_io_handle_t output) const; 117 118 virtual status_t setMasterVolume(float value); 119 virtual status_t setMasterMute(bool muted); 120 121 virtual float masterVolume() const; 122 virtual float masterVolumeSW() const; 123 virtual bool masterMute() const; 124 125 virtual status_t setStreamVolume(audio_stream_type_t stream, float value, 126 audio_io_handle_t output); 127 virtual status_t setStreamMute(audio_stream_type_t stream, bool muted); 128 129 virtual float streamVolume(audio_stream_type_t stream, 130 audio_io_handle_t output) const; 131 virtual bool streamMute(audio_stream_type_t stream) const; 132 133 virtual status_t setMode(audio_mode_t mode); 134 135 virtual status_t setMicMute(bool state); 136 virtual bool getMicMute() const; 137 138 virtual status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs); 139 virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) const; 140 141 virtual void registerClient(const sp<IAudioFlingerClient>& client); 142 143 virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const; 144 145 virtual audio_io_handle_t openOutput(audio_module_handle_t module, 146 audio_devices_t *pDevices, 147 uint32_t *pSamplingRate, 148 audio_format_t *pFormat, 149 audio_channel_mask_t *pChannelMask, 150 uint32_t *pLatencyMs, 151 audio_output_flags_t flags); 152 153 virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1, 154 audio_io_handle_t output2); 155 156 virtual status_t closeOutput(audio_io_handle_t output); 157 158 virtual status_t suspendOutput(audio_io_handle_t output); 159 160 virtual status_t restoreOutput(audio_io_handle_t output); 161 162 virtual audio_io_handle_t openInput(audio_module_handle_t module, 163 audio_devices_t *pDevices, 164 uint32_t *pSamplingRate, 165 audio_format_t *pFormat, 166 audio_channel_mask_t *pChannelMask); 167 168 virtual status_t closeInput(audio_io_handle_t input); 169 170 virtual status_t setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output); 171 172 virtual status_t setVoiceVolume(float volume); 173 174 virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 175 audio_io_handle_t output) const; 176 177 virtual unsigned int getInputFramesLost(audio_io_handle_t ioHandle) const; 178 179 virtual int newAudioSessionId(); 180 181 virtual void acquireAudioSessionId(int audioSession); 182 183 virtual void releaseAudioSessionId(int audioSession); 184 185 virtual status_t queryNumberEffects(uint32_t *numEffects) const; 186 187 virtual status_t queryEffect(uint32_t index, effect_descriptor_t *descriptor) const; 188 189 virtual status_t getEffectDescriptor(const effect_uuid_t *pUuid, 190 effect_descriptor_t *descriptor) const; 191 192 virtual sp<IEffect> createEffect(pid_t pid, 193 effect_descriptor_t *pDesc, 194 const sp<IEffectClient>& effectClient, 195 int32_t priority, 196 audio_io_handle_t io, 197 int sessionId, 198 status_t *status, 199 int *id, 200 int *enabled); 201 202 virtual status_t moveEffects(int sessionId, audio_io_handle_t srcOutput, 203 audio_io_handle_t dstOutput); 204 205 virtual audio_module_handle_t loadHwModule(const char *name); 206 207 virtual status_t onTransact( 208 uint32_t code, 209 const Parcel& data, 210 Parcel* reply, 211 uint32_t flags); 212 213 // end of IAudioFlinger interface 214 215 class SyncEvent; 216 217 typedef void (*sync_event_callback_t)(const wp<SyncEvent>& event) ; 218 219 class SyncEvent : public RefBase { 220 public: 221 SyncEvent(AudioSystem::sync_event_t type, 222 int triggerSession, 223 int listenerSession, 224 sync_event_callback_t callBack, 225 void *cookie) 226 : mType(type), mTriggerSession(triggerSession), mListenerSession(listenerSession), 227 mCallback(callBack), mCookie(cookie) 228 {} 229 230 virtual ~SyncEvent() {} 231 232 void trigger() { Mutex::Autolock _l(mLock); if (mCallback) mCallback(this); } 233 void cancel() {Mutex::Autolock _l(mLock); mCallback = NULL; } 234 AudioSystem::sync_event_t type() const { return mType; } 235 int triggerSession() const { return mTriggerSession; } 236 int listenerSession() const { return mListenerSession; } 237 void *cookie() const { return mCookie; } 238 239 private: 240 const AudioSystem::sync_event_t mType; 241 const int mTriggerSession; 242 const int mListenerSession; 243 sync_event_callback_t mCallback; 244 void * const mCookie; 245 Mutex mLock; 246 }; 247 248 sp<SyncEvent> createSyncEvent(AudioSystem::sync_event_t type, 249 int triggerSession, 250 int listenerSession, 251 sync_event_callback_t callBack, 252 void *cookie); 253private: 254 audio_mode_t getMode() const { return mMode; } 255 256 bool btNrecIsOff() const { return mBtNrecIsOff; } 257 258 AudioFlinger(); 259 virtual ~AudioFlinger(); 260 261 // call in any IAudioFlinger method that accesses mPrimaryHardwareDev 262 status_t initCheck() const { return mPrimaryHardwareDev == NULL ? NO_INIT : NO_ERROR; } 263 264 // RefBase 265 virtual void onFirstRef(); 266 267 audio_hw_device_t* findSuitableHwDev_l(audio_module_handle_t module, uint32_t devices); 268 void purgeStaleEffects_l(); 269 270 // standby delay for MIXER and DUPLICATING playback threads is read from property 271 // ro.audio.flinger_standbytime_ms or defaults to kDefaultStandbyTimeInNsecs 272 static nsecs_t mStandbyTimeInNsecs; 273 274 // Internal dump utilites. 275 status_t dumpPermissionDenial(int fd, const Vector<String16>& args); 276 status_t dumpClients(int fd, const Vector<String16>& args); 277 status_t dumpInternals(int fd, const Vector<String16>& args); 278 279 // --- Client --- 280 class Client : public RefBase { 281 public: 282 Client(const sp<AudioFlinger>& audioFlinger, pid_t pid); 283 virtual ~Client(); 284 sp<MemoryDealer> heap() const; 285 pid_t pid() const { return mPid; } 286 sp<AudioFlinger> audioFlinger() const { return mAudioFlinger; } 287 288 bool reserveTimedTrack(); 289 void releaseTimedTrack(); 290 291 private: 292 Client(const Client&); 293 Client& operator = (const Client&); 294 const sp<AudioFlinger> mAudioFlinger; 295 const sp<MemoryDealer> mMemoryDealer; 296 const pid_t mPid; 297 298 Mutex mTimedTrackLock; 299 int mTimedTrackCount; 300 }; 301 302 // --- Notification Client --- 303 class NotificationClient : public IBinder::DeathRecipient { 304 public: 305 NotificationClient(const sp<AudioFlinger>& audioFlinger, 306 const sp<IAudioFlingerClient>& client, 307 pid_t pid); 308 virtual ~NotificationClient(); 309 310 sp<IAudioFlingerClient> audioFlingerClient() const { return mAudioFlingerClient; } 311 312 // IBinder::DeathRecipient 313 virtual void binderDied(const wp<IBinder>& who); 314 315 private: 316 NotificationClient(const NotificationClient&); 317 NotificationClient& operator = (const NotificationClient&); 318 319 const sp<AudioFlinger> mAudioFlinger; 320 const pid_t mPid; 321 const sp<IAudioFlingerClient> mAudioFlingerClient; 322 }; 323 324 class TrackHandle; 325 class RecordHandle; 326 class RecordThread; 327 class PlaybackThread; 328 class MixerThread; 329 class DirectOutputThread; 330 class DuplicatingThread; 331 class Track; 332 class RecordTrack; 333 class EffectModule; 334 class EffectHandle; 335 class EffectChain; 336 struct AudioStreamOut; 337 struct AudioStreamIn; 338 339 class ThreadBase : public Thread { 340 public: 341 342 enum type_t { 343 MIXER, // Thread class is MixerThread 344 DIRECT, // Thread class is DirectOutputThread 345 DUPLICATING, // Thread class is DuplicatingThread 346 RECORD // Thread class is RecordThread 347 }; 348 349 ThreadBase (const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, uint32_t device, type_t type); 350 virtual ~ThreadBase(); 351 352 status_t dumpBase(int fd, const Vector<String16>& args); 353 status_t dumpEffectChains(int fd, const Vector<String16>& args); 354 355 void clearPowerManager(); 356 357 // base for record and playback 358 class TrackBase : public AudioBufferProvider, public RefBase { 359 360 public: 361 enum track_state { 362 IDLE, 363 TERMINATED, 364 // These are order-sensitive; do not change order without reviewing the impact. 365 // In particular there are assumptions about > STOPPED. 366 STOPPED, 367 RESUMING, 368 ACTIVE, 369 PAUSING, 370 PAUSED 371 }; 372 373 TrackBase(ThreadBase *thread, 374 const sp<Client>& client, 375 uint32_t sampleRate, 376 audio_format_t format, 377 uint32_t channelMask, 378 int frameCount, 379 const sp<IMemory>& sharedBuffer, 380 int sessionId); 381 virtual ~TrackBase(); 382 383 virtual status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 384 int triggerSession = 0) = 0; 385 virtual void stop() = 0; 386 sp<IMemory> getCblk() const { return mCblkMemory; } 387 audio_track_cblk_t* cblk() const { return mCblk; } 388 int sessionId() const { return mSessionId; } 389 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 390 391 protected: 392 TrackBase(const TrackBase&); 393 TrackBase& operator = (const TrackBase&); 394 395 // AudioBufferProvider interface 396 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) = 0; 397 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 398 399 audio_format_t format() const { 400 return mFormat; 401 } 402 403 int channelCount() const { return mChannelCount; } 404 405 uint32_t channelMask() const { return mChannelMask; } 406 407 int sampleRate() const; // FIXME inline after cblk sr moved 408 409 void* getBuffer(uint32_t offset, uint32_t frames) const; 410 411 bool isStopped() const { 412 return mState == STOPPED; 413 } 414 415 bool isTerminated() const { 416 return mState == TERMINATED; 417 } 418 419 bool step(); 420 void reset(); 421 422 const wp<ThreadBase> mThread; 423 /*const*/ sp<Client> mClient; // see explanation at ~TrackBase() why not const 424 sp<IMemory> mCblkMemory; 425 audio_track_cblk_t* mCblk; 426 void* mBuffer; 427 void* mBufferEnd; 428 uint32_t mFrameCount; 429 // we don't really need a lock for these 430 track_state mState; 431 const uint32_t mSampleRate; // initial sample rate only; for tracks which 432 // support dynamic rates, the current value is in control block 433 const audio_format_t mFormat; 434 bool mStepServerFailed; 435 const int mSessionId; 436 uint8_t mChannelCount; 437 uint32_t mChannelMask; 438 Vector < sp<SyncEvent> >mSyncEvents; 439 }; 440 441 class ConfigEvent { 442 public: 443 ConfigEvent() : mEvent(0), mParam(0) {} 444 445 int mEvent; 446 int mParam; 447 }; 448 449 class PMDeathRecipient : public IBinder::DeathRecipient { 450 public: 451 PMDeathRecipient(const wp<ThreadBase>& thread) : mThread(thread) {} 452 virtual ~PMDeathRecipient() {} 453 454 // IBinder::DeathRecipient 455 virtual void binderDied(const wp<IBinder>& who); 456 457 private: 458 PMDeathRecipient(const PMDeathRecipient&); 459 PMDeathRecipient& operator = (const PMDeathRecipient&); 460 461 wp<ThreadBase> mThread; 462 }; 463 464 virtual status_t initCheck() const = 0; 465 type_t type() const { return mType; } 466 uint32_t sampleRate() const { return mSampleRate; } 467 int channelCount() const { return mChannelCount; } 468 audio_format_t format() const { return mFormat; } 469 // Called by AudioFlinger::frameCount(audio_io_handle_t output) and effects, 470 // and returns the normal mix buffer's frame count. No API for HAL frame count. 471 size_t frameCount() const { return mNormalFrameCount; } 472 void wakeUp() { mWaitWorkCV.broadcast(); } 473 // Should be "virtual status_t requestExitAndWait()" and override same 474 // method in Thread, but Thread::requestExitAndWait() is not yet virtual. 475 void exit(); 476 virtual bool checkForNewParameters_l() = 0; 477 virtual status_t setParameters(const String8& keyValuePairs); 478 virtual String8 getParameters(const String8& keys) = 0; 479 virtual void audioConfigChanged_l(int event, int param = 0) = 0; 480 void sendConfigEvent(int event, int param = 0); 481 void sendConfigEvent_l(int event, int param = 0); 482 void processConfigEvents(); 483 audio_io_handle_t id() const { return mId;} 484 bool standby() const { return mStandby; } 485 uint32_t device() const { return mDevice; } 486 virtual audio_stream_t* stream() const = 0; 487 488 sp<EffectHandle> createEffect_l( 489 const sp<AudioFlinger::Client>& client, 490 const sp<IEffectClient>& effectClient, 491 int32_t priority, 492 int sessionId, 493 effect_descriptor_t *desc, 494 int *enabled, 495 status_t *status); 496 void disconnectEffect(const sp< EffectModule>& effect, 497 const wp<EffectHandle>& handle, 498 bool unpinIfLast); 499 500 // return values for hasAudioSession (bit field) 501 enum effect_state { 502 EFFECT_SESSION = 0x1, // the audio session corresponds to at least one 503 // effect 504 TRACK_SESSION = 0x2 // the audio session corresponds to at least one 505 // track 506 }; 507 508 // get effect chain corresponding to session Id. 509 sp<EffectChain> getEffectChain(int sessionId); 510 // same as getEffectChain() but must be called with ThreadBase mutex locked 511 sp<EffectChain> getEffectChain_l(int sessionId); 512 // add an effect chain to the chain list (mEffectChains) 513 virtual status_t addEffectChain_l(const sp<EffectChain>& chain) = 0; 514 // remove an effect chain from the chain list (mEffectChains) 515 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain) = 0; 516 // lock all effect chains Mutexes. Must be called before releasing the 517 // ThreadBase mutex before processing the mixer and effects. This guarantees the 518 // integrity of the chains during the process. 519 // Also sets the parameter 'effectChains' to current value of mEffectChains. 520 void lockEffectChains_l(Vector< sp<EffectChain> >& effectChains); 521 // unlock effect chains after process 522 void unlockEffectChains(const Vector< sp<EffectChain> >& effectChains); 523 // set audio mode to all effect chains 524 void setMode(audio_mode_t mode); 525 // get effect module with corresponding ID on specified audio session 526 sp<AudioFlinger::EffectModule> getEffect_l(int sessionId, int effectId); 527 // add and effect module. Also creates the effect chain is none exists for 528 // the effects audio session 529 status_t addEffect_l(const sp< EffectModule>& effect); 530 // remove and effect module. Also removes the effect chain is this was the last 531 // effect 532 void removeEffect_l(const sp< EffectModule>& effect); 533 // detach all tracks connected to an auxiliary effect 534 virtual void detachAuxEffect_l(int effectId) {} 535 // returns either EFFECT_SESSION if effects on this audio session exist in one 536 // chain, or TRACK_SESSION if tracks on this audio session exist, or both 537 virtual uint32_t hasAudioSession(int sessionId) = 0; 538 // the value returned by default implementation is not important as the 539 // strategy is only meaningful for PlaybackThread which implements this method 540 virtual uint32_t getStrategyForSession_l(int sessionId) { return 0; } 541 542 // suspend or restore effect according to the type of effect passed. a NULL 543 // type pointer means suspend all effects in the session 544 void setEffectSuspended(const effect_uuid_t *type, 545 bool suspend, 546 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 547 // check if some effects must be suspended/restored when an effect is enabled 548 // or disabled 549 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 550 bool enabled, 551 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 552 void checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 553 bool enabled, 554 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 555 556 virtual status_t setSyncEvent(const sp<SyncEvent>& event) = 0; 557 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) = 0; 558 559 560 mutable Mutex mLock; 561 562 protected: 563 564 // entry describing an effect being suspended in mSuspendedSessions keyed vector 565 class SuspendedSessionDesc : public RefBase { 566 public: 567 SuspendedSessionDesc() : mRefCount(0) {} 568 569 int mRefCount; // number of active suspend requests 570 effect_uuid_t mType; // effect type UUID 571 }; 572 573 void acquireWakeLock(); 574 void acquireWakeLock_l(); 575 void releaseWakeLock(); 576 void releaseWakeLock_l(); 577 void setEffectSuspended_l(const effect_uuid_t *type, 578 bool suspend, 579 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 580 // updated mSuspendedSessions when an effect suspended or restored 581 void updateSuspendedSessions_l(const effect_uuid_t *type, 582 bool suspend, 583 int sessionId); 584 // check if some effects must be suspended when an effect chain is added 585 void checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain); 586 587 friend class AudioFlinger; // for mEffectChains 588 589 const type_t mType; 590 591 // Used by parameters, config events, addTrack_l, exit 592 Condition mWaitWorkCV; 593 594 const sp<AudioFlinger> mAudioFlinger; 595 uint32_t mSampleRate; 596 size_t mFrameCount; // output HAL, direct output, record 597 size_t mNormalFrameCount; // normal mixer and effects 598 uint32_t mChannelMask; 599 uint16_t mChannelCount; 600 size_t mFrameSize; 601 audio_format_t mFormat; 602 603 // Parameter sequence by client: binder thread calling setParameters(): 604 // 1. Lock mLock 605 // 2. Append to mNewParameters 606 // 3. mWaitWorkCV.signal 607 // 4. mParamCond.waitRelative with timeout 608 // 5. read mParamStatus 609 // 6. mWaitWorkCV.signal 610 // 7. Unlock 611 // 612 // Parameter sequence by server: threadLoop calling checkForNewParameters_l(): 613 // 1. Lock mLock 614 // 2. If there is an entry in mNewParameters proceed ... 615 // 2. Read first entry in mNewParameters 616 // 3. Process 617 // 4. Remove first entry from mNewParameters 618 // 5. Set mParamStatus 619 // 6. mParamCond.signal 620 // 7. mWaitWorkCV.wait with timeout (this is to avoid overwriting mParamStatus) 621 // 8. Unlock 622 Condition mParamCond; 623 Vector<String8> mNewParameters; 624 status_t mParamStatus; 625 626 Vector<ConfigEvent> mConfigEvents; 627 bool mStandby; 628 const audio_io_handle_t mId; 629 Vector< sp<EffectChain> > mEffectChains; 630 uint32_t mDevice; // output device for PlaybackThread 631 // input + output devices for RecordThread 632 static const int kNameLength = 16; // prctl(PR_SET_NAME) limit 633 char mName[kNameLength]; 634 sp<IPowerManager> mPowerManager; 635 sp<IBinder> mWakeLockToken; 636 const sp<PMDeathRecipient> mDeathRecipient; 637 // list of suspended effects per session and per type. The first vector is 638 // keyed by session ID, the second by type UUID timeLow field 639 KeyedVector< int, KeyedVector< int, sp<SuspendedSessionDesc> > > mSuspendedSessions; 640 }; 641 642 struct stream_type_t { 643 stream_type_t() 644 : volume(1.0f), 645 mute(false) 646 { 647 } 648 float volume; 649 bool mute; 650 }; 651 652 // --- PlaybackThread --- 653 class PlaybackThread : public ThreadBase { 654 public: 655 656 enum mixer_state { 657 MIXER_IDLE, // no active tracks 658 MIXER_TRACKS_ENABLED, // at least one active track, but no track has any data ready 659 MIXER_TRACKS_READY // at least one active track, and at least one track has data 660 // standby mode does not have an enum value 661 // suspend by audio policy manager is orthogonal to mixer state 662 }; 663 664 // playback track 665 class Track : public TrackBase, public VolumeProvider { 666 public: 667 Track( PlaybackThread *thread, 668 const sp<Client>& client, 669 audio_stream_type_t streamType, 670 uint32_t sampleRate, 671 audio_format_t format, 672 uint32_t channelMask, 673 int frameCount, 674 const sp<IMemory>& sharedBuffer, 675 int sessionId, 676 IAudioFlinger::track_flags_t flags); 677 virtual ~Track(); 678 679 void dump(char* buffer, size_t size); 680 virtual status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 681 int triggerSession = 0); 682 virtual void stop(); 683 void pause(); 684 685 void flush(); 686 void destroy(); 687 void mute(bool); 688 int name() const { 689 return mName; 690 } 691 692 audio_stream_type_t streamType() const { 693 return mStreamType; 694 } 695 status_t attachAuxEffect(int EffectId); 696 void setAuxBuffer(int EffectId, int32_t *buffer); 697 int32_t *auxBuffer() const { return mAuxBuffer; } 698 void setMainBuffer(int16_t *buffer) { mMainBuffer = buffer; } 699 int16_t *mainBuffer() const { return mMainBuffer; } 700 int auxEffectId() const { return mAuxEffectId; } 701 702#if 0 703 bool isFastTrack() const 704 { return (mFlags & IAudioFlinger::TRACK_FAST) != 0; } 705#endif 706 707 // implement FastMixerState::VolumeProvider interface 708 virtual uint32_t getVolumeLR(); 709 710 protected: 711 // for numerous 712 friend class PlaybackThread; 713 friend class MixerThread; 714 friend class DirectOutputThread; 715 716 Track(const Track&); 717 Track& operator = (const Track&); 718 719 // AudioBufferProvider interface 720 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts = kInvalidPTS); 721 // releaseBuffer() not overridden 722 723 virtual uint32_t framesReady() const; 724 725 bool isMuted() const { return mMute; } 726 bool isPausing() const { 727 return mState == PAUSING; 728 } 729 bool isPaused() const { 730 return mState == PAUSED; 731 } 732 bool isReady() const; 733 void setPaused() { mState = PAUSED; } 734 void reset(); 735 736 bool isOutputTrack() const { 737 return (mStreamType == AUDIO_STREAM_CNT); 738 } 739 740 sp<IMemory> sharedBuffer() const { return mSharedBuffer; } 741 742 bool presentationComplete(size_t framesWritten, size_t audioHalFrames); 743 void triggerEvents(AudioSystem::sync_event_t type); 744 745 public: 746 virtual bool isTimedTrack() const { return false; } 747 bool isFastTrack() const { return (mFlags & IAudioFlinger::TRACK_FAST) != 0; } 748 protected: 749 750 // we don't really need a lock for these 751 volatile bool mMute; 752 // FILLED state is used for suppressing volume ramp at begin of playing 753 enum {FS_INVALID, FS_FILLING, FS_FILLED, FS_ACTIVE}; 754 mutable uint8_t mFillingUpStatus; 755 int8_t mRetryCount; 756 const sp<IMemory> mSharedBuffer; 757 bool mResetDone; 758 const audio_stream_type_t mStreamType; 759 int mName; // track name on the normal mixer 760 int16_t *mMainBuffer; 761 int32_t *mAuxBuffer; 762 int mAuxEffectId; 763 bool mHasVolumeController; 764 size_t mPresentationCompleteFrames; // number of frames written to the audio HAL 765 // when this track will be fully rendered 766 private: 767 IAudioFlinger::track_flags_t mFlags; 768 int mFastIndex; // index within FastMixerState::mFastTracks[] or -1 769 volatile float mCachedVolume; // combined master volume and stream type volume; 770 // 'volatile' means accessed without lock or 771 // barrier, but is read/written atomically 772 }; // end of Track 773 774 class TimedTrack : public Track { 775 public: 776 static sp<TimedTrack> create(PlaybackThread *thread, 777 const sp<Client>& client, 778 audio_stream_type_t streamType, 779 uint32_t sampleRate, 780 audio_format_t format, 781 uint32_t channelMask, 782 int frameCount, 783 const sp<IMemory>& sharedBuffer, 784 int sessionId); 785 ~TimedTrack(); 786 787 class TimedBuffer { 788 public: 789 TimedBuffer(); 790 TimedBuffer(const sp<IMemory>& buffer, int64_t pts); 791 const sp<IMemory>& buffer() const { return mBuffer; } 792 int64_t pts() const { return mPTS; } 793 uint32_t position() const { return mPosition; } 794 void setPosition(uint32_t pos) { mPosition = pos; } 795 private: 796 sp<IMemory> mBuffer; 797 int64_t mPTS; 798 uint32_t mPosition; 799 }; 800 801 // Mixer facing methods. 802 virtual bool isTimedTrack() const { return true; } 803 virtual uint32_t framesReady() const; 804 805 // AudioBufferProvider interface 806 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, 807 int64_t pts); 808 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 809 810 // Client/App facing methods. 811 status_t allocateTimedBuffer(size_t size, 812 sp<IMemory>* buffer); 813 status_t queueTimedBuffer(const sp<IMemory>& buffer, 814 int64_t pts); 815 status_t setMediaTimeTransform(const LinearTransform& xform, 816 TimedAudioTrack::TargetTimeline target); 817 818 private: 819 TimedTrack(PlaybackThread *thread, 820 const sp<Client>& client, 821 audio_stream_type_t streamType, 822 uint32_t sampleRate, 823 audio_format_t format, 824 uint32_t channelMask, 825 int frameCount, 826 const sp<IMemory>& sharedBuffer, 827 int sessionId); 828 829 void timedYieldSamples_l(AudioBufferProvider::Buffer* buffer); 830 void timedYieldSilence_l(uint32_t numFrames, 831 AudioBufferProvider::Buffer* buffer); 832 void trimTimedBufferQueue_l(); 833 void trimTimedBufferQueueHead_l(const char* logTag); 834 void updateFramesPendingAfterTrim_l(const TimedBuffer& buf, 835 const char* logTag); 836 837 uint64_t mLocalTimeFreq; 838 LinearTransform mLocalTimeToSampleTransform; 839 LinearTransform mMediaTimeToSampleTransform; 840 sp<MemoryDealer> mTimedMemoryDealer; 841 842 Vector<TimedBuffer> mTimedBufferQueue; 843 bool mQueueHeadInFlight; 844 bool mTrimQueueHeadOnRelease; 845 uint32_t mFramesPendingInQueue; 846 847 uint8_t* mTimedSilenceBuffer; 848 uint32_t mTimedSilenceBufferSize; 849 mutable Mutex mTimedBufferQueueLock; 850 bool mTimedAudioOutputOnTime; 851 CCHelper mCCHelper; 852 853 Mutex mMediaTimeTransformLock; 854 LinearTransform mMediaTimeTransform; 855 bool mMediaTimeTransformValid; 856 TimedAudioTrack::TargetTimeline mMediaTimeTransformTarget; 857 }; 858 859 860 // playback track 861 class OutputTrack : public Track { 862 public: 863 864 class Buffer: public AudioBufferProvider::Buffer { 865 public: 866 int16_t *mBuffer; 867 }; 868 869 OutputTrack(PlaybackThread *thread, 870 DuplicatingThread *sourceThread, 871 uint32_t sampleRate, 872 audio_format_t format, 873 uint32_t channelMask, 874 int frameCount); 875 virtual ~OutputTrack(); 876 877 virtual status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 878 int triggerSession = 0); 879 virtual void stop(); 880 bool write(int16_t* data, uint32_t frames); 881 bool bufferQueueEmpty() const { return mBufferQueue.size() == 0; } 882 bool isActive() const { return mActive; } 883 const wp<ThreadBase>& thread() const { return mThread; } 884 885 private: 886 887 enum { 888 NO_MORE_BUFFERS = 0x80000001, // same in AudioTrack.h, ok to be different value 889 }; 890 891 status_t obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs); 892 void clearBufferQueue(); 893 894 // Maximum number of pending buffers allocated by OutputTrack::write() 895 static const uint8_t kMaxOverFlowBuffers = 10; 896 897 Vector < Buffer* > mBufferQueue; 898 AudioBufferProvider::Buffer mOutBuffer; 899 bool mActive; 900 DuplicatingThread* const mSourceThread; // for waitTimeMs() in write() 901 }; // end of OutputTrack 902 903 PlaybackThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 904 audio_io_handle_t id, uint32_t device, type_t type); 905 virtual ~PlaybackThread(); 906 907 status_t dump(int fd, const Vector<String16>& args); 908 909 // Thread virtuals 910 virtual status_t readyToRun(); 911 virtual bool threadLoop(); 912 913 // RefBase 914 virtual void onFirstRef(); 915 916protected: 917 // Code snippets that were lifted up out of threadLoop() 918 virtual void threadLoop_mix() = 0; 919 virtual void threadLoop_sleepTime() = 0; 920 virtual void threadLoop_write(); 921 virtual void threadLoop_standby(); 922 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) { } 923 924 // prepareTracks_l reads and writes mActiveTracks, and also returns the 925 // pending set of tracks to remove via Vector 'tracksToRemove'. The caller is 926 // responsible for clearing or destroying this Vector later on, when it 927 // is safe to do so. That will drop the final ref count and destroy the tracks. 928 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove) = 0; 929 930public: 931 932 virtual status_t initCheck() const { return (mOutput == NULL) ? NO_INIT : NO_ERROR; } 933 934 // return estimated latency in milliseconds, as reported by HAL 935 uint32_t latency() const; 936 937 void setMasterVolume(float value); 938 void setMasterMute(bool muted); 939 940 void setStreamVolume(audio_stream_type_t stream, float value); 941 void setStreamMute(audio_stream_type_t stream, bool muted); 942 943 float streamVolume(audio_stream_type_t stream) const; 944 945 sp<Track> createTrack_l( 946 const sp<AudioFlinger::Client>& client, 947 audio_stream_type_t streamType, 948 uint32_t sampleRate, 949 audio_format_t format, 950 uint32_t channelMask, 951 int frameCount, 952 const sp<IMemory>& sharedBuffer, 953 int sessionId, 954 IAudioFlinger::track_flags_t flags, 955 pid_t tid, 956 status_t *status); 957 958 AudioStreamOut* getOutput() const; 959 AudioStreamOut* clearOutput(); 960 virtual audio_stream_t* stream() const; 961 962 void suspend() { mSuspended++; } 963 void restore() { if (mSuspended > 0) mSuspended--; } 964 bool isSuspended() const { return (mSuspended > 0); } 965 virtual String8 getParameters(const String8& keys); 966 virtual void audioConfigChanged_l(int event, int param = 0); 967 status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames); 968 int16_t *mixBuffer() const { return mMixBuffer; }; 969 970 virtual void detachAuxEffect_l(int effectId); 971 status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track> track, 972 int EffectId); 973 status_t attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track> track, 974 int EffectId); 975 976 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 977 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 978 virtual uint32_t hasAudioSession(int sessionId); 979 virtual uint32_t getStrategyForSession_l(int sessionId); 980 981 982 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 983 virtual bool isValidSyncEvent(const sp<SyncEvent>& event); 984 985 protected: 986 int16_t* mMixBuffer; 987 uint32_t mSuspended; // suspend count, > 0 means suspended 988 int mBytesWritten; 989 private: 990 // mMasterMute is in both PlaybackThread and in AudioFlinger. When a 991 // PlaybackThread needs to find out if master-muted, it checks it's local 992 // copy rather than the one in AudioFlinger. This optimization saves a lock. 993 bool mMasterMute; 994 void setMasterMute_l(bool muted) { mMasterMute = muted; } 995 protected: 996 SortedVector< wp<Track> > mActiveTracks; 997 998 // Allocate a track name for a given channel mask. 999 // Returns name >= 0 if successful, -1 on failure. 1000 virtual int getTrackName_l(audio_channel_mask_t channelMask) = 0; 1001 virtual void deleteTrackName_l(int name) = 0; 1002 1003 // Time to sleep between cycles when: 1004 virtual uint32_t activeSleepTimeUs() const; // mixer state MIXER_TRACKS_ENABLED 1005 virtual uint32_t idleSleepTimeUs() const = 0; // mixer state MIXER_IDLE 1006 virtual uint32_t suspendSleepTimeUs() const = 0; // audio policy manager suspended us 1007 // No sleep when mixer state == MIXER_TRACKS_READY; relies on audio HAL stream->write() 1008 // No sleep in standby mode; waits on a condition 1009 1010 // Code snippets that are temporarily lifted up out of threadLoop() until the merge 1011 void checkSilentMode_l(); 1012 1013 // Non-trivial for DUPLICATING only 1014 virtual void saveOutputTracks() { } 1015 virtual void clearOutputTracks() { } 1016 1017 // Cache various calculated values, at threadLoop() entry and after a parameter change 1018 virtual void cacheParameters_l(); 1019 1020 private: 1021 1022 friend class AudioFlinger; // for numerous 1023 1024 PlaybackThread(const Client&); 1025 PlaybackThread& operator = (const PlaybackThread&); 1026 1027 status_t addTrack_l(const sp<Track>& track); 1028 void destroyTrack_l(const sp<Track>& track); 1029 void removeTrack_l(const sp<Track>& track); 1030 1031 void readOutputParameters(); 1032 1033 virtual status_t dumpInternals(int fd, const Vector<String16>& args); 1034 status_t dumpTracks(int fd, const Vector<String16>& args); 1035 1036 SortedVector< sp<Track> > mTracks; 1037 // mStreamTypes[] uses 1 additional stream type internally for the OutputTrack used by DuplicatingThread 1038 stream_type_t mStreamTypes[AUDIO_STREAM_CNT + 1]; 1039 AudioStreamOut *mOutput; 1040 float mMasterVolume; 1041 nsecs_t mLastWriteTime; 1042 int mNumWrites; 1043 int mNumDelayedWrites; 1044 bool mInWrite; 1045 1046 // FIXME rename these former local variables of threadLoop to standard "m" names 1047 nsecs_t standbyTime; 1048 size_t mixBufferSize; 1049 1050 // cached copies of activeSleepTimeUs() and idleSleepTimeUs() made by cacheParameters_l() 1051 uint32_t activeSleepTime; 1052 uint32_t idleSleepTime; 1053 1054 uint32_t sleepTime; 1055 1056 // mixer status returned by prepareTracks_l() 1057 mixer_state mMixerStatus; // current cycle 1058 // previous cycle when in prepareTracks_l() 1059 1060 // FIXME move these declarations into the specific sub-class that needs them 1061 // MIXER only 1062 bool longStandbyExit; 1063 uint32_t sleepTimeShift; 1064 1065 // same as AudioFlinger::mStandbyTimeInNsecs except for DIRECT which uses a shorter value 1066 nsecs_t standbyDelay; 1067 1068 // MIXER only 1069 nsecs_t maxPeriod; 1070 1071 // DUPLICATING only 1072 uint32_t writeFrames; 1073 1074 private: 1075 // The HAL output sink is treated as non-blocking, but current implementation is blocking 1076 sp<NBAIO_Sink> mOutputSink; 1077 // If a fast mixer is present, the blocking pipe sink, otherwise clear 1078 sp<NBAIO_Sink> mPipeSink; 1079 // The current sink for the normal mixer to write it's (sub)mix, mOutputSink or mPipeSink 1080 sp<NBAIO_Sink> mNormalSink; 1081 public: 1082 virtual bool hasFastMixer() const = 0; 1083 1084 protected: 1085 // accessed by both binder threads and within threadLoop(), lock on mutex needed 1086 unsigned mFastTrackAvailMask; // bit i set if fast track [i] is available 1087 unsigned mFastTrackNewMask; // bit i set if fast track [i] just created 1088 Track* mFastTrackNewArray[FastMixerState::kMaxFastTracks]; 1089 1090 }; 1091 1092 class MixerThread : public PlaybackThread { 1093 public: 1094 MixerThread (const sp<AudioFlinger>& audioFlinger, 1095 AudioStreamOut* output, 1096 audio_io_handle_t id, 1097 uint32_t device, 1098 type_t type = MIXER); 1099 virtual ~MixerThread(); 1100 1101 // Thread virtuals 1102 1103 void invalidateTracks(audio_stream_type_t streamType); 1104 virtual bool checkForNewParameters_l(); 1105 virtual status_t dumpInternals(int fd, const Vector<String16>& args); 1106 1107 protected: 1108 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1109 virtual int getTrackName_l(audio_channel_mask_t channelMask); 1110 virtual void deleteTrackName_l(int name); 1111 virtual uint32_t idleSleepTimeUs() const; 1112 virtual uint32_t suspendSleepTimeUs() const; 1113 virtual void cacheParameters_l(); 1114 1115 // threadLoop snippets 1116 virtual void threadLoop_write(); 1117 virtual void threadLoop_standby(); 1118 virtual void threadLoop_mix(); 1119 virtual void threadLoop_sleepTime(); 1120 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 1121 1122 AudioMixer* mAudioMixer; // normal mixer 1123 private: 1124#ifdef SOAKER 1125 Thread* mSoaker; 1126#endif 1127 // one-time initialization, no locks required 1128 FastMixer* mFastMixer; // non-NULL if there is also a fast mixer 1129 1130 // contents are not guaranteed to be consistent, no locks required 1131 FastMixerDumpState mFastMixerDumpState; 1132 1133 // accessible only within the threadLoop(), no locks required 1134 // mFastMixer->sq() // for mutating and pushing state 1135 int32_t mFastMixerFutex; // for cold idle 1136 1137 public: 1138 virtual bool hasFastMixer() const { return mFastMixer != NULL; } 1139 }; 1140 1141 class DirectOutputThread : public PlaybackThread { 1142 public: 1143 1144 DirectOutputThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1145 audio_io_handle_t id, uint32_t device); 1146 virtual ~DirectOutputThread(); 1147 1148 // Thread virtuals 1149 1150 virtual bool checkForNewParameters_l(); 1151 1152 protected: 1153 virtual int getTrackName_l(audio_channel_mask_t channelMask); 1154 virtual void deleteTrackName_l(int name); 1155 virtual uint32_t activeSleepTimeUs() const; 1156 virtual uint32_t idleSleepTimeUs() const; 1157 virtual uint32_t suspendSleepTimeUs() const; 1158 virtual void cacheParameters_l(); 1159 1160 // threadLoop snippets 1161 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1162 virtual void threadLoop_mix(); 1163 virtual void threadLoop_sleepTime(); 1164 1165 // volumes last sent to audio HAL with stream->set_volume() 1166 // FIXME use standard representation and names 1167 float mLeftVolFloat; 1168 float mRightVolFloat; 1169 uint16_t mLeftVolShort; 1170 uint16_t mRightVolShort; 1171 1172 // FIXME rename these former local variables of threadLoop to standard names 1173 // next 3 were local to the while !exitingPending loop 1174 bool rampVolume; 1175 uint16_t leftVol; 1176 uint16_t rightVol; 1177 1178private: 1179 // prepareTracks_l() tells threadLoop_mix() the name of the single active track 1180 sp<Track> mActiveTrack; 1181 public: 1182 virtual bool hasFastMixer() const { return false; } 1183 }; 1184 1185 class DuplicatingThread : public MixerThread { 1186 public: 1187 DuplicatingThread (const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread, 1188 audio_io_handle_t id); 1189 virtual ~DuplicatingThread(); 1190 1191 // Thread virtuals 1192 void addOutputTrack(MixerThread* thread); 1193 void removeOutputTrack(MixerThread* thread); 1194 uint32_t waitTimeMs() const { return mWaitTimeMs; } 1195 protected: 1196 virtual uint32_t activeSleepTimeUs() const; 1197 1198 private: 1199 bool outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks); 1200 protected: 1201 // threadLoop snippets 1202 virtual void threadLoop_mix(); 1203 virtual void threadLoop_sleepTime(); 1204 virtual void threadLoop_write(); 1205 virtual void threadLoop_standby(); 1206 virtual void cacheParameters_l(); 1207 1208 private: 1209 // called from threadLoop, addOutputTrack, removeOutputTrack 1210 virtual void updateWaitTime_l(); 1211 protected: 1212 virtual void saveOutputTracks(); 1213 virtual void clearOutputTracks(); 1214 private: 1215 1216 uint32_t mWaitTimeMs; 1217 SortedVector < sp<OutputTrack> > outputTracks; 1218 SortedVector < sp<OutputTrack> > mOutputTracks; 1219 public: 1220 virtual bool hasFastMixer() const { return false; } 1221 }; 1222 1223 PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const; 1224 MixerThread *checkMixerThread_l(audio_io_handle_t output) const; 1225 RecordThread *checkRecordThread_l(audio_io_handle_t input) const; 1226 // no range check, AudioFlinger::mLock held 1227 bool streamMute_l(audio_stream_type_t stream) const 1228 { return mStreamTypes[stream].mute; } 1229 // no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held 1230 float streamVolume_l(audio_stream_type_t stream) const 1231 { return mStreamTypes[stream].volume; } 1232 void audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2); 1233 1234 // allocate an audio_io_handle_t, session ID, or effect ID 1235 uint32_t nextUniqueId(); 1236 1237 status_t moveEffectChain_l(int sessionId, 1238 PlaybackThread *srcThread, 1239 PlaybackThread *dstThread, 1240 bool reRegister); 1241 // return thread associated with primary hardware device, or NULL 1242 PlaybackThread *primaryPlaybackThread_l() const; 1243 uint32_t primaryOutputDevice_l() const; 1244 1245 // server side of the client's IAudioTrack 1246 class TrackHandle : public android::BnAudioTrack { 1247 public: 1248 TrackHandle(const sp<PlaybackThread::Track>& track); 1249 virtual ~TrackHandle(); 1250 virtual sp<IMemory> getCblk() const; 1251 virtual status_t start(); 1252 virtual void stop(); 1253 virtual void flush(); 1254 virtual void mute(bool); 1255 virtual void pause(); 1256 virtual status_t attachAuxEffect(int effectId); 1257 virtual status_t allocateTimedBuffer(size_t size, 1258 sp<IMemory>* buffer); 1259 virtual status_t queueTimedBuffer(const sp<IMemory>& buffer, 1260 int64_t pts); 1261 virtual status_t setMediaTimeTransform(const LinearTransform& xform, 1262 int target); 1263 virtual status_t onTransact( 1264 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 1265 private: 1266 const sp<PlaybackThread::Track> mTrack; 1267 }; 1268 1269 void removeClient_l(pid_t pid); 1270 void removeNotificationClient(pid_t pid); 1271 1272 1273 // record thread 1274 class RecordThread : public ThreadBase, public AudioBufferProvider 1275 { 1276 public: 1277 1278 // record track 1279 class RecordTrack : public TrackBase { 1280 public: 1281 RecordTrack(RecordThread *thread, 1282 const sp<Client>& client, 1283 uint32_t sampleRate, 1284 audio_format_t format, 1285 uint32_t channelMask, 1286 int frameCount, 1287 int sessionId); 1288 virtual ~RecordTrack(); 1289 1290 virtual status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 1291 int triggerSession = 0); 1292 virtual void stop(); 1293 1294 bool overflow() { bool tmp = mOverflow; mOverflow = false; return tmp; } 1295 bool setOverflow() { bool tmp = mOverflow; mOverflow = true; return tmp; } 1296 1297 void dump(char* buffer, size_t size); 1298 1299 private: 1300 friend class AudioFlinger; // for mState 1301 1302 RecordTrack(const RecordTrack&); 1303 RecordTrack& operator = (const RecordTrack&); 1304 1305 // AudioBufferProvider interface 1306 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts = kInvalidPTS); 1307 // releaseBuffer() not overridden 1308 1309 bool mOverflow; 1310 }; 1311 1312 1313 RecordThread(const sp<AudioFlinger>& audioFlinger, 1314 AudioStreamIn *input, 1315 uint32_t sampleRate, 1316 uint32_t channels, 1317 audio_io_handle_t id, 1318 uint32_t device); 1319 virtual ~RecordThread(); 1320 1321 // Thread 1322 virtual bool threadLoop(); 1323 virtual status_t readyToRun(); 1324 1325 // RefBase 1326 virtual void onFirstRef(); 1327 1328 virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; } 1329 sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l( 1330 const sp<AudioFlinger::Client>& client, 1331 uint32_t sampleRate, 1332 audio_format_t format, 1333 int channelMask, 1334 int frameCount, 1335 int sessionId, 1336 status_t *status); 1337 1338 status_t start(RecordTrack* recordTrack, 1339 AudioSystem::sync_event_t event, 1340 int triggerSession); 1341 void stop(RecordTrack* recordTrack); 1342 status_t dump(int fd, const Vector<String16>& args); 1343 AudioStreamIn* getInput() const; 1344 AudioStreamIn* clearInput(); 1345 virtual audio_stream_t* stream() const; 1346 1347 // AudioBufferProvider interface 1348 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts); 1349 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 1350 1351 virtual bool checkForNewParameters_l(); 1352 virtual String8 getParameters(const String8& keys); 1353 virtual void audioConfigChanged_l(int event, int param = 0); 1354 void readInputParameters(); 1355 virtual unsigned int getInputFramesLost(); 1356 1357 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1358 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1359 virtual uint32_t hasAudioSession(int sessionId); 1360 RecordTrack* track(); 1361 1362 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 1363 virtual bool isValidSyncEvent(const sp<SyncEvent>& event); 1364 1365 static void syncStartEventCallback(const wp<SyncEvent>& event); 1366 void handleSyncStartEvent(const sp<SyncEvent>& event); 1367 1368 private: 1369 void clearSyncStartEvent(); 1370 1371 RecordThread(); 1372 AudioStreamIn *mInput; 1373 RecordTrack* mTrack; 1374 sp<RecordTrack> mActiveTrack; 1375 Condition mStartStopCond; 1376 AudioResampler *mResampler; 1377 int32_t *mRsmpOutBuffer; 1378 int16_t *mRsmpInBuffer; 1379 size_t mRsmpInIndex; 1380 size_t mInputBytes; 1381 const int mReqChannelCount; 1382 const uint32_t mReqSampleRate; 1383 ssize_t mBytesRead; 1384 // sync event triggering actual audio capture. Frames read before this event will 1385 // be dropped and therefore not read by the application. 1386 sp<SyncEvent> mSyncStartEvent; 1387 // number of captured frames to drop after the start sync event has been received. 1388 ssize_t mFramestoDrop; 1389 }; 1390 1391 // server side of the client's IAudioRecord 1392 class RecordHandle : public android::BnAudioRecord { 1393 public: 1394 RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack); 1395 virtual ~RecordHandle(); 1396 virtual sp<IMemory> getCblk() const; 1397 virtual status_t start(int event, int triggerSession); 1398 virtual void stop(); 1399 virtual status_t onTransact( 1400 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 1401 private: 1402 const sp<RecordThread::RecordTrack> mRecordTrack; 1403 }; 1404 1405 //--- Audio Effect Management 1406 1407 // EffectModule and EffectChain classes both have their own mutex to protect 1408 // state changes or resource modifications. Always respect the following order 1409 // if multiple mutexes must be acquired to avoid cross deadlock: 1410 // AudioFlinger -> ThreadBase -> EffectChain -> EffectModule 1411 1412 // The EffectModule class is a wrapper object controlling the effect engine implementation 1413 // in the effect library. It prevents concurrent calls to process() and command() functions 1414 // from different client threads. It keeps a list of EffectHandle objects corresponding 1415 // to all client applications using this effect and notifies applications of effect state, 1416 // control or parameter changes. It manages the activation state machine to send appropriate 1417 // reset, enable, disable commands to effect engine and provide volume 1418 // ramping when effects are activated/deactivated. 1419 // When controlling an auxiliary effect, the EffectModule also provides an input buffer used by 1420 // the attached track(s) to accumulate their auxiliary channel. 1421 class EffectModule: public RefBase { 1422 public: 1423 EffectModule(ThreadBase *thread, 1424 const wp<AudioFlinger::EffectChain>& chain, 1425 effect_descriptor_t *desc, 1426 int id, 1427 int sessionId); 1428 virtual ~EffectModule(); 1429 1430 enum effect_state { 1431 IDLE, 1432 RESTART, 1433 STARTING, 1434 ACTIVE, 1435 STOPPING, 1436 STOPPED, 1437 DESTROYED 1438 }; 1439 1440 int id() const { return mId; } 1441 void process(); 1442 void updateState(); 1443 status_t command(uint32_t cmdCode, 1444 uint32_t cmdSize, 1445 void *pCmdData, 1446 uint32_t *replySize, 1447 void *pReplyData); 1448 1449 void reset_l(); 1450 status_t configure(); 1451 status_t init(); 1452 effect_state state() const { 1453 return mState; 1454 } 1455 uint32_t status() { 1456 return mStatus; 1457 } 1458 int sessionId() const { 1459 return mSessionId; 1460 } 1461 status_t setEnabled(bool enabled); 1462 bool isEnabled() const; 1463 bool isProcessEnabled() const; 1464 1465 void setInBuffer(int16_t *buffer) { mConfig.inputCfg.buffer.s16 = buffer; } 1466 int16_t *inBuffer() { return mConfig.inputCfg.buffer.s16; } 1467 void setOutBuffer(int16_t *buffer) { mConfig.outputCfg.buffer.s16 = buffer; } 1468 int16_t *outBuffer() { return mConfig.outputCfg.buffer.s16; } 1469 void setChain(const wp<EffectChain>& chain) { mChain = chain; } 1470 void setThread(const wp<ThreadBase>& thread) { mThread = thread; } 1471 const wp<ThreadBase>& thread() { return mThread; } 1472 1473 status_t addHandle(const sp<EffectHandle>& handle); 1474 void disconnect(const wp<EffectHandle>& handle, bool unpinIfLast); 1475 size_t removeHandle (const wp<EffectHandle>& handle); 1476 1477 effect_descriptor_t& desc() { return mDescriptor; } 1478 wp<EffectChain>& chain() { return mChain; } 1479 1480 status_t setDevice(uint32_t device); 1481 status_t setVolume(uint32_t *left, uint32_t *right, bool controller); 1482 status_t setMode(audio_mode_t mode); 1483 status_t start(); 1484 status_t stop(); 1485 void setSuspended(bool suspended); 1486 bool suspended() const; 1487 1488 sp<EffectHandle> controlHandle(); 1489 1490 bool isPinned() const { return mPinned; } 1491 void unPin() { mPinned = false; } 1492 1493 status_t dump(int fd, const Vector<String16>& args); 1494 1495 protected: 1496 friend class AudioFlinger; // for mHandles 1497 bool mPinned; 1498 1499 // Maximum time allocated to effect engines to complete the turn off sequence 1500 static const uint32_t MAX_DISABLE_TIME_MS = 10000; 1501 1502 EffectModule(const EffectModule&); 1503 EffectModule& operator = (const EffectModule&); 1504 1505 status_t start_l(); 1506 status_t stop_l(); 1507 1508mutable Mutex mLock; // mutex for process, commands and handles list protection 1509 wp<ThreadBase> mThread; // parent thread 1510 wp<EffectChain> mChain; // parent effect chain 1511 int mId; // this instance unique ID 1512 int mSessionId; // audio session ID 1513 effect_descriptor_t mDescriptor;// effect descriptor received from effect engine 1514 effect_config_t mConfig; // input and output audio configuration 1515 effect_handle_t mEffectInterface; // Effect module C API 1516 status_t mStatus; // initialization status 1517 effect_state mState; // current activation state 1518 Vector< wp<EffectHandle> > mHandles; // list of client handles 1519 // First handle in mHandles has highest priority and controls the effect module 1520 uint32_t mMaxDisableWaitCnt; // maximum grace period before forcing an effect off after 1521 // sending disable command. 1522 uint32_t mDisableWaitCnt; // current process() calls count during disable period. 1523 bool mSuspended; // effect is suspended: temporarily disabled by framework 1524 }; 1525 1526 // The EffectHandle class implements the IEffect interface. It provides resources 1527 // to receive parameter updates, keeps track of effect control 1528 // ownership and state and has a pointer to the EffectModule object it is controlling. 1529 // There is one EffectHandle object for each application controlling (or using) 1530 // an effect module. 1531 // The EffectHandle is obtained by calling AudioFlinger::createEffect(). 1532 class EffectHandle: public android::BnEffect { 1533 public: 1534 1535 EffectHandle(const sp<EffectModule>& effect, 1536 const sp<AudioFlinger::Client>& client, 1537 const sp<IEffectClient>& effectClient, 1538 int32_t priority); 1539 virtual ~EffectHandle(); 1540 1541 // IEffect 1542 virtual status_t enable(); 1543 virtual status_t disable(); 1544 virtual status_t command(uint32_t cmdCode, 1545 uint32_t cmdSize, 1546 void *pCmdData, 1547 uint32_t *replySize, 1548 void *pReplyData); 1549 virtual void disconnect(); 1550 private: 1551 void disconnect(bool unpinIfLast); 1552 public: 1553 virtual sp<IMemory> getCblk() const { return mCblkMemory; } 1554 virtual status_t onTransact(uint32_t code, const Parcel& data, 1555 Parcel* reply, uint32_t flags); 1556 1557 1558 // Give or take control of effect module 1559 // - hasControl: true if control is given, false if removed 1560 // - signal: true client app should be signaled of change, false otherwise 1561 // - enabled: state of the effect when control is passed 1562 void setControl(bool hasControl, bool signal, bool enabled); 1563 void commandExecuted(uint32_t cmdCode, 1564 uint32_t cmdSize, 1565 void *pCmdData, 1566 uint32_t replySize, 1567 void *pReplyData); 1568 void setEnabled(bool enabled); 1569 bool enabled() const { return mEnabled; } 1570 1571 // Getters 1572 int id() const { return mEffect->id(); } 1573 int priority() const { return mPriority; } 1574 bool hasControl() const { return mHasControl; } 1575 sp<EffectModule> effect() const { return mEffect; } 1576 1577 void dump(char* buffer, size_t size); 1578 1579 protected: 1580 friend class AudioFlinger; // for mEffect, mHasControl, mEnabled 1581 EffectHandle(const EffectHandle&); 1582 EffectHandle& operator =(const EffectHandle&); 1583 1584 sp<EffectModule> mEffect; // pointer to controlled EffectModule 1585 sp<IEffectClient> mEffectClient; // callback interface for client notifications 1586 /*const*/ sp<Client> mClient; // client for shared memory allocation, see disconnect() 1587 sp<IMemory> mCblkMemory; // shared memory for control block 1588 effect_param_cblk_t* mCblk; // control block for deferred parameter setting via shared memory 1589 uint8_t* mBuffer; // pointer to parameter area in shared memory 1590 int mPriority; // client application priority to control the effect 1591 bool mHasControl; // true if this handle is controlling the effect 1592 bool mEnabled; // cached enable state: needed when the effect is 1593 // restored after being suspended 1594 }; 1595 1596 // the EffectChain class represents a group of effects associated to one audio session. 1597 // There can be any number of EffectChain objects per output mixer thread (PlaybackThread). 1598 // The EffecChain with session ID 0 contains global effects applied to the output mix. 1599 // Effects in this chain can be insert or auxiliary. Effects in other chains (attached to tracks) 1600 // are insert only. The EffectChain maintains an ordered list of effect module, the order corresponding 1601 // in the effect process order. When attached to a track (session ID != 0), it also provide it's own 1602 // input buffer used by the track as accumulation buffer. 1603 class EffectChain: public RefBase { 1604 public: 1605 EffectChain(const wp<ThreadBase>& wThread, int sessionId); 1606 EffectChain(ThreadBase *thread, int sessionId); 1607 virtual ~EffectChain(); 1608 1609 // special key used for an entry in mSuspendedEffects keyed vector 1610 // corresponding to a suspend all request. 1611 static const int kKeyForSuspendAll = 0; 1612 1613 // minimum duration during which we force calling effect process when last track on 1614 // a session is stopped or removed to allow effect tail to be rendered 1615 static const int kProcessTailDurationMs = 1000; 1616 1617 void process_l(); 1618 1619 void lock() { 1620 mLock.lock(); 1621 } 1622 void unlock() { 1623 mLock.unlock(); 1624 } 1625 1626 status_t addEffect_l(const sp<EffectModule>& handle); 1627 size_t removeEffect_l(const sp<EffectModule>& handle); 1628 1629 int sessionId() const { return mSessionId; } 1630 void setSessionId(int sessionId) { mSessionId = sessionId; } 1631 1632 sp<EffectModule> getEffectFromDesc_l(effect_descriptor_t *descriptor); 1633 sp<EffectModule> getEffectFromId_l(int id); 1634 sp<EffectModule> getEffectFromType_l(const effect_uuid_t *type); 1635 bool setVolume_l(uint32_t *left, uint32_t *right); 1636 void setDevice_l(uint32_t device); 1637 void setMode_l(audio_mode_t mode); 1638 1639 void setInBuffer(int16_t *buffer, bool ownsBuffer = false) { 1640 mInBuffer = buffer; 1641 mOwnInBuffer = ownsBuffer; 1642 } 1643 int16_t *inBuffer() const { 1644 return mInBuffer; 1645 } 1646 void setOutBuffer(int16_t *buffer) { 1647 mOutBuffer = buffer; 1648 } 1649 int16_t *outBuffer() const { 1650 return mOutBuffer; 1651 } 1652 1653 void incTrackCnt() { android_atomic_inc(&mTrackCnt); } 1654 void decTrackCnt() { android_atomic_dec(&mTrackCnt); } 1655 int32_t trackCnt() const { return mTrackCnt;} 1656 1657 void incActiveTrackCnt() { android_atomic_inc(&mActiveTrackCnt); 1658 mTailBufferCount = mMaxTailBuffers; } 1659 void decActiveTrackCnt() { android_atomic_dec(&mActiveTrackCnt); } 1660 int32_t activeTrackCnt() const { return mActiveTrackCnt;} 1661 1662 uint32_t strategy() const { return mStrategy; } 1663 void setStrategy(uint32_t strategy) 1664 { mStrategy = strategy; } 1665 1666 // suspend effect of the given type 1667 void setEffectSuspended_l(const effect_uuid_t *type, 1668 bool suspend); 1669 // suspend all eligible effects 1670 void setEffectSuspendedAll_l(bool suspend); 1671 // check if effects should be suspend or restored when a given effect is enable or disabled 1672 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1673 bool enabled); 1674 1675 status_t dump(int fd, const Vector<String16>& args); 1676 1677 protected: 1678 friend class AudioFlinger; // for mThread, mEffects 1679 EffectChain(const EffectChain&); 1680 EffectChain& operator =(const EffectChain&); 1681 1682 class SuspendedEffectDesc : public RefBase { 1683 public: 1684 SuspendedEffectDesc() : mRefCount(0) {} 1685 1686 int mRefCount; 1687 effect_uuid_t mType; 1688 wp<EffectModule> mEffect; 1689 }; 1690 1691 // get a list of effect modules to suspend when an effect of the type 1692 // passed is enabled. 1693 void getSuspendEligibleEffects(Vector< sp<EffectModule> > &effects); 1694 1695 // get an effect module if it is currently enable 1696 sp<EffectModule> getEffectIfEnabled(const effect_uuid_t *type); 1697 // true if the effect whose descriptor is passed can be suspended 1698 // OEMs can modify the rules implemented in this method to exclude specific effect 1699 // types or implementations from the suspend/restore mechanism. 1700 bool isEffectEligibleForSuspend(const effect_descriptor_t& desc); 1701 1702 wp<ThreadBase> mThread; // parent mixer thread 1703 Mutex mLock; // mutex protecting effect list 1704 Vector< sp<EffectModule> > mEffects; // list of effect modules 1705 int mSessionId; // audio session ID 1706 int16_t *mInBuffer; // chain input buffer 1707 int16_t *mOutBuffer; // chain output buffer 1708 volatile int32_t mActiveTrackCnt; // number of active tracks connected 1709 volatile int32_t mTrackCnt; // number of tracks connected 1710 int32_t mTailBufferCount; // current effect tail buffer count 1711 int32_t mMaxTailBuffers; // maximum effect tail buffers 1712 bool mOwnInBuffer; // true if the chain owns its input buffer 1713 int mVolumeCtrlIdx; // index of insert effect having control over volume 1714 uint32_t mLeftVolume; // previous volume on left channel 1715 uint32_t mRightVolume; // previous volume on right channel 1716 uint32_t mNewLeftVolume; // new volume on left channel 1717 uint32_t mNewRightVolume; // new volume on right channel 1718 uint32_t mStrategy; // strategy for this effect chain 1719 // mSuspendedEffects lists all effects currently suspended in the chain. 1720 // Use effect type UUID timelow field as key. There is no real risk of identical 1721 // timeLow fields among effect type UUIDs. 1722 // Updated by updateSuspendedSessions_l() only. 1723 KeyedVector< int, sp<SuspendedEffectDesc> > mSuspendedEffects; 1724 }; 1725 1726 // AudioStreamOut and AudioStreamIn are immutable, so their fields are const. 1727 // For emphasis, we could also make all pointers to them be "const *", 1728 // but that would clutter the code unnecessarily. 1729 1730 struct AudioStreamOut { 1731 audio_hw_device_t* const hwDev; 1732 audio_stream_out_t* const stream; 1733 1734 AudioStreamOut(audio_hw_device_t *dev, audio_stream_out_t *out) : 1735 hwDev(dev), stream(out) {} 1736 }; 1737 1738 struct AudioStreamIn { 1739 audio_hw_device_t* const hwDev; 1740 audio_stream_in_t* const stream; 1741 1742 AudioStreamIn(audio_hw_device_t *dev, audio_stream_in_t *in) : 1743 hwDev(dev), stream(in) {} 1744 }; 1745 1746 // for mAudioSessionRefs only 1747 struct AudioSessionRef { 1748 AudioSessionRef(int sessionid, pid_t pid) : 1749 mSessionid(sessionid), mPid(pid), mCnt(1) {} 1750 const int mSessionid; 1751 const pid_t mPid; 1752 int mCnt; 1753 }; 1754 1755 enum master_volume_support { 1756 // MVS_NONE: 1757 // Audio HAL has no support for master volume, either setting or 1758 // getting. All master volume control must be implemented in SW by the 1759 // AudioFlinger mixing core. 1760 MVS_NONE, 1761 1762 // MVS_SETONLY: 1763 // Audio HAL has support for setting master volume, but not for getting 1764 // master volume (original HAL design did not include a getter). 1765 // AudioFlinger needs to keep track of the last set master volume in 1766 // addition to needing to set an initial, default, master volume at HAL 1767 // load time. 1768 MVS_SETONLY, 1769 1770 // MVS_FULL: 1771 // Audio HAL has support both for setting and getting master volume. 1772 // AudioFlinger should send all set and get master volume requests 1773 // directly to the HAL. 1774 MVS_FULL, 1775 }; 1776 1777 class AudioHwDevice { 1778 public: 1779 AudioHwDevice(const char *moduleName, audio_hw_device_t *hwDevice) : 1780 mModuleName(strdup(moduleName)), mHwDevice(hwDevice){} 1781 ~AudioHwDevice() { free((void *)mModuleName); } 1782 1783 const char *moduleName() const { return mModuleName; } 1784 audio_hw_device_t *hwDevice() const { return mHwDevice; } 1785 private: 1786 const char * const mModuleName; 1787 audio_hw_device_t * const mHwDevice; 1788 }; 1789 1790 mutable Mutex mLock; 1791 1792 DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client() 1793 1794 mutable Mutex mHardwareLock; 1795 // NOTE: If both mLock and mHardwareLock mutexes must be held, 1796 // always take mLock before mHardwareLock 1797 1798 // These two fields are immutable after onFirstRef(), so no lock needed to access 1799 audio_hw_device_t* mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL 1800 DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*> mAudioHwDevs; 1801 1802 // for dump, indicates which hardware operation is currently in progress (but not stream ops) 1803 enum hardware_call_state { 1804 AUDIO_HW_IDLE = 0, // no operation in progress 1805 AUDIO_HW_INIT, // init_check 1806 AUDIO_HW_OUTPUT_OPEN, // open_output_stream 1807 AUDIO_HW_OUTPUT_CLOSE, // unused 1808 AUDIO_HW_INPUT_OPEN, // unused 1809 AUDIO_HW_INPUT_CLOSE, // unused 1810 AUDIO_HW_STANDBY, // unused 1811 AUDIO_HW_SET_MASTER_VOLUME, // set_master_volume 1812 AUDIO_HW_GET_ROUTING, // unused 1813 AUDIO_HW_SET_ROUTING, // unused 1814 AUDIO_HW_GET_MODE, // unused 1815 AUDIO_HW_SET_MODE, // set_mode 1816 AUDIO_HW_GET_MIC_MUTE, // get_mic_mute 1817 AUDIO_HW_SET_MIC_MUTE, // set_mic_mute 1818 AUDIO_HW_SET_VOICE_VOLUME, // set_voice_volume 1819 AUDIO_HW_SET_PARAMETER, // set_parameters 1820 AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size 1821 AUDIO_HW_GET_MASTER_VOLUME, // get_master_volume 1822 AUDIO_HW_GET_PARAMETER, // get_parameters 1823 }; 1824 1825 mutable hardware_call_state mHardwareStatus; // for dump only 1826 1827 1828 DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads; 1829 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 1830 1831 // both are protected by mLock 1832 float mMasterVolume; 1833 float mMasterVolumeSW; 1834 master_volume_support mMasterVolumeSupportLvl; 1835 bool mMasterMute; 1836 1837 DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads; 1838 1839 DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients; 1840 volatile int32_t mNextUniqueId; // updated by android_atomic_inc 1841 audio_mode_t mMode; 1842 bool mBtNrecIsOff; 1843 1844 // protected by mLock 1845 Vector<AudioSessionRef*> mAudioSessionRefs; 1846 1847 float masterVolume_l() const; 1848 float masterVolumeSW_l() const { return mMasterVolumeSW; } 1849 bool masterMute_l() const { return mMasterMute; } 1850 audio_module_handle_t loadHwModule_l(const char *name); 1851 1852 Vector < sp<SyncEvent> > mPendingSyncEvents; // sync events awaiting for a session 1853 // to be created 1854 1855private: 1856 sp<Client> registerPid_l(pid_t pid); // always returns non-0 1857 1858}; 1859 1860 1861// ---------------------------------------------------------------------------- 1862 1863}; // namespace android 1864 1865#endif // ANDROID_AUDIO_FLINGER_H 1866