AudioFlinger.h revision 67c0a58e05f4c19d4a6f01fe6f06267d57b49305
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 53#include <powermanager/IPowerManager.h> 54 55namespace android { 56 57class audio_track_cblk_t; 58class effect_param_cblk_t; 59class AudioMixer; 60class AudioBuffer; 61class AudioResampler; 62class FastMixer; 63 64// ---------------------------------------------------------------------------- 65 66// AudioFlinger has a hard-coded upper limit of 2 channels for capture and playback. 67// There is support for > 2 channel tracks down-mixed to 2 channel output via a down-mix effect. 68// Adding full support for > 2 channel capture or playback would require more than simply changing 69// this #define. There is an independent hard-coded upper limit in AudioMixer; 70// removing that AudioMixer limit would be necessary but insufficient to support > 2 channels. 71// The macro FCC_2 highlights some (but not all) places where there is are 2-channel assumptions. 72// Search also for "2", "left", "right", "[0]", "[1]", ">> 16", "<< 16", etc. 73#define FCC_2 2 // FCC_2 = Fixed Channel Count 2 74 75static const nsecs_t kDefaultStandbyTimeInNsecs = seconds(3); 76 77class AudioFlinger : 78 public BinderService<AudioFlinger>, 79 public BnAudioFlinger 80{ 81 friend class BinderService<AudioFlinger>; // for AudioFlinger() 82public: 83 static const char* getServiceName() { return "media.audio_flinger"; } 84 85 virtual status_t dump(int fd, const Vector<String16>& args); 86 87 // IAudioFlinger interface, in binder opcode order 88 virtual sp<IAudioTrack> createTrack( 89 pid_t pid, 90 audio_stream_type_t streamType, 91 uint32_t sampleRate, 92 audio_format_t format, 93 uint32_t channelMask, 94 int frameCount, 95 IAudioFlinger::track_flags_t flags, 96 const sp<IMemory>& sharedBuffer, 97 audio_io_handle_t output, 98 pid_t tid, 99 int *sessionId, 100 status_t *status); 101 102 virtual sp<IAudioRecord> openRecord( 103 pid_t pid, 104 audio_io_handle_t input, 105 uint32_t sampleRate, 106 audio_format_t format, 107 uint32_t channelMask, 108 int frameCount, 109 IAudioFlinger::track_flags_t flags, 110 int *sessionId, 111 status_t *status); 112 113 virtual uint32_t sampleRate(audio_io_handle_t output) const; 114 virtual int channelCount(audio_io_handle_t output) const; 115 virtual audio_format_t format(audio_io_handle_t output) const; 116 virtual size_t frameCount(audio_io_handle_t output) const; 117 virtual uint32_t latency(audio_io_handle_t output) const; 118 119 virtual status_t setMasterVolume(float value); 120 virtual status_t setMasterMute(bool muted); 121 122 virtual float masterVolume() const; 123 virtual float masterVolumeSW() const; 124 virtual bool masterMute() const; 125 126 virtual status_t setStreamVolume(audio_stream_type_t stream, float value, 127 audio_io_handle_t output); 128 virtual status_t setStreamMute(audio_stream_type_t stream, bool muted); 129 130 virtual float streamVolume(audio_stream_type_t stream, 131 audio_io_handle_t output) const; 132 virtual bool streamMute(audio_stream_type_t stream) const; 133 134 virtual status_t setMode(audio_mode_t mode); 135 136 virtual status_t setMicMute(bool state); 137 virtual bool getMicMute() const; 138 139 virtual status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs); 140 virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) const; 141 142 virtual void registerClient(const sp<IAudioFlingerClient>& client); 143 144 virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const; 145 146 virtual audio_io_handle_t openOutput(audio_module_handle_t module, 147 audio_devices_t *pDevices, 148 uint32_t *pSamplingRate, 149 audio_format_t *pFormat, 150 audio_channel_mask_t *pChannelMask, 151 uint32_t *pLatencyMs, 152 audio_output_flags_t flags); 153 154 virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1, 155 audio_io_handle_t output2); 156 157 virtual status_t closeOutput(audio_io_handle_t output); 158 159 virtual status_t suspendOutput(audio_io_handle_t output); 160 161 virtual status_t restoreOutput(audio_io_handle_t output); 162 163 virtual audio_io_handle_t openInput(audio_module_handle_t module, 164 audio_devices_t *pDevices, 165 uint32_t *pSamplingRate, 166 audio_format_t *pFormat, 167 audio_channel_mask_t *pChannelMask); 168 169 virtual status_t closeInput(audio_io_handle_t input); 170 171 virtual status_t setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output); 172 173 virtual status_t setVoiceVolume(float volume); 174 175 virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 176 audio_io_handle_t output) const; 177 178 virtual unsigned int getInputFramesLost(audio_io_handle_t ioHandle) const; 179 180 virtual int newAudioSessionId(); 181 182 virtual void acquireAudioSessionId(int audioSession); 183 184 virtual void releaseAudioSessionId(int audioSession); 185 186 virtual status_t queryNumberEffects(uint32_t *numEffects) const; 187 188 virtual status_t queryEffect(uint32_t index, effect_descriptor_t *descriptor) const; 189 190 virtual status_t getEffectDescriptor(const effect_uuid_t *pUuid, 191 effect_descriptor_t *descriptor) const; 192 193 virtual sp<IEffect> createEffect(pid_t pid, 194 effect_descriptor_t *pDesc, 195 const sp<IEffectClient>& effectClient, 196 int32_t priority, 197 audio_io_handle_t io, 198 int sessionId, 199 status_t *status, 200 int *id, 201 int *enabled); 202 203 virtual status_t moveEffects(int sessionId, audio_io_handle_t srcOutput, 204 audio_io_handle_t dstOutput); 205 206 virtual audio_module_handle_t loadHwModule(const char *name); 207 208 virtual status_t onTransact( 209 uint32_t code, 210 const Parcel& data, 211 Parcel* reply, 212 uint32_t flags); 213 214 // end of IAudioFlinger interface 215 216 class SyncEvent; 217 218 typedef void (*sync_event_callback_t)(const wp<SyncEvent>& event) ; 219 220 class SyncEvent : public RefBase { 221 public: 222 SyncEvent(AudioSystem::sync_event_t type, 223 int triggerSession, 224 int listenerSession, 225 sync_event_callback_t callBack, 226 void *cookie) 227 : mType(type), mTriggerSession(triggerSession), mListenerSession(listenerSession), 228 mCallback(callBack), mCookie(cookie) 229 {} 230 231 virtual ~SyncEvent() {} 232 233 void trigger() { Mutex::Autolock _l(mLock); if (mCallback) mCallback(this); } 234 bool isCancelled() { Mutex::Autolock _l(mLock); return (mCallback == NULL); } 235 void cancel() {Mutex::Autolock _l(mLock); mCallback = NULL; } 236 AudioSystem::sync_event_t type() const { return mType; } 237 int triggerSession() const { return mTriggerSession; } 238 int listenerSession() const { return mListenerSession; } 239 void *cookie() const { return mCookie; } 240 241 private: 242 const AudioSystem::sync_event_t mType; 243 const int mTriggerSession; 244 const int mListenerSession; 245 sync_event_callback_t mCallback; 246 void * const mCookie; 247 Mutex mLock; 248 }; 249 250 sp<SyncEvent> createSyncEvent(AudioSystem::sync_event_t type, 251 int triggerSession, 252 int listenerSession, 253 sync_event_callback_t callBack, 254 void *cookie); 255private: 256 audio_mode_t getMode() const { return mMode; } 257 258 bool btNrecIsOff() const { return mBtNrecIsOff; } 259 260 AudioFlinger(); 261 virtual ~AudioFlinger(); 262 263 // call in any IAudioFlinger method that accesses mPrimaryHardwareDev 264 status_t initCheck() const { return mPrimaryHardwareDev == NULL ? NO_INIT : NO_ERROR; } 265 266 // RefBase 267 virtual void onFirstRef(); 268 269 audio_hw_device_t* findSuitableHwDev_l(audio_module_handle_t module, uint32_t devices); 270 void purgeStaleEffects_l(); 271 272 // standby delay for MIXER and DUPLICATING playback threads is read from property 273 // ro.audio.flinger_standbytime_ms or defaults to kDefaultStandbyTimeInNsecs 274 static nsecs_t mStandbyTimeInNsecs; 275 276 // Internal dump utilites. 277 status_t dumpPermissionDenial(int fd, const Vector<String16>& args); 278 status_t dumpClients(int fd, const Vector<String16>& args); 279 status_t dumpInternals(int fd, const Vector<String16>& args); 280 281 // --- Client --- 282 class Client : public RefBase { 283 public: 284 Client(const sp<AudioFlinger>& audioFlinger, pid_t pid); 285 virtual ~Client(); 286 sp<MemoryDealer> heap() const; 287 pid_t pid() const { return mPid; } 288 sp<AudioFlinger> audioFlinger() const { return mAudioFlinger; } 289 290 bool reserveTimedTrack(); 291 void releaseTimedTrack(); 292 293 private: 294 Client(const Client&); 295 Client& operator = (const Client&); 296 const sp<AudioFlinger> mAudioFlinger; 297 const sp<MemoryDealer> mMemoryDealer; 298 const pid_t mPid; 299 300 Mutex mTimedTrackLock; 301 int mTimedTrackCount; 302 }; 303 304 // --- Notification Client --- 305 class NotificationClient : public IBinder::DeathRecipient { 306 public: 307 NotificationClient(const sp<AudioFlinger>& audioFlinger, 308 const sp<IAudioFlingerClient>& client, 309 pid_t pid); 310 virtual ~NotificationClient(); 311 312 sp<IAudioFlingerClient> audioFlingerClient() const { return mAudioFlingerClient; } 313 314 // IBinder::DeathRecipient 315 virtual void binderDied(const wp<IBinder>& who); 316 317 private: 318 NotificationClient(const NotificationClient&); 319 NotificationClient& operator = (const NotificationClient&); 320 321 const sp<AudioFlinger> mAudioFlinger; 322 const pid_t mPid; 323 const sp<IAudioFlingerClient> mAudioFlingerClient; 324 }; 325 326 class TrackHandle; 327 class RecordHandle; 328 class RecordThread; 329 class PlaybackThread; 330 class MixerThread; 331 class DirectOutputThread; 332 class DuplicatingThread; 333 class Track; 334 class RecordTrack; 335 class EffectModule; 336 class EffectHandle; 337 class EffectChain; 338 struct AudioStreamOut; 339 struct AudioStreamIn; 340 341 class ThreadBase : public Thread { 342 public: 343 344 enum type_t { 345 MIXER, // Thread class is MixerThread 346 DIRECT, // Thread class is DirectOutputThread 347 DUPLICATING, // Thread class is DuplicatingThread 348 RECORD // Thread class is RecordThread 349 }; 350 351 ThreadBase (const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, uint32_t device, type_t type); 352 virtual ~ThreadBase(); 353 354 status_t dumpBase(int fd, const Vector<String16>& args); 355 status_t dumpEffectChains(int fd, const Vector<String16>& args); 356 357 void clearPowerManager(); 358 359 // base for record and playback 360 class TrackBase : public ExtendedAudioBufferProvider, public RefBase { 361 362 public: 363 enum track_state { 364 IDLE, 365 TERMINATED, 366 FLUSHED, 367 STOPPED, 368 // next 2 states are currently used for fast tracks only 369 STOPPING_1, // waiting for first underrun 370 STOPPING_2, // waiting for presentation complete 371 RESUMING, 372 ACTIVE, 373 PAUSING, 374 PAUSED 375 }; 376 377 TrackBase(ThreadBase *thread, 378 const sp<Client>& client, 379 uint32_t sampleRate, 380 audio_format_t format, 381 uint32_t channelMask, 382 int frameCount, 383 const sp<IMemory>& sharedBuffer, 384 int sessionId); 385 virtual ~TrackBase(); 386 387 virtual status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 388 int triggerSession = 0) = 0; 389 virtual void stop() = 0; 390 sp<IMemory> getCblk() const { return mCblkMemory; } 391 audio_track_cblk_t* cblk() const { return mCblk; } 392 int sessionId() const { return mSessionId; } 393 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 394 395 protected: 396 TrackBase(const TrackBase&); 397 TrackBase& operator = (const TrackBase&); 398 399 // AudioBufferProvider interface 400 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) = 0; 401 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 402 403 // ExtendedAudioBufferProvider interface is only needed for Track, 404 // but putting it in TrackBase avoids the complexity of virtual inheritance 405 virtual size_t framesReady() const { return SIZE_MAX; } 406 407 audio_format_t format() const { 408 return mFormat; 409 } 410 411 int channelCount() const { return mChannelCount; } 412 413 uint32_t channelMask() const { return mChannelMask; } 414 415 int sampleRate() const; // FIXME inline after cblk sr moved 416 417 void* getBuffer(uint32_t offset, uint32_t frames) const; 418 419 bool isStopped() const { 420 return (mState == STOPPED || mState == FLUSHED); 421 } 422 423 // for fast tracks only 424 bool isStopping() const { 425 return mState == STOPPING_1 || mState == STOPPING_2; 426 } 427 bool isStopping_1() const { 428 return mState == STOPPING_1; 429 } 430 bool isStopping_2() const { 431 return mState == STOPPING_2; 432 } 433 434 bool isTerminated() const { 435 return mState == TERMINATED; 436 } 437 438 bool step(); 439 void reset(); 440 441 const wp<ThreadBase> mThread; 442 /*const*/ sp<Client> mClient; // see explanation at ~TrackBase() why not const 443 sp<IMemory> mCblkMemory; 444 audio_track_cblk_t* mCblk; 445 void* mBuffer; 446 void* mBufferEnd; 447 uint32_t mFrameCount; 448 // we don't really need a lock for these 449 track_state mState; 450 const uint32_t mSampleRate; // initial sample rate only; for tracks which 451 // support dynamic rates, the current value is in control block 452 const audio_format_t mFormat; 453 bool mStepServerFailed; 454 const int mSessionId; 455 uint8_t mChannelCount; 456 uint32_t mChannelMask; 457 Vector < sp<SyncEvent> >mSyncEvents; 458 }; 459 460 class ConfigEvent { 461 public: 462 ConfigEvent() : mEvent(0), mParam(0) {} 463 464 int mEvent; 465 int mParam; 466 }; 467 468 class PMDeathRecipient : public IBinder::DeathRecipient { 469 public: 470 PMDeathRecipient(const wp<ThreadBase>& thread) : mThread(thread) {} 471 virtual ~PMDeathRecipient() {} 472 473 // IBinder::DeathRecipient 474 virtual void binderDied(const wp<IBinder>& who); 475 476 private: 477 PMDeathRecipient(const PMDeathRecipient&); 478 PMDeathRecipient& operator = (const PMDeathRecipient&); 479 480 wp<ThreadBase> mThread; 481 }; 482 483 virtual status_t initCheck() const = 0; 484 type_t type() const { return mType; } 485 uint32_t sampleRate() const { return mSampleRate; } 486 int channelCount() const { return mChannelCount; } 487 audio_format_t format() const { return mFormat; } 488 // Called by AudioFlinger::frameCount(audio_io_handle_t output) and effects, 489 // and returns the normal mix buffer's frame count. No API for HAL frame count. 490 size_t frameCount() const { return mNormalFrameCount; } 491 void wakeUp() { mWaitWorkCV.broadcast(); } 492 // Should be "virtual status_t requestExitAndWait()" and override same 493 // method in Thread, but Thread::requestExitAndWait() is not yet virtual. 494 void exit(); 495 virtual bool checkForNewParameters_l() = 0; 496 virtual status_t setParameters(const String8& keyValuePairs); 497 virtual String8 getParameters(const String8& keys) = 0; 498 virtual void audioConfigChanged_l(int event, int param = 0) = 0; 499 void sendConfigEvent(int event, int param = 0); 500 void sendConfigEvent_l(int event, int param = 0); 501 void processConfigEvents(); 502 audio_io_handle_t id() const { return mId;} 503 bool standby() const { return mStandby; } 504 uint32_t device() const { return mDevice; } 505 virtual audio_stream_t* stream() const = 0; 506 507 sp<EffectHandle> createEffect_l( 508 const sp<AudioFlinger::Client>& client, 509 const sp<IEffectClient>& effectClient, 510 int32_t priority, 511 int sessionId, 512 effect_descriptor_t *desc, 513 int *enabled, 514 status_t *status); 515 void disconnectEffect(const sp< EffectModule>& effect, 516 const wp<EffectHandle>& handle, 517 bool unpinIfLast); 518 519 // return values for hasAudioSession (bit field) 520 enum effect_state { 521 EFFECT_SESSION = 0x1, // the audio session corresponds to at least one 522 // effect 523 TRACK_SESSION = 0x2 // the audio session corresponds to at least one 524 // track 525 }; 526 527 // get effect chain corresponding to session Id. 528 sp<EffectChain> getEffectChain(int sessionId); 529 // same as getEffectChain() but must be called with ThreadBase mutex locked 530 sp<EffectChain> getEffectChain_l(int sessionId); 531 // add an effect chain to the chain list (mEffectChains) 532 virtual status_t addEffectChain_l(const sp<EffectChain>& chain) = 0; 533 // remove an effect chain from the chain list (mEffectChains) 534 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain) = 0; 535 // lock all effect chains Mutexes. Must be called before releasing the 536 // ThreadBase mutex before processing the mixer and effects. This guarantees the 537 // integrity of the chains during the process. 538 // Also sets the parameter 'effectChains' to current value of mEffectChains. 539 void lockEffectChains_l(Vector< sp<EffectChain> >& effectChains); 540 // unlock effect chains after process 541 void unlockEffectChains(const Vector< sp<EffectChain> >& effectChains); 542 // set audio mode to all effect chains 543 void setMode(audio_mode_t mode); 544 // get effect module with corresponding ID on specified audio session 545 sp<AudioFlinger::EffectModule> getEffect_l(int sessionId, int effectId); 546 // add and effect module. Also creates the effect chain is none exists for 547 // the effects audio session 548 status_t addEffect_l(const sp< EffectModule>& effect); 549 // remove and effect module. Also removes the effect chain is this was the last 550 // effect 551 void removeEffect_l(const sp< EffectModule>& effect); 552 // detach all tracks connected to an auxiliary effect 553 virtual void detachAuxEffect_l(int effectId) {} 554 // returns either EFFECT_SESSION if effects on this audio session exist in one 555 // chain, or TRACK_SESSION if tracks on this audio session exist, or both 556 virtual uint32_t hasAudioSession(int sessionId) = 0; 557 // the value returned by default implementation is not important as the 558 // strategy is only meaningful for PlaybackThread which implements this method 559 virtual uint32_t getStrategyForSession_l(int sessionId) { return 0; } 560 561 // suspend or restore effect according to the type of effect passed. a NULL 562 // type pointer means suspend all effects in the session 563 void setEffectSuspended(const effect_uuid_t *type, 564 bool suspend, 565 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 566 // check if some effects must be suspended/restored when an effect is enabled 567 // or disabled 568 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 569 bool enabled, 570 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 571 void checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 572 bool enabled, 573 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 574 575 virtual status_t setSyncEvent(const sp<SyncEvent>& event) = 0; 576 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) = 0; 577 578 579 mutable Mutex mLock; 580 581 protected: 582 583 // entry describing an effect being suspended in mSuspendedSessions keyed vector 584 class SuspendedSessionDesc : public RefBase { 585 public: 586 SuspendedSessionDesc() : mRefCount(0) {} 587 588 int mRefCount; // number of active suspend requests 589 effect_uuid_t mType; // effect type UUID 590 }; 591 592 void acquireWakeLock(); 593 void acquireWakeLock_l(); 594 void releaseWakeLock(); 595 void releaseWakeLock_l(); 596 void setEffectSuspended_l(const effect_uuid_t *type, 597 bool suspend, 598 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 599 // updated mSuspendedSessions when an effect suspended or restored 600 void updateSuspendedSessions_l(const effect_uuid_t *type, 601 bool suspend, 602 int sessionId); 603 // check if some effects must be suspended when an effect chain is added 604 void checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain); 605 606 friend class AudioFlinger; // for mEffectChains 607 608 const type_t mType; 609 610 // Used by parameters, config events, addTrack_l, exit 611 Condition mWaitWorkCV; 612 613 const sp<AudioFlinger> mAudioFlinger; 614 uint32_t mSampleRate; 615 size_t mFrameCount; // output HAL, direct output, record 616 size_t mNormalFrameCount; // normal mixer and effects 617 uint32_t mChannelMask; 618 uint16_t mChannelCount; 619 size_t mFrameSize; 620 audio_format_t mFormat; 621 622 // Parameter sequence by client: binder thread calling setParameters(): 623 // 1. Lock mLock 624 // 2. Append to mNewParameters 625 // 3. mWaitWorkCV.signal 626 // 4. mParamCond.waitRelative with timeout 627 // 5. read mParamStatus 628 // 6. mWaitWorkCV.signal 629 // 7. Unlock 630 // 631 // Parameter sequence by server: threadLoop calling checkForNewParameters_l(): 632 // 1. Lock mLock 633 // 2. If there is an entry in mNewParameters proceed ... 634 // 2. Read first entry in mNewParameters 635 // 3. Process 636 // 4. Remove first entry from mNewParameters 637 // 5. Set mParamStatus 638 // 6. mParamCond.signal 639 // 7. mWaitWorkCV.wait with timeout (this is to avoid overwriting mParamStatus) 640 // 8. Unlock 641 Condition mParamCond; 642 Vector<String8> mNewParameters; 643 status_t mParamStatus; 644 645 Vector<ConfigEvent> mConfigEvents; 646 bool mStandby; 647 const audio_io_handle_t mId; 648 Vector< sp<EffectChain> > mEffectChains; 649 uint32_t mDevice; // output device for PlaybackThread 650 // input + output devices for RecordThread 651 static const int kNameLength = 16; // prctl(PR_SET_NAME) limit 652 char mName[kNameLength]; 653 sp<IPowerManager> mPowerManager; 654 sp<IBinder> mWakeLockToken; 655 const sp<PMDeathRecipient> mDeathRecipient; 656 // list of suspended effects per session and per type. The first vector is 657 // keyed by session ID, the second by type UUID timeLow field 658 KeyedVector< int, KeyedVector< int, sp<SuspendedSessionDesc> > > mSuspendedSessions; 659 }; 660 661 struct stream_type_t { 662 stream_type_t() 663 : volume(1.0f), 664 mute(false) 665 { 666 } 667 float volume; 668 bool mute; 669 }; 670 671 // --- PlaybackThread --- 672 class PlaybackThread : public ThreadBase { 673 public: 674 675 enum mixer_state { 676 MIXER_IDLE, // no active tracks 677 MIXER_TRACKS_ENABLED, // at least one active track, but no track has any data ready 678 MIXER_TRACKS_READY // at least one active track, and at least one track has data 679 // standby mode does not have an enum value 680 // suspend by audio policy manager is orthogonal to mixer state 681 }; 682 683 // playback track 684 class Track : public TrackBase, public VolumeProvider { 685 public: 686 Track( PlaybackThread *thread, 687 const sp<Client>& client, 688 audio_stream_type_t streamType, 689 uint32_t sampleRate, 690 audio_format_t format, 691 uint32_t channelMask, 692 int frameCount, 693 const sp<IMemory>& sharedBuffer, 694 int sessionId, 695 IAudioFlinger::track_flags_t flags); 696 virtual ~Track(); 697 698 static void appendDumpHeader(String8& result); 699 void dump(char* buffer, size_t size); 700 virtual status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 701 int triggerSession = 0); 702 virtual void stop(); 703 void pause(); 704 705 void flush(); 706 void destroy(); 707 void mute(bool); 708 int name() const { 709 return mName; 710 } 711 712 audio_stream_type_t streamType() const { 713 return mStreamType; 714 } 715 status_t attachAuxEffect(int EffectId); 716 void setAuxBuffer(int EffectId, int32_t *buffer); 717 int32_t *auxBuffer() const { return mAuxBuffer; } 718 void setMainBuffer(int16_t *buffer) { mMainBuffer = buffer; } 719 int16_t *mainBuffer() const { return mMainBuffer; } 720 int auxEffectId() const { return mAuxEffectId; } 721 722 // implement FastMixerState::VolumeProvider interface 723 virtual uint32_t getVolumeLR(); 724 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 725 726 protected: 727 // for numerous 728 friend class PlaybackThread; 729 friend class MixerThread; 730 friend class DirectOutputThread; 731 732 Track(const Track&); 733 Track& operator = (const Track&); 734 735 // AudioBufferProvider interface 736 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts = kInvalidPTS); 737 // releaseBuffer() not overridden 738 739 virtual size_t framesReady() const; 740 741 bool isMuted() const { return mMute; } 742 bool isPausing() const { 743 return mState == PAUSING; 744 } 745 bool isPaused() const { 746 return mState == PAUSED; 747 } 748 bool isResuming() const { 749 return mState == RESUMING; 750 } 751 bool isReady() const; 752 void setPaused() { mState = PAUSED; } 753 void reset(); 754 755 bool isOutputTrack() const { 756 return (mStreamType == AUDIO_STREAM_CNT); 757 } 758 759 sp<IMemory> sharedBuffer() const { return mSharedBuffer; } 760 761 bool presentationComplete(size_t framesWritten, size_t audioHalFrames); 762 763 public: 764 void triggerEvents(AudioSystem::sync_event_t type); 765 virtual bool isTimedTrack() const { return false; } 766 bool isFastTrack() const { return (mFlags & IAudioFlinger::TRACK_FAST) != 0; } 767 protected: 768 769 // we don't really need a lock for these 770 volatile bool mMute; 771 // FILLED state is used for suppressing volume ramp at begin of playing 772 enum {FS_INVALID, FS_FILLING, FS_FILLED, FS_ACTIVE}; 773 mutable uint8_t mFillingUpStatus; 774 int8_t mRetryCount; 775 const sp<IMemory> mSharedBuffer; 776 bool mResetDone; 777 const audio_stream_type_t mStreamType; 778 int mName; // track name on the normal mixer, 779 // allocated statically at track creation time, 780 // and is even allocated (though unused) for fast tracks 781 // FIXME don't allocate track name for fast tracks 782 int16_t *mMainBuffer; 783 int32_t *mAuxBuffer; 784 int mAuxEffectId; 785 bool mHasVolumeController; 786 size_t mPresentationCompleteFrames; // number of frames written to the audio HAL 787 // when this track will be fully rendered 788 private: 789 IAudioFlinger::track_flags_t mFlags; 790 791 // The following fields are only for fast tracks, and should be in a subclass 792 int mFastIndex; // index within FastMixerState::mFastTracks[]; 793 // either mFastIndex == -1 if not isFastTrack() 794 // or 0 < mFastIndex < FastMixerState::kMaxFast because 795 // index 0 is reserved for normal mixer's submix; 796 // index is allocated statically at track creation time 797 // but the slot is only used if track is active 798 FastTrackUnderruns mObservedUnderruns; // Most recently observed value of 799 // mFastMixerDumpState.mTracks[mFastIndex].mUnderruns 800 uint32_t mUnderrunCount; // Counter of total number of underruns, never reset 801 volatile float mCachedVolume; // combined master volume and stream type volume; 802 // 'volatile' means accessed without lock or 803 // barrier, but is read/written atomically 804 }; // end of Track 805 806 class TimedTrack : public Track { 807 public: 808 static sp<TimedTrack> create(PlaybackThread *thread, 809 const sp<Client>& client, 810 audio_stream_type_t streamType, 811 uint32_t sampleRate, 812 audio_format_t format, 813 uint32_t channelMask, 814 int frameCount, 815 const sp<IMemory>& sharedBuffer, 816 int sessionId); 817 ~TimedTrack(); 818 819 class TimedBuffer { 820 public: 821 TimedBuffer(); 822 TimedBuffer(const sp<IMemory>& buffer, int64_t pts); 823 const sp<IMemory>& buffer() const { return mBuffer; } 824 int64_t pts() const { return mPTS; } 825 uint32_t position() const { return mPosition; } 826 void setPosition(uint32_t pos) { mPosition = pos; } 827 private: 828 sp<IMemory> mBuffer; 829 int64_t mPTS; 830 uint32_t mPosition; 831 }; 832 833 // Mixer facing methods. 834 virtual bool isTimedTrack() const { return true; } 835 virtual size_t framesReady() const; 836 837 // AudioBufferProvider interface 838 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, 839 int64_t pts); 840 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 841 842 // Client/App facing methods. 843 status_t allocateTimedBuffer(size_t size, 844 sp<IMemory>* buffer); 845 status_t queueTimedBuffer(const sp<IMemory>& buffer, 846 int64_t pts); 847 status_t setMediaTimeTransform(const LinearTransform& xform, 848 TimedAudioTrack::TargetTimeline target); 849 850 private: 851 TimedTrack(PlaybackThread *thread, 852 const sp<Client>& client, 853 audio_stream_type_t streamType, 854 uint32_t sampleRate, 855 audio_format_t format, 856 uint32_t channelMask, 857 int frameCount, 858 const sp<IMemory>& sharedBuffer, 859 int sessionId); 860 861 void timedYieldSamples_l(AudioBufferProvider::Buffer* buffer); 862 void timedYieldSilence_l(uint32_t numFrames, 863 AudioBufferProvider::Buffer* buffer); 864 void trimTimedBufferQueue_l(); 865 void trimTimedBufferQueueHead_l(const char* logTag); 866 void updateFramesPendingAfterTrim_l(const TimedBuffer& buf, 867 const char* logTag); 868 869 uint64_t mLocalTimeFreq; 870 LinearTransform mLocalTimeToSampleTransform; 871 LinearTransform mMediaTimeToSampleTransform; 872 sp<MemoryDealer> mTimedMemoryDealer; 873 874 Vector<TimedBuffer> mTimedBufferQueue; 875 bool mQueueHeadInFlight; 876 bool mTrimQueueHeadOnRelease; 877 uint32_t mFramesPendingInQueue; 878 879 uint8_t* mTimedSilenceBuffer; 880 uint32_t mTimedSilenceBufferSize; 881 mutable Mutex mTimedBufferQueueLock; 882 bool mTimedAudioOutputOnTime; 883 CCHelper mCCHelper; 884 885 Mutex mMediaTimeTransformLock; 886 LinearTransform mMediaTimeTransform; 887 bool mMediaTimeTransformValid; 888 TimedAudioTrack::TargetTimeline mMediaTimeTransformTarget; 889 }; 890 891 892 // playback track 893 class OutputTrack : public Track { 894 public: 895 896 class Buffer: public AudioBufferProvider::Buffer { 897 public: 898 int16_t *mBuffer; 899 }; 900 901 OutputTrack(PlaybackThread *thread, 902 DuplicatingThread *sourceThread, 903 uint32_t sampleRate, 904 audio_format_t format, 905 uint32_t channelMask, 906 int frameCount); 907 virtual ~OutputTrack(); 908 909 virtual status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 910 int triggerSession = 0); 911 virtual void stop(); 912 bool write(int16_t* data, uint32_t frames); 913 bool bufferQueueEmpty() const { return mBufferQueue.size() == 0; } 914 bool isActive() const { return mActive; } 915 const wp<ThreadBase>& thread() const { return mThread; } 916 917 private: 918 919 enum { 920 NO_MORE_BUFFERS = 0x80000001, // same in AudioTrack.h, ok to be different value 921 }; 922 923 status_t obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs); 924 void clearBufferQueue(); 925 926 // Maximum number of pending buffers allocated by OutputTrack::write() 927 static const uint8_t kMaxOverFlowBuffers = 10; 928 929 Vector < Buffer* > mBufferQueue; 930 AudioBufferProvider::Buffer mOutBuffer; 931 bool mActive; 932 DuplicatingThread* const mSourceThread; // for waitTimeMs() in write() 933 }; // end of OutputTrack 934 935 PlaybackThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 936 audio_io_handle_t id, uint32_t device, type_t type); 937 virtual ~PlaybackThread(); 938 939 status_t dump(int fd, const Vector<String16>& args); 940 941 // Thread virtuals 942 virtual status_t readyToRun(); 943 virtual bool threadLoop(); 944 945 // RefBase 946 virtual void onFirstRef(); 947 948protected: 949 // Code snippets that were lifted up out of threadLoop() 950 virtual void threadLoop_mix() = 0; 951 virtual void threadLoop_sleepTime() = 0; 952 virtual void threadLoop_write(); 953 virtual void threadLoop_standby(); 954 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 955 956 // prepareTracks_l reads and writes mActiveTracks, and returns 957 // the pending set of tracks to remove via Vector 'tracksToRemove'. The caller 958 // is responsible for clearing or destroying this Vector later on, when it 959 // is safe to do so. That will drop the final ref count and destroy the tracks. 960 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove) = 0; 961 962public: 963 964 virtual status_t initCheck() const { return (mOutput == NULL) ? NO_INIT : NO_ERROR; } 965 966 // return estimated latency in milliseconds, as reported by HAL 967 uint32_t latency() const; 968 969 void setMasterVolume(float value); 970 void setMasterMute(bool muted); 971 972 void setStreamVolume(audio_stream_type_t stream, float value); 973 void setStreamMute(audio_stream_type_t stream, bool muted); 974 975 float streamVolume(audio_stream_type_t stream) const; 976 977 sp<Track> createTrack_l( 978 const sp<AudioFlinger::Client>& client, 979 audio_stream_type_t streamType, 980 uint32_t sampleRate, 981 audio_format_t format, 982 uint32_t channelMask, 983 int frameCount, 984 const sp<IMemory>& sharedBuffer, 985 int sessionId, 986 IAudioFlinger::track_flags_t flags, 987 pid_t tid, 988 status_t *status); 989 990 AudioStreamOut* getOutput() const; 991 AudioStreamOut* clearOutput(); 992 virtual audio_stream_t* stream() const; 993 994 void suspend() { mSuspended++; } 995 void restore() { if (mSuspended > 0) mSuspended--; } 996 bool isSuspended() const { return (mSuspended > 0); } 997 virtual String8 getParameters(const String8& keys); 998 virtual void audioConfigChanged_l(int event, int param = 0); 999 status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames); 1000 int16_t *mixBuffer() const { return mMixBuffer; }; 1001 1002 virtual void detachAuxEffect_l(int effectId); 1003 status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track> track, 1004 int EffectId); 1005 status_t attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track> track, 1006 int EffectId); 1007 1008 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1009 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1010 virtual uint32_t hasAudioSession(int sessionId); 1011 virtual uint32_t getStrategyForSession_l(int sessionId); 1012 1013 1014 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 1015 virtual bool isValidSyncEvent(const sp<SyncEvent>& event); 1016 1017 protected: 1018 int16_t* mMixBuffer; 1019 uint32_t mSuspended; // suspend count, > 0 means suspended 1020 int mBytesWritten; 1021 private: 1022 // mMasterMute is in both PlaybackThread and in AudioFlinger. When a 1023 // PlaybackThread needs to find out if master-muted, it checks it's local 1024 // copy rather than the one in AudioFlinger. This optimization saves a lock. 1025 bool mMasterMute; 1026 void setMasterMute_l(bool muted) { mMasterMute = muted; } 1027 protected: 1028 SortedVector< wp<Track> > mActiveTracks; // FIXME check if this could be sp<> 1029 1030 // Allocate a track name for a given channel mask. 1031 // Returns name >= 0 if successful, -1 on failure. 1032 virtual int getTrackName_l(audio_channel_mask_t channelMask) = 0; 1033 virtual void deleteTrackName_l(int name) = 0; 1034 1035 // Time to sleep between cycles when: 1036 virtual uint32_t activeSleepTimeUs() const; // mixer state MIXER_TRACKS_ENABLED 1037 virtual uint32_t idleSleepTimeUs() const = 0; // mixer state MIXER_IDLE 1038 virtual uint32_t suspendSleepTimeUs() const = 0; // audio policy manager suspended us 1039 // No sleep when mixer state == MIXER_TRACKS_READY; relies on audio HAL stream->write() 1040 // No sleep in standby mode; waits on a condition 1041 1042 // Code snippets that are temporarily lifted up out of threadLoop() until the merge 1043 void checkSilentMode_l(); 1044 1045 // Non-trivial for DUPLICATING only 1046 virtual void saveOutputTracks() { } 1047 virtual void clearOutputTracks() { } 1048 1049 // Cache various calculated values, at threadLoop() entry and after a parameter change 1050 virtual void cacheParameters_l(); 1051 1052 virtual uint32_t correctLatency(uint32_t latency) const; 1053 1054 private: 1055 1056 friend class AudioFlinger; // for numerous 1057 1058 PlaybackThread(const Client&); 1059 PlaybackThread& operator = (const PlaybackThread&); 1060 1061 status_t addTrack_l(const sp<Track>& track); 1062 void destroyTrack_l(const sp<Track>& track); 1063 void removeTrack_l(const sp<Track>& track); 1064 1065 void readOutputParameters(); 1066 1067 virtual status_t dumpInternals(int fd, const Vector<String16>& args); 1068 status_t dumpTracks(int fd, const Vector<String16>& args); 1069 1070 SortedVector< sp<Track> > mTracks; 1071 // mStreamTypes[] uses 1 additional stream type internally for the OutputTrack used by DuplicatingThread 1072 stream_type_t mStreamTypes[AUDIO_STREAM_CNT + 1]; 1073 AudioStreamOut *mOutput; 1074 float mMasterVolume; 1075 nsecs_t mLastWriteTime; 1076 int mNumWrites; 1077 int mNumDelayedWrites; 1078 bool mInWrite; 1079 1080 // FIXME rename these former local variables of threadLoop to standard "m" names 1081 nsecs_t standbyTime; 1082 size_t mixBufferSize; 1083 1084 // cached copies of activeSleepTimeUs() and idleSleepTimeUs() made by cacheParameters_l() 1085 uint32_t activeSleepTime; 1086 uint32_t idleSleepTime; 1087 1088 uint32_t sleepTime; 1089 1090 // mixer status returned by prepareTracks_l() 1091 mixer_state mMixerStatus; // current cycle 1092 // previous cycle when in prepareTracks_l() 1093 mixer_state mMixerStatusIgnoringFastTracks; 1094 // FIXME or a separate ready state per track 1095 1096 // FIXME move these declarations into the specific sub-class that needs them 1097 // MIXER only 1098 bool longStandbyExit; 1099 uint32_t sleepTimeShift; 1100 1101 // same as AudioFlinger::mStandbyTimeInNsecs except for DIRECT which uses a shorter value 1102 nsecs_t standbyDelay; 1103 1104 // MIXER only 1105 nsecs_t maxPeriod; 1106 1107 // DUPLICATING only 1108 uint32_t writeFrames; 1109 1110 private: 1111 // The HAL output sink is treated as non-blocking, but current implementation is blocking 1112 sp<NBAIO_Sink> mOutputSink; 1113 // If a fast mixer is present, the blocking pipe sink, otherwise clear 1114 sp<NBAIO_Sink> mPipeSink; 1115 // The current sink for the normal mixer to write it's (sub)mix, mOutputSink or mPipeSink 1116 sp<NBAIO_Sink> mNormalSink; 1117 // For dumpsys 1118 sp<NBAIO_Sink> mTeeSink; 1119 sp<NBAIO_Source> mTeeSource; 1120 public: 1121 virtual bool hasFastMixer() const = 0; 1122 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const 1123 { FastTrackUnderruns dummy; return dummy; } 1124 1125 protected: 1126 // accessed by both binder threads and within threadLoop(), lock on mutex needed 1127 unsigned mFastTrackAvailMask; // bit i set if fast track [i] is available 1128 1129 }; 1130 1131 class MixerThread : public PlaybackThread { 1132 public: 1133 MixerThread (const sp<AudioFlinger>& audioFlinger, 1134 AudioStreamOut* output, 1135 audio_io_handle_t id, 1136 uint32_t device, 1137 type_t type = MIXER); 1138 virtual ~MixerThread(); 1139 1140 // Thread virtuals 1141 1142 void invalidateTracks(audio_stream_type_t streamType); 1143 virtual bool checkForNewParameters_l(); 1144 virtual status_t dumpInternals(int fd, const Vector<String16>& args); 1145 1146 protected: 1147 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1148 virtual int getTrackName_l(audio_channel_mask_t channelMask); 1149 virtual void deleteTrackName_l(int name); 1150 virtual uint32_t idleSleepTimeUs() const; 1151 virtual uint32_t suspendSleepTimeUs() const; 1152 virtual void cacheParameters_l(); 1153 1154 // threadLoop snippets 1155 virtual void threadLoop_write(); 1156 virtual void threadLoop_standby(); 1157 virtual void threadLoop_mix(); 1158 virtual void threadLoop_sleepTime(); 1159 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 1160 virtual uint32_t correctLatency(uint32_t latency) const; 1161 1162 AudioMixer* mAudioMixer; // normal mixer 1163 private: 1164#ifdef SOAKER 1165 Thread* mSoaker; 1166#endif 1167 // one-time initialization, no locks required 1168 FastMixer* mFastMixer; // non-NULL if there is also a fast mixer 1169 1170 // contents are not guaranteed to be consistent, no locks required 1171 FastMixerDumpState mFastMixerDumpState; 1172#ifdef STATE_QUEUE_DUMP 1173 StateQueueObserverDump mStateQueueObserverDump; 1174 StateQueueMutatorDump mStateQueueMutatorDump; 1175#endif 1176 1177 // accessible only within the threadLoop(), no locks required 1178 // mFastMixer->sq() // for mutating and pushing state 1179 int32_t mFastMixerFutex; // for cold idle 1180 1181 public: 1182 virtual bool hasFastMixer() const { return mFastMixer != NULL; } 1183 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const { 1184 ALOG_ASSERT(fastIndex < FastMixerState::kMaxFastTracks); 1185 return mFastMixerDumpState.mTracks[fastIndex].mUnderruns; 1186 } 1187 }; 1188 1189 class DirectOutputThread : public PlaybackThread { 1190 public: 1191 1192 DirectOutputThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1193 audio_io_handle_t id, uint32_t device); 1194 virtual ~DirectOutputThread(); 1195 1196 // Thread virtuals 1197 1198 virtual bool checkForNewParameters_l(); 1199 1200 protected: 1201 virtual int getTrackName_l(audio_channel_mask_t channelMask); 1202 virtual void deleteTrackName_l(int name); 1203 virtual uint32_t activeSleepTimeUs() const; 1204 virtual uint32_t idleSleepTimeUs() const; 1205 virtual uint32_t suspendSleepTimeUs() const; 1206 virtual void cacheParameters_l(); 1207 1208 // threadLoop snippets 1209 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1210 virtual void threadLoop_mix(); 1211 virtual void threadLoop_sleepTime(); 1212 1213 // volumes last sent to audio HAL with stream->set_volume() 1214 float mLeftVolFloat; 1215 float mRightVolFloat; 1216 1217private: 1218 // prepareTracks_l() tells threadLoop_mix() the name of the single active track 1219 sp<Track> mActiveTrack; 1220 public: 1221 virtual bool hasFastMixer() const { return false; } 1222 }; 1223 1224 class DuplicatingThread : public MixerThread { 1225 public: 1226 DuplicatingThread (const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread, 1227 audio_io_handle_t id); 1228 virtual ~DuplicatingThread(); 1229 1230 // Thread virtuals 1231 void addOutputTrack(MixerThread* thread); 1232 void removeOutputTrack(MixerThread* thread); 1233 uint32_t waitTimeMs() const { return mWaitTimeMs; } 1234 protected: 1235 virtual uint32_t activeSleepTimeUs() const; 1236 1237 private: 1238 bool outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks); 1239 protected: 1240 // threadLoop snippets 1241 virtual void threadLoop_mix(); 1242 virtual void threadLoop_sleepTime(); 1243 virtual void threadLoop_write(); 1244 virtual void threadLoop_standby(); 1245 virtual void cacheParameters_l(); 1246 1247 private: 1248 // called from threadLoop, addOutputTrack, removeOutputTrack 1249 virtual void updateWaitTime_l(); 1250 protected: 1251 virtual void saveOutputTracks(); 1252 virtual void clearOutputTracks(); 1253 private: 1254 1255 uint32_t mWaitTimeMs; 1256 SortedVector < sp<OutputTrack> > outputTracks; 1257 SortedVector < sp<OutputTrack> > mOutputTracks; 1258 public: 1259 virtual bool hasFastMixer() const { return false; } 1260 }; 1261 1262 PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const; 1263 MixerThread *checkMixerThread_l(audio_io_handle_t output) const; 1264 RecordThread *checkRecordThread_l(audio_io_handle_t input) const; 1265 // no range check, AudioFlinger::mLock held 1266 bool streamMute_l(audio_stream_type_t stream) const 1267 { return mStreamTypes[stream].mute; } 1268 // no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held 1269 float streamVolume_l(audio_stream_type_t stream) const 1270 { return mStreamTypes[stream].volume; } 1271 void audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2); 1272 1273 // allocate an audio_io_handle_t, session ID, or effect ID 1274 uint32_t nextUniqueId(); 1275 1276 status_t moveEffectChain_l(int sessionId, 1277 PlaybackThread *srcThread, 1278 PlaybackThread *dstThread, 1279 bool reRegister); 1280 // return thread associated with primary hardware device, or NULL 1281 PlaybackThread *primaryPlaybackThread_l() const; 1282 uint32_t primaryOutputDevice_l() const; 1283 1284 // server side of the client's IAudioTrack 1285 class TrackHandle : public android::BnAudioTrack { 1286 public: 1287 TrackHandle(const sp<PlaybackThread::Track>& track); 1288 virtual ~TrackHandle(); 1289 virtual sp<IMemory> getCblk() const; 1290 virtual status_t start(); 1291 virtual void stop(); 1292 virtual void flush(); 1293 virtual void mute(bool); 1294 virtual void pause(); 1295 virtual status_t attachAuxEffect(int effectId); 1296 virtual status_t allocateTimedBuffer(size_t size, 1297 sp<IMemory>* buffer); 1298 virtual status_t queueTimedBuffer(const sp<IMemory>& buffer, 1299 int64_t pts); 1300 virtual status_t setMediaTimeTransform(const LinearTransform& xform, 1301 int target); 1302 virtual status_t onTransact( 1303 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 1304 private: 1305 const sp<PlaybackThread::Track> mTrack; 1306 }; 1307 1308 void removeClient_l(pid_t pid); 1309 void removeNotificationClient(pid_t pid); 1310 1311 1312 // record thread 1313 class RecordThread : public ThreadBase, public AudioBufferProvider 1314 { 1315 public: 1316 1317 // record track 1318 class RecordTrack : public TrackBase { 1319 public: 1320 RecordTrack(RecordThread *thread, 1321 const sp<Client>& client, 1322 uint32_t sampleRate, 1323 audio_format_t format, 1324 uint32_t channelMask, 1325 int frameCount, 1326 int sessionId); 1327 virtual ~RecordTrack(); 1328 1329 virtual status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 1330 int triggerSession = 0); 1331 virtual void stop(); 1332 1333 bool overflow() { bool tmp = mOverflow; mOverflow = false; return tmp; } 1334 bool setOverflow() { bool tmp = mOverflow; mOverflow = true; return tmp; } 1335 1336 void dump(char* buffer, size_t size); 1337 1338 private: 1339 friend class AudioFlinger; // for mState 1340 1341 RecordTrack(const RecordTrack&); 1342 RecordTrack& operator = (const RecordTrack&); 1343 1344 // AudioBufferProvider interface 1345 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts = kInvalidPTS); 1346 // releaseBuffer() not overridden 1347 1348 bool mOverflow; 1349 }; 1350 1351 1352 RecordThread(const sp<AudioFlinger>& audioFlinger, 1353 AudioStreamIn *input, 1354 uint32_t sampleRate, 1355 uint32_t channels, 1356 audio_io_handle_t id, 1357 uint32_t device); 1358 virtual ~RecordThread(); 1359 1360 // Thread 1361 virtual bool threadLoop(); 1362 virtual status_t readyToRun(); 1363 1364 // RefBase 1365 virtual void onFirstRef(); 1366 1367 virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; } 1368 sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l( 1369 const sp<AudioFlinger::Client>& client, 1370 uint32_t sampleRate, 1371 audio_format_t format, 1372 int channelMask, 1373 int frameCount, 1374 int sessionId, 1375 status_t *status); 1376 1377 status_t start(RecordTrack* recordTrack, 1378 AudioSystem::sync_event_t event, 1379 int triggerSession); 1380 void stop(RecordTrack* recordTrack); 1381 status_t dump(int fd, const Vector<String16>& args); 1382 AudioStreamIn* getInput() const; 1383 AudioStreamIn* clearInput(); 1384 virtual audio_stream_t* stream() const; 1385 1386 // AudioBufferProvider interface 1387 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts); 1388 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 1389 1390 virtual bool checkForNewParameters_l(); 1391 virtual String8 getParameters(const String8& keys); 1392 virtual void audioConfigChanged_l(int event, int param = 0); 1393 void readInputParameters(); 1394 virtual unsigned int getInputFramesLost(); 1395 1396 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1397 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1398 virtual uint32_t hasAudioSession(int sessionId); 1399 RecordTrack* track(); 1400 1401 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 1402 virtual bool isValidSyncEvent(const sp<SyncEvent>& event); 1403 1404 static void syncStartEventCallback(const wp<SyncEvent>& event); 1405 void handleSyncStartEvent(const sp<SyncEvent>& event); 1406 1407 private: 1408 void clearSyncStartEvent(); 1409 1410 RecordThread(); 1411 AudioStreamIn *mInput; 1412 RecordTrack* mTrack; 1413 sp<RecordTrack> mActiveTrack; 1414 Condition mStartStopCond; 1415 AudioResampler *mResampler; 1416 int32_t *mRsmpOutBuffer; 1417 int16_t *mRsmpInBuffer; 1418 size_t mRsmpInIndex; 1419 size_t mInputBytes; 1420 const int mReqChannelCount; 1421 const uint32_t mReqSampleRate; 1422 ssize_t mBytesRead; 1423 // sync event triggering actual audio capture. Frames read before this event will 1424 // be dropped and therefore not read by the application. 1425 sp<SyncEvent> mSyncStartEvent; 1426 // number of captured frames to drop after the start sync event has been received. 1427 // when < 0, maximum frames to drop before starting capture even if sync event is 1428 // not received 1429 ssize_t mFramestoDrop; 1430 }; 1431 1432 // server side of the client's IAudioRecord 1433 class RecordHandle : public android::BnAudioRecord { 1434 public: 1435 RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack); 1436 virtual ~RecordHandle(); 1437 virtual sp<IMemory> getCblk() const; 1438 virtual status_t start(int event, int triggerSession); 1439 virtual void stop(); 1440 virtual status_t onTransact( 1441 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 1442 private: 1443 const sp<RecordThread::RecordTrack> mRecordTrack; 1444 }; 1445 1446 //--- Audio Effect Management 1447 1448 // EffectModule and EffectChain classes both have their own mutex to protect 1449 // state changes or resource modifications. Always respect the following order 1450 // if multiple mutexes must be acquired to avoid cross deadlock: 1451 // AudioFlinger -> ThreadBase -> EffectChain -> EffectModule 1452 1453 // The EffectModule class is a wrapper object controlling the effect engine implementation 1454 // in the effect library. It prevents concurrent calls to process() and command() functions 1455 // from different client threads. It keeps a list of EffectHandle objects corresponding 1456 // to all client applications using this effect and notifies applications of effect state, 1457 // control or parameter changes. It manages the activation state machine to send appropriate 1458 // reset, enable, disable commands to effect engine and provide volume 1459 // ramping when effects are activated/deactivated. 1460 // When controlling an auxiliary effect, the EffectModule also provides an input buffer used by 1461 // the attached track(s) to accumulate their auxiliary channel. 1462 class EffectModule: public RefBase { 1463 public: 1464 EffectModule(ThreadBase *thread, 1465 const wp<AudioFlinger::EffectChain>& chain, 1466 effect_descriptor_t *desc, 1467 int id, 1468 int sessionId); 1469 virtual ~EffectModule(); 1470 1471 enum effect_state { 1472 IDLE, 1473 RESTART, 1474 STARTING, 1475 ACTIVE, 1476 STOPPING, 1477 STOPPED, 1478 DESTROYED 1479 }; 1480 1481 int id() const { return mId; } 1482 void process(); 1483 void updateState(); 1484 status_t command(uint32_t cmdCode, 1485 uint32_t cmdSize, 1486 void *pCmdData, 1487 uint32_t *replySize, 1488 void *pReplyData); 1489 1490 void reset_l(); 1491 status_t configure(); 1492 status_t init(); 1493 effect_state state() const { 1494 return mState; 1495 } 1496 uint32_t status() { 1497 return mStatus; 1498 } 1499 int sessionId() const { 1500 return mSessionId; 1501 } 1502 status_t setEnabled(bool enabled); 1503 bool isEnabled() const; 1504 bool isProcessEnabled() const; 1505 1506 void setInBuffer(int16_t *buffer) { mConfig.inputCfg.buffer.s16 = buffer; } 1507 int16_t *inBuffer() { return mConfig.inputCfg.buffer.s16; } 1508 void setOutBuffer(int16_t *buffer) { mConfig.outputCfg.buffer.s16 = buffer; } 1509 int16_t *outBuffer() { return mConfig.outputCfg.buffer.s16; } 1510 void setChain(const wp<EffectChain>& chain) { mChain = chain; } 1511 void setThread(const wp<ThreadBase>& thread) { mThread = thread; } 1512 const wp<ThreadBase>& thread() { return mThread; } 1513 1514 status_t addHandle(const sp<EffectHandle>& handle); 1515 void disconnect(const wp<EffectHandle>& handle, bool unpinIfLast); 1516 size_t removeHandle (const wp<EffectHandle>& handle); 1517 1518 effect_descriptor_t& desc() { return mDescriptor; } 1519 wp<EffectChain>& chain() { return mChain; } 1520 1521 status_t setDevice(uint32_t device); 1522 status_t setVolume(uint32_t *left, uint32_t *right, bool controller); 1523 status_t setMode(audio_mode_t mode); 1524 status_t start(); 1525 status_t stop(); 1526 void setSuspended(bool suspended); 1527 bool suspended() const; 1528 1529 sp<EffectHandle> controlHandle(); 1530 1531 bool isPinned() const { return mPinned; } 1532 void unPin() { mPinned = false; } 1533 1534 status_t dump(int fd, const Vector<String16>& args); 1535 1536 protected: 1537 friend class AudioFlinger; // for mHandles 1538 bool mPinned; 1539 1540 // Maximum time allocated to effect engines to complete the turn off sequence 1541 static const uint32_t MAX_DISABLE_TIME_MS = 10000; 1542 1543 EffectModule(const EffectModule&); 1544 EffectModule& operator = (const EffectModule&); 1545 1546 status_t start_l(); 1547 status_t stop_l(); 1548 1549mutable Mutex mLock; // mutex for process, commands and handles list protection 1550 wp<ThreadBase> mThread; // parent thread 1551 wp<EffectChain> mChain; // parent effect chain 1552 int mId; // this instance unique ID 1553 int mSessionId; // audio session ID 1554 effect_descriptor_t mDescriptor;// effect descriptor received from effect engine 1555 effect_config_t mConfig; // input and output audio configuration 1556 effect_handle_t mEffectInterface; // Effect module C API 1557 status_t mStatus; // initialization status 1558 effect_state mState; // current activation state 1559 Vector< wp<EffectHandle> > mHandles; // list of client handles 1560 // First handle in mHandles has highest priority and controls the effect module 1561 uint32_t mMaxDisableWaitCnt; // maximum grace period before forcing an effect off after 1562 // sending disable command. 1563 uint32_t mDisableWaitCnt; // current process() calls count during disable period. 1564 bool mSuspended; // effect is suspended: temporarily disabled by framework 1565 }; 1566 1567 // The EffectHandle class implements the IEffect interface. It provides resources 1568 // to receive parameter updates, keeps track of effect control 1569 // ownership and state and has a pointer to the EffectModule object it is controlling. 1570 // There is one EffectHandle object for each application controlling (or using) 1571 // an effect module. 1572 // The EffectHandle is obtained by calling AudioFlinger::createEffect(). 1573 class EffectHandle: public android::BnEffect { 1574 public: 1575 1576 EffectHandle(const sp<EffectModule>& effect, 1577 const sp<AudioFlinger::Client>& client, 1578 const sp<IEffectClient>& effectClient, 1579 int32_t priority); 1580 virtual ~EffectHandle(); 1581 1582 // IEffect 1583 virtual status_t enable(); 1584 virtual status_t disable(); 1585 virtual status_t command(uint32_t cmdCode, 1586 uint32_t cmdSize, 1587 void *pCmdData, 1588 uint32_t *replySize, 1589 void *pReplyData); 1590 virtual void disconnect(); 1591 private: 1592 void disconnect(bool unpinIfLast); 1593 public: 1594 virtual sp<IMemory> getCblk() const { return mCblkMemory; } 1595 virtual status_t onTransact(uint32_t code, const Parcel& data, 1596 Parcel* reply, uint32_t flags); 1597 1598 1599 // Give or take control of effect module 1600 // - hasControl: true if control is given, false if removed 1601 // - signal: true client app should be signaled of change, false otherwise 1602 // - enabled: state of the effect when control is passed 1603 void setControl(bool hasControl, bool signal, bool enabled); 1604 void commandExecuted(uint32_t cmdCode, 1605 uint32_t cmdSize, 1606 void *pCmdData, 1607 uint32_t replySize, 1608 void *pReplyData); 1609 void setEnabled(bool enabled); 1610 bool enabled() const { return mEnabled; } 1611 1612 // Getters 1613 int id() const { return mEffect->id(); } 1614 int priority() const { return mPriority; } 1615 bool hasControl() const { return mHasControl; } 1616 sp<EffectModule> effect() const { return mEffect; } 1617 1618 void dump(char* buffer, size_t size); 1619 1620 protected: 1621 friend class AudioFlinger; // for mEffect, mHasControl, mEnabled 1622 EffectHandle(const EffectHandle&); 1623 EffectHandle& operator =(const EffectHandle&); 1624 1625 sp<EffectModule> mEffect; // pointer to controlled EffectModule 1626 sp<IEffectClient> mEffectClient; // callback interface for client notifications 1627 /*const*/ sp<Client> mClient; // client for shared memory allocation, see disconnect() 1628 sp<IMemory> mCblkMemory; // shared memory for control block 1629 effect_param_cblk_t* mCblk; // control block for deferred parameter setting via shared memory 1630 uint8_t* mBuffer; // pointer to parameter area in shared memory 1631 int mPriority; // client application priority to control the effect 1632 bool mHasControl; // true if this handle is controlling the effect 1633 bool mEnabled; // cached enable state: needed when the effect is 1634 // restored after being suspended 1635 }; 1636 1637 // the EffectChain class represents a group of effects associated to one audio session. 1638 // There can be any number of EffectChain objects per output mixer thread (PlaybackThread). 1639 // The EffecChain with session ID 0 contains global effects applied to the output mix. 1640 // Effects in this chain can be insert or auxiliary. Effects in other chains (attached to tracks) 1641 // are insert only. The EffectChain maintains an ordered list of effect module, the order corresponding 1642 // in the effect process order. When attached to a track (session ID != 0), it also provide it's own 1643 // input buffer used by the track as accumulation buffer. 1644 class EffectChain: public RefBase { 1645 public: 1646 EffectChain(const wp<ThreadBase>& wThread, int sessionId); 1647 EffectChain(ThreadBase *thread, int sessionId); 1648 virtual ~EffectChain(); 1649 1650 // special key used for an entry in mSuspendedEffects keyed vector 1651 // corresponding to a suspend all request. 1652 static const int kKeyForSuspendAll = 0; 1653 1654 // minimum duration during which we force calling effect process when last track on 1655 // a session is stopped or removed to allow effect tail to be rendered 1656 static const int kProcessTailDurationMs = 1000; 1657 1658 void process_l(); 1659 1660 void lock() { 1661 mLock.lock(); 1662 } 1663 void unlock() { 1664 mLock.unlock(); 1665 } 1666 1667 status_t addEffect_l(const sp<EffectModule>& handle); 1668 size_t removeEffect_l(const sp<EffectModule>& handle); 1669 1670 int sessionId() const { return mSessionId; } 1671 void setSessionId(int sessionId) { mSessionId = sessionId; } 1672 1673 sp<EffectModule> getEffectFromDesc_l(effect_descriptor_t *descriptor); 1674 sp<EffectModule> getEffectFromId_l(int id); 1675 sp<EffectModule> getEffectFromType_l(const effect_uuid_t *type); 1676 bool setVolume_l(uint32_t *left, uint32_t *right); 1677 void setDevice_l(uint32_t device); 1678 void setMode_l(audio_mode_t mode); 1679 1680 void setInBuffer(int16_t *buffer, bool ownsBuffer = false) { 1681 mInBuffer = buffer; 1682 mOwnInBuffer = ownsBuffer; 1683 } 1684 int16_t *inBuffer() const { 1685 return mInBuffer; 1686 } 1687 void setOutBuffer(int16_t *buffer) { 1688 mOutBuffer = buffer; 1689 } 1690 int16_t *outBuffer() const { 1691 return mOutBuffer; 1692 } 1693 1694 void incTrackCnt() { android_atomic_inc(&mTrackCnt); } 1695 void decTrackCnt() { android_atomic_dec(&mTrackCnt); } 1696 int32_t trackCnt() const { return mTrackCnt;} 1697 1698 void incActiveTrackCnt() { android_atomic_inc(&mActiveTrackCnt); 1699 mTailBufferCount = mMaxTailBuffers; } 1700 void decActiveTrackCnt() { android_atomic_dec(&mActiveTrackCnt); } 1701 int32_t activeTrackCnt() const { return mActiveTrackCnt;} 1702 1703 uint32_t strategy() const { return mStrategy; } 1704 void setStrategy(uint32_t strategy) 1705 { mStrategy = strategy; } 1706 1707 // suspend effect of the given type 1708 void setEffectSuspended_l(const effect_uuid_t *type, 1709 bool suspend); 1710 // suspend all eligible effects 1711 void setEffectSuspendedAll_l(bool suspend); 1712 // check if effects should be suspend or restored when a given effect is enable or disabled 1713 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1714 bool enabled); 1715 1716 void clearInputBuffer(); 1717 1718 status_t dump(int fd, const Vector<String16>& args); 1719 1720 protected: 1721 friend class AudioFlinger; // for mThread, mEffects 1722 EffectChain(const EffectChain&); 1723 EffectChain& operator =(const EffectChain&); 1724 1725 class SuspendedEffectDesc : public RefBase { 1726 public: 1727 SuspendedEffectDesc() : mRefCount(0) {} 1728 1729 int mRefCount; 1730 effect_uuid_t mType; 1731 wp<EffectModule> mEffect; 1732 }; 1733 1734 // get a list of effect modules to suspend when an effect of the type 1735 // passed is enabled. 1736 void getSuspendEligibleEffects(Vector< sp<EffectModule> > &effects); 1737 1738 // get an effect module if it is currently enable 1739 sp<EffectModule> getEffectIfEnabled(const effect_uuid_t *type); 1740 // true if the effect whose descriptor is passed can be suspended 1741 // OEMs can modify the rules implemented in this method to exclude specific effect 1742 // types or implementations from the suspend/restore mechanism. 1743 bool isEffectEligibleForSuspend(const effect_descriptor_t& desc); 1744 1745 void clearInputBuffer_l(sp<ThreadBase> thread); 1746 1747 wp<ThreadBase> mThread; // parent mixer thread 1748 Mutex mLock; // mutex protecting effect list 1749 Vector< sp<EffectModule> > mEffects; // list of effect modules 1750 int mSessionId; // audio session ID 1751 int16_t *mInBuffer; // chain input buffer 1752 int16_t *mOutBuffer; // chain output buffer 1753 volatile int32_t mActiveTrackCnt; // number of active tracks connected 1754 volatile int32_t mTrackCnt; // number of tracks connected 1755 int32_t mTailBufferCount; // current effect tail buffer count 1756 int32_t mMaxTailBuffers; // maximum effect tail buffers 1757 bool mOwnInBuffer; // true if the chain owns its input buffer 1758 int mVolumeCtrlIdx; // index of insert effect having control over volume 1759 uint32_t mLeftVolume; // previous volume on left channel 1760 uint32_t mRightVolume; // previous volume on right channel 1761 uint32_t mNewLeftVolume; // new volume on left channel 1762 uint32_t mNewRightVolume; // new volume on right channel 1763 uint32_t mStrategy; // strategy for this effect chain 1764 // mSuspendedEffects lists all effects currently suspended in the chain. 1765 // Use effect type UUID timelow field as key. There is no real risk of identical 1766 // timeLow fields among effect type UUIDs. 1767 // Updated by updateSuspendedSessions_l() only. 1768 KeyedVector< int, sp<SuspendedEffectDesc> > mSuspendedEffects; 1769 }; 1770 1771 // AudioStreamOut and AudioStreamIn are immutable, so their fields are const. 1772 // For emphasis, we could also make all pointers to them be "const *", 1773 // but that would clutter the code unnecessarily. 1774 1775 struct AudioStreamOut { 1776 audio_hw_device_t* const hwDev; 1777 audio_stream_out_t* const stream; 1778 1779 AudioStreamOut(audio_hw_device_t *dev, audio_stream_out_t *out) : 1780 hwDev(dev), stream(out) {} 1781 }; 1782 1783 struct AudioStreamIn { 1784 audio_hw_device_t* const hwDev; 1785 audio_stream_in_t* const stream; 1786 1787 AudioStreamIn(audio_hw_device_t *dev, audio_stream_in_t *in) : 1788 hwDev(dev), stream(in) {} 1789 }; 1790 1791 // for mAudioSessionRefs only 1792 struct AudioSessionRef { 1793 AudioSessionRef(int sessionid, pid_t pid) : 1794 mSessionid(sessionid), mPid(pid), mCnt(1) {} 1795 const int mSessionid; 1796 const pid_t mPid; 1797 int mCnt; 1798 }; 1799 1800 enum master_volume_support { 1801 // MVS_NONE: 1802 // Audio HAL has no support for master volume, either setting or 1803 // getting. All master volume control must be implemented in SW by the 1804 // AudioFlinger mixing core. 1805 MVS_NONE, 1806 1807 // MVS_SETONLY: 1808 // Audio HAL has support for setting master volume, but not for getting 1809 // master volume (original HAL design did not include a getter). 1810 // AudioFlinger needs to keep track of the last set master volume in 1811 // addition to needing to set an initial, default, master volume at HAL 1812 // load time. 1813 MVS_SETONLY, 1814 1815 // MVS_FULL: 1816 // Audio HAL has support both for setting and getting master volume. 1817 // AudioFlinger should send all set and get master volume requests 1818 // directly to the HAL. 1819 MVS_FULL, 1820 }; 1821 1822 class AudioHwDevice { 1823 public: 1824 AudioHwDevice(const char *moduleName, audio_hw_device_t *hwDevice) : 1825 mModuleName(strdup(moduleName)), mHwDevice(hwDevice){} 1826 ~AudioHwDevice() { free((void *)mModuleName); } 1827 1828 const char *moduleName() const { return mModuleName; } 1829 audio_hw_device_t *hwDevice() const { return mHwDevice; } 1830 private: 1831 const char * const mModuleName; 1832 audio_hw_device_t * const mHwDevice; 1833 }; 1834 1835 mutable Mutex mLock; 1836 1837 DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client() 1838 1839 mutable Mutex mHardwareLock; 1840 // NOTE: If both mLock and mHardwareLock mutexes must be held, 1841 // always take mLock before mHardwareLock 1842 1843 // These two fields are immutable after onFirstRef(), so no lock needed to access 1844 audio_hw_device_t* mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL 1845 DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*> mAudioHwDevs; 1846 1847 // for dump, indicates which hardware operation is currently in progress (but not stream ops) 1848 enum hardware_call_state { 1849 AUDIO_HW_IDLE = 0, // no operation in progress 1850 AUDIO_HW_INIT, // init_check 1851 AUDIO_HW_OUTPUT_OPEN, // open_output_stream 1852 AUDIO_HW_OUTPUT_CLOSE, // unused 1853 AUDIO_HW_INPUT_OPEN, // unused 1854 AUDIO_HW_INPUT_CLOSE, // unused 1855 AUDIO_HW_STANDBY, // unused 1856 AUDIO_HW_SET_MASTER_VOLUME, // set_master_volume 1857 AUDIO_HW_GET_ROUTING, // unused 1858 AUDIO_HW_SET_ROUTING, // unused 1859 AUDIO_HW_GET_MODE, // unused 1860 AUDIO_HW_SET_MODE, // set_mode 1861 AUDIO_HW_GET_MIC_MUTE, // get_mic_mute 1862 AUDIO_HW_SET_MIC_MUTE, // set_mic_mute 1863 AUDIO_HW_SET_VOICE_VOLUME, // set_voice_volume 1864 AUDIO_HW_SET_PARAMETER, // set_parameters 1865 AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size 1866 AUDIO_HW_GET_MASTER_VOLUME, // get_master_volume 1867 AUDIO_HW_GET_PARAMETER, // get_parameters 1868 }; 1869 1870 mutable hardware_call_state mHardwareStatus; // for dump only 1871 1872 1873 DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads; 1874 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 1875 1876 // both are protected by mLock 1877 float mMasterVolume; 1878 float mMasterVolumeSW; 1879 master_volume_support mMasterVolumeSupportLvl; 1880 bool mMasterMute; 1881 1882 DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads; 1883 1884 DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients; 1885 volatile int32_t mNextUniqueId; // updated by android_atomic_inc 1886 audio_mode_t mMode; 1887 bool mBtNrecIsOff; 1888 1889 // protected by mLock 1890 Vector<AudioSessionRef*> mAudioSessionRefs; 1891 1892 float masterVolume_l() const; 1893 float masterVolumeSW_l() const { return mMasterVolumeSW; } 1894 bool masterMute_l() const { return mMasterMute; } 1895 audio_module_handle_t loadHwModule_l(const char *name); 1896 1897 Vector < sp<SyncEvent> > mPendingSyncEvents; // sync events awaiting for a session 1898 // to be created 1899 1900private: 1901 sp<Client> registerPid_l(pid_t pid); // always returns non-0 1902 1903}; 1904 1905 1906// ---------------------------------------------------------------------------- 1907 1908}; // namespace android 1909 1910#endif // ANDROID_AUDIO_FLINGER_H 1911