AudioFlinger.h revision c1dae24a08b67b98e18e4239d4f3a74d600d353c
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 audio_channel_mask_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 audio_channel_mask_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, 146 audio_channel_mask_t channelMask) const; 147 148 virtual audio_io_handle_t openOutput(audio_module_handle_t module, 149 audio_devices_t *pDevices, 150 uint32_t *pSamplingRate, 151 audio_format_t *pFormat, 152 audio_channel_mask_t *pChannelMask, 153 uint32_t *pLatencyMs, 154 audio_output_flags_t flags); 155 156 virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1, 157 audio_io_handle_t output2); 158 159 virtual status_t closeOutput(audio_io_handle_t output); 160 161 virtual status_t suspendOutput(audio_io_handle_t output); 162 163 virtual status_t restoreOutput(audio_io_handle_t output); 164 165 virtual audio_io_handle_t openInput(audio_module_handle_t module, 166 audio_devices_t *pDevices, 167 uint32_t *pSamplingRate, 168 audio_format_t *pFormat, 169 audio_channel_mask_t *pChannelMask); 170 171 virtual status_t closeInput(audio_io_handle_t input); 172 173 virtual status_t setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output); 174 175 virtual status_t setVoiceVolume(float volume); 176 177 virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 178 audio_io_handle_t output) const; 179 180 virtual unsigned int getInputFramesLost(audio_io_handle_t ioHandle) const; 181 182 virtual int newAudioSessionId(); 183 184 virtual void acquireAudioSessionId(int audioSession); 185 186 virtual void releaseAudioSessionId(int audioSession); 187 188 virtual status_t queryNumberEffects(uint32_t *numEffects) const; 189 190 virtual status_t queryEffect(uint32_t index, effect_descriptor_t *descriptor) const; 191 192 virtual status_t getEffectDescriptor(const effect_uuid_t *pUuid, 193 effect_descriptor_t *descriptor) const; 194 195 virtual sp<IEffect> createEffect(pid_t pid, 196 effect_descriptor_t *pDesc, 197 const sp<IEffectClient>& effectClient, 198 int32_t priority, 199 audio_io_handle_t io, 200 int sessionId, 201 status_t *status, 202 int *id, 203 int *enabled); 204 205 virtual status_t moveEffects(int sessionId, audio_io_handle_t srcOutput, 206 audio_io_handle_t dstOutput); 207 208 virtual audio_module_handle_t loadHwModule(const char *name); 209 210 virtual status_t onTransact( 211 uint32_t code, 212 const Parcel& data, 213 Parcel* reply, 214 uint32_t flags); 215 216 // end of IAudioFlinger interface 217 218 class SyncEvent; 219 220 typedef void (*sync_event_callback_t)(const wp<SyncEvent>& event) ; 221 222 class SyncEvent : public RefBase { 223 public: 224 SyncEvent(AudioSystem::sync_event_t type, 225 int triggerSession, 226 int listenerSession, 227 sync_event_callback_t callBack, 228 void *cookie) 229 : mType(type), mTriggerSession(triggerSession), mListenerSession(listenerSession), 230 mCallback(callBack), mCookie(cookie) 231 {} 232 233 virtual ~SyncEvent() {} 234 235 void trigger() { Mutex::Autolock _l(mLock); if (mCallback) mCallback(this); } 236 bool isCancelled() { Mutex::Autolock _l(mLock); return (mCallback == NULL); } 237 void cancel() {Mutex::Autolock _l(mLock); mCallback = NULL; } 238 AudioSystem::sync_event_t type() const { return mType; } 239 int triggerSession() const { return mTriggerSession; } 240 int listenerSession() const { return mListenerSession; } 241 void *cookie() const { return mCookie; } 242 243 private: 244 const AudioSystem::sync_event_t mType; 245 const int mTriggerSession; 246 const int mListenerSession; 247 sync_event_callback_t mCallback; 248 void * const mCookie; 249 Mutex mLock; 250 }; 251 252 sp<SyncEvent> createSyncEvent(AudioSystem::sync_event_t type, 253 int triggerSession, 254 int listenerSession, 255 sync_event_callback_t callBack, 256 void *cookie); 257 258private: 259 audio_mode_t getMode() const { return mMode; } 260 261 bool btNrecIsOff() const { return mBtNrecIsOff; } 262 263 AudioFlinger(); 264 virtual ~AudioFlinger(); 265 266 // call in any IAudioFlinger method that accesses mPrimaryHardwareDev 267 status_t initCheck() const { return mPrimaryHardwareDev == NULL ? NO_INIT : NO_ERROR; } 268 269 // RefBase 270 virtual void onFirstRef(); 271 272 audio_hw_device_t* findSuitableHwDev_l(audio_module_handle_t module, uint32_t devices); 273 void purgeStaleEffects_l(); 274 275 // standby delay for MIXER and DUPLICATING playback threads is read from property 276 // ro.audio.flinger_standbytime_ms or defaults to kDefaultStandbyTimeInNsecs 277 static nsecs_t mStandbyTimeInNsecs; 278 279 // Internal dump utilites. 280 status_t dumpPermissionDenial(int fd, const Vector<String16>& args); 281 status_t dumpClients(int fd, const Vector<String16>& args); 282 status_t dumpInternals(int fd, const Vector<String16>& args); 283 284 // --- Client --- 285 class Client : public RefBase { 286 public: 287 Client(const sp<AudioFlinger>& audioFlinger, pid_t pid); 288 virtual ~Client(); 289 sp<MemoryDealer> heap() const; 290 pid_t pid() const { return mPid; } 291 sp<AudioFlinger> audioFlinger() const { return mAudioFlinger; } 292 293 bool reserveTimedTrack(); 294 void releaseTimedTrack(); 295 296 private: 297 Client(const Client&); 298 Client& operator = (const Client&); 299 const sp<AudioFlinger> mAudioFlinger; 300 const sp<MemoryDealer> mMemoryDealer; 301 const pid_t mPid; 302 303 Mutex mTimedTrackLock; 304 int mTimedTrackCount; 305 }; 306 307 // --- Notification Client --- 308 class NotificationClient : public IBinder::DeathRecipient { 309 public: 310 NotificationClient(const sp<AudioFlinger>& audioFlinger, 311 const sp<IAudioFlingerClient>& client, 312 pid_t pid); 313 virtual ~NotificationClient(); 314 315 sp<IAudioFlingerClient> audioFlingerClient() const { return mAudioFlingerClient; } 316 317 // IBinder::DeathRecipient 318 virtual void binderDied(const wp<IBinder>& who); 319 320 private: 321 NotificationClient(const NotificationClient&); 322 NotificationClient& operator = (const NotificationClient&); 323 324 const sp<AudioFlinger> mAudioFlinger; 325 const pid_t mPid; 326 const sp<IAudioFlingerClient> mAudioFlingerClient; 327 }; 328 329 class TrackHandle; 330 class RecordHandle; 331 class RecordThread; 332 class PlaybackThread; 333 class MixerThread; 334 class DirectOutputThread; 335 class DuplicatingThread; 336 class Track; 337 class RecordTrack; 338 class EffectModule; 339 class EffectHandle; 340 class EffectChain; 341 struct AudioStreamOut; 342 struct AudioStreamIn; 343 344 class ThreadBase : public Thread { 345 public: 346 347 enum type_t { 348 MIXER, // Thread class is MixerThread 349 DIRECT, // Thread class is DirectOutputThread 350 DUPLICATING, // Thread class is DuplicatingThread 351 RECORD // Thread class is RecordThread 352 }; 353 354 ThreadBase (const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, uint32_t device, type_t type); 355 virtual ~ThreadBase(); 356 357 status_t dumpBase(int fd, const Vector<String16>& args); 358 status_t dumpEffectChains(int fd, const Vector<String16>& args); 359 360 void clearPowerManager(); 361 362 // base for record and playback 363 class TrackBase : public ExtendedAudioBufferProvider, public RefBase { 364 365 public: 366 enum track_state { 367 IDLE, 368 TERMINATED, 369 FLUSHED, 370 STOPPED, 371 // next 2 states are currently used for fast tracks only 372 STOPPING_1, // waiting for first underrun 373 STOPPING_2, // waiting for presentation complete 374 RESUMING, 375 ACTIVE, 376 PAUSING, 377 PAUSED 378 }; 379 380 TrackBase(ThreadBase *thread, 381 const sp<Client>& client, 382 uint32_t sampleRate, 383 audio_format_t format, 384 uint32_t channelMask, 385 int frameCount, 386 const sp<IMemory>& sharedBuffer, 387 int sessionId); 388 virtual ~TrackBase(); 389 390 virtual status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 391 int triggerSession = 0) = 0; 392 virtual void stop() = 0; 393 sp<IMemory> getCblk() const { return mCblkMemory; } 394 audio_track_cblk_t* cblk() const { return mCblk; } 395 int sessionId() const { return mSessionId; } 396 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 397 398 protected: 399 TrackBase(const TrackBase&); 400 TrackBase& operator = (const TrackBase&); 401 402 // AudioBufferProvider interface 403 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) = 0; 404 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 405 406 // ExtendedAudioBufferProvider interface is only needed for Track, 407 // but putting it in TrackBase avoids the complexity of virtual inheritance 408 virtual size_t framesReady() const { return SIZE_MAX; } 409 410 audio_format_t format() const { 411 return mFormat; 412 } 413 414 int channelCount() const { return mChannelCount; } 415 416 uint32_t channelMask() const { return mChannelMask; } 417 418 int sampleRate() const; // FIXME inline after cblk sr moved 419 420 void* getBuffer(uint32_t offset, uint32_t frames) const; 421 422 bool isStopped() const { 423 return (mState == STOPPED || mState == FLUSHED); 424 } 425 426 // for fast tracks only 427 bool isStopping() const { 428 return mState == STOPPING_1 || mState == STOPPING_2; 429 } 430 bool isStopping_1() const { 431 return mState == STOPPING_1; 432 } 433 bool isStopping_2() const { 434 return mState == STOPPING_2; 435 } 436 437 bool isTerminated() const { 438 return mState == TERMINATED; 439 } 440 441 bool step(); 442 void reset(); 443 444 const wp<ThreadBase> mThread; 445 /*const*/ sp<Client> mClient; // see explanation at ~TrackBase() why not const 446 sp<IMemory> mCblkMemory; 447 audio_track_cblk_t* mCblk; 448 void* mBuffer; 449 void* mBufferEnd; 450 uint32_t mFrameCount; 451 // we don't really need a lock for these 452 track_state mState; 453 const uint32_t mSampleRate; // initial sample rate only; for tracks which 454 // support dynamic rates, the current value is in control block 455 const audio_format_t mFormat; 456 bool mStepServerFailed; 457 const int mSessionId; 458 uint8_t mChannelCount; 459 uint32_t mChannelMask; 460 Vector < sp<SyncEvent> >mSyncEvents; 461 }; 462 463 class ConfigEvent { 464 public: 465 ConfigEvent() : mEvent(0), mParam(0) {} 466 467 int mEvent; 468 int mParam; 469 }; 470 471 class PMDeathRecipient : public IBinder::DeathRecipient { 472 public: 473 PMDeathRecipient(const wp<ThreadBase>& thread) : mThread(thread) {} 474 virtual ~PMDeathRecipient() {} 475 476 // IBinder::DeathRecipient 477 virtual void binderDied(const wp<IBinder>& who); 478 479 private: 480 PMDeathRecipient(const PMDeathRecipient&); 481 PMDeathRecipient& operator = (const PMDeathRecipient&); 482 483 wp<ThreadBase> mThread; 484 }; 485 486 virtual status_t initCheck() const = 0; 487 type_t type() const { return mType; } 488 uint32_t sampleRate() const { return mSampleRate; } 489 int channelCount() const { return mChannelCount; } 490 audio_format_t format() const { return mFormat; } 491 // Called by AudioFlinger::frameCount(audio_io_handle_t output) and effects, 492 // and returns the normal mix buffer's frame count. No API for HAL frame count. 493 size_t frameCount() const { return mNormalFrameCount; } 494 void wakeUp() { mWaitWorkCV.broadcast(); } 495 // Should be "virtual status_t requestExitAndWait()" and override same 496 // method in Thread, but Thread::requestExitAndWait() is not yet virtual. 497 void exit(); 498 virtual bool checkForNewParameters_l() = 0; 499 virtual status_t setParameters(const String8& keyValuePairs); 500 virtual String8 getParameters(const String8& keys) = 0; 501 virtual void audioConfigChanged_l(int event, int param = 0) = 0; 502 void sendConfigEvent(int event, int param = 0); 503 void sendConfigEvent_l(int event, int param = 0); 504 void processConfigEvents(); 505 audio_io_handle_t id() const { return mId;} 506 bool standby() const { return mStandby; } 507 uint32_t device() const { return mDevice; } 508 virtual audio_stream_t* stream() const = 0; 509 510 sp<EffectHandle> createEffect_l( 511 const sp<AudioFlinger::Client>& client, 512 const sp<IEffectClient>& effectClient, 513 int32_t priority, 514 int sessionId, 515 effect_descriptor_t *desc, 516 int *enabled, 517 status_t *status); 518 void disconnectEffect(const sp< EffectModule>& effect, 519 EffectHandle *handle, 520 bool unpinIfLast); 521 522 // return values for hasAudioSession (bit field) 523 enum effect_state { 524 EFFECT_SESSION = 0x1, // the audio session corresponds to at least one 525 // effect 526 TRACK_SESSION = 0x2 // the audio session corresponds to at least one 527 // track 528 }; 529 530 // get effect chain corresponding to session Id. 531 sp<EffectChain> getEffectChain(int sessionId); 532 // same as getEffectChain() but must be called with ThreadBase mutex locked 533 sp<EffectChain> getEffectChain_l(int sessionId); 534 // add an effect chain to the chain list (mEffectChains) 535 virtual status_t addEffectChain_l(const sp<EffectChain>& chain) = 0; 536 // remove an effect chain from the chain list (mEffectChains) 537 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain) = 0; 538 // lock all effect chains Mutexes. Must be called before releasing the 539 // ThreadBase mutex before processing the mixer and effects. This guarantees the 540 // integrity of the chains during the process. 541 // Also sets the parameter 'effectChains' to current value of mEffectChains. 542 void lockEffectChains_l(Vector< sp<EffectChain> >& effectChains); 543 // unlock effect chains after process 544 void unlockEffectChains(const Vector< sp<EffectChain> >& effectChains); 545 // set audio mode to all effect chains 546 void setMode(audio_mode_t mode); 547 // get effect module with corresponding ID on specified audio session 548 sp<AudioFlinger::EffectModule> getEffect(int sessionId, int effectId); 549 sp<AudioFlinger::EffectModule> getEffect_l(int sessionId, int effectId); 550 // add and effect module. Also creates the effect chain is none exists for 551 // the effects audio session 552 status_t addEffect_l(const sp< EffectModule>& effect); 553 // remove and effect module. Also removes the effect chain is this was the last 554 // effect 555 void removeEffect_l(const sp< EffectModule>& effect); 556 // detach all tracks connected to an auxiliary effect 557 virtual void detachAuxEffect_l(int effectId) {} 558 // returns either EFFECT_SESSION if effects on this audio session exist in one 559 // chain, or TRACK_SESSION if tracks on this audio session exist, or both 560 virtual uint32_t hasAudioSession(int sessionId) = 0; 561 // the value returned by default implementation is not important as the 562 // strategy is only meaningful for PlaybackThread which implements this method 563 virtual uint32_t getStrategyForSession_l(int sessionId) { return 0; } 564 565 // suspend or restore effect according to the type of effect passed. a NULL 566 // type pointer means suspend all effects in the session 567 void setEffectSuspended(const effect_uuid_t *type, 568 bool suspend, 569 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 570 // check if some effects must be suspended/restored when an effect is enabled 571 // or disabled 572 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 573 bool enabled, 574 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 575 void checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 576 bool enabled, 577 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 578 579 virtual status_t setSyncEvent(const sp<SyncEvent>& event) = 0; 580 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) = 0; 581 582 583 mutable Mutex mLock; 584 585 protected: 586 587 // entry describing an effect being suspended in mSuspendedSessions keyed vector 588 class SuspendedSessionDesc : public RefBase { 589 public: 590 SuspendedSessionDesc() : mRefCount(0) {} 591 592 int mRefCount; // number of active suspend requests 593 effect_uuid_t mType; // effect type UUID 594 }; 595 596 void acquireWakeLock(); 597 void acquireWakeLock_l(); 598 void releaseWakeLock(); 599 void releaseWakeLock_l(); 600 void setEffectSuspended_l(const effect_uuid_t *type, 601 bool suspend, 602 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 603 // updated mSuspendedSessions when an effect suspended or restored 604 void updateSuspendedSessions_l(const effect_uuid_t *type, 605 bool suspend, 606 int sessionId); 607 // check if some effects must be suspended when an effect chain is added 608 void checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain); 609 610 friend class AudioFlinger; // for mEffectChains 611 612 const type_t mType; 613 614 // Used by parameters, config events, addTrack_l, exit 615 Condition mWaitWorkCV; 616 617 const sp<AudioFlinger> mAudioFlinger; 618 uint32_t mSampleRate; 619 size_t mFrameCount; // output HAL, direct output, record 620 size_t mNormalFrameCount; // normal mixer and effects 621 uint32_t mChannelMask; 622 uint16_t mChannelCount; 623 size_t mFrameSize; 624 audio_format_t mFormat; 625 626 // Parameter sequence by client: binder thread calling setParameters(): 627 // 1. Lock mLock 628 // 2. Append to mNewParameters 629 // 3. mWaitWorkCV.signal 630 // 4. mParamCond.waitRelative with timeout 631 // 5. read mParamStatus 632 // 6. mWaitWorkCV.signal 633 // 7. Unlock 634 // 635 // Parameter sequence by server: threadLoop calling checkForNewParameters_l(): 636 // 1. Lock mLock 637 // 2. If there is an entry in mNewParameters proceed ... 638 // 2. Read first entry in mNewParameters 639 // 3. Process 640 // 4. Remove first entry from mNewParameters 641 // 5. Set mParamStatus 642 // 6. mParamCond.signal 643 // 7. mWaitWorkCV.wait with timeout (this is to avoid overwriting mParamStatus) 644 // 8. Unlock 645 Condition mParamCond; 646 Vector<String8> mNewParameters; 647 status_t mParamStatus; 648 649 Vector<ConfigEvent> mConfigEvents; 650 bool mStandby; 651 const audio_io_handle_t mId; 652 Vector< sp<EffectChain> > mEffectChains; 653 uint32_t mDevice; // output device for PlaybackThread 654 // input + output devices for RecordThread 655 static const int kNameLength = 16; // prctl(PR_SET_NAME) limit 656 char mName[kNameLength]; 657 sp<IPowerManager> mPowerManager; 658 sp<IBinder> mWakeLockToken; 659 const sp<PMDeathRecipient> mDeathRecipient; 660 // list of suspended effects per session and per type. The first vector is 661 // keyed by session ID, the second by type UUID timeLow field 662 KeyedVector< int, KeyedVector< int, sp<SuspendedSessionDesc> > > mSuspendedSessions; 663 }; 664 665 struct stream_type_t { 666 stream_type_t() 667 : volume(1.0f), 668 mute(false) 669 { 670 } 671 float volume; 672 bool mute; 673 }; 674 675 // --- PlaybackThread --- 676 class PlaybackThread : public ThreadBase { 677 public: 678 679 enum mixer_state { 680 MIXER_IDLE, // no active tracks 681 MIXER_TRACKS_ENABLED, // at least one active track, but no track has any data ready 682 MIXER_TRACKS_READY // at least one active track, and at least one track has data 683 // standby mode does not have an enum value 684 // suspend by audio policy manager is orthogonal to mixer state 685 }; 686 687 // playback track 688 class Track : public TrackBase, public VolumeProvider { 689 public: 690 Track( PlaybackThread *thread, 691 const sp<Client>& client, 692 audio_stream_type_t streamType, 693 uint32_t sampleRate, 694 audio_format_t format, 695 uint32_t channelMask, 696 int frameCount, 697 const sp<IMemory>& sharedBuffer, 698 int sessionId, 699 IAudioFlinger::track_flags_t flags); 700 virtual ~Track(); 701 702 static void appendDumpHeader(String8& result); 703 void dump(char* buffer, size_t size); 704 virtual status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 705 int triggerSession = 0); 706 virtual void stop(); 707 void pause(); 708 709 void flush(); 710 void destroy(); 711 void mute(bool); 712 int name() const { 713 return mName; 714 } 715 716 audio_stream_type_t streamType() const { 717 return mStreamType; 718 } 719 status_t attachAuxEffect(int EffectId); 720 void setAuxBuffer(int EffectId, int32_t *buffer); 721 int32_t *auxBuffer() const { return mAuxBuffer; } 722 void setMainBuffer(int16_t *buffer) { mMainBuffer = buffer; } 723 int16_t *mainBuffer() const { return mMainBuffer; } 724 int auxEffectId() const { return mAuxEffectId; } 725 726 // implement FastMixerState::VolumeProvider interface 727 virtual uint32_t getVolumeLR(); 728 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 729 730 protected: 731 // for numerous 732 friend class PlaybackThread; 733 friend class MixerThread; 734 friend class DirectOutputThread; 735 736 Track(const Track&); 737 Track& operator = (const Track&); 738 739 // AudioBufferProvider interface 740 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts = kInvalidPTS); 741 // releaseBuffer() not overridden 742 743 virtual size_t framesReady() const; 744 745 bool isMuted() const { return mMute; } 746 bool isPausing() const { 747 return mState == PAUSING; 748 } 749 bool isPaused() const { 750 return mState == PAUSED; 751 } 752 bool isResuming() const { 753 return mState == RESUMING; 754 } 755 bool isReady() const; 756 void setPaused() { mState = PAUSED; } 757 void reset(); 758 759 bool isOutputTrack() const { 760 return (mStreamType == AUDIO_STREAM_CNT); 761 } 762 763 sp<IMemory> sharedBuffer() const { return mSharedBuffer; } 764 765 bool presentationComplete(size_t framesWritten, size_t audioHalFrames); 766 767 public: 768 void triggerEvents(AudioSystem::sync_event_t type); 769 virtual bool isTimedTrack() const { return false; } 770 bool isFastTrack() const { return (mFlags & IAudioFlinger::TRACK_FAST) != 0; } 771 protected: 772 773 // we don't really need a lock for these 774 volatile bool mMute; 775 // FILLED state is used for suppressing volume ramp at begin of playing 776 enum {FS_INVALID, FS_FILLING, FS_FILLED, FS_ACTIVE}; 777 mutable uint8_t mFillingUpStatus; 778 int8_t mRetryCount; 779 const sp<IMemory> mSharedBuffer; 780 bool mResetDone; 781 const audio_stream_type_t mStreamType; 782 int mName; // track name on the normal mixer, 783 // allocated statically at track creation time, 784 // and is even allocated (though unused) for fast tracks 785 // FIXME don't allocate track name for fast tracks 786 int16_t *mMainBuffer; 787 int32_t *mAuxBuffer; 788 int mAuxEffectId; 789 bool mHasVolumeController; 790 size_t mPresentationCompleteFrames; // number of frames written to the audio HAL 791 // when this track will be fully rendered 792 private: 793 IAudioFlinger::track_flags_t mFlags; 794 795 // The following fields are only for fast tracks, and should be in a subclass 796 int mFastIndex; // index within FastMixerState::mFastTracks[]; 797 // either mFastIndex == -1 if not isFastTrack() 798 // or 0 < mFastIndex < FastMixerState::kMaxFast because 799 // index 0 is reserved for normal mixer's submix; 800 // index is allocated statically at track creation time 801 // but the slot is only used if track is active 802 FastTrackUnderruns mObservedUnderruns; // Most recently observed value of 803 // mFastMixerDumpState.mTracks[mFastIndex].mUnderruns 804 uint32_t mUnderrunCount; // Counter of total number of underruns, never reset 805 volatile float mCachedVolume; // combined master volume and stream type volume; 806 // 'volatile' means accessed without lock or 807 // barrier, but is read/written atomically 808 }; // end of Track 809 810 class TimedTrack : public Track { 811 public: 812 static sp<TimedTrack> create(PlaybackThread *thread, 813 const sp<Client>& client, 814 audio_stream_type_t streamType, 815 uint32_t sampleRate, 816 audio_format_t format, 817 uint32_t channelMask, 818 int frameCount, 819 const sp<IMemory>& sharedBuffer, 820 int sessionId); 821 ~TimedTrack(); 822 823 class TimedBuffer { 824 public: 825 TimedBuffer(); 826 TimedBuffer(const sp<IMemory>& buffer, int64_t pts); 827 const sp<IMemory>& buffer() const { return mBuffer; } 828 int64_t pts() const { return mPTS; } 829 uint32_t position() const { return mPosition; } 830 void setPosition(uint32_t pos) { mPosition = pos; } 831 private: 832 sp<IMemory> mBuffer; 833 int64_t mPTS; 834 uint32_t mPosition; 835 }; 836 837 // Mixer facing methods. 838 virtual bool isTimedTrack() const { return true; } 839 virtual size_t framesReady() const; 840 841 // AudioBufferProvider interface 842 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, 843 int64_t pts); 844 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 845 846 // Client/App facing methods. 847 status_t allocateTimedBuffer(size_t size, 848 sp<IMemory>* buffer); 849 status_t queueTimedBuffer(const sp<IMemory>& buffer, 850 int64_t pts); 851 status_t setMediaTimeTransform(const LinearTransform& xform, 852 TimedAudioTrack::TargetTimeline target); 853 854 private: 855 TimedTrack(PlaybackThread *thread, 856 const sp<Client>& client, 857 audio_stream_type_t streamType, 858 uint32_t sampleRate, 859 audio_format_t format, 860 uint32_t channelMask, 861 int frameCount, 862 const sp<IMemory>& sharedBuffer, 863 int sessionId); 864 865 void timedYieldSamples_l(AudioBufferProvider::Buffer* buffer); 866 void timedYieldSilence_l(uint32_t numFrames, 867 AudioBufferProvider::Buffer* buffer); 868 void trimTimedBufferQueue_l(); 869 void trimTimedBufferQueueHead_l(const char* logTag); 870 void updateFramesPendingAfterTrim_l(const TimedBuffer& buf, 871 const char* logTag); 872 873 uint64_t mLocalTimeFreq; 874 LinearTransform mLocalTimeToSampleTransform; 875 LinearTransform mMediaTimeToSampleTransform; 876 sp<MemoryDealer> mTimedMemoryDealer; 877 878 Vector<TimedBuffer> mTimedBufferQueue; 879 bool mQueueHeadInFlight; 880 bool mTrimQueueHeadOnRelease; 881 uint32_t mFramesPendingInQueue; 882 883 uint8_t* mTimedSilenceBuffer; 884 uint32_t mTimedSilenceBufferSize; 885 mutable Mutex mTimedBufferQueueLock; 886 bool mTimedAudioOutputOnTime; 887 CCHelper mCCHelper; 888 889 Mutex mMediaTimeTransformLock; 890 LinearTransform mMediaTimeTransform; 891 bool mMediaTimeTransformValid; 892 TimedAudioTrack::TargetTimeline mMediaTimeTransformTarget; 893 }; 894 895 896 // playback track 897 class OutputTrack : public Track { 898 public: 899 900 class Buffer: public AudioBufferProvider::Buffer { 901 public: 902 int16_t *mBuffer; 903 }; 904 905 OutputTrack(PlaybackThread *thread, 906 DuplicatingThread *sourceThread, 907 uint32_t sampleRate, 908 audio_format_t format, 909 uint32_t channelMask, 910 int frameCount); 911 virtual ~OutputTrack(); 912 913 virtual status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 914 int triggerSession = 0); 915 virtual void stop(); 916 bool write(int16_t* data, uint32_t frames); 917 bool bufferQueueEmpty() const { return mBufferQueue.size() == 0; } 918 bool isActive() const { return mActive; } 919 const wp<ThreadBase>& thread() const { return mThread; } 920 921 private: 922 923 enum { 924 NO_MORE_BUFFERS = 0x80000001, // same in AudioTrack.h, ok to be different value 925 }; 926 927 status_t obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs); 928 void clearBufferQueue(); 929 930 // Maximum number of pending buffers allocated by OutputTrack::write() 931 static const uint8_t kMaxOverFlowBuffers = 10; 932 933 Vector < Buffer* > mBufferQueue; 934 AudioBufferProvider::Buffer mOutBuffer; 935 bool mActive; 936 DuplicatingThread* const mSourceThread; // for waitTimeMs() in write() 937 }; // end of OutputTrack 938 939 PlaybackThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 940 audio_io_handle_t id, uint32_t device, type_t type); 941 virtual ~PlaybackThread(); 942 943 status_t dump(int fd, const Vector<String16>& args); 944 945 // Thread virtuals 946 virtual status_t readyToRun(); 947 virtual bool threadLoop(); 948 949 // RefBase 950 virtual void onFirstRef(); 951 952protected: 953 // Code snippets that were lifted up out of threadLoop() 954 virtual void threadLoop_mix() = 0; 955 virtual void threadLoop_sleepTime() = 0; 956 virtual void threadLoop_write(); 957 virtual void threadLoop_standby(); 958 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 959 960 // prepareTracks_l reads and writes mActiveTracks, and returns 961 // the pending set of tracks to remove via Vector 'tracksToRemove'. The caller 962 // is responsible for clearing or destroying this Vector later on, when it 963 // is safe to do so. That will drop the final ref count and destroy the tracks. 964 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove) = 0; 965 966public: 967 968 virtual status_t initCheck() const { return (mOutput == NULL) ? NO_INIT : NO_ERROR; } 969 970 // return estimated latency in milliseconds, as reported by HAL 971 uint32_t latency() const; 972 // same, but lock must already be held 973 uint32_t latency_l() const; 974 975 void setMasterVolume(float value); 976 void setMasterMute(bool muted); 977 978 void setStreamVolume(audio_stream_type_t stream, float value); 979 void setStreamMute(audio_stream_type_t stream, bool muted); 980 981 float streamVolume(audio_stream_type_t stream) const; 982 983 sp<Track> createTrack_l( 984 const sp<AudioFlinger::Client>& client, 985 audio_stream_type_t streamType, 986 uint32_t sampleRate, 987 audio_format_t format, 988 uint32_t channelMask, 989 int frameCount, 990 const sp<IMemory>& sharedBuffer, 991 int sessionId, 992 IAudioFlinger::track_flags_t flags, 993 pid_t tid, 994 status_t *status); 995 996 AudioStreamOut* getOutput() const; 997 AudioStreamOut* clearOutput(); 998 virtual audio_stream_t* stream() const; 999 1000 void suspend() { mSuspended++; } 1001 void restore() { if (mSuspended > 0) mSuspended--; } 1002 bool isSuspended() const { return (mSuspended > 0); } 1003 virtual String8 getParameters(const String8& keys); 1004 virtual void audioConfigChanged_l(int event, int param = 0); 1005 status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames); 1006 int16_t *mixBuffer() const { return mMixBuffer; }; 1007 1008 virtual void detachAuxEffect_l(int effectId); 1009 status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track> track, 1010 int EffectId); 1011 status_t attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track> track, 1012 int EffectId); 1013 1014 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1015 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1016 virtual uint32_t hasAudioSession(int sessionId); 1017 virtual uint32_t getStrategyForSession_l(int sessionId); 1018 1019 1020 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 1021 virtual bool isValidSyncEvent(const sp<SyncEvent>& event); 1022 void invalidateTracks(audio_stream_type_t streamType); 1023 1024 1025 protected: 1026 int16_t* mMixBuffer; 1027 uint32_t mSuspended; // suspend count, > 0 means suspended 1028 int mBytesWritten; 1029 private: 1030 // mMasterMute is in both PlaybackThread and in AudioFlinger. When a 1031 // PlaybackThread needs to find out if master-muted, it checks it's local 1032 // copy rather than the one in AudioFlinger. This optimization saves a lock. 1033 bool mMasterMute; 1034 void setMasterMute_l(bool muted) { mMasterMute = muted; } 1035 protected: 1036 SortedVector< wp<Track> > mActiveTracks; // FIXME check if this could be sp<> 1037 1038 // Allocate a track name for a given channel mask. 1039 // Returns name >= 0 if successful, -1 on failure. 1040 virtual int getTrackName_l(audio_channel_mask_t channelMask) = 0; 1041 virtual void deleteTrackName_l(int name) = 0; 1042 1043 // Time to sleep between cycles when: 1044 virtual uint32_t activeSleepTimeUs() const; // mixer state MIXER_TRACKS_ENABLED 1045 virtual uint32_t idleSleepTimeUs() const = 0; // mixer state MIXER_IDLE 1046 virtual uint32_t suspendSleepTimeUs() const = 0; // audio policy manager suspended us 1047 // No sleep when mixer state == MIXER_TRACKS_READY; relies on audio HAL stream->write() 1048 // No sleep in standby mode; waits on a condition 1049 1050 // Code snippets that are temporarily lifted up out of threadLoop() until the merge 1051 void checkSilentMode_l(); 1052 1053 // Non-trivial for DUPLICATING only 1054 virtual void saveOutputTracks() { } 1055 virtual void clearOutputTracks() { } 1056 1057 // Cache various calculated values, at threadLoop() entry and after a parameter change 1058 virtual void cacheParameters_l(); 1059 1060 virtual uint32_t correctLatency(uint32_t latency) const; 1061 1062 private: 1063 1064 friend class AudioFlinger; // for numerous 1065 1066 PlaybackThread(const Client&); 1067 PlaybackThread& operator = (const PlaybackThread&); 1068 1069 status_t addTrack_l(const sp<Track>& track); 1070 void destroyTrack_l(const sp<Track>& track); 1071 void removeTrack_l(const sp<Track>& track); 1072 1073 void readOutputParameters(); 1074 1075 virtual status_t dumpInternals(int fd, const Vector<String16>& args); 1076 status_t dumpTracks(int fd, const Vector<String16>& args); 1077 1078 SortedVector< sp<Track> > mTracks; 1079 // mStreamTypes[] uses 1 additional stream type internally for the OutputTrack used by DuplicatingThread 1080 stream_type_t mStreamTypes[AUDIO_STREAM_CNT + 1]; 1081 AudioStreamOut *mOutput; 1082 float mMasterVolume; 1083 nsecs_t mLastWriteTime; 1084 int mNumWrites; 1085 int mNumDelayedWrites; 1086 bool mInWrite; 1087 1088 // FIXME rename these former local variables of threadLoop to standard "m" names 1089 nsecs_t standbyTime; 1090 size_t mixBufferSize; 1091 1092 // cached copies of activeSleepTimeUs() and idleSleepTimeUs() made by cacheParameters_l() 1093 uint32_t activeSleepTime; 1094 uint32_t idleSleepTime; 1095 1096 uint32_t sleepTime; 1097 1098 // mixer status returned by prepareTracks_l() 1099 mixer_state mMixerStatus; // current cycle 1100 // previous cycle when in prepareTracks_l() 1101 mixer_state mMixerStatusIgnoringFastTracks; 1102 // FIXME or a separate ready state per track 1103 1104 // FIXME move these declarations into the specific sub-class that needs them 1105 // MIXER only 1106 uint32_t sleepTimeShift; 1107 1108 // same as AudioFlinger::mStandbyTimeInNsecs except for DIRECT which uses a shorter value 1109 nsecs_t standbyDelay; 1110 1111 // MIXER only 1112 nsecs_t maxPeriod; 1113 1114 // DUPLICATING only 1115 uint32_t writeFrames; 1116 1117 private: 1118 // The HAL output sink is treated as non-blocking, but current implementation is blocking 1119 sp<NBAIO_Sink> mOutputSink; 1120 // If a fast mixer is present, the blocking pipe sink, otherwise clear 1121 sp<NBAIO_Sink> mPipeSink; 1122 // The current sink for the normal mixer to write it's (sub)mix, mOutputSink or mPipeSink 1123 sp<NBAIO_Sink> mNormalSink; 1124 // For dumpsys 1125 sp<NBAIO_Sink> mTeeSink; 1126 sp<NBAIO_Source> mTeeSource; 1127 uint32_t mScreenState; // cached copy of gScreenState 1128 public: 1129 virtual bool hasFastMixer() const = 0; 1130 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const 1131 { FastTrackUnderruns dummy; return dummy; } 1132 1133 protected: 1134 // accessed by both binder threads and within threadLoop(), lock on mutex needed 1135 unsigned mFastTrackAvailMask; // bit i set if fast track [i] is available 1136 1137 }; 1138 1139 class MixerThread : public PlaybackThread { 1140 public: 1141 MixerThread (const sp<AudioFlinger>& audioFlinger, 1142 AudioStreamOut* output, 1143 audio_io_handle_t id, 1144 uint32_t device, 1145 type_t type = MIXER); 1146 virtual ~MixerThread(); 1147 1148 // Thread virtuals 1149 1150 virtual bool checkForNewParameters_l(); 1151 virtual status_t dumpInternals(int fd, const Vector<String16>& args); 1152 1153 protected: 1154 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1155 virtual int getTrackName_l(audio_channel_mask_t channelMask); 1156 virtual void deleteTrackName_l(int name); 1157 virtual uint32_t idleSleepTimeUs() const; 1158 virtual uint32_t suspendSleepTimeUs() const; 1159 virtual void cacheParameters_l(); 1160 1161 // threadLoop snippets 1162 virtual void threadLoop_write(); 1163 virtual void threadLoop_standby(); 1164 virtual void threadLoop_mix(); 1165 virtual void threadLoop_sleepTime(); 1166 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 1167 virtual uint32_t correctLatency(uint32_t latency) const; 1168 1169 AudioMixer* mAudioMixer; // normal mixer 1170 private: 1171 // one-time initialization, no locks required 1172 FastMixer* mFastMixer; // non-NULL if there is also a fast mixer 1173 sp<AudioWatchdog> mAudioWatchdog; // non-0 if there is an audio watchdog thread 1174 1175 // contents are not guaranteed to be consistent, no locks required 1176 FastMixerDumpState mFastMixerDumpState; 1177#ifdef STATE_QUEUE_DUMP 1178 StateQueueObserverDump mStateQueueObserverDump; 1179 StateQueueMutatorDump mStateQueueMutatorDump; 1180#endif 1181 AudioWatchdogDump mAudioWatchdogDump; 1182 1183 // accessible only within the threadLoop(), no locks required 1184 // mFastMixer->sq() // for mutating and pushing state 1185 int32_t mFastMixerFutex; // for cold idle 1186 1187 public: 1188 virtual bool hasFastMixer() const { return mFastMixer != NULL; } 1189 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const { 1190 ALOG_ASSERT(fastIndex < FastMixerState::kMaxFastTracks); 1191 return mFastMixerDumpState.mTracks[fastIndex].mUnderruns; 1192 } 1193 }; 1194 1195 class DirectOutputThread : public PlaybackThread { 1196 public: 1197 1198 DirectOutputThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1199 audio_io_handle_t id, uint32_t device); 1200 virtual ~DirectOutputThread(); 1201 1202 // Thread virtuals 1203 1204 virtual bool checkForNewParameters_l(); 1205 1206 protected: 1207 virtual int getTrackName_l(audio_channel_mask_t channelMask); 1208 virtual void deleteTrackName_l(int name); 1209 virtual uint32_t activeSleepTimeUs() const; 1210 virtual uint32_t idleSleepTimeUs() const; 1211 virtual uint32_t suspendSleepTimeUs() const; 1212 virtual void cacheParameters_l(); 1213 1214 // threadLoop snippets 1215 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1216 virtual void threadLoop_mix(); 1217 virtual void threadLoop_sleepTime(); 1218 1219 // volumes last sent to audio HAL with stream->set_volume() 1220 float mLeftVolFloat; 1221 float mRightVolFloat; 1222 1223private: 1224 // prepareTracks_l() tells threadLoop_mix() the name of the single active track 1225 sp<Track> mActiveTrack; 1226 public: 1227 virtual bool hasFastMixer() const { return false; } 1228 }; 1229 1230 class DuplicatingThread : public MixerThread { 1231 public: 1232 DuplicatingThread (const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread, 1233 audio_io_handle_t id); 1234 virtual ~DuplicatingThread(); 1235 1236 // Thread virtuals 1237 void addOutputTrack(MixerThread* thread); 1238 void removeOutputTrack(MixerThread* thread); 1239 uint32_t waitTimeMs() const { return mWaitTimeMs; } 1240 protected: 1241 virtual uint32_t activeSleepTimeUs() const; 1242 1243 private: 1244 bool outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks); 1245 protected: 1246 // threadLoop snippets 1247 virtual void threadLoop_mix(); 1248 virtual void threadLoop_sleepTime(); 1249 virtual void threadLoop_write(); 1250 virtual void threadLoop_standby(); 1251 virtual void cacheParameters_l(); 1252 1253 private: 1254 // called from threadLoop, addOutputTrack, removeOutputTrack 1255 virtual void updateWaitTime_l(); 1256 protected: 1257 virtual void saveOutputTracks(); 1258 virtual void clearOutputTracks(); 1259 private: 1260 1261 uint32_t mWaitTimeMs; 1262 SortedVector < sp<OutputTrack> > outputTracks; 1263 SortedVector < sp<OutputTrack> > mOutputTracks; 1264 public: 1265 virtual bool hasFastMixer() const { return false; } 1266 }; 1267 1268 PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const; 1269 MixerThread *checkMixerThread_l(audio_io_handle_t output) const; 1270 RecordThread *checkRecordThread_l(audio_io_handle_t input) const; 1271 // no range check, AudioFlinger::mLock held 1272 bool streamMute_l(audio_stream_type_t stream) const 1273 { return mStreamTypes[stream].mute; } 1274 // no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held 1275 float streamVolume_l(audio_stream_type_t stream) const 1276 { return mStreamTypes[stream].volume; } 1277 void audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2); 1278 1279 // allocate an audio_io_handle_t, session ID, or effect ID 1280 uint32_t nextUniqueId(); 1281 1282 status_t moveEffectChain_l(int sessionId, 1283 PlaybackThread *srcThread, 1284 PlaybackThread *dstThread, 1285 bool reRegister); 1286 // return thread associated with primary hardware device, or NULL 1287 PlaybackThread *primaryPlaybackThread_l() const; 1288 uint32_t primaryOutputDevice_l() const; 1289 1290 sp<PlaybackThread> getEffectThread_l(int sessionId, int EffectId); 1291 1292 // server side of the client's IAudioTrack 1293 class TrackHandle : public android::BnAudioTrack { 1294 public: 1295 TrackHandle(const sp<PlaybackThread::Track>& track); 1296 virtual ~TrackHandle(); 1297 virtual sp<IMemory> getCblk() const; 1298 virtual status_t start(); 1299 virtual void stop(); 1300 virtual void flush(); 1301 virtual void mute(bool); 1302 virtual void pause(); 1303 virtual status_t attachAuxEffect(int effectId); 1304 virtual status_t allocateTimedBuffer(size_t size, 1305 sp<IMemory>* buffer); 1306 virtual status_t queueTimedBuffer(const sp<IMemory>& buffer, 1307 int64_t pts); 1308 virtual status_t setMediaTimeTransform(const LinearTransform& xform, 1309 int target); 1310 virtual status_t onTransact( 1311 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 1312 private: 1313 const sp<PlaybackThread::Track> mTrack; 1314 }; 1315 1316 void removeClient_l(pid_t pid); 1317 void removeNotificationClient(pid_t pid); 1318 1319 1320 // record thread 1321 class RecordThread : public ThreadBase, public AudioBufferProvider 1322 { 1323 public: 1324 1325 // record track 1326 class RecordTrack : public TrackBase { 1327 public: 1328 RecordTrack(RecordThread *thread, 1329 const sp<Client>& client, 1330 uint32_t sampleRate, 1331 audio_format_t format, 1332 uint32_t channelMask, 1333 int frameCount, 1334 int sessionId); 1335 virtual ~RecordTrack(); 1336 1337 virtual status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 1338 int triggerSession = 0); 1339 virtual void stop(); 1340 1341 bool overflow() { bool tmp = mOverflow; mOverflow = false; return tmp; } 1342 bool setOverflow() { bool tmp = mOverflow; mOverflow = true; return tmp; } 1343 1344 void dump(char* buffer, size_t size); 1345 1346 private: 1347 friend class AudioFlinger; // for mState 1348 1349 RecordTrack(const RecordTrack&); 1350 RecordTrack& operator = (const RecordTrack&); 1351 1352 // AudioBufferProvider interface 1353 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts = kInvalidPTS); 1354 // releaseBuffer() not overridden 1355 1356 bool mOverflow; 1357 }; 1358 1359 1360 RecordThread(const sp<AudioFlinger>& audioFlinger, 1361 AudioStreamIn *input, 1362 uint32_t sampleRate, 1363 uint32_t channels, 1364 audio_io_handle_t id, 1365 uint32_t device); 1366 virtual ~RecordThread(); 1367 1368 // Thread 1369 virtual bool threadLoop(); 1370 virtual status_t readyToRun(); 1371 1372 // RefBase 1373 virtual void onFirstRef(); 1374 1375 virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; } 1376 sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l( 1377 const sp<AudioFlinger::Client>& client, 1378 uint32_t sampleRate, 1379 audio_format_t format, 1380 int channelMask, 1381 int frameCount, 1382 int sessionId, 1383 status_t *status); 1384 1385 status_t start(RecordTrack* recordTrack, 1386 AudioSystem::sync_event_t event, 1387 int triggerSession); 1388 void stop(RecordTrack* recordTrack); 1389 status_t dump(int fd, const Vector<String16>& args); 1390 AudioStreamIn* getInput() const; 1391 AudioStreamIn* clearInput(); 1392 virtual audio_stream_t* stream() const; 1393 1394 // AudioBufferProvider interface 1395 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts); 1396 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 1397 1398 virtual bool checkForNewParameters_l(); 1399 virtual String8 getParameters(const String8& keys); 1400 virtual void audioConfigChanged_l(int event, int param = 0); 1401 void readInputParameters(); 1402 virtual unsigned int getInputFramesLost(); 1403 1404 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1405 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1406 virtual uint32_t hasAudioSession(int sessionId); 1407 RecordTrack* track(); 1408 1409 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 1410 virtual bool isValidSyncEvent(const sp<SyncEvent>& event); 1411 1412 static void syncStartEventCallback(const wp<SyncEvent>& event); 1413 void handleSyncStartEvent(const sp<SyncEvent>& event); 1414 1415 private: 1416 void clearSyncStartEvent(); 1417 1418 RecordThread(); 1419 AudioStreamIn *mInput; 1420 RecordTrack* mTrack; 1421 sp<RecordTrack> mActiveTrack; 1422 Condition mStartStopCond; 1423 AudioResampler *mResampler; 1424 int32_t *mRsmpOutBuffer; 1425 int16_t *mRsmpInBuffer; 1426 size_t mRsmpInIndex; 1427 size_t mInputBytes; 1428 const int mReqChannelCount; 1429 const uint32_t mReqSampleRate; 1430 ssize_t mBytesRead; 1431 // sync event triggering actual audio capture. Frames read before this event will 1432 // be dropped and therefore not read by the application. 1433 sp<SyncEvent> mSyncStartEvent; 1434 // number of captured frames to drop after the start sync event has been received. 1435 // when < 0, maximum frames to drop before starting capture even if sync event is 1436 // not received 1437 ssize_t mFramestoDrop; 1438 }; 1439 1440 // server side of the client's IAudioRecord 1441 class RecordHandle : public android::BnAudioRecord { 1442 public: 1443 RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack); 1444 virtual ~RecordHandle(); 1445 virtual sp<IMemory> getCblk() const; 1446 virtual status_t start(int event, int triggerSession); 1447 virtual void stop(); 1448 virtual status_t onTransact( 1449 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 1450 private: 1451 const sp<RecordThread::RecordTrack> mRecordTrack; 1452 }; 1453 1454 //--- Audio Effect Management 1455 1456 // EffectModule and EffectChain classes both have their own mutex to protect 1457 // state changes or resource modifications. Always respect the following order 1458 // if multiple mutexes must be acquired to avoid cross deadlock: 1459 // AudioFlinger -> ThreadBase -> EffectChain -> EffectModule 1460 1461 // The EffectModule class is a wrapper object controlling the effect engine implementation 1462 // in the effect library. It prevents concurrent calls to process() and command() functions 1463 // from different client threads. It keeps a list of EffectHandle objects corresponding 1464 // to all client applications using this effect and notifies applications of effect state, 1465 // control or parameter changes. It manages the activation state machine to send appropriate 1466 // reset, enable, disable commands to effect engine and provide volume 1467 // ramping when effects are activated/deactivated. 1468 // When controlling an auxiliary effect, the EffectModule also provides an input buffer used by 1469 // the attached track(s) to accumulate their auxiliary channel. 1470 class EffectModule: public RefBase { 1471 public: 1472 EffectModule(ThreadBase *thread, 1473 const wp<AudioFlinger::EffectChain>& chain, 1474 effect_descriptor_t *desc, 1475 int id, 1476 int sessionId); 1477 virtual ~EffectModule(); 1478 1479 enum effect_state { 1480 IDLE, 1481 RESTART, 1482 STARTING, 1483 ACTIVE, 1484 STOPPING, 1485 STOPPED, 1486 DESTROYED 1487 }; 1488 1489 int id() const { return mId; } 1490 void process(); 1491 void updateState(); 1492 status_t command(uint32_t cmdCode, 1493 uint32_t cmdSize, 1494 void *pCmdData, 1495 uint32_t *replySize, 1496 void *pReplyData); 1497 1498 void reset_l(); 1499 status_t configure(); 1500 status_t init(); 1501 effect_state state() const { 1502 return mState; 1503 } 1504 uint32_t status() { 1505 return mStatus; 1506 } 1507 int sessionId() const { 1508 return mSessionId; 1509 } 1510 status_t setEnabled(bool enabled); 1511 status_t setEnabled_l(bool enabled); 1512 bool isEnabled() const; 1513 bool isProcessEnabled() const; 1514 1515 void setInBuffer(int16_t *buffer) { mConfig.inputCfg.buffer.s16 = buffer; } 1516 int16_t *inBuffer() { return mConfig.inputCfg.buffer.s16; } 1517 void setOutBuffer(int16_t *buffer) { mConfig.outputCfg.buffer.s16 = buffer; } 1518 int16_t *outBuffer() { return mConfig.outputCfg.buffer.s16; } 1519 void setChain(const wp<EffectChain>& chain) { mChain = chain; } 1520 void setThread(const wp<ThreadBase>& thread) { mThread = thread; } 1521 const wp<ThreadBase>& thread() { return mThread; } 1522 1523 status_t addHandle(EffectHandle *handle); 1524 size_t disconnect(EffectHandle *handle, bool unpinIfLast); 1525 size_t removeHandle(EffectHandle *handle); 1526 1527 effect_descriptor_t& desc() { return mDescriptor; } 1528 wp<EffectChain>& chain() { return mChain; } 1529 1530 status_t setDevice(uint32_t device); 1531 status_t setVolume(uint32_t *left, uint32_t *right, bool controller); 1532 status_t setMode(audio_mode_t mode); 1533 status_t start(); 1534 status_t stop(); 1535 void setSuspended(bool suspended); 1536 bool suspended() const; 1537 1538 EffectHandle* controlHandle_l(); 1539 1540 bool isPinned() const { return mPinned; } 1541 void unPin() { mPinned = false; } 1542 bool purgeHandles(); 1543 void lock() { mLock.lock(); } 1544 void unlock() { mLock.unlock(); } 1545 1546 status_t dump(int fd, const Vector<String16>& args); 1547 1548 protected: 1549 friend class AudioFlinger; // for mHandles 1550 bool mPinned; 1551 1552 // Maximum time allocated to effect engines to complete the turn off sequence 1553 static const uint32_t MAX_DISABLE_TIME_MS = 10000; 1554 1555 EffectModule(const EffectModule&); 1556 EffectModule& operator = (const EffectModule&); 1557 1558 status_t start_l(); 1559 status_t stop_l(); 1560 1561mutable Mutex mLock; // mutex for process, commands and handles list protection 1562 wp<ThreadBase> mThread; // parent thread 1563 wp<EffectChain> mChain; // parent effect chain 1564 const int mId; // this instance unique ID 1565 const int mSessionId; // audio session ID 1566 effect_descriptor_t mDescriptor;// effect descriptor received from effect engine 1567 effect_config_t mConfig; // input and output audio configuration 1568 effect_handle_t mEffectInterface; // Effect module C API 1569 status_t mStatus; // initialization status 1570 effect_state mState; // current activation state 1571 Vector<EffectHandle *> mHandles; // list of client handles 1572 // First handle in mHandles has highest priority and controls the effect module 1573 uint32_t mMaxDisableWaitCnt; // maximum grace period before forcing an effect off after 1574 // sending disable command. 1575 uint32_t mDisableWaitCnt; // current process() calls count during disable period. 1576 bool mSuspended; // effect is suspended: temporarily disabled by framework 1577 }; 1578 1579 // The EffectHandle class implements the IEffect interface. It provides resources 1580 // to receive parameter updates, keeps track of effect control 1581 // ownership and state and has a pointer to the EffectModule object it is controlling. 1582 // There is one EffectHandle object for each application controlling (or using) 1583 // an effect module. 1584 // The EffectHandle is obtained by calling AudioFlinger::createEffect(). 1585 class EffectHandle: public android::BnEffect { 1586 public: 1587 1588 EffectHandle(const sp<EffectModule>& effect, 1589 const sp<AudioFlinger::Client>& client, 1590 const sp<IEffectClient>& effectClient, 1591 int32_t priority); 1592 virtual ~EffectHandle(); 1593 1594 // IEffect 1595 virtual status_t enable(); 1596 virtual status_t disable(); 1597 virtual status_t command(uint32_t cmdCode, 1598 uint32_t cmdSize, 1599 void *pCmdData, 1600 uint32_t *replySize, 1601 void *pReplyData); 1602 virtual void disconnect(); 1603 private: 1604 void disconnect(bool unpinIfLast); 1605 public: 1606 virtual sp<IMemory> getCblk() const { return mCblkMemory; } 1607 virtual status_t onTransact(uint32_t code, const Parcel& data, 1608 Parcel* reply, uint32_t flags); 1609 1610 1611 // Give or take control of effect module 1612 // - hasControl: true if control is given, false if removed 1613 // - signal: true client app should be signaled of change, false otherwise 1614 // - enabled: state of the effect when control is passed 1615 void setControl(bool hasControl, bool signal, bool enabled); 1616 void commandExecuted(uint32_t cmdCode, 1617 uint32_t cmdSize, 1618 void *pCmdData, 1619 uint32_t replySize, 1620 void *pReplyData); 1621 void setEnabled(bool enabled); 1622 bool enabled() const { return mEnabled; } 1623 1624 // Getters 1625 int id() const { return mEffect->id(); } 1626 int priority() const { return mPriority; } 1627 bool hasControl() const { return mHasControl; } 1628 sp<EffectModule> effect() const { return mEffect; } 1629 // destroyed_l() must be called with the associated EffectModule mLock held 1630 bool destroyed_l() const { return mDestroyed; } 1631 1632 void dump(char* buffer, size_t size); 1633 1634 protected: 1635 friend class AudioFlinger; // for mEffect, mHasControl, mEnabled 1636 EffectHandle(const EffectHandle&); 1637 EffectHandle& operator =(const EffectHandle&); 1638 1639 sp<EffectModule> mEffect; // pointer to controlled EffectModule 1640 sp<IEffectClient> mEffectClient; // callback interface for client notifications 1641 /*const*/ sp<Client> mClient; // client for shared memory allocation, see disconnect() 1642 sp<IMemory> mCblkMemory; // shared memory for control block 1643 effect_param_cblk_t* mCblk; // control block for deferred parameter setting via shared memory 1644 uint8_t* mBuffer; // pointer to parameter area in shared memory 1645 int mPriority; // client application priority to control the effect 1646 bool mHasControl; // true if this handle is controlling the effect 1647 bool mEnabled; // cached enable state: needed when the effect is 1648 // restored after being suspended 1649 bool mDestroyed; // Set to true by destructor. Access with EffectModule 1650 // mLock held 1651 }; 1652 1653 // the EffectChain class represents a group of effects associated to one audio session. 1654 // There can be any number of EffectChain objects per output mixer thread (PlaybackThread). 1655 // The EffecChain with session ID 0 contains global effects applied to the output mix. 1656 // Effects in this chain can be insert or auxiliary. Effects in other chains (attached to tracks) 1657 // are insert only. The EffectChain maintains an ordered list of effect module, the order corresponding 1658 // in the effect process order. When attached to a track (session ID != 0), it also provide it's own 1659 // input buffer used by the track as accumulation buffer. 1660 class EffectChain: public RefBase { 1661 public: 1662 EffectChain(const wp<ThreadBase>& wThread, int sessionId); 1663 EffectChain(ThreadBase *thread, int sessionId); 1664 virtual ~EffectChain(); 1665 1666 // special key used for an entry in mSuspendedEffects keyed vector 1667 // corresponding to a suspend all request. 1668 static const int kKeyForSuspendAll = 0; 1669 1670 // minimum duration during which we force calling effect process when last track on 1671 // a session is stopped or removed to allow effect tail to be rendered 1672 static const int kProcessTailDurationMs = 1000; 1673 1674 void process_l(); 1675 1676 void lock() { 1677 mLock.lock(); 1678 } 1679 void unlock() { 1680 mLock.unlock(); 1681 } 1682 1683 status_t addEffect_l(const sp<EffectModule>& handle); 1684 size_t removeEffect_l(const sp<EffectModule>& handle); 1685 1686 int sessionId() const { return mSessionId; } 1687 void setSessionId(int sessionId) { mSessionId = sessionId; } 1688 1689 sp<EffectModule> getEffectFromDesc_l(effect_descriptor_t *descriptor); 1690 sp<EffectModule> getEffectFromId_l(int id); 1691 sp<EffectModule> getEffectFromType_l(const effect_uuid_t *type); 1692 bool setVolume_l(uint32_t *left, uint32_t *right); 1693 void setDevice_l(uint32_t device); 1694 void setMode_l(audio_mode_t mode); 1695 1696 void setInBuffer(int16_t *buffer, bool ownsBuffer = false) { 1697 mInBuffer = buffer; 1698 mOwnInBuffer = ownsBuffer; 1699 } 1700 int16_t *inBuffer() const { 1701 return mInBuffer; 1702 } 1703 void setOutBuffer(int16_t *buffer) { 1704 mOutBuffer = buffer; 1705 } 1706 int16_t *outBuffer() const { 1707 return mOutBuffer; 1708 } 1709 1710 void incTrackCnt() { android_atomic_inc(&mTrackCnt); } 1711 void decTrackCnt() { android_atomic_dec(&mTrackCnt); } 1712 int32_t trackCnt() const { return mTrackCnt;} 1713 1714 void incActiveTrackCnt() { android_atomic_inc(&mActiveTrackCnt); 1715 mTailBufferCount = mMaxTailBuffers; } 1716 void decActiveTrackCnt() { android_atomic_dec(&mActiveTrackCnt); } 1717 int32_t activeTrackCnt() const { return mActiveTrackCnt;} 1718 1719 uint32_t strategy() const { return mStrategy; } 1720 void setStrategy(uint32_t strategy) 1721 { mStrategy = strategy; } 1722 1723 // suspend effect of the given type 1724 void setEffectSuspended_l(const effect_uuid_t *type, 1725 bool suspend); 1726 // suspend all eligible effects 1727 void setEffectSuspendedAll_l(bool suspend); 1728 // check if effects should be suspend or restored when a given effect is enable or disabled 1729 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1730 bool enabled); 1731 1732 void clearInputBuffer(); 1733 1734 status_t dump(int fd, const Vector<String16>& args); 1735 1736 protected: 1737 friend class AudioFlinger; // for mThread, mEffects 1738 EffectChain(const EffectChain&); 1739 EffectChain& operator =(const EffectChain&); 1740 1741 class SuspendedEffectDesc : public RefBase { 1742 public: 1743 SuspendedEffectDesc() : mRefCount(0) {} 1744 1745 int mRefCount; 1746 effect_uuid_t mType; 1747 wp<EffectModule> mEffect; 1748 }; 1749 1750 // get a list of effect modules to suspend when an effect of the type 1751 // passed is enabled. 1752 void getSuspendEligibleEffects(Vector< sp<EffectModule> > &effects); 1753 1754 // get an effect module if it is currently enable 1755 sp<EffectModule> getEffectIfEnabled(const effect_uuid_t *type); 1756 // true if the effect whose descriptor is passed can be suspended 1757 // OEMs can modify the rules implemented in this method to exclude specific effect 1758 // types or implementations from the suspend/restore mechanism. 1759 bool isEffectEligibleForSuspend(const effect_descriptor_t& desc); 1760 1761 void clearInputBuffer_l(sp<ThreadBase> thread); 1762 1763 wp<ThreadBase> mThread; // parent mixer thread 1764 Mutex mLock; // mutex protecting effect list 1765 Vector< sp<EffectModule> > mEffects; // list of effect modules 1766 int mSessionId; // audio session ID 1767 int16_t *mInBuffer; // chain input buffer 1768 int16_t *mOutBuffer; // chain output buffer 1769 volatile int32_t mActiveTrackCnt; // number of active tracks connected 1770 volatile int32_t mTrackCnt; // number of tracks connected 1771 int32_t mTailBufferCount; // current effect tail buffer count 1772 int32_t mMaxTailBuffers; // maximum effect tail buffers 1773 bool mOwnInBuffer; // true if the chain owns its input buffer 1774 int mVolumeCtrlIdx; // index of insert effect having control over volume 1775 uint32_t mLeftVolume; // previous volume on left channel 1776 uint32_t mRightVolume; // previous volume on right channel 1777 uint32_t mNewLeftVolume; // new volume on left channel 1778 uint32_t mNewRightVolume; // new volume on right channel 1779 uint32_t mStrategy; // strategy for this effect chain 1780 // mSuspendedEffects lists all effects currently suspended in the chain. 1781 // Use effect type UUID timelow field as key. There is no real risk of identical 1782 // timeLow fields among effect type UUIDs. 1783 // Updated by updateSuspendedSessions_l() only. 1784 KeyedVector< int, sp<SuspendedEffectDesc> > mSuspendedEffects; 1785 }; 1786 1787 // AudioStreamOut and AudioStreamIn are immutable, so their fields are const. 1788 // For emphasis, we could also make all pointers to them be "const *", 1789 // but that would clutter the code unnecessarily. 1790 1791 struct AudioStreamOut { 1792 audio_hw_device_t* const hwDev; 1793 audio_stream_out_t* const stream; 1794 1795 AudioStreamOut(audio_hw_device_t *dev, audio_stream_out_t *out) : 1796 hwDev(dev), stream(out) {} 1797 }; 1798 1799 struct AudioStreamIn { 1800 audio_hw_device_t* const hwDev; 1801 audio_stream_in_t* const stream; 1802 1803 AudioStreamIn(audio_hw_device_t *dev, audio_stream_in_t *in) : 1804 hwDev(dev), stream(in) {} 1805 }; 1806 1807 // for mAudioSessionRefs only 1808 struct AudioSessionRef { 1809 AudioSessionRef(int sessionid, pid_t pid) : 1810 mSessionid(sessionid), mPid(pid), mCnt(1) {} 1811 const int mSessionid; 1812 const pid_t mPid; 1813 int mCnt; 1814 }; 1815 1816 enum master_volume_support { 1817 // MVS_NONE: 1818 // Audio HAL has no support for master volume, either setting or 1819 // getting. All master volume control must be implemented in SW by the 1820 // AudioFlinger mixing core. 1821 MVS_NONE, 1822 1823 // MVS_SETONLY: 1824 // Audio HAL has support for setting master volume, but not for getting 1825 // master volume (original HAL design did not include a getter). 1826 // AudioFlinger needs to keep track of the last set master volume in 1827 // addition to needing to set an initial, default, master volume at HAL 1828 // load time. 1829 MVS_SETONLY, 1830 1831 // MVS_FULL: 1832 // Audio HAL has support both for setting and getting master volume. 1833 // AudioFlinger should send all set and get master volume requests 1834 // directly to the HAL. 1835 MVS_FULL, 1836 }; 1837 1838 class AudioHwDevice { 1839 public: 1840 AudioHwDevice(const char *moduleName, audio_hw_device_t *hwDevice) : 1841 mModuleName(strdup(moduleName)), mHwDevice(hwDevice){} 1842 ~AudioHwDevice() { free((void *)mModuleName); } 1843 1844 const char *moduleName() const { return mModuleName; } 1845 audio_hw_device_t *hwDevice() const { return mHwDevice; } 1846 private: 1847 const char * const mModuleName; 1848 audio_hw_device_t * const mHwDevice; 1849 }; 1850 1851 mutable Mutex mLock; 1852 1853 DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client() 1854 1855 mutable Mutex mHardwareLock; 1856 // NOTE: If both mLock and mHardwareLock mutexes must be held, 1857 // always take mLock before mHardwareLock 1858 1859 // These two fields are immutable after onFirstRef(), so no lock needed to access 1860 audio_hw_device_t* mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL 1861 DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*> mAudioHwDevs; 1862 1863 // for dump, indicates which hardware operation is currently in progress (but not stream ops) 1864 enum hardware_call_state { 1865 AUDIO_HW_IDLE = 0, // no operation in progress 1866 AUDIO_HW_INIT, // init_check 1867 AUDIO_HW_OUTPUT_OPEN, // open_output_stream 1868 AUDIO_HW_OUTPUT_CLOSE, // unused 1869 AUDIO_HW_INPUT_OPEN, // unused 1870 AUDIO_HW_INPUT_CLOSE, // unused 1871 AUDIO_HW_STANDBY, // unused 1872 AUDIO_HW_SET_MASTER_VOLUME, // set_master_volume 1873 AUDIO_HW_GET_ROUTING, // unused 1874 AUDIO_HW_SET_ROUTING, // unused 1875 AUDIO_HW_GET_MODE, // unused 1876 AUDIO_HW_SET_MODE, // set_mode 1877 AUDIO_HW_GET_MIC_MUTE, // get_mic_mute 1878 AUDIO_HW_SET_MIC_MUTE, // set_mic_mute 1879 AUDIO_HW_SET_VOICE_VOLUME, // set_voice_volume 1880 AUDIO_HW_SET_PARAMETER, // set_parameters 1881 AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size 1882 AUDIO_HW_GET_MASTER_VOLUME, // get_master_volume 1883 AUDIO_HW_GET_PARAMETER, // get_parameters 1884 }; 1885 1886 mutable hardware_call_state mHardwareStatus; // for dump only 1887 1888 1889 DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads; 1890 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 1891 1892 // both are protected by mLock 1893 float mMasterVolume; 1894 float mMasterVolumeSW; 1895 master_volume_support mMasterVolumeSupportLvl; 1896 bool mMasterMute; 1897 1898 DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads; 1899 1900 DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients; 1901 volatile int32_t mNextUniqueId; // updated by android_atomic_inc 1902 audio_mode_t mMode; 1903 bool mBtNrecIsOff; 1904 1905 // protected by mLock 1906 Vector<AudioSessionRef*> mAudioSessionRefs; 1907 1908 float masterVolume_l() const; 1909 float masterVolumeSW_l() const { return mMasterVolumeSW; } 1910 bool masterMute_l() const { return mMasterMute; } 1911 audio_module_handle_t loadHwModule_l(const char *name); 1912 1913 Vector < sp<SyncEvent> > mPendingSyncEvents; // sync events awaiting for a session 1914 // to be created 1915 1916private: 1917 sp<Client> registerPid_l(pid_t pid); // always returns non-0 1918 1919}; 1920 1921 1922// ---------------------------------------------------------------------------- 1923 1924}; // namespace android 1925 1926#endif // ANDROID_AUDIO_FLINGER_H 1927