AudioFlinger.h revision 717e128691f083a9469a1d0e363ac6ecd5c65d58
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); 256 257private: 258 audio_mode_t getMode() const { return mMode; } 259 260 bool btNrecIsOff() const { return mBtNrecIsOff; } 261 262 AudioFlinger(); 263 virtual ~AudioFlinger(); 264 265 // call in any IAudioFlinger method that accesses mPrimaryHardwareDev 266 status_t initCheck() const { return mPrimaryHardwareDev == NULL ? NO_INIT : NO_ERROR; } 267 268 // RefBase 269 virtual void onFirstRef(); 270 271 audio_hw_device_t* findSuitableHwDev_l(audio_module_handle_t module, uint32_t devices); 272 void purgeStaleEffects_l(); 273 274 // standby delay for MIXER and DUPLICATING playback threads is read from property 275 // ro.audio.flinger_standbytime_ms or defaults to kDefaultStandbyTimeInNsecs 276 static nsecs_t mStandbyTimeInNsecs; 277 278 // Internal dump utilites. 279 status_t dumpPermissionDenial(int fd, const Vector<String16>& args); 280 status_t dumpClients(int fd, const Vector<String16>& args); 281 status_t dumpInternals(int fd, const Vector<String16>& args); 282 283 // --- Client --- 284 class Client : public RefBase { 285 public: 286 Client(const sp<AudioFlinger>& audioFlinger, pid_t pid); 287 virtual ~Client(); 288 sp<MemoryDealer> heap() const; 289 pid_t pid() const { return mPid; } 290 sp<AudioFlinger> audioFlinger() const { return mAudioFlinger; } 291 292 bool reserveTimedTrack(); 293 void releaseTimedTrack(); 294 295 private: 296 Client(const Client&); 297 Client& operator = (const Client&); 298 const sp<AudioFlinger> mAudioFlinger; 299 const sp<MemoryDealer> mMemoryDealer; 300 const pid_t mPid; 301 302 Mutex mTimedTrackLock; 303 int mTimedTrackCount; 304 }; 305 306 // --- Notification Client --- 307 class NotificationClient : public IBinder::DeathRecipient { 308 public: 309 NotificationClient(const sp<AudioFlinger>& audioFlinger, 310 const sp<IAudioFlingerClient>& client, 311 pid_t pid); 312 virtual ~NotificationClient(); 313 314 sp<IAudioFlingerClient> audioFlingerClient() const { return mAudioFlingerClient; } 315 316 // IBinder::DeathRecipient 317 virtual void binderDied(const wp<IBinder>& who); 318 319 private: 320 NotificationClient(const NotificationClient&); 321 NotificationClient& operator = (const NotificationClient&); 322 323 const sp<AudioFlinger> mAudioFlinger; 324 const pid_t mPid; 325 const sp<IAudioFlingerClient> mAudioFlingerClient; 326 }; 327 328 class TrackHandle; 329 class RecordHandle; 330 class RecordThread; 331 class PlaybackThread; 332 class MixerThread; 333 class DirectOutputThread; 334 class DuplicatingThread; 335 class Track; 336 class RecordTrack; 337 class EffectModule; 338 class EffectHandle; 339 class EffectChain; 340 struct AudioStreamOut; 341 struct AudioStreamIn; 342 343 class ThreadBase : public Thread { 344 public: 345 346 enum type_t { 347 MIXER, // Thread class is MixerThread 348 DIRECT, // Thread class is DirectOutputThread 349 DUPLICATING, // Thread class is DuplicatingThread 350 RECORD // Thread class is RecordThread 351 }; 352 353 ThreadBase (const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, uint32_t device, type_t type); 354 virtual ~ThreadBase(); 355 356 status_t dumpBase(int fd, const Vector<String16>& args); 357 status_t dumpEffectChains(int fd, const Vector<String16>& args); 358 359 void clearPowerManager(); 360 361 // base for record and playback 362 class TrackBase : public ExtendedAudioBufferProvider, public RefBase { 363 364 public: 365 enum track_state { 366 IDLE, 367 TERMINATED, 368 FLUSHED, 369 STOPPED, 370 // next 2 states are currently used for fast tracks only 371 STOPPING_1, // waiting for first underrun 372 STOPPING_2, // waiting for presentation complete 373 RESUMING, 374 ACTIVE, 375 PAUSING, 376 PAUSED 377 }; 378 379 TrackBase(ThreadBase *thread, 380 const sp<Client>& client, 381 uint32_t sampleRate, 382 audio_format_t format, 383 uint32_t channelMask, 384 int frameCount, 385 const sp<IMemory>& sharedBuffer, 386 int sessionId); 387 virtual ~TrackBase(); 388 389 virtual status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 390 int triggerSession = 0) = 0; 391 virtual void stop() = 0; 392 sp<IMemory> getCblk() const { return mCblkMemory; } 393 audio_track_cblk_t* cblk() const { return mCblk; } 394 int sessionId() const { return mSessionId; } 395 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 396 397 protected: 398 TrackBase(const TrackBase&); 399 TrackBase& operator = (const TrackBase&); 400 401 // AudioBufferProvider interface 402 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) = 0; 403 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 404 405 // ExtendedAudioBufferProvider interface is only needed for Track, 406 // but putting it in TrackBase avoids the complexity of virtual inheritance 407 virtual size_t framesReady() const { return SIZE_MAX; } 408 409 audio_format_t format() const { 410 return mFormat; 411 } 412 413 int channelCount() const { return mChannelCount; } 414 415 uint32_t channelMask() const { return mChannelMask; } 416 417 int sampleRate() const; // FIXME inline after cblk sr moved 418 419 void* getBuffer(uint32_t offset, uint32_t frames) const; 420 421 bool isStopped() const { 422 return (mState == STOPPED || mState == FLUSHED); 423 } 424 425 // for fast tracks only 426 bool isStopping() const { 427 return mState == STOPPING_1 || mState == STOPPING_2; 428 } 429 bool isStopping_1() const { 430 return mState == STOPPING_1; 431 } 432 bool isStopping_2() const { 433 return mState == STOPPING_2; 434 } 435 436 bool isTerminated() const { 437 return mState == TERMINATED; 438 } 439 440 bool step(); 441 void reset(); 442 443 const wp<ThreadBase> mThread; 444 /*const*/ sp<Client> mClient; // see explanation at ~TrackBase() why not const 445 sp<IMemory> mCblkMemory; 446 audio_track_cblk_t* mCblk; 447 void* mBuffer; 448 void* mBufferEnd; 449 uint32_t mFrameCount; 450 // we don't really need a lock for these 451 track_state mState; 452 const uint32_t mSampleRate; // initial sample rate only; for tracks which 453 // support dynamic rates, the current value is in control block 454 const audio_format_t mFormat; 455 bool mStepServerFailed; 456 const int mSessionId; 457 uint8_t mChannelCount; 458 uint32_t mChannelMask; 459 Vector < sp<SyncEvent> >mSyncEvents; 460 }; 461 462 class ConfigEvent { 463 public: 464 ConfigEvent() : mEvent(0), mParam(0) {} 465 466 int mEvent; 467 int mParam; 468 }; 469 470 class PMDeathRecipient : public IBinder::DeathRecipient { 471 public: 472 PMDeathRecipient(const wp<ThreadBase>& thread) : mThread(thread) {} 473 virtual ~PMDeathRecipient() {} 474 475 // IBinder::DeathRecipient 476 virtual void binderDied(const wp<IBinder>& who); 477 478 private: 479 PMDeathRecipient(const PMDeathRecipient&); 480 PMDeathRecipient& operator = (const PMDeathRecipient&); 481 482 wp<ThreadBase> mThread; 483 }; 484 485 virtual status_t initCheck() const = 0; 486 type_t type() const { return mType; } 487 uint32_t sampleRate() const { return mSampleRate; } 488 int channelCount() const { return mChannelCount; } 489 audio_format_t format() const { return mFormat; } 490 // Called by AudioFlinger::frameCount(audio_io_handle_t output) and effects, 491 // and returns the normal mix buffer's frame count. No API for HAL frame count. 492 size_t frameCount() const { return mNormalFrameCount; } 493 void wakeUp() { mWaitWorkCV.broadcast(); } 494 // Should be "virtual status_t requestExitAndWait()" and override same 495 // method in Thread, but Thread::requestExitAndWait() is not yet virtual. 496 void exit(); 497 virtual bool checkForNewParameters_l() = 0; 498 virtual status_t setParameters(const String8& keyValuePairs); 499 virtual String8 getParameters(const String8& keys) = 0; 500 virtual void audioConfigChanged_l(int event, int param = 0) = 0; 501 void sendConfigEvent(int event, int param = 0); 502 void sendConfigEvent_l(int event, int param = 0); 503 void processConfigEvents(); 504 audio_io_handle_t id() const { return mId;} 505 bool standby() const { return mStandby; } 506 uint32_t device() const { return mDevice; } 507 virtual audio_stream_t* stream() const = 0; 508 509 sp<EffectHandle> createEffect_l( 510 const sp<AudioFlinger::Client>& client, 511 const sp<IEffectClient>& effectClient, 512 int32_t priority, 513 int sessionId, 514 effect_descriptor_t *desc, 515 int *enabled, 516 status_t *status); 517 void disconnectEffect(const sp< EffectModule>& effect, 518 const wp<EffectHandle>& handle, 519 bool unpinIfLast); 520 521 // return values for hasAudioSession (bit field) 522 enum effect_state { 523 EFFECT_SESSION = 0x1, // the audio session corresponds to at least one 524 // effect 525 TRACK_SESSION = 0x2 // the audio session corresponds to at least one 526 // track 527 }; 528 529 // get effect chain corresponding to session Id. 530 sp<EffectChain> getEffectChain(int sessionId); 531 // same as getEffectChain() but must be called with ThreadBase mutex locked 532 sp<EffectChain> getEffectChain_l(int sessionId); 533 // add an effect chain to the chain list (mEffectChains) 534 virtual status_t addEffectChain_l(const sp<EffectChain>& chain) = 0; 535 // remove an effect chain from the chain list (mEffectChains) 536 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain) = 0; 537 // lock all effect chains Mutexes. Must be called before releasing the 538 // ThreadBase mutex before processing the mixer and effects. This guarantees the 539 // integrity of the chains during the process. 540 // Also sets the parameter 'effectChains' to current value of mEffectChains. 541 void lockEffectChains_l(Vector< sp<EffectChain> >& effectChains); 542 // unlock effect chains after process 543 void unlockEffectChains(const Vector< sp<EffectChain> >& effectChains); 544 // set audio mode to all effect chains 545 void setMode(audio_mode_t mode); 546 // get effect module with corresponding ID on specified audio session 547 sp<AudioFlinger::EffectModule> getEffect(int sessionId, int effectId); 548 sp<AudioFlinger::EffectModule> getEffect_l(int sessionId, int effectId); 549 // add and effect module. Also creates the effect chain is none exists for 550 // the effects audio session 551 status_t addEffect_l(const sp< EffectModule>& effect); 552 // remove and effect module. Also removes the effect chain is this was the last 553 // effect 554 void removeEffect_l(const sp< EffectModule>& effect); 555 // detach all tracks connected to an auxiliary effect 556 virtual void detachAuxEffect_l(int effectId) {} 557 // returns either EFFECT_SESSION if effects on this audio session exist in one 558 // chain, or TRACK_SESSION if tracks on this audio session exist, or both 559 virtual uint32_t hasAudioSession(int sessionId) = 0; 560 // the value returned by default implementation is not important as the 561 // strategy is only meaningful for PlaybackThread which implements this method 562 virtual uint32_t getStrategyForSession_l(int sessionId) { return 0; } 563 564 // suspend or restore effect according to the type of effect passed. a NULL 565 // type pointer means suspend all effects in the session 566 void setEffectSuspended(const effect_uuid_t *type, 567 bool suspend, 568 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 569 // check if some effects must be suspended/restored when an effect is enabled 570 // or disabled 571 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 572 bool enabled, 573 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 574 void checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 575 bool enabled, 576 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 577 578 virtual status_t setSyncEvent(const sp<SyncEvent>& event) = 0; 579 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) = 0; 580 581 582 mutable Mutex mLock; 583 584 protected: 585 586 // entry describing an effect being suspended in mSuspendedSessions keyed vector 587 class SuspendedSessionDesc : public RefBase { 588 public: 589 SuspendedSessionDesc() : mRefCount(0) {} 590 591 int mRefCount; // number of active suspend requests 592 effect_uuid_t mType; // effect type UUID 593 }; 594 595 void acquireWakeLock(); 596 void acquireWakeLock_l(); 597 void releaseWakeLock(); 598 void releaseWakeLock_l(); 599 void setEffectSuspended_l(const effect_uuid_t *type, 600 bool suspend, 601 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 602 // updated mSuspendedSessions when an effect suspended or restored 603 void updateSuspendedSessions_l(const effect_uuid_t *type, 604 bool suspend, 605 int sessionId); 606 // check if some effects must be suspended when an effect chain is added 607 void checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain); 608 609 friend class AudioFlinger; // for mEffectChains 610 611 const type_t mType; 612 613 // Used by parameters, config events, addTrack_l, exit 614 Condition mWaitWorkCV; 615 616 const sp<AudioFlinger> mAudioFlinger; 617 uint32_t mSampleRate; 618 size_t mFrameCount; // output HAL, direct output, record 619 size_t mNormalFrameCount; // normal mixer and effects 620 uint32_t mChannelMask; 621 uint16_t mChannelCount; 622 size_t mFrameSize; 623 audio_format_t mFormat; 624 625 // Parameter sequence by client: binder thread calling setParameters(): 626 // 1. Lock mLock 627 // 2. Append to mNewParameters 628 // 3. mWaitWorkCV.signal 629 // 4. mParamCond.waitRelative with timeout 630 // 5. read mParamStatus 631 // 6. mWaitWorkCV.signal 632 // 7. Unlock 633 // 634 // Parameter sequence by server: threadLoop calling checkForNewParameters_l(): 635 // 1. Lock mLock 636 // 2. If there is an entry in mNewParameters proceed ... 637 // 2. Read first entry in mNewParameters 638 // 3. Process 639 // 4. Remove first entry from mNewParameters 640 // 5. Set mParamStatus 641 // 6. mParamCond.signal 642 // 7. mWaitWorkCV.wait with timeout (this is to avoid overwriting mParamStatus) 643 // 8. Unlock 644 Condition mParamCond; 645 Vector<String8> mNewParameters; 646 status_t mParamStatus; 647 648 Vector<ConfigEvent> mConfigEvents; 649 bool mStandby; 650 const audio_io_handle_t mId; 651 Vector< sp<EffectChain> > mEffectChains; 652 uint32_t mDevice; // output device for PlaybackThread 653 // input + output devices for RecordThread 654 static const int kNameLength = 16; // prctl(PR_SET_NAME) limit 655 char mName[kNameLength]; 656 sp<IPowerManager> mPowerManager; 657 sp<IBinder> mWakeLockToken; 658 const sp<PMDeathRecipient> mDeathRecipient; 659 // list of suspended effects per session and per type. The first vector is 660 // keyed by session ID, the second by type UUID timeLow field 661 KeyedVector< int, KeyedVector< int, sp<SuspendedSessionDesc> > > mSuspendedSessions; 662 }; 663 664 struct stream_type_t { 665 stream_type_t() 666 : volume(1.0f), 667 mute(false) 668 { 669 } 670 float volume; 671 bool mute; 672 }; 673 674 // --- PlaybackThread --- 675 class PlaybackThread : public ThreadBase { 676 public: 677 678 enum mixer_state { 679 MIXER_IDLE, // no active tracks 680 MIXER_TRACKS_ENABLED, // at least one active track, but no track has any data ready 681 MIXER_TRACKS_READY // at least one active track, and at least one track has data 682 // standby mode does not have an enum value 683 // suspend by audio policy manager is orthogonal to mixer state 684 }; 685 686 // playback track 687 class Track : public TrackBase, public VolumeProvider { 688 public: 689 Track( PlaybackThread *thread, 690 const sp<Client>& client, 691 audio_stream_type_t streamType, 692 uint32_t sampleRate, 693 audio_format_t format, 694 uint32_t channelMask, 695 int frameCount, 696 const sp<IMemory>& sharedBuffer, 697 int sessionId, 698 IAudioFlinger::track_flags_t flags); 699 virtual ~Track(); 700 701 static void appendDumpHeader(String8& result); 702 void dump(char* buffer, size_t size); 703 virtual status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 704 int triggerSession = 0); 705 virtual void stop(); 706 void pause(); 707 708 void flush(); 709 void destroy(); 710 void mute(bool); 711 int name() const { 712 return mName; 713 } 714 715 audio_stream_type_t streamType() const { 716 return mStreamType; 717 } 718 status_t attachAuxEffect(int EffectId); 719 void setAuxBuffer(int EffectId, int32_t *buffer); 720 int32_t *auxBuffer() const { return mAuxBuffer; } 721 void setMainBuffer(int16_t *buffer) { mMainBuffer = buffer; } 722 int16_t *mainBuffer() const { return mMainBuffer; } 723 int auxEffectId() const { return mAuxEffectId; } 724 725 // implement FastMixerState::VolumeProvider interface 726 virtual uint32_t getVolumeLR(); 727 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 728 729 protected: 730 // for numerous 731 friend class PlaybackThread; 732 friend class MixerThread; 733 friend class DirectOutputThread; 734 735 Track(const Track&); 736 Track& operator = (const Track&); 737 738 // AudioBufferProvider interface 739 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts = kInvalidPTS); 740 // releaseBuffer() not overridden 741 742 virtual size_t framesReady() const; 743 744 bool isMuted() const { return mMute; } 745 bool isPausing() const { 746 return mState == PAUSING; 747 } 748 bool isPaused() const { 749 return mState == PAUSED; 750 } 751 bool isResuming() const { 752 return mState == RESUMING; 753 } 754 bool isReady() const; 755 void setPaused() { mState = PAUSED; } 756 void reset(); 757 758 bool isOutputTrack() const { 759 return (mStreamType == AUDIO_STREAM_CNT); 760 } 761 762 sp<IMemory> sharedBuffer() const { return mSharedBuffer; } 763 764 bool presentationComplete(size_t framesWritten, size_t audioHalFrames); 765 766 public: 767 void triggerEvents(AudioSystem::sync_event_t type); 768 virtual bool isTimedTrack() const { return false; } 769 bool isFastTrack() const { return (mFlags & IAudioFlinger::TRACK_FAST) != 0; } 770 protected: 771 772 // we don't really need a lock for these 773 volatile bool mMute; 774 // FILLED state is used for suppressing volume ramp at begin of playing 775 enum {FS_INVALID, FS_FILLING, FS_FILLED, FS_ACTIVE}; 776 mutable uint8_t mFillingUpStatus; 777 int8_t mRetryCount; 778 const sp<IMemory> mSharedBuffer; 779 bool mResetDone; 780 const audio_stream_type_t mStreamType; 781 int mName; // track name on the normal mixer, 782 // allocated statically at track creation time, 783 // and is even allocated (though unused) for fast tracks 784 // FIXME don't allocate track name for fast tracks 785 int16_t *mMainBuffer; 786 int32_t *mAuxBuffer; 787 int mAuxEffectId; 788 bool mHasVolumeController; 789 size_t mPresentationCompleteFrames; // number of frames written to the audio HAL 790 // when this track will be fully rendered 791 private: 792 IAudioFlinger::track_flags_t mFlags; 793 794 // The following fields are only for fast tracks, and should be in a subclass 795 int mFastIndex; // index within FastMixerState::mFastTracks[]; 796 // either mFastIndex == -1 if not isFastTrack() 797 // or 0 < mFastIndex < FastMixerState::kMaxFast because 798 // index 0 is reserved for normal mixer's submix; 799 // index is allocated statically at track creation time 800 // but the slot is only used if track is active 801 FastTrackUnderruns mObservedUnderruns; // Most recently observed value of 802 // mFastMixerDumpState.mTracks[mFastIndex].mUnderruns 803 uint32_t mUnderrunCount; // Counter of total number of underruns, never reset 804 volatile float mCachedVolume; // combined master volume and stream type volume; 805 // 'volatile' means accessed without lock or 806 // barrier, but is read/written atomically 807 }; // end of Track 808 809 class TimedTrack : public Track { 810 public: 811 static sp<TimedTrack> create(PlaybackThread *thread, 812 const sp<Client>& client, 813 audio_stream_type_t streamType, 814 uint32_t sampleRate, 815 audio_format_t format, 816 uint32_t channelMask, 817 int frameCount, 818 const sp<IMemory>& sharedBuffer, 819 int sessionId); 820 ~TimedTrack(); 821 822 class TimedBuffer { 823 public: 824 TimedBuffer(); 825 TimedBuffer(const sp<IMemory>& buffer, int64_t pts); 826 const sp<IMemory>& buffer() const { return mBuffer; } 827 int64_t pts() const { return mPTS; } 828 uint32_t position() const { return mPosition; } 829 void setPosition(uint32_t pos) { mPosition = pos; } 830 private: 831 sp<IMemory> mBuffer; 832 int64_t mPTS; 833 uint32_t mPosition; 834 }; 835 836 // Mixer facing methods. 837 virtual bool isTimedTrack() const { return true; } 838 virtual size_t framesReady() const; 839 840 // AudioBufferProvider interface 841 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, 842 int64_t pts); 843 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 844 845 // Client/App facing methods. 846 status_t allocateTimedBuffer(size_t size, 847 sp<IMemory>* buffer); 848 status_t queueTimedBuffer(const sp<IMemory>& buffer, 849 int64_t pts); 850 status_t setMediaTimeTransform(const LinearTransform& xform, 851 TimedAudioTrack::TargetTimeline target); 852 853 private: 854 TimedTrack(PlaybackThread *thread, 855 const sp<Client>& client, 856 audio_stream_type_t streamType, 857 uint32_t sampleRate, 858 audio_format_t format, 859 uint32_t channelMask, 860 int frameCount, 861 const sp<IMemory>& sharedBuffer, 862 int sessionId); 863 864 void timedYieldSamples_l(AudioBufferProvider::Buffer* buffer); 865 void timedYieldSilence_l(uint32_t numFrames, 866 AudioBufferProvider::Buffer* buffer); 867 void trimTimedBufferQueue_l(); 868 void trimTimedBufferQueueHead_l(const char* logTag); 869 void updateFramesPendingAfterTrim_l(const TimedBuffer& buf, 870 const char* logTag); 871 872 uint64_t mLocalTimeFreq; 873 LinearTransform mLocalTimeToSampleTransform; 874 LinearTransform mMediaTimeToSampleTransform; 875 sp<MemoryDealer> mTimedMemoryDealer; 876 877 Vector<TimedBuffer> mTimedBufferQueue; 878 bool mQueueHeadInFlight; 879 bool mTrimQueueHeadOnRelease; 880 uint32_t mFramesPendingInQueue; 881 882 uint8_t* mTimedSilenceBuffer; 883 uint32_t mTimedSilenceBufferSize; 884 mutable Mutex mTimedBufferQueueLock; 885 bool mTimedAudioOutputOnTime; 886 CCHelper mCCHelper; 887 888 Mutex mMediaTimeTransformLock; 889 LinearTransform mMediaTimeTransform; 890 bool mMediaTimeTransformValid; 891 TimedAudioTrack::TargetTimeline mMediaTimeTransformTarget; 892 }; 893 894 895 // playback track 896 class OutputTrack : public Track { 897 public: 898 899 class Buffer: public AudioBufferProvider::Buffer { 900 public: 901 int16_t *mBuffer; 902 }; 903 904 OutputTrack(PlaybackThread *thread, 905 DuplicatingThread *sourceThread, 906 uint32_t sampleRate, 907 audio_format_t format, 908 uint32_t channelMask, 909 int frameCount); 910 virtual ~OutputTrack(); 911 912 virtual status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 913 int triggerSession = 0); 914 virtual void stop(); 915 bool write(int16_t* data, uint32_t frames); 916 bool bufferQueueEmpty() const { return mBufferQueue.size() == 0; } 917 bool isActive() const { return mActive; } 918 const wp<ThreadBase>& thread() const { return mThread; } 919 920 private: 921 922 enum { 923 NO_MORE_BUFFERS = 0x80000001, // same in AudioTrack.h, ok to be different value 924 }; 925 926 status_t obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs); 927 void clearBufferQueue(); 928 929 // Maximum number of pending buffers allocated by OutputTrack::write() 930 static const uint8_t kMaxOverFlowBuffers = 10; 931 932 Vector < Buffer* > mBufferQueue; 933 AudioBufferProvider::Buffer mOutBuffer; 934 bool mActive; 935 DuplicatingThread* const mSourceThread; // for waitTimeMs() in write() 936 }; // end of OutputTrack 937 938 PlaybackThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 939 audio_io_handle_t id, uint32_t device, type_t type); 940 virtual ~PlaybackThread(); 941 942 status_t dump(int fd, const Vector<String16>& args); 943 944 // Thread virtuals 945 virtual status_t readyToRun(); 946 virtual bool threadLoop(); 947 948 // RefBase 949 virtual void onFirstRef(); 950 951protected: 952 // Code snippets that were lifted up out of threadLoop() 953 virtual void threadLoop_mix() = 0; 954 virtual void threadLoop_sleepTime() = 0; 955 virtual void threadLoop_write(); 956 virtual void threadLoop_standby(); 957 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 958 959 // prepareTracks_l reads and writes mActiveTracks, and returns 960 // the pending set of tracks to remove via Vector 'tracksToRemove'. The caller 961 // is responsible for clearing or destroying this Vector later on, when it 962 // is safe to do so. That will drop the final ref count and destroy the tracks. 963 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove) = 0; 964 965public: 966 967 virtual status_t initCheck() const { return (mOutput == NULL) ? NO_INIT : NO_ERROR; } 968 969 // return estimated latency in milliseconds, as reported by HAL 970 uint32_t latency() const; 971 // same, but lock must already be held 972 uint32_t latency_l() const; 973 974 void setMasterVolume(float value); 975 void setMasterMute(bool muted); 976 977 void setStreamVolume(audio_stream_type_t stream, float value); 978 void setStreamMute(audio_stream_type_t stream, bool muted); 979 980 float streamVolume(audio_stream_type_t stream) const; 981 982 sp<Track> createTrack_l( 983 const sp<AudioFlinger::Client>& client, 984 audio_stream_type_t streamType, 985 uint32_t sampleRate, 986 audio_format_t format, 987 uint32_t channelMask, 988 int frameCount, 989 const sp<IMemory>& sharedBuffer, 990 int sessionId, 991 IAudioFlinger::track_flags_t flags, 992 pid_t tid, 993 status_t *status); 994 995 AudioStreamOut* getOutput() const; 996 AudioStreamOut* clearOutput(); 997 virtual audio_stream_t* stream() const; 998 999 void suspend() { mSuspended++; } 1000 void restore() { if (mSuspended > 0) mSuspended--; } 1001 bool isSuspended() const { return (mSuspended > 0); } 1002 virtual String8 getParameters(const String8& keys); 1003 virtual void audioConfigChanged_l(int event, int param = 0); 1004 status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames); 1005 int16_t *mixBuffer() const { return mMixBuffer; }; 1006 1007 virtual void detachAuxEffect_l(int effectId); 1008 status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track> track, 1009 int EffectId); 1010 status_t attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track> track, 1011 int EffectId); 1012 1013 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1014 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1015 virtual uint32_t hasAudioSession(int sessionId); 1016 virtual uint32_t getStrategyForSession_l(int sessionId); 1017 1018 1019 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 1020 virtual bool isValidSyncEvent(const sp<SyncEvent>& event); 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 void invalidateTracks(audio_stream_type_t streamType); 1149 virtual bool checkForNewParameters_l(); 1150 virtual status_t dumpInternals(int fd, const Vector<String16>& args); 1151 1152 protected: 1153 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1154 virtual int getTrackName_l(audio_channel_mask_t channelMask); 1155 virtual void deleteTrackName_l(int name); 1156 virtual uint32_t idleSleepTimeUs() const; 1157 virtual uint32_t suspendSleepTimeUs() const; 1158 virtual void cacheParameters_l(); 1159 1160 // threadLoop snippets 1161 virtual void threadLoop_write(); 1162 virtual void threadLoop_standby(); 1163 virtual void threadLoop_mix(); 1164 virtual void threadLoop_sleepTime(); 1165 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 1166 virtual uint32_t correctLatency(uint32_t latency) const; 1167 1168 AudioMixer* mAudioMixer; // normal mixer 1169 private: 1170#ifdef SOAKER 1171 Thread* mSoaker; 1172#endif 1173 // one-time initialization, no locks required 1174 FastMixer* mFastMixer; // non-NULL if there is also a fast mixer 1175 sp<AudioWatchdog> mAudioWatchdog; // non-0 if there is an audio watchdog thread 1176 1177 // contents are not guaranteed to be consistent, no locks required 1178 FastMixerDumpState mFastMixerDumpState; 1179#ifdef STATE_QUEUE_DUMP 1180 StateQueueObserverDump mStateQueueObserverDump; 1181 StateQueueMutatorDump mStateQueueMutatorDump; 1182#endif 1183 AudioWatchdogDump mAudioWatchdogDump; 1184 1185 // accessible only within the threadLoop(), no locks required 1186 // mFastMixer->sq() // for mutating and pushing state 1187 int32_t mFastMixerFutex; // for cold idle 1188 1189 public: 1190 virtual bool hasFastMixer() const { return mFastMixer != NULL; } 1191 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const { 1192 ALOG_ASSERT(fastIndex < FastMixerState::kMaxFastTracks); 1193 return mFastMixerDumpState.mTracks[fastIndex].mUnderruns; 1194 } 1195 }; 1196 1197 class DirectOutputThread : public PlaybackThread { 1198 public: 1199 1200 DirectOutputThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1201 audio_io_handle_t id, uint32_t device); 1202 virtual ~DirectOutputThread(); 1203 1204 // Thread virtuals 1205 1206 virtual bool checkForNewParameters_l(); 1207 1208 protected: 1209 virtual int getTrackName_l(audio_channel_mask_t channelMask); 1210 virtual void deleteTrackName_l(int name); 1211 virtual uint32_t activeSleepTimeUs() const; 1212 virtual uint32_t idleSleepTimeUs() const; 1213 virtual uint32_t suspendSleepTimeUs() const; 1214 virtual void cacheParameters_l(); 1215 1216 // threadLoop snippets 1217 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1218 virtual void threadLoop_mix(); 1219 virtual void threadLoop_sleepTime(); 1220 1221 // volumes last sent to audio HAL with stream->set_volume() 1222 float mLeftVolFloat; 1223 float mRightVolFloat; 1224 1225private: 1226 // prepareTracks_l() tells threadLoop_mix() the name of the single active track 1227 sp<Track> mActiveTrack; 1228 public: 1229 virtual bool hasFastMixer() const { return false; } 1230 }; 1231 1232 class DuplicatingThread : public MixerThread { 1233 public: 1234 DuplicatingThread (const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread, 1235 audio_io_handle_t id); 1236 virtual ~DuplicatingThread(); 1237 1238 // Thread virtuals 1239 void addOutputTrack(MixerThread* thread); 1240 void removeOutputTrack(MixerThread* thread); 1241 uint32_t waitTimeMs() const { return mWaitTimeMs; } 1242 protected: 1243 virtual uint32_t activeSleepTimeUs() const; 1244 1245 private: 1246 bool outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks); 1247 protected: 1248 // threadLoop snippets 1249 virtual void threadLoop_mix(); 1250 virtual void threadLoop_sleepTime(); 1251 virtual void threadLoop_write(); 1252 virtual void threadLoop_standby(); 1253 virtual void cacheParameters_l(); 1254 1255 private: 1256 // called from threadLoop, addOutputTrack, removeOutputTrack 1257 virtual void updateWaitTime_l(); 1258 protected: 1259 virtual void saveOutputTracks(); 1260 virtual void clearOutputTracks(); 1261 private: 1262 1263 uint32_t mWaitTimeMs; 1264 SortedVector < sp<OutputTrack> > outputTracks; 1265 SortedVector < sp<OutputTrack> > mOutputTracks; 1266 public: 1267 virtual bool hasFastMixer() const { return false; } 1268 }; 1269 1270 PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const; 1271 MixerThread *checkMixerThread_l(audio_io_handle_t output) const; 1272 RecordThread *checkRecordThread_l(audio_io_handle_t input) const; 1273 // no range check, AudioFlinger::mLock held 1274 bool streamMute_l(audio_stream_type_t stream) const 1275 { return mStreamTypes[stream].mute; } 1276 // no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held 1277 float streamVolume_l(audio_stream_type_t stream) const 1278 { return mStreamTypes[stream].volume; } 1279 void audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2); 1280 1281 // allocate an audio_io_handle_t, session ID, or effect ID 1282 uint32_t nextUniqueId(); 1283 1284 status_t moveEffectChain_l(int sessionId, 1285 PlaybackThread *srcThread, 1286 PlaybackThread *dstThread, 1287 bool reRegister); 1288 // return thread associated with primary hardware device, or NULL 1289 PlaybackThread *primaryPlaybackThread_l() const; 1290 uint32_t primaryOutputDevice_l() const; 1291 1292 sp<PlaybackThread> getEffectThread_l(int sessionId, int EffectId); 1293 1294 // server side of the client's IAudioTrack 1295 class TrackHandle : public android::BnAudioTrack { 1296 public: 1297 TrackHandle(const sp<PlaybackThread::Track>& track); 1298 virtual ~TrackHandle(); 1299 virtual sp<IMemory> getCblk() const; 1300 virtual status_t start(); 1301 virtual void stop(); 1302 virtual void flush(); 1303 virtual void mute(bool); 1304 virtual void pause(); 1305 virtual status_t attachAuxEffect(int effectId); 1306 virtual status_t allocateTimedBuffer(size_t size, 1307 sp<IMemory>* buffer); 1308 virtual status_t queueTimedBuffer(const sp<IMemory>& buffer, 1309 int64_t pts); 1310 virtual status_t setMediaTimeTransform(const LinearTransform& xform, 1311 int target); 1312 virtual status_t onTransact( 1313 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 1314 private: 1315 const sp<PlaybackThread::Track> mTrack; 1316 }; 1317 1318 void removeClient_l(pid_t pid); 1319 void removeNotificationClient(pid_t pid); 1320 1321 1322 // record thread 1323 class RecordThread : public ThreadBase, public AudioBufferProvider 1324 { 1325 public: 1326 1327 // record track 1328 class RecordTrack : public TrackBase { 1329 public: 1330 RecordTrack(RecordThread *thread, 1331 const sp<Client>& client, 1332 uint32_t sampleRate, 1333 audio_format_t format, 1334 uint32_t channelMask, 1335 int frameCount, 1336 int sessionId); 1337 virtual ~RecordTrack(); 1338 1339 virtual status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 1340 int triggerSession = 0); 1341 virtual void stop(); 1342 1343 bool overflow() { bool tmp = mOverflow; mOverflow = false; return tmp; } 1344 bool setOverflow() { bool tmp = mOverflow; mOverflow = true; return tmp; } 1345 1346 void dump(char* buffer, size_t size); 1347 1348 private: 1349 friend class AudioFlinger; // for mState 1350 1351 RecordTrack(const RecordTrack&); 1352 RecordTrack& operator = (const RecordTrack&); 1353 1354 // AudioBufferProvider interface 1355 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts = kInvalidPTS); 1356 // releaseBuffer() not overridden 1357 1358 bool mOverflow; 1359 }; 1360 1361 1362 RecordThread(const sp<AudioFlinger>& audioFlinger, 1363 AudioStreamIn *input, 1364 uint32_t sampleRate, 1365 uint32_t channels, 1366 audio_io_handle_t id, 1367 uint32_t device); 1368 virtual ~RecordThread(); 1369 1370 // Thread 1371 virtual bool threadLoop(); 1372 virtual status_t readyToRun(); 1373 1374 // RefBase 1375 virtual void onFirstRef(); 1376 1377 virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; } 1378 sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l( 1379 const sp<AudioFlinger::Client>& client, 1380 uint32_t sampleRate, 1381 audio_format_t format, 1382 int channelMask, 1383 int frameCount, 1384 int sessionId, 1385 status_t *status); 1386 1387 status_t start(RecordTrack* recordTrack, 1388 AudioSystem::sync_event_t event, 1389 int triggerSession); 1390 void stop(RecordTrack* recordTrack); 1391 status_t dump(int fd, const Vector<String16>& args); 1392 AudioStreamIn* getInput() const; 1393 AudioStreamIn* clearInput(); 1394 virtual audio_stream_t* stream() const; 1395 1396 // AudioBufferProvider interface 1397 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts); 1398 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 1399 1400 virtual bool checkForNewParameters_l(); 1401 virtual String8 getParameters(const String8& keys); 1402 virtual void audioConfigChanged_l(int event, int param = 0); 1403 void readInputParameters(); 1404 virtual unsigned int getInputFramesLost(); 1405 1406 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1407 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1408 virtual uint32_t hasAudioSession(int sessionId); 1409 RecordTrack* track(); 1410 1411 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 1412 virtual bool isValidSyncEvent(const sp<SyncEvent>& event); 1413 1414 static void syncStartEventCallback(const wp<SyncEvent>& event); 1415 void handleSyncStartEvent(const sp<SyncEvent>& event); 1416 1417 private: 1418 void clearSyncStartEvent(); 1419 1420 RecordThread(); 1421 AudioStreamIn *mInput; 1422 RecordTrack* mTrack; 1423 sp<RecordTrack> mActiveTrack; 1424 Condition mStartStopCond; 1425 AudioResampler *mResampler; 1426 int32_t *mRsmpOutBuffer; 1427 int16_t *mRsmpInBuffer; 1428 size_t mRsmpInIndex; 1429 size_t mInputBytes; 1430 const int mReqChannelCount; 1431 const uint32_t mReqSampleRate; 1432 ssize_t mBytesRead; 1433 // sync event triggering actual audio capture. Frames read before this event will 1434 // be dropped and therefore not read by the application. 1435 sp<SyncEvent> mSyncStartEvent; 1436 // number of captured frames to drop after the start sync event has been received. 1437 // when < 0, maximum frames to drop before starting capture even if sync event is 1438 // not received 1439 ssize_t mFramestoDrop; 1440 }; 1441 1442 // server side of the client's IAudioRecord 1443 class RecordHandle : public android::BnAudioRecord { 1444 public: 1445 RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack); 1446 virtual ~RecordHandle(); 1447 virtual sp<IMemory> getCblk() const; 1448 virtual status_t start(int event, int triggerSession); 1449 virtual void stop(); 1450 virtual status_t onTransact( 1451 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 1452 private: 1453 const sp<RecordThread::RecordTrack> mRecordTrack; 1454 }; 1455 1456 //--- Audio Effect Management 1457 1458 // EffectModule and EffectChain classes both have their own mutex to protect 1459 // state changes or resource modifications. Always respect the following order 1460 // if multiple mutexes must be acquired to avoid cross deadlock: 1461 // AudioFlinger -> ThreadBase -> EffectChain -> EffectModule 1462 1463 // The EffectModule class is a wrapper object controlling the effect engine implementation 1464 // in the effect library. It prevents concurrent calls to process() and command() functions 1465 // from different client threads. It keeps a list of EffectHandle objects corresponding 1466 // to all client applications using this effect and notifies applications of effect state, 1467 // control or parameter changes. It manages the activation state machine to send appropriate 1468 // reset, enable, disable commands to effect engine and provide volume 1469 // ramping when effects are activated/deactivated. 1470 // When controlling an auxiliary effect, the EffectModule also provides an input buffer used by 1471 // the attached track(s) to accumulate their auxiliary channel. 1472 class EffectModule: public RefBase { 1473 public: 1474 EffectModule(ThreadBase *thread, 1475 const wp<AudioFlinger::EffectChain>& chain, 1476 effect_descriptor_t *desc, 1477 int id, 1478 int sessionId); 1479 virtual ~EffectModule(); 1480 1481 enum effect_state { 1482 IDLE, 1483 RESTART, 1484 STARTING, 1485 ACTIVE, 1486 STOPPING, 1487 STOPPED, 1488 DESTROYED 1489 }; 1490 1491 int id() const { return mId; } 1492 void process(); 1493 void updateState(); 1494 status_t command(uint32_t cmdCode, 1495 uint32_t cmdSize, 1496 void *pCmdData, 1497 uint32_t *replySize, 1498 void *pReplyData); 1499 1500 void reset_l(); 1501 status_t configure(); 1502 status_t init(); 1503 effect_state state() const { 1504 return mState; 1505 } 1506 uint32_t status() { 1507 return mStatus; 1508 } 1509 int sessionId() const { 1510 return mSessionId; 1511 } 1512 status_t setEnabled(bool enabled); 1513 bool isEnabled() const; 1514 bool isProcessEnabled() const; 1515 1516 void setInBuffer(int16_t *buffer) { mConfig.inputCfg.buffer.s16 = buffer; } 1517 int16_t *inBuffer() { return mConfig.inputCfg.buffer.s16; } 1518 void setOutBuffer(int16_t *buffer) { mConfig.outputCfg.buffer.s16 = buffer; } 1519 int16_t *outBuffer() { return mConfig.outputCfg.buffer.s16; } 1520 void setChain(const wp<EffectChain>& chain) { mChain = chain; } 1521 void setThread(const wp<ThreadBase>& thread) { mThread = thread; } 1522 const wp<ThreadBase>& thread() { return mThread; } 1523 1524 status_t addHandle(const sp<EffectHandle>& handle); 1525 void disconnect(const wp<EffectHandle>& handle, bool unpinIfLast); 1526 size_t removeHandle (const wp<EffectHandle>& handle); 1527 1528 effect_descriptor_t& desc() { return mDescriptor; } 1529 wp<EffectChain>& chain() { return mChain; } 1530 1531 status_t setDevice(uint32_t device); 1532 status_t setVolume(uint32_t *left, uint32_t *right, bool controller); 1533 status_t setMode(audio_mode_t mode); 1534 status_t start(); 1535 status_t stop(); 1536 void setSuspended(bool suspended); 1537 bool suspended() const; 1538 1539 sp<EffectHandle> controlHandle(); 1540 1541 bool isPinned() const { return mPinned; } 1542 void unPin() { mPinned = false; } 1543 1544 status_t dump(int fd, const Vector<String16>& args); 1545 1546 protected: 1547 friend class AudioFlinger; // for mHandles 1548 bool mPinned; 1549 1550 // Maximum time allocated to effect engines to complete the turn off sequence 1551 static const uint32_t MAX_DISABLE_TIME_MS = 10000; 1552 1553 EffectModule(const EffectModule&); 1554 EffectModule& operator = (const EffectModule&); 1555 1556 status_t start_l(); 1557 status_t stop_l(); 1558 1559mutable Mutex mLock; // mutex for process, commands and handles list protection 1560 wp<ThreadBase> mThread; // parent thread 1561 wp<EffectChain> mChain; // parent effect chain 1562 int mId; // this instance unique ID 1563 int mSessionId; // audio session ID 1564 effect_descriptor_t mDescriptor;// effect descriptor received from effect engine 1565 effect_config_t mConfig; // input and output audio configuration 1566 effect_handle_t mEffectInterface; // Effect module C API 1567 status_t mStatus; // initialization status 1568 effect_state mState; // current activation state 1569 Vector< wp<EffectHandle> > mHandles; // list of client handles 1570 // First handle in mHandles has highest priority and controls the effect module 1571 uint32_t mMaxDisableWaitCnt; // maximum grace period before forcing an effect off after 1572 // sending disable command. 1573 uint32_t mDisableWaitCnt; // current process() calls count during disable period. 1574 bool mSuspended; // effect is suspended: temporarily disabled by framework 1575 }; 1576 1577 // The EffectHandle class implements the IEffect interface. It provides resources 1578 // to receive parameter updates, keeps track of effect control 1579 // ownership and state and has a pointer to the EffectModule object it is controlling. 1580 // There is one EffectHandle object for each application controlling (or using) 1581 // an effect module. 1582 // The EffectHandle is obtained by calling AudioFlinger::createEffect(). 1583 class EffectHandle: public android::BnEffect { 1584 public: 1585 1586 EffectHandle(const sp<EffectModule>& effect, 1587 const sp<AudioFlinger::Client>& client, 1588 const sp<IEffectClient>& effectClient, 1589 int32_t priority); 1590 virtual ~EffectHandle(); 1591 1592 // IEffect 1593 virtual status_t enable(); 1594 virtual status_t disable(); 1595 virtual status_t command(uint32_t cmdCode, 1596 uint32_t cmdSize, 1597 void *pCmdData, 1598 uint32_t *replySize, 1599 void *pReplyData); 1600 virtual void disconnect(); 1601 private: 1602 void disconnect(bool unpinIfLast); 1603 public: 1604 virtual sp<IMemory> getCblk() const { return mCblkMemory; } 1605 virtual status_t onTransact(uint32_t code, const Parcel& data, 1606 Parcel* reply, uint32_t flags); 1607 1608 1609 // Give or take control of effect module 1610 // - hasControl: true if control is given, false if removed 1611 // - signal: true client app should be signaled of change, false otherwise 1612 // - enabled: state of the effect when control is passed 1613 void setControl(bool hasControl, bool signal, bool enabled); 1614 void commandExecuted(uint32_t cmdCode, 1615 uint32_t cmdSize, 1616 void *pCmdData, 1617 uint32_t replySize, 1618 void *pReplyData); 1619 void setEnabled(bool enabled); 1620 bool enabled() const { return mEnabled; } 1621 1622 // Getters 1623 int id() const { return mEffect->id(); } 1624 int priority() const { return mPriority; } 1625 bool hasControl() const { return mHasControl; } 1626 sp<EffectModule> effect() const { return mEffect; } 1627 1628 void dump(char* buffer, size_t size); 1629 1630 protected: 1631 friend class AudioFlinger; // for mEffect, mHasControl, mEnabled 1632 EffectHandle(const EffectHandle&); 1633 EffectHandle& operator =(const EffectHandle&); 1634 1635 sp<EffectModule> mEffect; // pointer to controlled EffectModule 1636 sp<IEffectClient> mEffectClient; // callback interface for client notifications 1637 /*const*/ sp<Client> mClient; // client for shared memory allocation, see disconnect() 1638 sp<IMemory> mCblkMemory; // shared memory for control block 1639 effect_param_cblk_t* mCblk; // control block for deferred parameter setting via shared memory 1640 uint8_t* mBuffer; // pointer to parameter area in shared memory 1641 int mPriority; // client application priority to control the effect 1642 bool mHasControl; // true if this handle is controlling the effect 1643 bool mEnabled; // cached enable state: needed when the effect is 1644 // restored after being suspended 1645 }; 1646 1647 // the EffectChain class represents a group of effects associated to one audio session. 1648 // There can be any number of EffectChain objects per output mixer thread (PlaybackThread). 1649 // The EffecChain with session ID 0 contains global effects applied to the output mix. 1650 // Effects in this chain can be insert or auxiliary. Effects in other chains (attached to tracks) 1651 // are insert only. The EffectChain maintains an ordered list of effect module, the order corresponding 1652 // in the effect process order. When attached to a track (session ID != 0), it also provide it's own 1653 // input buffer used by the track as accumulation buffer. 1654 class EffectChain: public RefBase { 1655 public: 1656 EffectChain(const wp<ThreadBase>& wThread, int sessionId); 1657 EffectChain(ThreadBase *thread, int sessionId); 1658 virtual ~EffectChain(); 1659 1660 // special key used for an entry in mSuspendedEffects keyed vector 1661 // corresponding to a suspend all request. 1662 static const int kKeyForSuspendAll = 0; 1663 1664 // minimum duration during which we force calling effect process when last track on 1665 // a session is stopped or removed to allow effect tail to be rendered 1666 static const int kProcessTailDurationMs = 1000; 1667 1668 void process_l(); 1669 1670 void lock() { 1671 mLock.lock(); 1672 } 1673 void unlock() { 1674 mLock.unlock(); 1675 } 1676 1677 status_t addEffect_l(const sp<EffectModule>& handle); 1678 size_t removeEffect_l(const sp<EffectModule>& handle); 1679 1680 int sessionId() const { return mSessionId; } 1681 void setSessionId(int sessionId) { mSessionId = sessionId; } 1682 1683 sp<EffectModule> getEffectFromDesc_l(effect_descriptor_t *descriptor); 1684 sp<EffectModule> getEffectFromId_l(int id); 1685 sp<EffectModule> getEffectFromType_l(const effect_uuid_t *type); 1686 bool setVolume_l(uint32_t *left, uint32_t *right); 1687 void setDevice_l(uint32_t device); 1688 void setMode_l(audio_mode_t mode); 1689 1690 void setInBuffer(int16_t *buffer, bool ownsBuffer = false) { 1691 mInBuffer = buffer; 1692 mOwnInBuffer = ownsBuffer; 1693 } 1694 int16_t *inBuffer() const { 1695 return mInBuffer; 1696 } 1697 void setOutBuffer(int16_t *buffer) { 1698 mOutBuffer = buffer; 1699 } 1700 int16_t *outBuffer() const { 1701 return mOutBuffer; 1702 } 1703 1704 void incTrackCnt() { android_atomic_inc(&mTrackCnt); } 1705 void decTrackCnt() { android_atomic_dec(&mTrackCnt); } 1706 int32_t trackCnt() const { return mTrackCnt;} 1707 1708 void incActiveTrackCnt() { android_atomic_inc(&mActiveTrackCnt); 1709 mTailBufferCount = mMaxTailBuffers; } 1710 void decActiveTrackCnt() { android_atomic_dec(&mActiveTrackCnt); } 1711 int32_t activeTrackCnt() const { return mActiveTrackCnt;} 1712 1713 uint32_t strategy() const { return mStrategy; } 1714 void setStrategy(uint32_t strategy) 1715 { mStrategy = strategy; } 1716 1717 // suspend effect of the given type 1718 void setEffectSuspended_l(const effect_uuid_t *type, 1719 bool suspend); 1720 // suspend all eligible effects 1721 void setEffectSuspendedAll_l(bool suspend); 1722 // check if effects should be suspend or restored when a given effect is enable or disabled 1723 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1724 bool enabled); 1725 1726 void clearInputBuffer(); 1727 1728 status_t dump(int fd, const Vector<String16>& args); 1729 1730 protected: 1731 friend class AudioFlinger; // for mThread, mEffects 1732 EffectChain(const EffectChain&); 1733 EffectChain& operator =(const EffectChain&); 1734 1735 class SuspendedEffectDesc : public RefBase { 1736 public: 1737 SuspendedEffectDesc() : mRefCount(0) {} 1738 1739 int mRefCount; 1740 effect_uuid_t mType; 1741 wp<EffectModule> mEffect; 1742 }; 1743 1744 // get a list of effect modules to suspend when an effect of the type 1745 // passed is enabled. 1746 void getSuspendEligibleEffects(Vector< sp<EffectModule> > &effects); 1747 1748 // get an effect module if it is currently enable 1749 sp<EffectModule> getEffectIfEnabled(const effect_uuid_t *type); 1750 // true if the effect whose descriptor is passed can be suspended 1751 // OEMs can modify the rules implemented in this method to exclude specific effect 1752 // types or implementations from the suspend/restore mechanism. 1753 bool isEffectEligibleForSuspend(const effect_descriptor_t& desc); 1754 1755 void clearInputBuffer_l(sp<ThreadBase> thread); 1756 1757 wp<ThreadBase> mThread; // parent mixer thread 1758 Mutex mLock; // mutex protecting effect list 1759 Vector< sp<EffectModule> > mEffects; // list of effect modules 1760 int mSessionId; // audio session ID 1761 int16_t *mInBuffer; // chain input buffer 1762 int16_t *mOutBuffer; // chain output buffer 1763 volatile int32_t mActiveTrackCnt; // number of active tracks connected 1764 volatile int32_t mTrackCnt; // number of tracks connected 1765 int32_t mTailBufferCount; // current effect tail buffer count 1766 int32_t mMaxTailBuffers; // maximum effect tail buffers 1767 bool mOwnInBuffer; // true if the chain owns its input buffer 1768 int mVolumeCtrlIdx; // index of insert effect having control over volume 1769 uint32_t mLeftVolume; // previous volume on left channel 1770 uint32_t mRightVolume; // previous volume on right channel 1771 uint32_t mNewLeftVolume; // new volume on left channel 1772 uint32_t mNewRightVolume; // new volume on right channel 1773 uint32_t mStrategy; // strategy for this effect chain 1774 // mSuspendedEffects lists all effects currently suspended in the chain. 1775 // Use effect type UUID timelow field as key. There is no real risk of identical 1776 // timeLow fields among effect type UUIDs. 1777 // Updated by updateSuspendedSessions_l() only. 1778 KeyedVector< int, sp<SuspendedEffectDesc> > mSuspendedEffects; 1779 }; 1780 1781 // AudioStreamOut and AudioStreamIn are immutable, so their fields are const. 1782 // For emphasis, we could also make all pointers to them be "const *", 1783 // but that would clutter the code unnecessarily. 1784 1785 struct AudioStreamOut { 1786 audio_hw_device_t* const hwDev; 1787 audio_stream_out_t* const stream; 1788 1789 AudioStreamOut(audio_hw_device_t *dev, audio_stream_out_t *out) : 1790 hwDev(dev), stream(out) {} 1791 }; 1792 1793 struct AudioStreamIn { 1794 audio_hw_device_t* const hwDev; 1795 audio_stream_in_t* const stream; 1796 1797 AudioStreamIn(audio_hw_device_t *dev, audio_stream_in_t *in) : 1798 hwDev(dev), stream(in) {} 1799 }; 1800 1801 // for mAudioSessionRefs only 1802 struct AudioSessionRef { 1803 AudioSessionRef(int sessionid, pid_t pid) : 1804 mSessionid(sessionid), mPid(pid), mCnt(1) {} 1805 const int mSessionid; 1806 const pid_t mPid; 1807 int mCnt; 1808 }; 1809 1810 enum master_volume_support { 1811 // MVS_NONE: 1812 // Audio HAL has no support for master volume, either setting or 1813 // getting. All master volume control must be implemented in SW by the 1814 // AudioFlinger mixing core. 1815 MVS_NONE, 1816 1817 // MVS_SETONLY: 1818 // Audio HAL has support for setting master volume, but not for getting 1819 // master volume (original HAL design did not include a getter). 1820 // AudioFlinger needs to keep track of the last set master volume in 1821 // addition to needing to set an initial, default, master volume at HAL 1822 // load time. 1823 MVS_SETONLY, 1824 1825 // MVS_FULL: 1826 // Audio HAL has support both for setting and getting master volume. 1827 // AudioFlinger should send all set and get master volume requests 1828 // directly to the HAL. 1829 MVS_FULL, 1830 }; 1831 1832 class AudioHwDevice { 1833 public: 1834 AudioHwDevice(const char *moduleName, audio_hw_device_t *hwDevice) : 1835 mModuleName(strdup(moduleName)), mHwDevice(hwDevice){} 1836 ~AudioHwDevice() { free((void *)mModuleName); } 1837 1838 const char *moduleName() const { return mModuleName; } 1839 audio_hw_device_t *hwDevice() const { return mHwDevice; } 1840 private: 1841 const char * const mModuleName; 1842 audio_hw_device_t * const mHwDevice; 1843 }; 1844 1845 mutable Mutex mLock; 1846 1847 DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client() 1848 1849 mutable Mutex mHardwareLock; 1850 // NOTE: If both mLock and mHardwareLock mutexes must be held, 1851 // always take mLock before mHardwareLock 1852 1853 // These two fields are immutable after onFirstRef(), so no lock needed to access 1854 audio_hw_device_t* mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL 1855 DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*> mAudioHwDevs; 1856 1857 // for dump, indicates which hardware operation is currently in progress (but not stream ops) 1858 enum hardware_call_state { 1859 AUDIO_HW_IDLE = 0, // no operation in progress 1860 AUDIO_HW_INIT, // init_check 1861 AUDIO_HW_OUTPUT_OPEN, // open_output_stream 1862 AUDIO_HW_OUTPUT_CLOSE, // unused 1863 AUDIO_HW_INPUT_OPEN, // unused 1864 AUDIO_HW_INPUT_CLOSE, // unused 1865 AUDIO_HW_STANDBY, // unused 1866 AUDIO_HW_SET_MASTER_VOLUME, // set_master_volume 1867 AUDIO_HW_GET_ROUTING, // unused 1868 AUDIO_HW_SET_ROUTING, // unused 1869 AUDIO_HW_GET_MODE, // unused 1870 AUDIO_HW_SET_MODE, // set_mode 1871 AUDIO_HW_GET_MIC_MUTE, // get_mic_mute 1872 AUDIO_HW_SET_MIC_MUTE, // set_mic_mute 1873 AUDIO_HW_SET_VOICE_VOLUME, // set_voice_volume 1874 AUDIO_HW_SET_PARAMETER, // set_parameters 1875 AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size 1876 AUDIO_HW_GET_MASTER_VOLUME, // get_master_volume 1877 AUDIO_HW_GET_PARAMETER, // get_parameters 1878 }; 1879 1880 mutable hardware_call_state mHardwareStatus; // for dump only 1881 1882 1883 DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads; 1884 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 1885 1886 // both are protected by mLock 1887 float mMasterVolume; 1888 float mMasterVolumeSW; 1889 master_volume_support mMasterVolumeSupportLvl; 1890 bool mMasterMute; 1891 1892 DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads; 1893 1894 DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients; 1895 volatile int32_t mNextUniqueId; // updated by android_atomic_inc 1896 audio_mode_t mMode; 1897 bool mBtNrecIsOff; 1898 1899 // protected by mLock 1900 Vector<AudioSessionRef*> mAudioSessionRefs; 1901 1902 float masterVolume_l() const; 1903 float masterVolumeSW_l() const { return mMasterVolumeSW; } 1904 bool masterMute_l() const { return mMasterMute; } 1905 audio_module_handle_t loadHwModule_l(const char *name); 1906 1907 Vector < sp<SyncEvent> > mPendingSyncEvents; // sync events awaiting for a session 1908 // to be created 1909 1910private: 1911 sp<Client> registerPid_l(pid_t pid); // always returns non-0 1912 1913}; 1914 1915 1916// ---------------------------------------------------------------------------- 1917 1918}; // namespace android 1919 1920#endif // ANDROID_AUDIO_FLINGER_H 1921