AudioFlinger.h revision d08f48c2ad2941d62b313007955c7145075d562c
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 void cancel() {Mutex::Autolock _l(mLock); mCallback = NULL; } 235 AudioSystem::sync_event_t type() const { return mType; } 236 int triggerSession() const { return mTriggerSession; } 237 int listenerSession() const { return mListenerSession; } 238 void *cookie() const { return mCookie; } 239 240 private: 241 const AudioSystem::sync_event_t mType; 242 const int mTriggerSession; 243 const int mListenerSession; 244 sync_event_callback_t mCallback; 245 void * const mCookie; 246 Mutex mLock; 247 }; 248 249 sp<SyncEvent> createSyncEvent(AudioSystem::sync_event_t type, 250 int triggerSession, 251 int listenerSession, 252 sync_event_callback_t callBack, 253 void *cookie); 254private: 255 audio_mode_t getMode() const { return mMode; } 256 257 bool btNrecIsOff() const { return mBtNrecIsOff; } 258 259 AudioFlinger(); 260 virtual ~AudioFlinger(); 261 262 // call in any IAudioFlinger method that accesses mPrimaryHardwareDev 263 status_t initCheck() const { return mPrimaryHardwareDev == NULL ? NO_INIT : NO_ERROR; } 264 265 // RefBase 266 virtual void onFirstRef(); 267 268 audio_hw_device_t* findSuitableHwDev_l(audio_module_handle_t module, uint32_t devices); 269 void purgeStaleEffects_l(); 270 271 // standby delay for MIXER and DUPLICATING playback threads is read from property 272 // ro.audio.flinger_standbytime_ms or defaults to kDefaultStandbyTimeInNsecs 273 static nsecs_t mStandbyTimeInNsecs; 274 275 // Internal dump utilites. 276 status_t dumpPermissionDenial(int fd, const Vector<String16>& args); 277 status_t dumpClients(int fd, const Vector<String16>& args); 278 status_t dumpInternals(int fd, const Vector<String16>& args); 279 280 // --- Client --- 281 class Client : public RefBase { 282 public: 283 Client(const sp<AudioFlinger>& audioFlinger, pid_t pid); 284 virtual ~Client(); 285 sp<MemoryDealer> heap() const; 286 pid_t pid() const { return mPid; } 287 sp<AudioFlinger> audioFlinger() const { return mAudioFlinger; } 288 289 bool reserveTimedTrack(); 290 void releaseTimedTrack(); 291 292 private: 293 Client(const Client&); 294 Client& operator = (const Client&); 295 const sp<AudioFlinger> mAudioFlinger; 296 const sp<MemoryDealer> mMemoryDealer; 297 const pid_t mPid; 298 299 Mutex mTimedTrackLock; 300 int mTimedTrackCount; 301 }; 302 303 // --- Notification Client --- 304 class NotificationClient : public IBinder::DeathRecipient { 305 public: 306 NotificationClient(const sp<AudioFlinger>& audioFlinger, 307 const sp<IAudioFlingerClient>& client, 308 pid_t pid); 309 virtual ~NotificationClient(); 310 311 sp<IAudioFlingerClient> audioFlingerClient() const { return mAudioFlingerClient; } 312 313 // IBinder::DeathRecipient 314 virtual void binderDied(const wp<IBinder>& who); 315 316 private: 317 NotificationClient(const NotificationClient&); 318 NotificationClient& operator = (const NotificationClient&); 319 320 const sp<AudioFlinger> mAudioFlinger; 321 const pid_t mPid; 322 const sp<IAudioFlingerClient> mAudioFlingerClient; 323 }; 324 325 class TrackHandle; 326 class RecordHandle; 327 class RecordThread; 328 class PlaybackThread; 329 class MixerThread; 330 class DirectOutputThread; 331 class DuplicatingThread; 332 class Track; 333 class RecordTrack; 334 class EffectModule; 335 class EffectHandle; 336 class EffectChain; 337 struct AudioStreamOut; 338 struct AudioStreamIn; 339 340 class ThreadBase : public Thread { 341 public: 342 343 enum type_t { 344 MIXER, // Thread class is MixerThread 345 DIRECT, // Thread class is DirectOutputThread 346 DUPLICATING, // Thread class is DuplicatingThread 347 RECORD // Thread class is RecordThread 348 }; 349 350 ThreadBase (const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, uint32_t device, type_t type); 351 virtual ~ThreadBase(); 352 353 status_t dumpBase(int fd, const Vector<String16>& args); 354 status_t dumpEffectChains(int fd, const Vector<String16>& args); 355 356 void clearPowerManager(); 357 358 // base for record and playback 359 class TrackBase : public ExtendedAudioBufferProvider, public RefBase { 360 361 public: 362 enum track_state { 363 IDLE, 364 TERMINATED, 365 // These are order-sensitive; do not change order without reviewing the impact. 366 // In particular there are assumptions about > STOPPED. 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; 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(); 724 725 protected: 726 // for numerous 727 friend class PlaybackThread; 728 friend class MixerThread; 729 friend class DirectOutputThread; 730 731 Track(const Track&); 732 Track& operator = (const Track&); 733 734 // AudioBufferProvider interface 735 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts = kInvalidPTS); 736 // releaseBuffer() not overridden 737 738 virtual size_t framesReady() const; 739 740 bool isMuted() const { return mMute; } 741 bool isPausing() const { 742 return mState == PAUSING; 743 } 744 bool isPaused() const { 745 return mState == PAUSED; 746 } 747 bool isResuming() const { 748 return mState == RESUMING; 749 } 750 bool isReady() const; 751 void setPaused() { mState = PAUSED; } 752 void reset(); 753 754 bool isOutputTrack() const { 755 return (mStreamType == AUDIO_STREAM_CNT); 756 } 757 758 sp<IMemory> sharedBuffer() const { return mSharedBuffer; } 759 760 bool presentationComplete(size_t framesWritten, size_t audioHalFrames); 761 void triggerEvents(AudioSystem::sync_event_t type); 762 763 public: 764 virtual bool isTimedTrack() const { return false; } 765 bool isFastTrack() const { return (mFlags & IAudioFlinger::TRACK_FAST) != 0; } 766 protected: 767 768 // we don't really need a lock for these 769 volatile bool mMute; 770 // FILLED state is used for suppressing volume ramp at begin of playing 771 enum {FS_INVALID, FS_FILLING, FS_FILLED, FS_ACTIVE}; 772 mutable uint8_t mFillingUpStatus; 773 int8_t mRetryCount; 774 const sp<IMemory> mSharedBuffer; 775 bool mResetDone; 776 const audio_stream_type_t mStreamType; 777 int mName; // track name on the normal mixer, 778 // allocated statically at track creation time, 779 // and is even allocated (though unused) for fast tracks 780 int16_t *mMainBuffer; 781 int32_t *mAuxBuffer; 782 int mAuxEffectId; 783 bool mHasVolumeController; 784 size_t mPresentationCompleteFrames; // number of frames written to the audio HAL 785 // when this track will be fully rendered 786 private: 787 IAudioFlinger::track_flags_t mFlags; 788 789 // The following fields are only for fast tracks, and should be in a subclass 790 int mFastIndex; // index within FastMixerState::mFastTracks[]; 791 // either mFastIndex == -1 792 // or 0 < mFastIndex < FastMixerState::kMaxFast because 793 // index 0 is reserved for normal mixer's submix; 794 // index is allocated statically at track creation time 795 // but the slot is only used if track is active 796 uint32_t mObservedUnderruns; // Most recently observed value of 797 // mFastMixerDumpState.mTracks[mFastIndex].mUnderruns 798 uint32_t mUnderrunCount; // Counter of total number of underruns, never reset 799 volatile float mCachedVolume; // combined master volume and stream type volume; 800 // 'volatile' means accessed without lock or 801 // barrier, but is read/written atomically 802 }; // end of Track 803 804 class TimedTrack : public Track { 805 public: 806 static sp<TimedTrack> create(PlaybackThread *thread, 807 const sp<Client>& client, 808 audio_stream_type_t streamType, 809 uint32_t sampleRate, 810 audio_format_t format, 811 uint32_t channelMask, 812 int frameCount, 813 const sp<IMemory>& sharedBuffer, 814 int sessionId); 815 ~TimedTrack(); 816 817 class TimedBuffer { 818 public: 819 TimedBuffer(); 820 TimedBuffer(const sp<IMemory>& buffer, int64_t pts); 821 const sp<IMemory>& buffer() const { return mBuffer; } 822 int64_t pts() const { return mPTS; } 823 uint32_t position() const { return mPosition; } 824 void setPosition(uint32_t pos) { mPosition = pos; } 825 private: 826 sp<IMemory> mBuffer; 827 int64_t mPTS; 828 uint32_t mPosition; 829 }; 830 831 // Mixer facing methods. 832 virtual bool isTimedTrack() const { return true; } 833 virtual size_t framesReady() const; 834 835 // AudioBufferProvider interface 836 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, 837 int64_t pts); 838 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 839 840 // Client/App facing methods. 841 status_t allocateTimedBuffer(size_t size, 842 sp<IMemory>* buffer); 843 status_t queueTimedBuffer(const sp<IMemory>& buffer, 844 int64_t pts); 845 status_t setMediaTimeTransform(const LinearTransform& xform, 846 TimedAudioTrack::TargetTimeline target); 847 848 private: 849 TimedTrack(PlaybackThread *thread, 850 const sp<Client>& client, 851 audio_stream_type_t streamType, 852 uint32_t sampleRate, 853 audio_format_t format, 854 uint32_t channelMask, 855 int frameCount, 856 const sp<IMemory>& sharedBuffer, 857 int sessionId); 858 859 void timedYieldSamples_l(AudioBufferProvider::Buffer* buffer); 860 void timedYieldSilence_l(uint32_t numFrames, 861 AudioBufferProvider::Buffer* buffer); 862 void trimTimedBufferQueue_l(); 863 void trimTimedBufferQueueHead_l(const char* logTag); 864 void updateFramesPendingAfterTrim_l(const TimedBuffer& buf, 865 const char* logTag); 866 867 uint64_t mLocalTimeFreq; 868 LinearTransform mLocalTimeToSampleTransform; 869 LinearTransform mMediaTimeToSampleTransform; 870 sp<MemoryDealer> mTimedMemoryDealer; 871 872 Vector<TimedBuffer> mTimedBufferQueue; 873 bool mQueueHeadInFlight; 874 bool mTrimQueueHeadOnRelease; 875 uint32_t mFramesPendingInQueue; 876 877 uint8_t* mTimedSilenceBuffer; 878 uint32_t mTimedSilenceBufferSize; 879 mutable Mutex mTimedBufferQueueLock; 880 bool mTimedAudioOutputOnTime; 881 CCHelper mCCHelper; 882 883 Mutex mMediaTimeTransformLock; 884 LinearTransform mMediaTimeTransform; 885 bool mMediaTimeTransformValid; 886 TimedAudioTrack::TargetTimeline mMediaTimeTransformTarget; 887 }; 888 889 890 // playback track 891 class OutputTrack : public Track { 892 public: 893 894 class Buffer: public AudioBufferProvider::Buffer { 895 public: 896 int16_t *mBuffer; 897 }; 898 899 OutputTrack(PlaybackThread *thread, 900 DuplicatingThread *sourceThread, 901 uint32_t sampleRate, 902 audio_format_t format, 903 uint32_t channelMask, 904 int frameCount); 905 virtual ~OutputTrack(); 906 907 virtual status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 908 int triggerSession = 0); 909 virtual void stop(); 910 bool write(int16_t* data, uint32_t frames); 911 bool bufferQueueEmpty() const { return mBufferQueue.size() == 0; } 912 bool isActive() const { return mActive; } 913 const wp<ThreadBase>& thread() const { return mThread; } 914 915 private: 916 917 enum { 918 NO_MORE_BUFFERS = 0x80000001, // same in AudioTrack.h, ok to be different value 919 }; 920 921 status_t obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs); 922 void clearBufferQueue(); 923 924 // Maximum number of pending buffers allocated by OutputTrack::write() 925 static const uint8_t kMaxOverFlowBuffers = 10; 926 927 Vector < Buffer* > mBufferQueue; 928 AudioBufferProvider::Buffer mOutBuffer; 929 bool mActive; 930 DuplicatingThread* const mSourceThread; // for waitTimeMs() in write() 931 }; // end of OutputTrack 932 933 PlaybackThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 934 audio_io_handle_t id, uint32_t device, type_t type); 935 virtual ~PlaybackThread(); 936 937 status_t dump(int fd, const Vector<String16>& args); 938 939 // Thread virtuals 940 virtual status_t readyToRun(); 941 virtual bool threadLoop(); 942 943 // RefBase 944 virtual void onFirstRef(); 945 946protected: 947 // Code snippets that were lifted up out of threadLoop() 948 virtual void threadLoop_mix() = 0; 949 virtual void threadLoop_sleepTime() = 0; 950 virtual void threadLoop_write(); 951 virtual void threadLoop_standby(); 952 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) { } 953 954 // prepareTracks_l reads and writes mActiveTracks, and returns 955 // the pending set of tracks to remove via Vector 'tracksToRemove'. The caller 956 // is responsible for clearing or destroying this Vector later on, when it 957 // is safe to do so. That will drop the final ref count and destroy the tracks. 958 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove) = 0; 959 960public: 961 962 virtual status_t initCheck() const { return (mOutput == NULL) ? NO_INIT : NO_ERROR; } 963 964 // return estimated latency in milliseconds, as reported by HAL 965 uint32_t latency() const; 966 967 void setMasterVolume(float value); 968 void setMasterMute(bool muted); 969 970 void setStreamVolume(audio_stream_type_t stream, float value); 971 void setStreamMute(audio_stream_type_t stream, bool muted); 972 973 float streamVolume(audio_stream_type_t stream) const; 974 975 sp<Track> createTrack_l( 976 const sp<AudioFlinger::Client>& client, 977 audio_stream_type_t streamType, 978 uint32_t sampleRate, 979 audio_format_t format, 980 uint32_t channelMask, 981 int frameCount, 982 const sp<IMemory>& sharedBuffer, 983 int sessionId, 984 IAudioFlinger::track_flags_t flags, 985 pid_t tid, 986 status_t *status); 987 988 AudioStreamOut* getOutput() const; 989 AudioStreamOut* clearOutput(); 990 virtual audio_stream_t* stream() const; 991 992 void suspend() { mSuspended++; } 993 void restore() { if (mSuspended > 0) mSuspended--; } 994 bool isSuspended() const { return (mSuspended > 0); } 995 virtual String8 getParameters(const String8& keys); 996 virtual void audioConfigChanged_l(int event, int param = 0); 997 status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames); 998 int16_t *mixBuffer() const { return mMixBuffer; }; 999 1000 virtual void detachAuxEffect_l(int effectId); 1001 status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track> track, 1002 int EffectId); 1003 status_t attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track> track, 1004 int EffectId); 1005 1006 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1007 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1008 virtual uint32_t hasAudioSession(int sessionId); 1009 virtual uint32_t getStrategyForSession_l(int sessionId); 1010 1011 1012 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 1013 virtual bool isValidSyncEvent(const sp<SyncEvent>& event); 1014 1015 protected: 1016 int16_t* mMixBuffer; 1017 uint32_t mSuspended; // suspend count, > 0 means suspended 1018 int mBytesWritten; 1019 private: 1020 // mMasterMute is in both PlaybackThread and in AudioFlinger. When a 1021 // PlaybackThread needs to find out if master-muted, it checks it's local 1022 // copy rather than the one in AudioFlinger. This optimization saves a lock. 1023 bool mMasterMute; 1024 void setMasterMute_l(bool muted) { mMasterMute = muted; } 1025 protected: 1026 SortedVector< wp<Track> > mActiveTracks; // FIXME check if this could be sp<> 1027 1028 // Allocate a track name for a given channel mask. 1029 // Returns name >= 0 if successful, -1 on failure. 1030 virtual int getTrackName_l(audio_channel_mask_t channelMask) = 0; 1031 virtual void deleteTrackName_l(int name) = 0; 1032 1033 // Time to sleep between cycles when: 1034 virtual uint32_t activeSleepTimeUs() const; // mixer state MIXER_TRACKS_ENABLED 1035 virtual uint32_t idleSleepTimeUs() const = 0; // mixer state MIXER_IDLE 1036 virtual uint32_t suspendSleepTimeUs() const = 0; // audio policy manager suspended us 1037 // No sleep when mixer state == MIXER_TRACKS_READY; relies on audio HAL stream->write() 1038 // No sleep in standby mode; waits on a condition 1039 1040 // Code snippets that are temporarily lifted up out of threadLoop() until the merge 1041 void checkSilentMode_l(); 1042 1043 // Non-trivial for DUPLICATING only 1044 virtual void saveOutputTracks() { } 1045 virtual void clearOutputTracks() { } 1046 1047 // Cache various calculated values, at threadLoop() entry and after a parameter change 1048 virtual void cacheParameters_l(); 1049 1050 private: 1051 1052 friend class AudioFlinger; // for numerous 1053 1054 PlaybackThread(const Client&); 1055 PlaybackThread& operator = (const PlaybackThread&); 1056 1057 status_t addTrack_l(const sp<Track>& track); 1058 void destroyTrack_l(const sp<Track>& track); 1059 void removeTrack_l(const sp<Track>& track); 1060 1061 void readOutputParameters(); 1062 1063 virtual status_t dumpInternals(int fd, const Vector<String16>& args); 1064 status_t dumpTracks(int fd, const Vector<String16>& args); 1065 1066 SortedVector< sp<Track> > mTracks; 1067 // mStreamTypes[] uses 1 additional stream type internally for the OutputTrack used by DuplicatingThread 1068 stream_type_t mStreamTypes[AUDIO_STREAM_CNT + 1]; 1069 AudioStreamOut *mOutput; 1070 float mMasterVolume; 1071 nsecs_t mLastWriteTime; 1072 int mNumWrites; 1073 int mNumDelayedWrites; 1074 bool mInWrite; 1075 1076 // FIXME rename these former local variables of threadLoop to standard "m" names 1077 nsecs_t standbyTime; 1078 size_t mixBufferSize; 1079 1080 // cached copies of activeSleepTimeUs() and idleSleepTimeUs() made by cacheParameters_l() 1081 uint32_t activeSleepTime; 1082 uint32_t idleSleepTime; 1083 1084 uint32_t sleepTime; 1085 1086 // mixer status returned by prepareTracks_l() 1087 mixer_state mMixerStatus; // current cycle 1088 // previous cycle when in prepareTracks_l() 1089 mixer_state mMixerStatusIgnoringFastTracks; 1090 // FIXME or a separate ready state per track 1091 1092 // FIXME move these declarations into the specific sub-class that needs them 1093 // MIXER only 1094 bool longStandbyExit; 1095 uint32_t sleepTimeShift; 1096 1097 // same as AudioFlinger::mStandbyTimeInNsecs except for DIRECT which uses a shorter value 1098 nsecs_t standbyDelay; 1099 1100 // MIXER only 1101 nsecs_t maxPeriod; 1102 1103 // DUPLICATING only 1104 uint32_t writeFrames; 1105 1106 private: 1107 // The HAL output sink is treated as non-blocking, but current implementation is blocking 1108 sp<NBAIO_Sink> mOutputSink; 1109 // If a fast mixer is present, the blocking pipe sink, otherwise clear 1110 sp<NBAIO_Sink> mPipeSink; 1111 // The current sink for the normal mixer to write it's (sub)mix, mOutputSink or mPipeSink 1112 sp<NBAIO_Sink> mNormalSink; 1113 public: 1114 virtual bool hasFastMixer() const = 0; 1115 virtual uint32_t getFastTrackUnderruns(size_t fastIndex) const { return 0; } 1116 1117 protected: 1118 // accessed by both binder threads and within threadLoop(), lock on mutex needed 1119 unsigned mFastTrackAvailMask; // bit i set if fast track [i] is available 1120 1121 }; 1122 1123 class MixerThread : public PlaybackThread { 1124 public: 1125 MixerThread (const sp<AudioFlinger>& audioFlinger, 1126 AudioStreamOut* output, 1127 audio_io_handle_t id, 1128 uint32_t device, 1129 type_t type = MIXER); 1130 virtual ~MixerThread(); 1131 1132 // Thread virtuals 1133 1134 void invalidateTracks(audio_stream_type_t streamType); 1135 virtual bool checkForNewParameters_l(); 1136 virtual status_t dumpInternals(int fd, const Vector<String16>& args); 1137 1138 protected: 1139 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1140 virtual int getTrackName_l(audio_channel_mask_t channelMask); 1141 virtual void deleteTrackName_l(int name); 1142 virtual uint32_t idleSleepTimeUs() const; 1143 virtual uint32_t suspendSleepTimeUs() const; 1144 virtual void cacheParameters_l(); 1145 1146 // threadLoop snippets 1147 virtual void threadLoop_write(); 1148 virtual void threadLoop_standby(); 1149 virtual void threadLoop_mix(); 1150 virtual void threadLoop_sleepTime(); 1151 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 1152 1153 AudioMixer* mAudioMixer; // normal mixer 1154 private: 1155#ifdef SOAKER 1156 Thread* mSoaker; 1157#endif 1158 // one-time initialization, no locks required 1159 FastMixer* mFastMixer; // non-NULL if there is also a fast mixer 1160 1161 // contents are not guaranteed to be consistent, no locks required 1162 FastMixerDumpState mFastMixerDumpState; 1163 1164 // accessible only within the threadLoop(), no locks required 1165 // mFastMixer->sq() // for mutating and pushing state 1166 int32_t mFastMixerFutex; // for cold idle 1167 1168 public: 1169 virtual bool hasFastMixer() const { return mFastMixer != NULL; } 1170 virtual uint32_t getFastTrackUnderruns(size_t fastIndex) const { 1171 ALOG_ASSERT(0 < fastIndex && 1172 fastIndex < FastMixerState::kMaxFastTracks); 1173 return mFastMixerDumpState.mTracks[fastIndex].mUnderruns; 1174 } 1175 }; 1176 1177 class DirectOutputThread : public PlaybackThread { 1178 public: 1179 1180 DirectOutputThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1181 audio_io_handle_t id, uint32_t device); 1182 virtual ~DirectOutputThread(); 1183 1184 // Thread virtuals 1185 1186 virtual bool checkForNewParameters_l(); 1187 1188 protected: 1189 virtual int getTrackName_l(audio_channel_mask_t channelMask); 1190 virtual void deleteTrackName_l(int name); 1191 virtual uint32_t activeSleepTimeUs() const; 1192 virtual uint32_t idleSleepTimeUs() const; 1193 virtual uint32_t suspendSleepTimeUs() const; 1194 virtual void cacheParameters_l(); 1195 1196 // threadLoop snippets 1197 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1198 virtual void threadLoop_mix(); 1199 virtual void threadLoop_sleepTime(); 1200 1201 // volumes last sent to audio HAL with stream->set_volume() 1202 // FIXME use standard representation and names 1203 float mLeftVolFloat; 1204 float mRightVolFloat; 1205 uint16_t mLeftVolShort; 1206 uint16_t mRightVolShort; 1207 1208 // FIXME rename these former local variables of threadLoop to standard names 1209 // next 3 were local to the while !exitingPending loop 1210 bool rampVolume; 1211 uint16_t leftVol; 1212 uint16_t rightVol; 1213 1214private: 1215 // prepareTracks_l() tells threadLoop_mix() the name of the single active track 1216 sp<Track> mActiveTrack; 1217 public: 1218 virtual bool hasFastMixer() const { return false; } 1219 }; 1220 1221 class DuplicatingThread : public MixerThread { 1222 public: 1223 DuplicatingThread (const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread, 1224 audio_io_handle_t id); 1225 virtual ~DuplicatingThread(); 1226 1227 // Thread virtuals 1228 void addOutputTrack(MixerThread* thread); 1229 void removeOutputTrack(MixerThread* thread); 1230 uint32_t waitTimeMs() const { return mWaitTimeMs; } 1231 protected: 1232 virtual uint32_t activeSleepTimeUs() const; 1233 1234 private: 1235 bool outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks); 1236 protected: 1237 // threadLoop snippets 1238 virtual void threadLoop_mix(); 1239 virtual void threadLoop_sleepTime(); 1240 virtual void threadLoop_write(); 1241 virtual void threadLoop_standby(); 1242 virtual void cacheParameters_l(); 1243 1244 private: 1245 // called from threadLoop, addOutputTrack, removeOutputTrack 1246 virtual void updateWaitTime_l(); 1247 protected: 1248 virtual void saveOutputTracks(); 1249 virtual void clearOutputTracks(); 1250 private: 1251 1252 uint32_t mWaitTimeMs; 1253 SortedVector < sp<OutputTrack> > outputTracks; 1254 SortedVector < sp<OutputTrack> > mOutputTracks; 1255 public: 1256 virtual bool hasFastMixer() const { return false; } 1257 }; 1258 1259 PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const; 1260 MixerThread *checkMixerThread_l(audio_io_handle_t output) const; 1261 RecordThread *checkRecordThread_l(audio_io_handle_t input) const; 1262 // no range check, AudioFlinger::mLock held 1263 bool streamMute_l(audio_stream_type_t stream) const 1264 { return mStreamTypes[stream].mute; } 1265 // no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held 1266 float streamVolume_l(audio_stream_type_t stream) const 1267 { return mStreamTypes[stream].volume; } 1268 void audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2); 1269 1270 // allocate an audio_io_handle_t, session ID, or effect ID 1271 uint32_t nextUniqueId(); 1272 1273 status_t moveEffectChain_l(int sessionId, 1274 PlaybackThread *srcThread, 1275 PlaybackThread *dstThread, 1276 bool reRegister); 1277 // return thread associated with primary hardware device, or NULL 1278 PlaybackThread *primaryPlaybackThread_l() const; 1279 uint32_t primaryOutputDevice_l() const; 1280 1281 // server side of the client's IAudioTrack 1282 class TrackHandle : public android::BnAudioTrack { 1283 public: 1284 TrackHandle(const sp<PlaybackThread::Track>& track); 1285 virtual ~TrackHandle(); 1286 virtual sp<IMemory> getCblk() const; 1287 virtual status_t start(); 1288 virtual void stop(); 1289 virtual void flush(); 1290 virtual void mute(bool); 1291 virtual void pause(); 1292 virtual status_t attachAuxEffect(int effectId); 1293 virtual status_t allocateTimedBuffer(size_t size, 1294 sp<IMemory>* buffer); 1295 virtual status_t queueTimedBuffer(const sp<IMemory>& buffer, 1296 int64_t pts); 1297 virtual status_t setMediaTimeTransform(const LinearTransform& xform, 1298 int target); 1299 virtual status_t onTransact( 1300 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 1301 private: 1302 const sp<PlaybackThread::Track> mTrack; 1303 }; 1304 1305 void removeClient_l(pid_t pid); 1306 void removeNotificationClient(pid_t pid); 1307 1308 1309 // record thread 1310 class RecordThread : public ThreadBase, public AudioBufferProvider 1311 { 1312 public: 1313 1314 // record track 1315 class RecordTrack : public TrackBase { 1316 public: 1317 RecordTrack(RecordThread *thread, 1318 const sp<Client>& client, 1319 uint32_t sampleRate, 1320 audio_format_t format, 1321 uint32_t channelMask, 1322 int frameCount, 1323 int sessionId); 1324 virtual ~RecordTrack(); 1325 1326 virtual status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 1327 int triggerSession = 0); 1328 virtual void stop(); 1329 1330 bool overflow() { bool tmp = mOverflow; mOverflow = false; return tmp; } 1331 bool setOverflow() { bool tmp = mOverflow; mOverflow = true; return tmp; } 1332 1333 void dump(char* buffer, size_t size); 1334 1335 private: 1336 friend class AudioFlinger; // for mState 1337 1338 RecordTrack(const RecordTrack&); 1339 RecordTrack& operator = (const RecordTrack&); 1340 1341 // AudioBufferProvider interface 1342 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts = kInvalidPTS); 1343 // releaseBuffer() not overridden 1344 1345 bool mOverflow; 1346 }; 1347 1348 1349 RecordThread(const sp<AudioFlinger>& audioFlinger, 1350 AudioStreamIn *input, 1351 uint32_t sampleRate, 1352 uint32_t channels, 1353 audio_io_handle_t id, 1354 uint32_t device); 1355 virtual ~RecordThread(); 1356 1357 // Thread 1358 virtual bool threadLoop(); 1359 virtual status_t readyToRun(); 1360 1361 // RefBase 1362 virtual void onFirstRef(); 1363 1364 virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; } 1365 sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l( 1366 const sp<AudioFlinger::Client>& client, 1367 uint32_t sampleRate, 1368 audio_format_t format, 1369 int channelMask, 1370 int frameCount, 1371 int sessionId, 1372 status_t *status); 1373 1374 status_t start(RecordTrack* recordTrack, 1375 AudioSystem::sync_event_t event, 1376 int triggerSession); 1377 void stop(RecordTrack* recordTrack); 1378 status_t dump(int fd, const Vector<String16>& args); 1379 AudioStreamIn* getInput() const; 1380 AudioStreamIn* clearInput(); 1381 virtual audio_stream_t* stream() const; 1382 1383 // AudioBufferProvider interface 1384 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts); 1385 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 1386 1387 virtual bool checkForNewParameters_l(); 1388 virtual String8 getParameters(const String8& keys); 1389 virtual void audioConfigChanged_l(int event, int param = 0); 1390 void readInputParameters(); 1391 virtual unsigned int getInputFramesLost(); 1392 1393 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1394 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1395 virtual uint32_t hasAudioSession(int sessionId); 1396 RecordTrack* track(); 1397 1398 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 1399 virtual bool isValidSyncEvent(const sp<SyncEvent>& event); 1400 1401 static void syncStartEventCallback(const wp<SyncEvent>& event); 1402 void handleSyncStartEvent(const sp<SyncEvent>& event); 1403 1404 private: 1405 void clearSyncStartEvent(); 1406 1407 RecordThread(); 1408 AudioStreamIn *mInput; 1409 RecordTrack* mTrack; 1410 sp<RecordTrack> mActiveTrack; 1411 Condition mStartStopCond; 1412 AudioResampler *mResampler; 1413 int32_t *mRsmpOutBuffer; 1414 int16_t *mRsmpInBuffer; 1415 size_t mRsmpInIndex; 1416 size_t mInputBytes; 1417 const int mReqChannelCount; 1418 const uint32_t mReqSampleRate; 1419 ssize_t mBytesRead; 1420 // sync event triggering actual audio capture. Frames read before this event will 1421 // be dropped and therefore not read by the application. 1422 sp<SyncEvent> mSyncStartEvent; 1423 // number of captured frames to drop after the start sync event has been received. 1424 ssize_t mFramestoDrop; 1425 }; 1426 1427 // server side of the client's IAudioRecord 1428 class RecordHandle : public android::BnAudioRecord { 1429 public: 1430 RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack); 1431 virtual ~RecordHandle(); 1432 virtual sp<IMemory> getCblk() const; 1433 virtual status_t start(int event, int triggerSession); 1434 virtual void stop(); 1435 virtual status_t onTransact( 1436 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 1437 private: 1438 const sp<RecordThread::RecordTrack> mRecordTrack; 1439 }; 1440 1441 //--- Audio Effect Management 1442 1443 // EffectModule and EffectChain classes both have their own mutex to protect 1444 // state changes or resource modifications. Always respect the following order 1445 // if multiple mutexes must be acquired to avoid cross deadlock: 1446 // AudioFlinger -> ThreadBase -> EffectChain -> EffectModule 1447 1448 // The EffectModule class is a wrapper object controlling the effect engine implementation 1449 // in the effect library. It prevents concurrent calls to process() and command() functions 1450 // from different client threads. It keeps a list of EffectHandle objects corresponding 1451 // to all client applications using this effect and notifies applications of effect state, 1452 // control or parameter changes. It manages the activation state machine to send appropriate 1453 // reset, enable, disable commands to effect engine and provide volume 1454 // ramping when effects are activated/deactivated. 1455 // When controlling an auxiliary effect, the EffectModule also provides an input buffer used by 1456 // the attached track(s) to accumulate their auxiliary channel. 1457 class EffectModule: public RefBase { 1458 public: 1459 EffectModule(ThreadBase *thread, 1460 const wp<AudioFlinger::EffectChain>& chain, 1461 effect_descriptor_t *desc, 1462 int id, 1463 int sessionId); 1464 virtual ~EffectModule(); 1465 1466 enum effect_state { 1467 IDLE, 1468 RESTART, 1469 STARTING, 1470 ACTIVE, 1471 STOPPING, 1472 STOPPED, 1473 DESTROYED 1474 }; 1475 1476 int id() const { return mId; } 1477 void process(); 1478 void updateState(); 1479 status_t command(uint32_t cmdCode, 1480 uint32_t cmdSize, 1481 void *pCmdData, 1482 uint32_t *replySize, 1483 void *pReplyData); 1484 1485 void reset_l(); 1486 status_t configure(); 1487 status_t init(); 1488 effect_state state() const { 1489 return mState; 1490 } 1491 uint32_t status() { 1492 return mStatus; 1493 } 1494 int sessionId() const { 1495 return mSessionId; 1496 } 1497 status_t setEnabled(bool enabled); 1498 bool isEnabled() const; 1499 bool isProcessEnabled() const; 1500 1501 void setInBuffer(int16_t *buffer) { mConfig.inputCfg.buffer.s16 = buffer; } 1502 int16_t *inBuffer() { return mConfig.inputCfg.buffer.s16; } 1503 void setOutBuffer(int16_t *buffer) { mConfig.outputCfg.buffer.s16 = buffer; } 1504 int16_t *outBuffer() { return mConfig.outputCfg.buffer.s16; } 1505 void setChain(const wp<EffectChain>& chain) { mChain = chain; } 1506 void setThread(const wp<ThreadBase>& thread) { mThread = thread; } 1507 const wp<ThreadBase>& thread() { return mThread; } 1508 1509 status_t addHandle(const sp<EffectHandle>& handle); 1510 void disconnect(const wp<EffectHandle>& handle, bool unpinIfLast); 1511 size_t removeHandle (const wp<EffectHandle>& handle); 1512 1513 effect_descriptor_t& desc() { return mDescriptor; } 1514 wp<EffectChain>& chain() { return mChain; } 1515 1516 status_t setDevice(uint32_t device); 1517 status_t setVolume(uint32_t *left, uint32_t *right, bool controller); 1518 status_t setMode(audio_mode_t mode); 1519 status_t start(); 1520 status_t stop(); 1521 void setSuspended(bool suspended); 1522 bool suspended() const; 1523 1524 sp<EffectHandle> controlHandle(); 1525 1526 bool isPinned() const { return mPinned; } 1527 void unPin() { mPinned = false; } 1528 1529 status_t dump(int fd, const Vector<String16>& args); 1530 1531 protected: 1532 friend class AudioFlinger; // for mHandles 1533 bool mPinned; 1534 1535 // Maximum time allocated to effect engines to complete the turn off sequence 1536 static const uint32_t MAX_DISABLE_TIME_MS = 10000; 1537 1538 EffectModule(const EffectModule&); 1539 EffectModule& operator = (const EffectModule&); 1540 1541 status_t start_l(); 1542 status_t stop_l(); 1543 1544mutable Mutex mLock; // mutex for process, commands and handles list protection 1545 wp<ThreadBase> mThread; // parent thread 1546 wp<EffectChain> mChain; // parent effect chain 1547 int mId; // this instance unique ID 1548 int mSessionId; // audio session ID 1549 effect_descriptor_t mDescriptor;// effect descriptor received from effect engine 1550 effect_config_t mConfig; // input and output audio configuration 1551 effect_handle_t mEffectInterface; // Effect module C API 1552 status_t mStatus; // initialization status 1553 effect_state mState; // current activation state 1554 Vector< wp<EffectHandle> > mHandles; // list of client handles 1555 // First handle in mHandles has highest priority and controls the effect module 1556 uint32_t mMaxDisableWaitCnt; // maximum grace period before forcing an effect off after 1557 // sending disable command. 1558 uint32_t mDisableWaitCnt; // current process() calls count during disable period. 1559 bool mSuspended; // effect is suspended: temporarily disabled by framework 1560 }; 1561 1562 // The EffectHandle class implements the IEffect interface. It provides resources 1563 // to receive parameter updates, keeps track of effect control 1564 // ownership and state and has a pointer to the EffectModule object it is controlling. 1565 // There is one EffectHandle object for each application controlling (or using) 1566 // an effect module. 1567 // The EffectHandle is obtained by calling AudioFlinger::createEffect(). 1568 class EffectHandle: public android::BnEffect { 1569 public: 1570 1571 EffectHandle(const sp<EffectModule>& effect, 1572 const sp<AudioFlinger::Client>& client, 1573 const sp<IEffectClient>& effectClient, 1574 int32_t priority); 1575 virtual ~EffectHandle(); 1576 1577 // IEffect 1578 virtual status_t enable(); 1579 virtual status_t disable(); 1580 virtual status_t command(uint32_t cmdCode, 1581 uint32_t cmdSize, 1582 void *pCmdData, 1583 uint32_t *replySize, 1584 void *pReplyData); 1585 virtual void disconnect(); 1586 private: 1587 void disconnect(bool unpinIfLast); 1588 public: 1589 virtual sp<IMemory> getCblk() const { return mCblkMemory; } 1590 virtual status_t onTransact(uint32_t code, const Parcel& data, 1591 Parcel* reply, uint32_t flags); 1592 1593 1594 // Give or take control of effect module 1595 // - hasControl: true if control is given, false if removed 1596 // - signal: true client app should be signaled of change, false otherwise 1597 // - enabled: state of the effect when control is passed 1598 void setControl(bool hasControl, bool signal, bool enabled); 1599 void commandExecuted(uint32_t cmdCode, 1600 uint32_t cmdSize, 1601 void *pCmdData, 1602 uint32_t replySize, 1603 void *pReplyData); 1604 void setEnabled(bool enabled); 1605 bool enabled() const { return mEnabled; } 1606 1607 // Getters 1608 int id() const { return mEffect->id(); } 1609 int priority() const { return mPriority; } 1610 bool hasControl() const { return mHasControl; } 1611 sp<EffectModule> effect() const { return mEffect; } 1612 1613 void dump(char* buffer, size_t size); 1614 1615 protected: 1616 friend class AudioFlinger; // for mEffect, mHasControl, mEnabled 1617 EffectHandle(const EffectHandle&); 1618 EffectHandle& operator =(const EffectHandle&); 1619 1620 sp<EffectModule> mEffect; // pointer to controlled EffectModule 1621 sp<IEffectClient> mEffectClient; // callback interface for client notifications 1622 /*const*/ sp<Client> mClient; // client for shared memory allocation, see disconnect() 1623 sp<IMemory> mCblkMemory; // shared memory for control block 1624 effect_param_cblk_t* mCblk; // control block for deferred parameter setting via shared memory 1625 uint8_t* mBuffer; // pointer to parameter area in shared memory 1626 int mPriority; // client application priority to control the effect 1627 bool mHasControl; // true if this handle is controlling the effect 1628 bool mEnabled; // cached enable state: needed when the effect is 1629 // restored after being suspended 1630 }; 1631 1632 // the EffectChain class represents a group of effects associated to one audio session. 1633 // There can be any number of EffectChain objects per output mixer thread (PlaybackThread). 1634 // The EffecChain with session ID 0 contains global effects applied to the output mix. 1635 // Effects in this chain can be insert or auxiliary. Effects in other chains (attached to tracks) 1636 // are insert only. The EffectChain maintains an ordered list of effect module, the order corresponding 1637 // in the effect process order. When attached to a track (session ID != 0), it also provide it's own 1638 // input buffer used by the track as accumulation buffer. 1639 class EffectChain: public RefBase { 1640 public: 1641 EffectChain(const wp<ThreadBase>& wThread, int sessionId); 1642 EffectChain(ThreadBase *thread, int sessionId); 1643 virtual ~EffectChain(); 1644 1645 // special key used for an entry in mSuspendedEffects keyed vector 1646 // corresponding to a suspend all request. 1647 static const int kKeyForSuspendAll = 0; 1648 1649 // minimum duration during which we force calling effect process when last track on 1650 // a session is stopped or removed to allow effect tail to be rendered 1651 static const int kProcessTailDurationMs = 1000; 1652 1653 void process_l(); 1654 1655 void lock() { 1656 mLock.lock(); 1657 } 1658 void unlock() { 1659 mLock.unlock(); 1660 } 1661 1662 status_t addEffect_l(const sp<EffectModule>& handle); 1663 size_t removeEffect_l(const sp<EffectModule>& handle); 1664 1665 int sessionId() const { return mSessionId; } 1666 void setSessionId(int sessionId) { mSessionId = sessionId; } 1667 1668 sp<EffectModule> getEffectFromDesc_l(effect_descriptor_t *descriptor); 1669 sp<EffectModule> getEffectFromId_l(int id); 1670 sp<EffectModule> getEffectFromType_l(const effect_uuid_t *type); 1671 bool setVolume_l(uint32_t *left, uint32_t *right); 1672 void setDevice_l(uint32_t device); 1673 void setMode_l(audio_mode_t mode); 1674 1675 void setInBuffer(int16_t *buffer, bool ownsBuffer = false) { 1676 mInBuffer = buffer; 1677 mOwnInBuffer = ownsBuffer; 1678 } 1679 int16_t *inBuffer() const { 1680 return mInBuffer; 1681 } 1682 void setOutBuffer(int16_t *buffer) { 1683 mOutBuffer = buffer; 1684 } 1685 int16_t *outBuffer() const { 1686 return mOutBuffer; 1687 } 1688 1689 void incTrackCnt() { android_atomic_inc(&mTrackCnt); } 1690 void decTrackCnt() { android_atomic_dec(&mTrackCnt); } 1691 int32_t trackCnt() const { return mTrackCnt;} 1692 1693 void incActiveTrackCnt() { android_atomic_inc(&mActiveTrackCnt); 1694 mTailBufferCount = mMaxTailBuffers; } 1695 void decActiveTrackCnt() { android_atomic_dec(&mActiveTrackCnt); } 1696 int32_t activeTrackCnt() const { return mActiveTrackCnt;} 1697 1698 uint32_t strategy() const { return mStrategy; } 1699 void setStrategy(uint32_t strategy) 1700 { mStrategy = strategy; } 1701 1702 // suspend effect of the given type 1703 void setEffectSuspended_l(const effect_uuid_t *type, 1704 bool suspend); 1705 // suspend all eligible effects 1706 void setEffectSuspendedAll_l(bool suspend); 1707 // check if effects should be suspend or restored when a given effect is enable or disabled 1708 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1709 bool enabled); 1710 1711 status_t dump(int fd, const Vector<String16>& args); 1712 1713 protected: 1714 friend class AudioFlinger; // for mThread, mEffects 1715 EffectChain(const EffectChain&); 1716 EffectChain& operator =(const EffectChain&); 1717 1718 class SuspendedEffectDesc : public RefBase { 1719 public: 1720 SuspendedEffectDesc() : mRefCount(0) {} 1721 1722 int mRefCount; 1723 effect_uuid_t mType; 1724 wp<EffectModule> mEffect; 1725 }; 1726 1727 // get a list of effect modules to suspend when an effect of the type 1728 // passed is enabled. 1729 void getSuspendEligibleEffects(Vector< sp<EffectModule> > &effects); 1730 1731 // get an effect module if it is currently enable 1732 sp<EffectModule> getEffectIfEnabled(const effect_uuid_t *type); 1733 // true if the effect whose descriptor is passed can be suspended 1734 // OEMs can modify the rules implemented in this method to exclude specific effect 1735 // types or implementations from the suspend/restore mechanism. 1736 bool isEffectEligibleForSuspend(const effect_descriptor_t& desc); 1737 1738 wp<ThreadBase> mThread; // parent mixer thread 1739 Mutex mLock; // mutex protecting effect list 1740 Vector< sp<EffectModule> > mEffects; // list of effect modules 1741 int mSessionId; // audio session ID 1742 int16_t *mInBuffer; // chain input buffer 1743 int16_t *mOutBuffer; // chain output buffer 1744 volatile int32_t mActiveTrackCnt; // number of active tracks connected 1745 volatile int32_t mTrackCnt; // number of tracks connected 1746 int32_t mTailBufferCount; // current effect tail buffer count 1747 int32_t mMaxTailBuffers; // maximum effect tail buffers 1748 bool mOwnInBuffer; // true if the chain owns its input buffer 1749 int mVolumeCtrlIdx; // index of insert effect having control over volume 1750 uint32_t mLeftVolume; // previous volume on left channel 1751 uint32_t mRightVolume; // previous volume on right channel 1752 uint32_t mNewLeftVolume; // new volume on left channel 1753 uint32_t mNewRightVolume; // new volume on right channel 1754 uint32_t mStrategy; // strategy for this effect chain 1755 // mSuspendedEffects lists all effects currently suspended in the chain. 1756 // Use effect type UUID timelow field as key. There is no real risk of identical 1757 // timeLow fields among effect type UUIDs. 1758 // Updated by updateSuspendedSessions_l() only. 1759 KeyedVector< int, sp<SuspendedEffectDesc> > mSuspendedEffects; 1760 }; 1761 1762 // AudioStreamOut and AudioStreamIn are immutable, so their fields are const. 1763 // For emphasis, we could also make all pointers to them be "const *", 1764 // but that would clutter the code unnecessarily. 1765 1766 struct AudioStreamOut { 1767 audio_hw_device_t* const hwDev; 1768 audio_stream_out_t* const stream; 1769 1770 AudioStreamOut(audio_hw_device_t *dev, audio_stream_out_t *out) : 1771 hwDev(dev), stream(out) {} 1772 }; 1773 1774 struct AudioStreamIn { 1775 audio_hw_device_t* const hwDev; 1776 audio_stream_in_t* const stream; 1777 1778 AudioStreamIn(audio_hw_device_t *dev, audio_stream_in_t *in) : 1779 hwDev(dev), stream(in) {} 1780 }; 1781 1782 // for mAudioSessionRefs only 1783 struct AudioSessionRef { 1784 AudioSessionRef(int sessionid, pid_t pid) : 1785 mSessionid(sessionid), mPid(pid), mCnt(1) {} 1786 const int mSessionid; 1787 const pid_t mPid; 1788 int mCnt; 1789 }; 1790 1791 enum master_volume_support { 1792 // MVS_NONE: 1793 // Audio HAL has no support for master volume, either setting or 1794 // getting. All master volume control must be implemented in SW by the 1795 // AudioFlinger mixing core. 1796 MVS_NONE, 1797 1798 // MVS_SETONLY: 1799 // Audio HAL has support for setting master volume, but not for getting 1800 // master volume (original HAL design did not include a getter). 1801 // AudioFlinger needs to keep track of the last set master volume in 1802 // addition to needing to set an initial, default, master volume at HAL 1803 // load time. 1804 MVS_SETONLY, 1805 1806 // MVS_FULL: 1807 // Audio HAL has support both for setting and getting master volume. 1808 // AudioFlinger should send all set and get master volume requests 1809 // directly to the HAL. 1810 MVS_FULL, 1811 }; 1812 1813 class AudioHwDevice { 1814 public: 1815 AudioHwDevice(const char *moduleName, audio_hw_device_t *hwDevice) : 1816 mModuleName(strdup(moduleName)), mHwDevice(hwDevice){} 1817 ~AudioHwDevice() { free((void *)mModuleName); } 1818 1819 const char *moduleName() const { return mModuleName; } 1820 audio_hw_device_t *hwDevice() const { return mHwDevice; } 1821 private: 1822 const char * const mModuleName; 1823 audio_hw_device_t * const mHwDevice; 1824 }; 1825 1826 mutable Mutex mLock; 1827 1828 DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client() 1829 1830 mutable Mutex mHardwareLock; 1831 // NOTE: If both mLock and mHardwareLock mutexes must be held, 1832 // always take mLock before mHardwareLock 1833 1834 // These two fields are immutable after onFirstRef(), so no lock needed to access 1835 audio_hw_device_t* mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL 1836 DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*> mAudioHwDevs; 1837 1838 // for dump, indicates which hardware operation is currently in progress (but not stream ops) 1839 enum hardware_call_state { 1840 AUDIO_HW_IDLE = 0, // no operation in progress 1841 AUDIO_HW_INIT, // init_check 1842 AUDIO_HW_OUTPUT_OPEN, // open_output_stream 1843 AUDIO_HW_OUTPUT_CLOSE, // unused 1844 AUDIO_HW_INPUT_OPEN, // unused 1845 AUDIO_HW_INPUT_CLOSE, // unused 1846 AUDIO_HW_STANDBY, // unused 1847 AUDIO_HW_SET_MASTER_VOLUME, // set_master_volume 1848 AUDIO_HW_GET_ROUTING, // unused 1849 AUDIO_HW_SET_ROUTING, // unused 1850 AUDIO_HW_GET_MODE, // unused 1851 AUDIO_HW_SET_MODE, // set_mode 1852 AUDIO_HW_GET_MIC_MUTE, // get_mic_mute 1853 AUDIO_HW_SET_MIC_MUTE, // set_mic_mute 1854 AUDIO_HW_SET_VOICE_VOLUME, // set_voice_volume 1855 AUDIO_HW_SET_PARAMETER, // set_parameters 1856 AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size 1857 AUDIO_HW_GET_MASTER_VOLUME, // get_master_volume 1858 AUDIO_HW_GET_PARAMETER, // get_parameters 1859 }; 1860 1861 mutable hardware_call_state mHardwareStatus; // for dump only 1862 1863 1864 DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads; 1865 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 1866 1867 // both are protected by mLock 1868 float mMasterVolume; 1869 float mMasterVolumeSW; 1870 master_volume_support mMasterVolumeSupportLvl; 1871 bool mMasterMute; 1872 1873 DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads; 1874 1875 DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients; 1876 volatile int32_t mNextUniqueId; // updated by android_atomic_inc 1877 audio_mode_t mMode; 1878 bool mBtNrecIsOff; 1879 1880 // protected by mLock 1881 Vector<AudioSessionRef*> mAudioSessionRefs; 1882 1883 float masterVolume_l() const; 1884 float masterVolumeSW_l() const { return mMasterVolumeSW; } 1885 bool masterMute_l() const { return mMasterMute; } 1886 audio_module_handle_t loadHwModule_l(const char *name); 1887 1888 Vector < sp<SyncEvent> > mPendingSyncEvents; // sync events awaiting for a session 1889 // to be created 1890 1891private: 1892 sp<Client> registerPid_l(pid_t pid); // always returns non-0 1893 1894}; 1895 1896 1897// ---------------------------------------------------------------------------- 1898 1899}; // namespace android 1900 1901#endif // ANDROID_AUDIO_FLINGER_H 1902