AudioFlinger.h revision 22167855ff9af7b13fda669ca27c67a037a7d585
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 "ExtendedAudioBufferProvider.h" 50#include "FastMixer.h" 51#include "NBAIO.h" 52#include "AudioWatchdog.h" 53 54#include <powermanager/IPowerManager.h> 55 56namespace android { 57 58class audio_track_cblk_t; 59class effect_param_cblk_t; 60class AudioMixer; 61class AudioBuffer; 62class AudioResampler; 63class FastMixer; 64 65// ---------------------------------------------------------------------------- 66 67// AudioFlinger has a hard-coded upper limit of 2 channels for capture and playback. 68// There is support for > 2 channel tracks down-mixed to 2 channel output via a down-mix effect. 69// Adding full support for > 2 channel capture or playback would require more than simply changing 70// this #define. There is an independent hard-coded upper limit in AudioMixer; 71// removing that AudioMixer limit would be necessary but insufficient to support > 2 channels. 72// The macro FCC_2 highlights some (but not all) places where there is are 2-channel assumptions. 73// Search also for "2", "left", "right", "[0]", "[1]", ">> 16", "<< 16", etc. 74#define FCC_2 2 // FCC_2 = Fixed Channel Count 2 75 76static const nsecs_t kDefaultStandbyTimeInNsecs = seconds(3); 77 78class AudioFlinger : 79 public BinderService<AudioFlinger>, 80 public BnAudioFlinger 81{ 82 friend class BinderService<AudioFlinger>; // for AudioFlinger() 83public: 84 static const char* getServiceName() { return "media.audio_flinger"; } 85 86 virtual status_t dump(int fd, const Vector<String16>& args); 87 88 // IAudioFlinger interface, in binder opcode order 89 virtual sp<IAudioTrack> createTrack( 90 pid_t pid, 91 audio_stream_type_t streamType, 92 uint32_t sampleRate, 93 audio_format_t format, 94 uint32_t channelMask, 95 int frameCount, 96 IAudioFlinger::track_flags_t flags, 97 const sp<IMemory>& sharedBuffer, 98 audio_io_handle_t output, 99 pid_t tid, 100 int *sessionId, 101 status_t *status); 102 103 virtual sp<IAudioRecord> openRecord( 104 pid_t pid, 105 audio_io_handle_t input, 106 uint32_t sampleRate, 107 audio_format_t format, 108 uint32_t channelMask, 109 int frameCount, 110 IAudioFlinger::track_flags_t flags, 111 int *sessionId, 112 status_t *status); 113 114 virtual uint32_t sampleRate(audio_io_handle_t output) const; 115 virtual int channelCount(audio_io_handle_t output) const; 116 virtual audio_format_t format(audio_io_handle_t output) const; 117 virtual size_t frameCount(audio_io_handle_t output) const; 118 virtual uint32_t latency(audio_io_handle_t output) const; 119 120 virtual status_t setMasterVolume(float value); 121 virtual status_t setMasterMute(bool muted); 122 123 virtual float masterVolume() const; 124 virtual float masterVolumeSW() const; 125 virtual bool masterMute() const; 126 127 virtual status_t setStreamVolume(audio_stream_type_t stream, float value, 128 audio_io_handle_t output); 129 virtual status_t setStreamMute(audio_stream_type_t stream, bool muted); 130 131 virtual float streamVolume(audio_stream_type_t stream, 132 audio_io_handle_t output) const; 133 virtual bool streamMute(audio_stream_type_t stream) const; 134 135 virtual status_t setMode(audio_mode_t mode); 136 137 virtual status_t setMicMute(bool state); 138 virtual bool getMicMute() const; 139 140 virtual status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs); 141 virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) const; 142 143 virtual void registerClient(const sp<IAudioFlingerClient>& client); 144 145 virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const; 146 147 virtual audio_io_handle_t openOutput(audio_module_handle_t module, 148 audio_devices_t *pDevices, 149 uint32_t *pSamplingRate, 150 audio_format_t *pFormat, 151 audio_channel_mask_t *pChannelMask, 152 uint32_t *pLatencyMs, 153 audio_output_flags_t flags); 154 155 virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1, 156 audio_io_handle_t output2); 157 158 virtual status_t closeOutput(audio_io_handle_t output); 159 160 virtual status_t suspendOutput(audio_io_handle_t output); 161 162 virtual status_t restoreOutput(audio_io_handle_t output); 163 164 virtual audio_io_handle_t openInput(audio_module_handle_t module, 165 audio_devices_t *pDevices, 166 uint32_t *pSamplingRate, 167 audio_format_t *pFormat, 168 audio_channel_mask_t *pChannelMask); 169 170 virtual status_t closeInput(audio_io_handle_t input); 171 172 virtual status_t setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output); 173 174 virtual status_t setVoiceVolume(float volume); 175 176 virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 177 audio_io_handle_t output) const; 178 179 virtual unsigned int getInputFramesLost(audio_io_handle_t ioHandle) const; 180 181 virtual int newAudioSessionId(); 182 183 virtual void acquireAudioSessionId(int audioSession); 184 185 virtual void releaseAudioSessionId(int audioSession); 186 187 virtual status_t queryNumberEffects(uint32_t *numEffects) const; 188 189 virtual status_t queryEffect(uint32_t index, effect_descriptor_t *descriptor) const; 190 191 virtual status_t getEffectDescriptor(const effect_uuid_t *pUuid, 192 effect_descriptor_t *descriptor) const; 193 194 virtual sp<IEffect> createEffect(pid_t pid, 195 effect_descriptor_t *pDesc, 196 const sp<IEffectClient>& effectClient, 197 int32_t priority, 198 audio_io_handle_t io, 199 int sessionId, 200 status_t *status, 201 int *id, 202 int *enabled); 203 204 virtual status_t moveEffects(int sessionId, audio_io_handle_t srcOutput, 205 audio_io_handle_t dstOutput); 206 207 virtual audio_module_handle_t loadHwModule(const char *name); 208 209 virtual status_t onTransact( 210 uint32_t code, 211 const Parcel& data, 212 Parcel* reply, 213 uint32_t flags); 214 215 // end of IAudioFlinger interface 216 217 class SyncEvent; 218 219 typedef void (*sync_event_callback_t)(const wp<SyncEvent>& event) ; 220 221 class SyncEvent : public RefBase { 222 public: 223 SyncEvent(AudioSystem::sync_event_t type, 224 int triggerSession, 225 int listenerSession, 226 sync_event_callback_t callBack, 227 void *cookie) 228 : mType(type), mTriggerSession(triggerSession), mListenerSession(listenerSession), 229 mCallback(callBack), mCookie(cookie) 230 {} 231 232 virtual ~SyncEvent() {} 233 234 void trigger() { Mutex::Autolock _l(mLock); if (mCallback) mCallback(this); } 235 bool isCancelled() { Mutex::Autolock _l(mLock); return (mCallback == NULL); } 236 void cancel() {Mutex::Autolock _l(mLock); mCallback = NULL; } 237 AudioSystem::sync_event_t type() const { return mType; } 238 int triggerSession() const { return mTriggerSession; } 239 int listenerSession() const { return mListenerSession; } 240 void *cookie() const { return mCookie; } 241 242 private: 243 const AudioSystem::sync_event_t mType; 244 const int mTriggerSession; 245 const int mListenerSession; 246 sync_event_callback_t mCallback; 247 void * const mCookie; 248 Mutex mLock; 249 }; 250 251 sp<SyncEvent> createSyncEvent(AudioSystem::sync_event_t type, 252 int triggerSession, 253 int listenerSession, 254 sync_event_callback_t callBack, 255 void *cookie); 256private: 257 audio_mode_t getMode() const { return mMode; } 258 259 bool btNrecIsOff() const { return mBtNrecIsOff; } 260 261 AudioFlinger(); 262 virtual ~AudioFlinger(); 263 264 // call in any IAudioFlinger method that accesses mPrimaryHardwareDev 265 status_t initCheck() const { return mPrimaryHardwareDev == NULL ? NO_INIT : NO_ERROR; } 266 267 // RefBase 268 virtual void onFirstRef(); 269 270 audio_hw_device_t* findSuitableHwDev_l(audio_module_handle_t module, uint32_t devices); 271 void purgeStaleEffects_l(); 272 273 // standby delay for MIXER and DUPLICATING playback threads is read from property 274 // ro.audio.flinger_standbytime_ms or defaults to kDefaultStandbyTimeInNsecs 275 static nsecs_t mStandbyTimeInNsecs; 276 277 // Internal dump utilites. 278 status_t dumpPermissionDenial(int fd, const Vector<String16>& args); 279 status_t dumpClients(int fd, const Vector<String16>& args); 280 status_t dumpInternals(int fd, const Vector<String16>& args); 281 282 // --- Client --- 283 class Client : public RefBase { 284 public: 285 Client(const sp<AudioFlinger>& audioFlinger, pid_t pid); 286 virtual ~Client(); 287 sp<MemoryDealer> heap() const; 288 pid_t pid() const { return mPid; } 289 sp<AudioFlinger> audioFlinger() const { return mAudioFlinger; } 290 291 bool reserveTimedTrack(); 292 void releaseTimedTrack(); 293 294 private: 295 Client(const Client&); 296 Client& operator = (const Client&); 297 const sp<AudioFlinger> mAudioFlinger; 298 const sp<MemoryDealer> mMemoryDealer; 299 const pid_t mPid; 300 301 Mutex mTimedTrackLock; 302 int mTimedTrackCount; 303 }; 304 305 // --- Notification Client --- 306 class NotificationClient : public IBinder::DeathRecipient { 307 public: 308 NotificationClient(const sp<AudioFlinger>& audioFlinger, 309 const sp<IAudioFlingerClient>& client, 310 pid_t pid); 311 virtual ~NotificationClient(); 312 313 sp<IAudioFlingerClient> audioFlingerClient() const { return mAudioFlingerClient; } 314 315 // IBinder::DeathRecipient 316 virtual void binderDied(const wp<IBinder>& who); 317 318 private: 319 NotificationClient(const NotificationClient&); 320 NotificationClient& operator = (const NotificationClient&); 321 322 const sp<AudioFlinger> mAudioFlinger; 323 const pid_t mPid; 324 const sp<IAudioFlingerClient> mAudioFlingerClient; 325 }; 326 327 class TrackHandle; 328 class RecordHandle; 329 class RecordThread; 330 class PlaybackThread; 331 class MixerThread; 332 class DirectOutputThread; 333 class DuplicatingThread; 334 class Track; 335 class RecordTrack; 336 class EffectModule; 337 class EffectHandle; 338 class EffectChain; 339 struct AudioStreamOut; 340 struct AudioStreamIn; 341 342 class ThreadBase : public Thread { 343 public: 344 345 enum type_t { 346 MIXER, // Thread class is MixerThread 347 DIRECT, // Thread class is DirectOutputThread 348 DUPLICATING, // Thread class is DuplicatingThread 349 RECORD // Thread class is RecordThread 350 }; 351 352 ThreadBase (const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, uint32_t device, type_t type); 353 virtual ~ThreadBase(); 354 355 status_t dumpBase(int fd, const Vector<String16>& args); 356 status_t dumpEffectChains(int fd, const Vector<String16>& args); 357 358 void clearPowerManager(); 359 360 // base for record and playback 361 class TrackBase : public ExtendedAudioBufferProvider, public RefBase { 362 363 public: 364 enum track_state { 365 IDLE, 366 TERMINATED, 367 FLUSHED, 368 STOPPED, 369 // next 2 states are currently used for fast tracks only 370 STOPPING_1, // waiting for first underrun 371 STOPPING_2, // waiting for presentation complete 372 RESUMING, 373 ACTIVE, 374 PAUSING, 375 PAUSED 376 }; 377 378 TrackBase(ThreadBase *thread, 379 const sp<Client>& client, 380 uint32_t sampleRate, 381 audio_format_t format, 382 uint32_t channelMask, 383 int frameCount, 384 const sp<IMemory>& sharedBuffer, 385 int sessionId); 386 virtual ~TrackBase(); 387 388 virtual status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 389 int triggerSession = 0) = 0; 390 virtual void stop() = 0; 391 sp<IMemory> getCblk() const { return mCblkMemory; } 392 audio_track_cblk_t* cblk() const { return mCblk; } 393 int sessionId() const { return mSessionId; } 394 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 395 396 protected: 397 TrackBase(const TrackBase&); 398 TrackBase& operator = (const TrackBase&); 399 400 // AudioBufferProvider interface 401 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) = 0; 402 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 403 404 // ExtendedAudioBufferProvider interface is only needed for Track, 405 // but putting it in TrackBase avoids the complexity of virtual inheritance 406 virtual size_t framesReady() const { return SIZE_MAX; } 407 408 audio_format_t format() const { 409 return mFormat; 410 } 411 412 int channelCount() const { return mChannelCount; } 413 414 uint32_t channelMask() const { return mChannelMask; } 415 416 int sampleRate() const; // FIXME inline after cblk sr moved 417 418 void* getBuffer(uint32_t offset, uint32_t frames) const; 419 420 bool isStopped() const { 421 return (mState == STOPPED || mState == FLUSHED); 422 } 423 424 // for fast tracks only 425 bool isStopping() const { 426 return mState == STOPPING_1 || mState == STOPPING_2; 427 } 428 bool isStopping_1() const { 429 return mState == STOPPING_1; 430 } 431 bool isStopping_2() const { 432 return mState == STOPPING_2; 433 } 434 435 bool isTerminated() const { 436 return mState == TERMINATED; 437 } 438 439 bool step(); 440 void reset(); 441 442 const wp<ThreadBase> mThread; 443 /*const*/ sp<Client> mClient; // see explanation at ~TrackBase() why not const 444 sp<IMemory> mCblkMemory; 445 audio_track_cblk_t* mCblk; 446 void* mBuffer; 447 void* mBufferEnd; 448 uint32_t mFrameCount; 449 // we don't really need a lock for these 450 track_state mState; 451 const uint32_t mSampleRate; // initial sample rate only; for tracks which 452 // support dynamic rates, the current value is in control block 453 const audio_format_t mFormat; 454 bool mStepServerFailed; 455 const int mSessionId; 456 uint8_t mChannelCount; 457 uint32_t mChannelMask; 458 Vector < sp<SyncEvent> >mSyncEvents; 459 }; 460 461 class ConfigEvent { 462 public: 463 ConfigEvent() : mEvent(0), mParam(0) {} 464 465 int mEvent; 466 int mParam; 467 }; 468 469 class PMDeathRecipient : public IBinder::DeathRecipient { 470 public: 471 PMDeathRecipient(const wp<ThreadBase>& thread) : mThread(thread) {} 472 virtual ~PMDeathRecipient() {} 473 474 // IBinder::DeathRecipient 475 virtual void binderDied(const wp<IBinder>& who); 476 477 private: 478 PMDeathRecipient(const PMDeathRecipient&); 479 PMDeathRecipient& operator = (const PMDeathRecipient&); 480 481 wp<ThreadBase> mThread; 482 }; 483 484 virtual status_t initCheck() const = 0; 485 type_t type() const { return mType; } 486 uint32_t sampleRate() const { return mSampleRate; } 487 int channelCount() const { return mChannelCount; } 488 audio_format_t format() const { return mFormat; } 489 // Called by AudioFlinger::frameCount(audio_io_handle_t output) and effects, 490 // and returns the normal mix buffer's frame count. No API for HAL frame count. 491 size_t frameCount() const { return mNormalFrameCount; } 492 void wakeUp() { mWaitWorkCV.broadcast(); } 493 // Should be "virtual status_t requestExitAndWait()" and override same 494 // method in Thread, but Thread::requestExitAndWait() is not yet virtual. 495 void exit(); 496 virtual bool checkForNewParameters_l() = 0; 497 virtual status_t setParameters(const String8& keyValuePairs); 498 virtual String8 getParameters(const String8& keys) = 0; 499 virtual void audioConfigChanged_l(int event, int param = 0) = 0; 500 void sendConfigEvent(int event, int param = 0); 501 void sendConfigEvent_l(int event, int param = 0); 502 void processConfigEvents(); 503 audio_io_handle_t id() const { return mId;} 504 bool standby() const { return mStandby; } 505 uint32_t device() const { return mDevice; } 506 virtual audio_stream_t* stream() const = 0; 507 508 sp<EffectHandle> createEffect_l( 509 const sp<AudioFlinger::Client>& client, 510 const sp<IEffectClient>& effectClient, 511 int32_t priority, 512 int sessionId, 513 effect_descriptor_t *desc, 514 int *enabled, 515 status_t *status); 516 void disconnectEffect(const sp< EffectModule>& effect, 517 const wp<EffectHandle>& handle, 518 bool unpinIfLast); 519 520 // return values for hasAudioSession (bit field) 521 enum effect_state { 522 EFFECT_SESSION = 0x1, // the audio session corresponds to at least one 523 // effect 524 TRACK_SESSION = 0x2 // the audio session corresponds to at least one 525 // track 526 }; 527 528 // get effect chain corresponding to session Id. 529 sp<EffectChain> getEffectChain(int sessionId); 530 // same as getEffectChain() but must be called with ThreadBase mutex locked 531 sp<EffectChain> getEffectChain_l(int sessionId); 532 // add an effect chain to the chain list (mEffectChains) 533 virtual status_t addEffectChain_l(const sp<EffectChain>& chain) = 0; 534 // remove an effect chain from the chain list (mEffectChains) 535 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain) = 0; 536 // lock all effect chains Mutexes. Must be called before releasing the 537 // ThreadBase mutex before processing the mixer and effects. This guarantees the 538 // integrity of the chains during the process. 539 // Also sets the parameter 'effectChains' to current value of mEffectChains. 540 void lockEffectChains_l(Vector< sp<EffectChain> >& effectChains); 541 // unlock effect chains after process 542 void unlockEffectChains(const Vector< sp<EffectChain> >& effectChains); 543 // set audio mode to all effect chains 544 void setMode(audio_mode_t mode); 545 // get effect module with corresponding ID on specified audio session 546 sp<AudioFlinger::EffectModule> getEffect_l(int sessionId, int effectId); 547 // add and effect module. Also creates the effect chain is none exists for 548 // the effects audio session 549 status_t addEffect_l(const sp< EffectModule>& effect); 550 // remove and effect module. Also removes the effect chain is this was the last 551 // effect 552 void removeEffect_l(const sp< EffectModule>& effect); 553 // detach all tracks connected to an auxiliary effect 554 virtual void detachAuxEffect_l(int effectId) {} 555 // returns either EFFECT_SESSION if effects on this audio session exist in one 556 // chain, or TRACK_SESSION if tracks on this audio session exist, or both 557 virtual uint32_t hasAudioSession(int sessionId) = 0; 558 // the value returned by default implementation is not important as the 559 // strategy is only meaningful for PlaybackThread which implements this method 560 virtual uint32_t getStrategyForSession_l(int sessionId) { return 0; } 561 562 // suspend or restore effect according to the type of effect passed. a NULL 563 // type pointer means suspend all effects in the session 564 void setEffectSuspended(const effect_uuid_t *type, 565 bool suspend, 566 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 567 // check if some effects must be suspended/restored when an effect is enabled 568 // or disabled 569 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 570 bool enabled, 571 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 572 void checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 573 bool enabled, 574 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 575 576 virtual status_t setSyncEvent(const sp<SyncEvent>& event) = 0; 577 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) = 0; 578 579 580 mutable Mutex mLock; 581 582 protected: 583 584 // entry describing an effect being suspended in mSuspendedSessions keyed vector 585 class SuspendedSessionDesc : public RefBase { 586 public: 587 SuspendedSessionDesc() : mRefCount(0) {} 588 589 int mRefCount; // number of active suspend requests 590 effect_uuid_t mType; // effect type UUID 591 }; 592 593 void acquireWakeLock(); 594 void acquireWakeLock_l(); 595 void releaseWakeLock(); 596 void releaseWakeLock_l(); 597 void setEffectSuspended_l(const effect_uuid_t *type, 598 bool suspend, 599 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 600 // updated mSuspendedSessions when an effect suspended or restored 601 void updateSuspendedSessions_l(const effect_uuid_t *type, 602 bool suspend, 603 int sessionId); 604 // check if some effects must be suspended when an effect chain is added 605 void checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain); 606 607 friend class AudioFlinger; // for mEffectChains 608 609 const type_t mType; 610 611 // Used by parameters, config events, addTrack_l, exit 612 Condition mWaitWorkCV; 613 614 const sp<AudioFlinger> mAudioFlinger; 615 uint32_t mSampleRate; 616 size_t mFrameCount; // output HAL, direct output, record 617 size_t mNormalFrameCount; // normal mixer and effects 618 uint32_t mChannelMask; 619 uint16_t mChannelCount; 620 size_t mFrameSize; 621 audio_format_t mFormat; 622 623 // Parameter sequence by client: binder thread calling setParameters(): 624 // 1. Lock mLock 625 // 2. Append to mNewParameters 626 // 3. mWaitWorkCV.signal 627 // 4. mParamCond.waitRelative with timeout 628 // 5. read mParamStatus 629 // 6. mWaitWorkCV.signal 630 // 7. Unlock 631 // 632 // Parameter sequence by server: threadLoop calling checkForNewParameters_l(): 633 // 1. Lock mLock 634 // 2. If there is an entry in mNewParameters proceed ... 635 // 2. Read first entry in mNewParameters 636 // 3. Process 637 // 4. Remove first entry from mNewParameters 638 // 5. Set mParamStatus 639 // 6. mParamCond.signal 640 // 7. mWaitWorkCV.wait with timeout (this is to avoid overwriting mParamStatus) 641 // 8. Unlock 642 Condition mParamCond; 643 Vector<String8> mNewParameters; 644 status_t mParamStatus; 645 646 Vector<ConfigEvent> mConfigEvents; 647 bool mStandby; 648 const audio_io_handle_t mId; 649 Vector< sp<EffectChain> > mEffectChains; 650 uint32_t mDevice; // output device for PlaybackThread 651 // input + output devices for RecordThread 652 static const int kNameLength = 16; // prctl(PR_SET_NAME) limit 653 char mName[kNameLength]; 654 sp<IPowerManager> mPowerManager; 655 sp<IBinder> mWakeLockToken; 656 const sp<PMDeathRecipient> mDeathRecipient; 657 // list of suspended effects per session and per type. The first vector is 658 // keyed by session ID, the second by type UUID timeLow field 659 KeyedVector< int, KeyedVector< int, sp<SuspendedSessionDesc> > > mSuspendedSessions; 660 }; 661 662 struct stream_type_t { 663 stream_type_t() 664 : volume(1.0f), 665 mute(false) 666 { 667 } 668 float volume; 669 bool mute; 670 }; 671 672 // --- PlaybackThread --- 673 class PlaybackThread : public ThreadBase { 674 public: 675 676 enum mixer_state { 677 MIXER_IDLE, // no active tracks 678 MIXER_TRACKS_ENABLED, // at least one active track, but no track has any data ready 679 MIXER_TRACKS_READY // at least one active track, and at least one track has data 680 // standby mode does not have an enum value 681 // suspend by audio policy manager is orthogonal to mixer state 682 }; 683 684 // playback track 685 class Track : public TrackBase, public VolumeProvider { 686 public: 687 Track( PlaybackThread *thread, 688 const sp<Client>& client, 689 audio_stream_type_t streamType, 690 uint32_t sampleRate, 691 audio_format_t format, 692 uint32_t channelMask, 693 int frameCount, 694 const sp<IMemory>& sharedBuffer, 695 int sessionId, 696 IAudioFlinger::track_flags_t flags); 697 virtual ~Track(); 698 699 static void appendDumpHeader(String8& result); 700 void dump(char* buffer, size_t size); 701 virtual status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 702 int triggerSession = 0); 703 virtual void stop(); 704 void pause(); 705 706 void flush(); 707 void destroy(); 708 void mute(bool); 709 int name() const { 710 return mName; 711 } 712 713 audio_stream_type_t streamType() const { 714 return mStreamType; 715 } 716 status_t attachAuxEffect(int EffectId); 717 void setAuxBuffer(int EffectId, int32_t *buffer); 718 int32_t *auxBuffer() const { return mAuxBuffer; } 719 void setMainBuffer(int16_t *buffer) { mMainBuffer = buffer; } 720 int16_t *mainBuffer() const { return mMainBuffer; } 721 int auxEffectId() const { return mAuxEffectId; } 722 723 // implement FastMixerState::VolumeProvider interface 724 virtual uint32_t getVolumeLR(); 725 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 726 727 protected: 728 // for numerous 729 friend class PlaybackThread; 730 friend class MixerThread; 731 friend class DirectOutputThread; 732 733 Track(const Track&); 734 Track& operator = (const Track&); 735 736 // AudioBufferProvider interface 737 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts = kInvalidPTS); 738 // releaseBuffer() not overridden 739 740 virtual size_t framesReady() const; 741 742 bool isMuted() const { return mMute; } 743 bool isPausing() const { 744 return mState == PAUSING; 745 } 746 bool isPaused() const { 747 return mState == PAUSED; 748 } 749 bool isResuming() const { 750 return mState == RESUMING; 751 } 752 bool isReady() const; 753 void setPaused() { mState = PAUSED; } 754 void reset(); 755 756 bool isOutputTrack() const { 757 return (mStreamType == AUDIO_STREAM_CNT); 758 } 759 760 sp<IMemory> sharedBuffer() const { return mSharedBuffer; } 761 762 bool presentationComplete(size_t framesWritten, size_t audioHalFrames); 763 764 public: 765 void triggerEvents(AudioSystem::sync_event_t type); 766 virtual bool isTimedTrack() const { return false; } 767 bool isFastTrack() const { return (mFlags & IAudioFlinger::TRACK_FAST) != 0; } 768 protected: 769 770 // we don't really need a lock for these 771 volatile bool mMute; 772 // FILLED state is used for suppressing volume ramp at begin of playing 773 enum {FS_INVALID, FS_FILLING, FS_FILLED, FS_ACTIVE}; 774 mutable uint8_t mFillingUpStatus; 775 int8_t mRetryCount; 776 const sp<IMemory> mSharedBuffer; 777 bool mResetDone; 778 const audio_stream_type_t mStreamType; 779 int mName; // track name on the normal mixer, 780 // allocated statically at track creation time, 781 // and is even allocated (though unused) for fast tracks 782 // FIXME don't allocate track name for fast tracks 783 int16_t *mMainBuffer; 784 int32_t *mAuxBuffer; 785 int mAuxEffectId; 786 bool mHasVolumeController; 787 size_t mPresentationCompleteFrames; // number of frames written to the audio HAL 788 // when this track will be fully rendered 789 private: 790 IAudioFlinger::track_flags_t mFlags; 791 792 // The following fields are only for fast tracks, and should be in a subclass 793 int mFastIndex; // index within FastMixerState::mFastTracks[]; 794 // either mFastIndex == -1 if not isFastTrack() 795 // or 0 < mFastIndex < FastMixerState::kMaxFast because 796 // index 0 is reserved for normal mixer's submix; 797 // index is allocated statically at track creation time 798 // but the slot is only used if track is active 799 FastTrackUnderruns mObservedUnderruns; // Most recently observed value of 800 // mFastMixerDumpState.mTracks[mFastIndex].mUnderruns 801 uint32_t mUnderrunCount; // Counter of total number of underruns, never reset 802 volatile float mCachedVolume; // combined master volume and stream type volume; 803 // 'volatile' means accessed without lock or 804 // barrier, but is read/written atomically 805 }; // end of Track 806 807 class TimedTrack : public Track { 808 public: 809 static sp<TimedTrack> create(PlaybackThread *thread, 810 const sp<Client>& client, 811 audio_stream_type_t streamType, 812 uint32_t sampleRate, 813 audio_format_t format, 814 uint32_t channelMask, 815 int frameCount, 816 const sp<IMemory>& sharedBuffer, 817 int sessionId); 818 ~TimedTrack(); 819 820 class TimedBuffer { 821 public: 822 TimedBuffer(); 823 TimedBuffer(const sp<IMemory>& buffer, int64_t pts); 824 const sp<IMemory>& buffer() const { return mBuffer; } 825 int64_t pts() const { return mPTS; } 826 uint32_t position() const { return mPosition; } 827 void setPosition(uint32_t pos) { mPosition = pos; } 828 private: 829 sp<IMemory> mBuffer; 830 int64_t mPTS; 831 uint32_t mPosition; 832 }; 833 834 // Mixer facing methods. 835 virtual bool isTimedTrack() const { return true; } 836 virtual size_t framesReady() const; 837 838 // AudioBufferProvider interface 839 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, 840 int64_t pts); 841 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 842 843 // Client/App facing methods. 844 status_t allocateTimedBuffer(size_t size, 845 sp<IMemory>* buffer); 846 status_t queueTimedBuffer(const sp<IMemory>& buffer, 847 int64_t pts); 848 status_t setMediaTimeTransform(const LinearTransform& xform, 849 TimedAudioTrack::TargetTimeline target); 850 851 private: 852 TimedTrack(PlaybackThread *thread, 853 const sp<Client>& client, 854 audio_stream_type_t streamType, 855 uint32_t sampleRate, 856 audio_format_t format, 857 uint32_t channelMask, 858 int frameCount, 859 const sp<IMemory>& sharedBuffer, 860 int sessionId); 861 862 void timedYieldSamples_l(AudioBufferProvider::Buffer* buffer); 863 void timedYieldSilence_l(uint32_t numFrames, 864 AudioBufferProvider::Buffer* buffer); 865 void trimTimedBufferQueue_l(); 866 void trimTimedBufferQueueHead_l(const char* logTag); 867 void updateFramesPendingAfterTrim_l(const TimedBuffer& buf, 868 const char* logTag); 869 870 uint64_t mLocalTimeFreq; 871 LinearTransform mLocalTimeToSampleTransform; 872 LinearTransform mMediaTimeToSampleTransform; 873 sp<MemoryDealer> mTimedMemoryDealer; 874 875 Vector<TimedBuffer> mTimedBufferQueue; 876 bool mQueueHeadInFlight; 877 bool mTrimQueueHeadOnRelease; 878 uint32_t mFramesPendingInQueue; 879 880 uint8_t* mTimedSilenceBuffer; 881 uint32_t mTimedSilenceBufferSize; 882 mutable Mutex mTimedBufferQueueLock; 883 bool mTimedAudioOutputOnTime; 884 CCHelper mCCHelper; 885 886 Mutex mMediaTimeTransformLock; 887 LinearTransform mMediaTimeTransform; 888 bool mMediaTimeTransformValid; 889 TimedAudioTrack::TargetTimeline mMediaTimeTransformTarget; 890 }; 891 892 893 // playback track 894 class OutputTrack : public Track { 895 public: 896 897 class Buffer: public AudioBufferProvider::Buffer { 898 public: 899 int16_t *mBuffer; 900 }; 901 902 OutputTrack(PlaybackThread *thread, 903 DuplicatingThread *sourceThread, 904 uint32_t sampleRate, 905 audio_format_t format, 906 uint32_t channelMask, 907 int frameCount); 908 virtual ~OutputTrack(); 909 910 virtual status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 911 int triggerSession = 0); 912 virtual void stop(); 913 bool write(int16_t* data, uint32_t frames); 914 bool bufferQueueEmpty() const { return mBufferQueue.size() == 0; } 915 bool isActive() const { return mActive; } 916 const wp<ThreadBase>& thread() const { return mThread; } 917 918 private: 919 920 enum { 921 NO_MORE_BUFFERS = 0x80000001, // same in AudioTrack.h, ok to be different value 922 }; 923 924 status_t obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs); 925 void clearBufferQueue(); 926 927 // Maximum number of pending buffers allocated by OutputTrack::write() 928 static const uint8_t kMaxOverFlowBuffers = 10; 929 930 Vector < Buffer* > mBufferQueue; 931 AudioBufferProvider::Buffer mOutBuffer; 932 bool mActive; 933 DuplicatingThread* const mSourceThread; // for waitTimeMs() in write() 934 }; // end of OutputTrack 935 936 PlaybackThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 937 audio_io_handle_t id, uint32_t device, type_t type); 938 virtual ~PlaybackThread(); 939 940 status_t dump(int fd, const Vector<String16>& args); 941 942 // Thread virtuals 943 virtual status_t readyToRun(); 944 virtual bool threadLoop(); 945 946 // RefBase 947 virtual void onFirstRef(); 948 949protected: 950 // Code snippets that were lifted up out of threadLoop() 951 virtual void threadLoop_mix() = 0; 952 virtual void threadLoop_sleepTime() = 0; 953 virtual void threadLoop_write(); 954 virtual void threadLoop_standby(); 955 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 956 957 // prepareTracks_l reads and writes mActiveTracks, and returns 958 // the pending set of tracks to remove via Vector 'tracksToRemove'. The caller 959 // is responsible for clearing or destroying this Vector later on, when it 960 // is safe to do so. That will drop the final ref count and destroy the tracks. 961 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove) = 0; 962 963public: 964 965 virtual status_t initCheck() const { return (mOutput == NULL) ? NO_INIT : NO_ERROR; } 966 967 // return estimated latency in milliseconds, as reported by HAL 968 uint32_t latency() const; 969 // same, but lock must already be held 970 uint32_t latency_l() const; 971 972 void setMasterVolume(float value); 973 void setMasterMute(bool muted); 974 975 void setStreamVolume(audio_stream_type_t stream, float value); 976 void setStreamMute(audio_stream_type_t stream, bool muted); 977 978 float streamVolume(audio_stream_type_t stream) const; 979 980 sp<Track> createTrack_l( 981 const sp<AudioFlinger::Client>& client, 982 audio_stream_type_t streamType, 983 uint32_t sampleRate, 984 audio_format_t format, 985 uint32_t channelMask, 986 int frameCount, 987 const sp<IMemory>& sharedBuffer, 988 int sessionId, 989 IAudioFlinger::track_flags_t flags, 990 pid_t tid, 991 status_t *status); 992 993 AudioStreamOut* getOutput() const; 994 AudioStreamOut* clearOutput(); 995 virtual audio_stream_t* stream() const; 996 997 void suspend() { mSuspended++; } 998 void restore() { if (mSuspended > 0) mSuspended--; } 999 bool isSuspended() const { return (mSuspended > 0); } 1000 virtual String8 getParameters(const String8& keys); 1001 virtual void audioConfigChanged_l(int event, int param = 0); 1002 status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames); 1003 int16_t *mixBuffer() const { return mMixBuffer; }; 1004 1005 virtual void detachAuxEffect_l(int effectId); 1006 status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track> track, 1007 int EffectId); 1008 status_t attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track> track, 1009 int EffectId); 1010 1011 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1012 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1013 virtual uint32_t hasAudioSession(int sessionId); 1014 virtual uint32_t getStrategyForSession_l(int sessionId); 1015 1016 1017 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 1018 virtual bool isValidSyncEvent(const sp<SyncEvent>& event); 1019 void invalidateTracks(audio_stream_type_t streamType); 1020 1021 1022 protected: 1023 int16_t* mMixBuffer; 1024 uint32_t mSuspended; // suspend count, > 0 means suspended 1025 int mBytesWritten; 1026 private: 1027 // mMasterMute is in both PlaybackThread and in AudioFlinger. When a 1028 // PlaybackThread needs to find out if master-muted, it checks it's local 1029 // copy rather than the one in AudioFlinger. This optimization saves a lock. 1030 bool mMasterMute; 1031 void setMasterMute_l(bool muted) { mMasterMute = muted; } 1032 protected: 1033 SortedVector< wp<Track> > mActiveTracks; // FIXME check if this could be sp<> 1034 1035 // Allocate a track name for a given channel mask. 1036 // Returns name >= 0 if successful, -1 on failure. 1037 virtual int getTrackName_l(audio_channel_mask_t channelMask) = 0; 1038 virtual void deleteTrackName_l(int name) = 0; 1039 1040 // Time to sleep between cycles when: 1041 virtual uint32_t activeSleepTimeUs() const; // mixer state MIXER_TRACKS_ENABLED 1042 virtual uint32_t idleSleepTimeUs() const = 0; // mixer state MIXER_IDLE 1043 virtual uint32_t suspendSleepTimeUs() const = 0; // audio policy manager suspended us 1044 // No sleep when mixer state == MIXER_TRACKS_READY; relies on audio HAL stream->write() 1045 // No sleep in standby mode; waits on a condition 1046 1047 // Code snippets that are temporarily lifted up out of threadLoop() until the merge 1048 void checkSilentMode_l(); 1049 1050 // Non-trivial for DUPLICATING only 1051 virtual void saveOutputTracks() { } 1052 virtual void clearOutputTracks() { } 1053 1054 // Cache various calculated values, at threadLoop() entry and after a parameter change 1055 virtual void cacheParameters_l(); 1056 1057 virtual uint32_t correctLatency(uint32_t latency) const; 1058 1059 private: 1060 1061 friend class AudioFlinger; // for numerous 1062 1063 PlaybackThread(const Client&); 1064 PlaybackThread& operator = (const PlaybackThread&); 1065 1066 status_t addTrack_l(const sp<Track>& track); 1067 void destroyTrack_l(const sp<Track>& track); 1068 void removeTrack_l(const sp<Track>& track); 1069 1070 void readOutputParameters(); 1071 1072 virtual status_t dumpInternals(int fd, const Vector<String16>& args); 1073 status_t dumpTracks(int fd, const Vector<String16>& args); 1074 1075 SortedVector< sp<Track> > mTracks; 1076 // mStreamTypes[] uses 1 additional stream type internally for the OutputTrack used by DuplicatingThread 1077 stream_type_t mStreamTypes[AUDIO_STREAM_CNT + 1]; 1078 AudioStreamOut *mOutput; 1079 float mMasterVolume; 1080 nsecs_t mLastWriteTime; 1081 int mNumWrites; 1082 int mNumDelayedWrites; 1083 bool mInWrite; 1084 1085 // FIXME rename these former local variables of threadLoop to standard "m" names 1086 nsecs_t standbyTime; 1087 size_t mixBufferSize; 1088 1089 // cached copies of activeSleepTimeUs() and idleSleepTimeUs() made by cacheParameters_l() 1090 uint32_t activeSleepTime; 1091 uint32_t idleSleepTime; 1092 1093 uint32_t sleepTime; 1094 1095 // mixer status returned by prepareTracks_l() 1096 mixer_state mMixerStatus; // current cycle 1097 // previous cycle when in prepareTracks_l() 1098 mixer_state mMixerStatusIgnoringFastTracks; 1099 // FIXME or a separate ready state per track 1100 1101 // FIXME move these declarations into the specific sub-class that needs them 1102 // MIXER only 1103 bool longStandbyExit; 1104 uint32_t sleepTimeShift; 1105 1106 // same as AudioFlinger::mStandbyTimeInNsecs except for DIRECT which uses a shorter value 1107 nsecs_t standbyDelay; 1108 1109 // MIXER only 1110 nsecs_t maxPeriod; 1111 1112 // DUPLICATING only 1113 uint32_t writeFrames; 1114 1115 private: 1116 // The HAL output sink is treated as non-blocking, but current implementation is blocking 1117 sp<NBAIO_Sink> mOutputSink; 1118 // If a fast mixer is present, the blocking pipe sink, otherwise clear 1119 sp<NBAIO_Sink> mPipeSink; 1120 // The current sink for the normal mixer to write it's (sub)mix, mOutputSink or mPipeSink 1121 sp<NBAIO_Sink> mNormalSink; 1122 // For dumpsys 1123 sp<NBAIO_Sink> mTeeSink; 1124 sp<NBAIO_Source> mTeeSource; 1125 uint32_t mScreenState; // cached copy of gScreenState 1126 public: 1127 virtual bool hasFastMixer() const = 0; 1128 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const 1129 { FastTrackUnderruns dummy; return dummy; } 1130 1131 protected: 1132 // accessed by both binder threads and within threadLoop(), lock on mutex needed 1133 unsigned mFastTrackAvailMask; // bit i set if fast track [i] is available 1134 1135 }; 1136 1137 class MixerThread : public PlaybackThread { 1138 public: 1139 MixerThread (const sp<AudioFlinger>& audioFlinger, 1140 AudioStreamOut* output, 1141 audio_io_handle_t id, 1142 uint32_t device, 1143 type_t type = MIXER); 1144 virtual ~MixerThread(); 1145 1146 // Thread virtuals 1147 1148 virtual bool checkForNewParameters_l(); 1149 virtual status_t dumpInternals(int fd, const Vector<String16>& args); 1150 1151 protected: 1152 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1153 virtual int getTrackName_l(audio_channel_mask_t channelMask); 1154 virtual void deleteTrackName_l(int name); 1155 virtual uint32_t idleSleepTimeUs() const; 1156 virtual uint32_t suspendSleepTimeUs() const; 1157 virtual void cacheParameters_l(); 1158 1159 // threadLoop snippets 1160 virtual void threadLoop_write(); 1161 virtual void threadLoop_standby(); 1162 virtual void threadLoop_mix(); 1163 virtual void threadLoop_sleepTime(); 1164 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 1165 virtual uint32_t correctLatency(uint32_t latency) const; 1166 1167 AudioMixer* mAudioMixer; // normal mixer 1168 private: 1169#ifdef SOAKER 1170 Thread* mSoaker; 1171#endif 1172 // one-time initialization, no locks required 1173 FastMixer* mFastMixer; // non-NULL if there is also a fast mixer 1174 sp<AudioWatchdog> mAudioWatchdog; // non-0 if there is an audio watchdog thread 1175 1176 // contents are not guaranteed to be consistent, no locks required 1177 FastMixerDumpState mFastMixerDumpState; 1178#ifdef STATE_QUEUE_DUMP 1179 StateQueueObserverDump mStateQueueObserverDump; 1180 StateQueueMutatorDump mStateQueueMutatorDump; 1181#endif 1182 AudioWatchdogDump mAudioWatchdogDump; 1183 1184 // accessible only within the threadLoop(), no locks required 1185 // mFastMixer->sq() // for mutating and pushing state 1186 int32_t mFastMixerFutex; // for cold idle 1187 1188 public: 1189 virtual bool hasFastMixer() const { return mFastMixer != NULL; } 1190 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const { 1191 ALOG_ASSERT(fastIndex < FastMixerState::kMaxFastTracks); 1192 return mFastMixerDumpState.mTracks[fastIndex].mUnderruns; 1193 } 1194 }; 1195 1196 class DirectOutputThread : public PlaybackThread { 1197 public: 1198 1199 DirectOutputThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1200 audio_io_handle_t id, uint32_t device); 1201 virtual ~DirectOutputThread(); 1202 1203 // Thread virtuals 1204 1205 virtual bool checkForNewParameters_l(); 1206 1207 protected: 1208 virtual int getTrackName_l(audio_channel_mask_t channelMask); 1209 virtual void deleteTrackName_l(int name); 1210 virtual uint32_t activeSleepTimeUs() const; 1211 virtual uint32_t idleSleepTimeUs() const; 1212 virtual uint32_t suspendSleepTimeUs() const; 1213 virtual void cacheParameters_l(); 1214 1215 // threadLoop snippets 1216 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1217 virtual void threadLoop_mix(); 1218 virtual void threadLoop_sleepTime(); 1219 1220 // volumes last sent to audio HAL with stream->set_volume() 1221 float mLeftVolFloat; 1222 float mRightVolFloat; 1223 1224private: 1225 // prepareTracks_l() tells threadLoop_mix() the name of the single active track 1226 sp<Track> mActiveTrack; 1227 public: 1228 virtual bool hasFastMixer() const { return false; } 1229 }; 1230 1231 class DuplicatingThread : public MixerThread { 1232 public: 1233 DuplicatingThread (const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread, 1234 audio_io_handle_t id); 1235 virtual ~DuplicatingThread(); 1236 1237 // Thread virtuals 1238 void addOutputTrack(MixerThread* thread); 1239 void removeOutputTrack(MixerThread* thread); 1240 uint32_t waitTimeMs() const { return mWaitTimeMs; } 1241 protected: 1242 virtual uint32_t activeSleepTimeUs() const; 1243 1244 private: 1245 bool outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks); 1246 protected: 1247 // threadLoop snippets 1248 virtual void threadLoop_mix(); 1249 virtual void threadLoop_sleepTime(); 1250 virtual void threadLoop_write(); 1251 virtual void threadLoop_standby(); 1252 virtual void cacheParameters_l(); 1253 1254 private: 1255 // called from threadLoop, addOutputTrack, removeOutputTrack 1256 virtual void updateWaitTime_l(); 1257 protected: 1258 virtual void saveOutputTracks(); 1259 virtual void clearOutputTracks(); 1260 private: 1261 1262 uint32_t mWaitTimeMs; 1263 SortedVector < sp<OutputTrack> > outputTracks; 1264 SortedVector < sp<OutputTrack> > mOutputTracks; 1265 public: 1266 virtual bool hasFastMixer() const { return false; } 1267 }; 1268 1269 PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const; 1270 MixerThread *checkMixerThread_l(audio_io_handle_t output) const; 1271 RecordThread *checkRecordThread_l(audio_io_handle_t input) const; 1272 // no range check, AudioFlinger::mLock held 1273 bool streamMute_l(audio_stream_type_t stream) const 1274 { return mStreamTypes[stream].mute; } 1275 // no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held 1276 float streamVolume_l(audio_stream_type_t stream) const 1277 { return mStreamTypes[stream].volume; } 1278 void audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2); 1279 1280 // allocate an audio_io_handle_t, session ID, or effect ID 1281 uint32_t nextUniqueId(); 1282 1283 status_t moveEffectChain_l(int sessionId, 1284 PlaybackThread *srcThread, 1285 PlaybackThread *dstThread, 1286 bool reRegister); 1287 // return thread associated with primary hardware device, or NULL 1288 PlaybackThread *primaryPlaybackThread_l() const; 1289 uint32_t primaryOutputDevice_l() const; 1290 1291 // server side of the client's IAudioTrack 1292 class TrackHandle : public android::BnAudioTrack { 1293 public: 1294 TrackHandle(const sp<PlaybackThread::Track>& track); 1295 virtual ~TrackHandle(); 1296 virtual sp<IMemory> getCblk() const; 1297 virtual status_t start(); 1298 virtual void stop(); 1299 virtual void flush(); 1300 virtual void mute(bool); 1301 virtual void pause(); 1302 virtual status_t attachAuxEffect(int effectId); 1303 virtual status_t allocateTimedBuffer(size_t size, 1304 sp<IMemory>* buffer); 1305 virtual status_t queueTimedBuffer(const sp<IMemory>& buffer, 1306 int64_t pts); 1307 virtual status_t setMediaTimeTransform(const LinearTransform& xform, 1308 int target); 1309 virtual status_t onTransact( 1310 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 1311 private: 1312 const sp<PlaybackThread::Track> mTrack; 1313 }; 1314 1315 void removeClient_l(pid_t pid); 1316 void removeNotificationClient(pid_t pid); 1317 1318 1319 // record thread 1320 class RecordThread : public ThreadBase, public AudioBufferProvider 1321 { 1322 public: 1323 1324 // record track 1325 class RecordTrack : public TrackBase { 1326 public: 1327 RecordTrack(RecordThread *thread, 1328 const sp<Client>& client, 1329 uint32_t sampleRate, 1330 audio_format_t format, 1331 uint32_t channelMask, 1332 int frameCount, 1333 int sessionId); 1334 virtual ~RecordTrack(); 1335 1336 virtual status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 1337 int triggerSession = 0); 1338 virtual void stop(); 1339 1340 bool overflow() { bool tmp = mOverflow; mOverflow = false; return tmp; } 1341 bool setOverflow() { bool tmp = mOverflow; mOverflow = true; return tmp; } 1342 1343 void dump(char* buffer, size_t size); 1344 1345 private: 1346 friend class AudioFlinger; // for mState 1347 1348 RecordTrack(const RecordTrack&); 1349 RecordTrack& operator = (const RecordTrack&); 1350 1351 // AudioBufferProvider interface 1352 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts = kInvalidPTS); 1353 // releaseBuffer() not overridden 1354 1355 bool mOverflow; 1356 }; 1357 1358 1359 RecordThread(const sp<AudioFlinger>& audioFlinger, 1360 AudioStreamIn *input, 1361 uint32_t sampleRate, 1362 uint32_t channels, 1363 audio_io_handle_t id, 1364 uint32_t device); 1365 virtual ~RecordThread(); 1366 1367 // Thread 1368 virtual bool threadLoop(); 1369 virtual status_t readyToRun(); 1370 1371 // RefBase 1372 virtual void onFirstRef(); 1373 1374 virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; } 1375 sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l( 1376 const sp<AudioFlinger::Client>& client, 1377 uint32_t sampleRate, 1378 audio_format_t format, 1379 int channelMask, 1380 int frameCount, 1381 int sessionId, 1382 status_t *status); 1383 1384 status_t start(RecordTrack* recordTrack, 1385 AudioSystem::sync_event_t event, 1386 int triggerSession); 1387 void stop(RecordTrack* recordTrack); 1388 status_t dump(int fd, const Vector<String16>& args); 1389 AudioStreamIn* getInput() const; 1390 AudioStreamIn* clearInput(); 1391 virtual audio_stream_t* stream() const; 1392 1393 // AudioBufferProvider interface 1394 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts); 1395 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 1396 1397 virtual bool checkForNewParameters_l(); 1398 virtual String8 getParameters(const String8& keys); 1399 virtual void audioConfigChanged_l(int event, int param = 0); 1400 void readInputParameters(); 1401 virtual unsigned int getInputFramesLost(); 1402 1403 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1404 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1405 virtual uint32_t hasAudioSession(int sessionId); 1406 RecordTrack* track(); 1407 1408 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 1409 virtual bool isValidSyncEvent(const sp<SyncEvent>& event); 1410 1411 static void syncStartEventCallback(const wp<SyncEvent>& event); 1412 void handleSyncStartEvent(const sp<SyncEvent>& event); 1413 1414 private: 1415 void clearSyncStartEvent(); 1416 1417 RecordThread(); 1418 AudioStreamIn *mInput; 1419 RecordTrack* mTrack; 1420 sp<RecordTrack> mActiveTrack; 1421 Condition mStartStopCond; 1422 AudioResampler *mResampler; 1423 int32_t *mRsmpOutBuffer; 1424 int16_t *mRsmpInBuffer; 1425 size_t mRsmpInIndex; 1426 size_t mInputBytes; 1427 const int mReqChannelCount; 1428 const uint32_t mReqSampleRate; 1429 ssize_t mBytesRead; 1430 // sync event triggering actual audio capture. Frames read before this event will 1431 // be dropped and therefore not read by the application. 1432 sp<SyncEvent> mSyncStartEvent; 1433 // number of captured frames to drop after the start sync event has been received. 1434 // when < 0, maximum frames to drop before starting capture even if sync event is 1435 // not received 1436 ssize_t mFramestoDrop; 1437 }; 1438 1439 // server side of the client's IAudioRecord 1440 class RecordHandle : public android::BnAudioRecord { 1441 public: 1442 RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack); 1443 virtual ~RecordHandle(); 1444 virtual sp<IMemory> getCblk() const; 1445 virtual status_t start(int event, int triggerSession); 1446 virtual void stop(); 1447 virtual status_t onTransact( 1448 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 1449 private: 1450 const sp<RecordThread::RecordTrack> mRecordTrack; 1451 }; 1452 1453 //--- Audio Effect Management 1454 1455 // EffectModule and EffectChain classes both have their own mutex to protect 1456 // state changes or resource modifications. Always respect the following order 1457 // if multiple mutexes must be acquired to avoid cross deadlock: 1458 // AudioFlinger -> ThreadBase -> EffectChain -> EffectModule 1459 1460 // The EffectModule class is a wrapper object controlling the effect engine implementation 1461 // in the effect library. It prevents concurrent calls to process() and command() functions 1462 // from different client threads. It keeps a list of EffectHandle objects corresponding 1463 // to all client applications using this effect and notifies applications of effect state, 1464 // control or parameter changes. It manages the activation state machine to send appropriate 1465 // reset, enable, disable commands to effect engine and provide volume 1466 // ramping when effects are activated/deactivated. 1467 // When controlling an auxiliary effect, the EffectModule also provides an input buffer used by 1468 // the attached track(s) to accumulate their auxiliary channel. 1469 class EffectModule: public RefBase { 1470 public: 1471 EffectModule(ThreadBase *thread, 1472 const wp<AudioFlinger::EffectChain>& chain, 1473 effect_descriptor_t *desc, 1474 int id, 1475 int sessionId); 1476 virtual ~EffectModule(); 1477 1478 enum effect_state { 1479 IDLE, 1480 RESTART, 1481 STARTING, 1482 ACTIVE, 1483 STOPPING, 1484 STOPPED, 1485 DESTROYED 1486 }; 1487 1488 int id() const { return mId; } 1489 void process(); 1490 void updateState(); 1491 status_t command(uint32_t cmdCode, 1492 uint32_t cmdSize, 1493 void *pCmdData, 1494 uint32_t *replySize, 1495 void *pReplyData); 1496 1497 void reset_l(); 1498 status_t configure(); 1499 status_t init(); 1500 effect_state state() const { 1501 return mState; 1502 } 1503 uint32_t status() { 1504 return mStatus; 1505 } 1506 int sessionId() const { 1507 return mSessionId; 1508 } 1509 status_t setEnabled(bool enabled); 1510 bool isEnabled() const; 1511 bool isProcessEnabled() const; 1512 1513 void setInBuffer(int16_t *buffer) { mConfig.inputCfg.buffer.s16 = buffer; } 1514 int16_t *inBuffer() { return mConfig.inputCfg.buffer.s16; } 1515 void setOutBuffer(int16_t *buffer) { mConfig.outputCfg.buffer.s16 = buffer; } 1516 int16_t *outBuffer() { return mConfig.outputCfg.buffer.s16; } 1517 void setChain(const wp<EffectChain>& chain) { mChain = chain; } 1518 void setThread(const wp<ThreadBase>& thread) { mThread = thread; } 1519 const wp<ThreadBase>& thread() { return mThread; } 1520 1521 status_t addHandle(const sp<EffectHandle>& handle); 1522 void disconnect(const wp<EffectHandle>& handle, bool unpinIfLast); 1523 size_t removeHandle (const wp<EffectHandle>& handle); 1524 1525 effect_descriptor_t& desc() { return mDescriptor; } 1526 wp<EffectChain>& chain() { return mChain; } 1527 1528 status_t setDevice(uint32_t device); 1529 status_t setVolume(uint32_t *left, uint32_t *right, bool controller); 1530 status_t setMode(audio_mode_t mode); 1531 status_t start(); 1532 status_t stop(); 1533 void setSuspended(bool suspended); 1534 bool suspended() const; 1535 1536 sp<EffectHandle> controlHandle(); 1537 1538 bool isPinned() const { return mPinned; } 1539 void unPin() { mPinned = false; } 1540 1541 status_t dump(int fd, const Vector<String16>& args); 1542 1543 protected: 1544 friend class AudioFlinger; // for mHandles 1545 bool mPinned; 1546 1547 // Maximum time allocated to effect engines to complete the turn off sequence 1548 static const uint32_t MAX_DISABLE_TIME_MS = 10000; 1549 1550 EffectModule(const EffectModule&); 1551 EffectModule& operator = (const EffectModule&); 1552 1553 status_t start_l(); 1554 status_t stop_l(); 1555 1556mutable Mutex mLock; // mutex for process, commands and handles list protection 1557 wp<ThreadBase> mThread; // parent thread 1558 wp<EffectChain> mChain; // parent effect chain 1559 int mId; // this instance unique ID 1560 int mSessionId; // audio session ID 1561 effect_descriptor_t mDescriptor;// effect descriptor received from effect engine 1562 effect_config_t mConfig; // input and output audio configuration 1563 effect_handle_t mEffectInterface; // Effect module C API 1564 status_t mStatus; // initialization status 1565 effect_state mState; // current activation state 1566 Vector< wp<EffectHandle> > mHandles; // list of client handles 1567 // First handle in mHandles has highest priority and controls the effect module 1568 uint32_t mMaxDisableWaitCnt; // maximum grace period before forcing an effect off after 1569 // sending disable command. 1570 uint32_t mDisableWaitCnt; // current process() calls count during disable period. 1571 bool mSuspended; // effect is suspended: temporarily disabled by framework 1572 }; 1573 1574 // The EffectHandle class implements the IEffect interface. It provides resources 1575 // to receive parameter updates, keeps track of effect control 1576 // ownership and state and has a pointer to the EffectModule object it is controlling. 1577 // There is one EffectHandle object for each application controlling (or using) 1578 // an effect module. 1579 // The EffectHandle is obtained by calling AudioFlinger::createEffect(). 1580 class EffectHandle: public android::BnEffect { 1581 public: 1582 1583 EffectHandle(const sp<EffectModule>& effect, 1584 const sp<AudioFlinger::Client>& client, 1585 const sp<IEffectClient>& effectClient, 1586 int32_t priority); 1587 virtual ~EffectHandle(); 1588 1589 // IEffect 1590 virtual status_t enable(); 1591 virtual status_t disable(); 1592 virtual status_t command(uint32_t cmdCode, 1593 uint32_t cmdSize, 1594 void *pCmdData, 1595 uint32_t *replySize, 1596 void *pReplyData); 1597 virtual void disconnect(); 1598 private: 1599 void disconnect(bool unpinIfLast); 1600 public: 1601 virtual sp<IMemory> getCblk() const { return mCblkMemory; } 1602 virtual status_t onTransact(uint32_t code, const Parcel& data, 1603 Parcel* reply, uint32_t flags); 1604 1605 1606 // Give or take control of effect module 1607 // - hasControl: true if control is given, false if removed 1608 // - signal: true client app should be signaled of change, false otherwise 1609 // - enabled: state of the effect when control is passed 1610 void setControl(bool hasControl, bool signal, bool enabled); 1611 void commandExecuted(uint32_t cmdCode, 1612 uint32_t cmdSize, 1613 void *pCmdData, 1614 uint32_t replySize, 1615 void *pReplyData); 1616 void setEnabled(bool enabled); 1617 bool enabled() const { return mEnabled; } 1618 1619 // Getters 1620 int id() const { return mEffect->id(); } 1621 int priority() const { return mPriority; } 1622 bool hasControl() const { return mHasControl; } 1623 sp<EffectModule> effect() const { return mEffect; } 1624 1625 void dump(char* buffer, size_t size); 1626 1627 protected: 1628 friend class AudioFlinger; // for mEffect, mHasControl, mEnabled 1629 EffectHandle(const EffectHandle&); 1630 EffectHandle& operator =(const EffectHandle&); 1631 1632 sp<EffectModule> mEffect; // pointer to controlled EffectModule 1633 sp<IEffectClient> mEffectClient; // callback interface for client notifications 1634 /*const*/ sp<Client> mClient; // client for shared memory allocation, see disconnect() 1635 sp<IMemory> mCblkMemory; // shared memory for control block 1636 effect_param_cblk_t* mCblk; // control block for deferred parameter setting via shared memory 1637 uint8_t* mBuffer; // pointer to parameter area in shared memory 1638 int mPriority; // client application priority to control the effect 1639 bool mHasControl; // true if this handle is controlling the effect 1640 bool mEnabled; // cached enable state: needed when the effect is 1641 // restored after being suspended 1642 }; 1643 1644 // the EffectChain class represents a group of effects associated to one audio session. 1645 // There can be any number of EffectChain objects per output mixer thread (PlaybackThread). 1646 // The EffecChain with session ID 0 contains global effects applied to the output mix. 1647 // Effects in this chain can be insert or auxiliary. Effects in other chains (attached to tracks) 1648 // are insert only. The EffectChain maintains an ordered list of effect module, the order corresponding 1649 // in the effect process order. When attached to a track (session ID != 0), it also provide it's own 1650 // input buffer used by the track as accumulation buffer. 1651 class EffectChain: public RefBase { 1652 public: 1653 EffectChain(const wp<ThreadBase>& wThread, int sessionId); 1654 EffectChain(ThreadBase *thread, int sessionId); 1655 virtual ~EffectChain(); 1656 1657 // special key used for an entry in mSuspendedEffects keyed vector 1658 // corresponding to a suspend all request. 1659 static const int kKeyForSuspendAll = 0; 1660 1661 // minimum duration during which we force calling effect process when last track on 1662 // a session is stopped or removed to allow effect tail to be rendered 1663 static const int kProcessTailDurationMs = 1000; 1664 1665 void process_l(); 1666 1667 void lock() { 1668 mLock.lock(); 1669 } 1670 void unlock() { 1671 mLock.unlock(); 1672 } 1673 1674 status_t addEffect_l(const sp<EffectModule>& handle); 1675 size_t removeEffect_l(const sp<EffectModule>& handle); 1676 1677 int sessionId() const { return mSessionId; } 1678 void setSessionId(int sessionId) { mSessionId = sessionId; } 1679 1680 sp<EffectModule> getEffectFromDesc_l(effect_descriptor_t *descriptor); 1681 sp<EffectModule> getEffectFromId_l(int id); 1682 sp<EffectModule> getEffectFromType_l(const effect_uuid_t *type); 1683 bool setVolume_l(uint32_t *left, uint32_t *right); 1684 void setDevice_l(uint32_t device); 1685 void setMode_l(audio_mode_t mode); 1686 1687 void setInBuffer(int16_t *buffer, bool ownsBuffer = false) { 1688 mInBuffer = buffer; 1689 mOwnInBuffer = ownsBuffer; 1690 } 1691 int16_t *inBuffer() const { 1692 return mInBuffer; 1693 } 1694 void setOutBuffer(int16_t *buffer) { 1695 mOutBuffer = buffer; 1696 } 1697 int16_t *outBuffer() const { 1698 return mOutBuffer; 1699 } 1700 1701 void incTrackCnt() { android_atomic_inc(&mTrackCnt); } 1702 void decTrackCnt() { android_atomic_dec(&mTrackCnt); } 1703 int32_t trackCnt() const { return mTrackCnt;} 1704 1705 void incActiveTrackCnt() { android_atomic_inc(&mActiveTrackCnt); 1706 mTailBufferCount = mMaxTailBuffers; } 1707 void decActiveTrackCnt() { android_atomic_dec(&mActiveTrackCnt); } 1708 int32_t activeTrackCnt() const { return mActiveTrackCnt;} 1709 1710 uint32_t strategy() const { return mStrategy; } 1711 void setStrategy(uint32_t strategy) 1712 { mStrategy = strategy; } 1713 1714 // suspend effect of the given type 1715 void setEffectSuspended_l(const effect_uuid_t *type, 1716 bool suspend); 1717 // suspend all eligible effects 1718 void setEffectSuspendedAll_l(bool suspend); 1719 // check if effects should be suspend or restored when a given effect is enable or disabled 1720 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1721 bool enabled); 1722 1723 void clearInputBuffer(); 1724 1725 status_t dump(int fd, const Vector<String16>& args); 1726 1727 protected: 1728 friend class AudioFlinger; // for mThread, mEffects 1729 EffectChain(const EffectChain&); 1730 EffectChain& operator =(const EffectChain&); 1731 1732 class SuspendedEffectDesc : public RefBase { 1733 public: 1734 SuspendedEffectDesc() : mRefCount(0) {} 1735 1736 int mRefCount; 1737 effect_uuid_t mType; 1738 wp<EffectModule> mEffect; 1739 }; 1740 1741 // get a list of effect modules to suspend when an effect of the type 1742 // passed is enabled. 1743 void getSuspendEligibleEffects(Vector< sp<EffectModule> > &effects); 1744 1745 // get an effect module if it is currently enable 1746 sp<EffectModule> getEffectIfEnabled(const effect_uuid_t *type); 1747 // true if the effect whose descriptor is passed can be suspended 1748 // OEMs can modify the rules implemented in this method to exclude specific effect 1749 // types or implementations from the suspend/restore mechanism. 1750 bool isEffectEligibleForSuspend(const effect_descriptor_t& desc); 1751 1752 void clearInputBuffer_l(sp<ThreadBase> thread); 1753 1754 wp<ThreadBase> mThread; // parent mixer thread 1755 Mutex mLock; // mutex protecting effect list 1756 Vector< sp<EffectModule> > mEffects; // list of effect modules 1757 int mSessionId; // audio session ID 1758 int16_t *mInBuffer; // chain input buffer 1759 int16_t *mOutBuffer; // chain output buffer 1760 volatile int32_t mActiveTrackCnt; // number of active tracks connected 1761 volatile int32_t mTrackCnt; // number of tracks connected 1762 int32_t mTailBufferCount; // current effect tail buffer count 1763 int32_t mMaxTailBuffers; // maximum effect tail buffers 1764 bool mOwnInBuffer; // true if the chain owns its input buffer 1765 int mVolumeCtrlIdx; // index of insert effect having control over volume 1766 uint32_t mLeftVolume; // previous volume on left channel 1767 uint32_t mRightVolume; // previous volume on right channel 1768 uint32_t mNewLeftVolume; // new volume on left channel 1769 uint32_t mNewRightVolume; // new volume on right channel 1770 uint32_t mStrategy; // strategy for this effect chain 1771 // mSuspendedEffects lists all effects currently suspended in the chain. 1772 // Use effect type UUID timelow field as key. There is no real risk of identical 1773 // timeLow fields among effect type UUIDs. 1774 // Updated by updateSuspendedSessions_l() only. 1775 KeyedVector< int, sp<SuspendedEffectDesc> > mSuspendedEffects; 1776 }; 1777 1778 // AudioStreamOut and AudioStreamIn are immutable, so their fields are const. 1779 // For emphasis, we could also make all pointers to them be "const *", 1780 // but that would clutter the code unnecessarily. 1781 1782 struct AudioStreamOut { 1783 audio_hw_device_t* const hwDev; 1784 audio_stream_out_t* const stream; 1785 1786 AudioStreamOut(audio_hw_device_t *dev, audio_stream_out_t *out) : 1787 hwDev(dev), stream(out) {} 1788 }; 1789 1790 struct AudioStreamIn { 1791 audio_hw_device_t* const hwDev; 1792 audio_stream_in_t* const stream; 1793 1794 AudioStreamIn(audio_hw_device_t *dev, audio_stream_in_t *in) : 1795 hwDev(dev), stream(in) {} 1796 }; 1797 1798 // for mAudioSessionRefs only 1799 struct AudioSessionRef { 1800 AudioSessionRef(int sessionid, pid_t pid) : 1801 mSessionid(sessionid), mPid(pid), mCnt(1) {} 1802 const int mSessionid; 1803 const pid_t mPid; 1804 int mCnt; 1805 }; 1806 1807 enum master_volume_support { 1808 // MVS_NONE: 1809 // Audio HAL has no support for master volume, either setting or 1810 // getting. All master volume control must be implemented in SW by the 1811 // AudioFlinger mixing core. 1812 MVS_NONE, 1813 1814 // MVS_SETONLY: 1815 // Audio HAL has support for setting master volume, but not for getting 1816 // master volume (original HAL design did not include a getter). 1817 // AudioFlinger needs to keep track of the last set master volume in 1818 // addition to needing to set an initial, default, master volume at HAL 1819 // load time. 1820 MVS_SETONLY, 1821 1822 // MVS_FULL: 1823 // Audio HAL has support both for setting and getting master volume. 1824 // AudioFlinger should send all set and get master volume requests 1825 // directly to the HAL. 1826 MVS_FULL, 1827 }; 1828 1829 class AudioHwDevice { 1830 public: 1831 AudioHwDevice(const char *moduleName, audio_hw_device_t *hwDevice) : 1832 mModuleName(strdup(moduleName)), mHwDevice(hwDevice){} 1833 ~AudioHwDevice() { free((void *)mModuleName); } 1834 1835 const char *moduleName() const { return mModuleName; } 1836 audio_hw_device_t *hwDevice() const { return mHwDevice; } 1837 private: 1838 const char * const mModuleName; 1839 audio_hw_device_t * const mHwDevice; 1840 }; 1841 1842 mutable Mutex mLock; 1843 1844 DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client() 1845 1846 mutable Mutex mHardwareLock; 1847 // NOTE: If both mLock and mHardwareLock mutexes must be held, 1848 // always take mLock before mHardwareLock 1849 1850 // These two fields are immutable after onFirstRef(), so no lock needed to access 1851 audio_hw_device_t* mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL 1852 DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*> mAudioHwDevs; 1853 1854 // for dump, indicates which hardware operation is currently in progress (but not stream ops) 1855 enum hardware_call_state { 1856 AUDIO_HW_IDLE = 0, // no operation in progress 1857 AUDIO_HW_INIT, // init_check 1858 AUDIO_HW_OUTPUT_OPEN, // open_output_stream 1859 AUDIO_HW_OUTPUT_CLOSE, // unused 1860 AUDIO_HW_INPUT_OPEN, // unused 1861 AUDIO_HW_INPUT_CLOSE, // unused 1862 AUDIO_HW_STANDBY, // unused 1863 AUDIO_HW_SET_MASTER_VOLUME, // set_master_volume 1864 AUDIO_HW_GET_ROUTING, // unused 1865 AUDIO_HW_SET_ROUTING, // unused 1866 AUDIO_HW_GET_MODE, // unused 1867 AUDIO_HW_SET_MODE, // set_mode 1868 AUDIO_HW_GET_MIC_MUTE, // get_mic_mute 1869 AUDIO_HW_SET_MIC_MUTE, // set_mic_mute 1870 AUDIO_HW_SET_VOICE_VOLUME, // set_voice_volume 1871 AUDIO_HW_SET_PARAMETER, // set_parameters 1872 AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size 1873 AUDIO_HW_GET_MASTER_VOLUME, // get_master_volume 1874 AUDIO_HW_GET_PARAMETER, // get_parameters 1875 }; 1876 1877 mutable hardware_call_state mHardwareStatus; // for dump only 1878 1879 1880 DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads; 1881 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 1882 1883 // both are protected by mLock 1884 float mMasterVolume; 1885 float mMasterVolumeSW; 1886 master_volume_support mMasterVolumeSupportLvl; 1887 bool mMasterMute; 1888 1889 DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads; 1890 1891 DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients; 1892 volatile int32_t mNextUniqueId; // updated by android_atomic_inc 1893 audio_mode_t mMode; 1894 bool mBtNrecIsOff; 1895 1896 // protected by mLock 1897 Vector<AudioSessionRef*> mAudioSessionRefs; 1898 1899 float masterVolume_l() const; 1900 float masterVolumeSW_l() const { return mMasterVolumeSW; } 1901 bool masterMute_l() const { return mMasterMute; } 1902 audio_module_handle_t loadHwModule_l(const char *name); 1903 1904 Vector < sp<SyncEvent> > mPendingSyncEvents; // sync events awaiting for a session 1905 // to be created 1906 1907private: 1908 sp<Client> registerPid_l(pid_t pid); // always returns non-0 1909 1910}; 1911 1912 1913// ---------------------------------------------------------------------------- 1914 1915}; // namespace android 1916 1917#endif // ANDROID_AUDIO_FLINGER_H 1918