AudioFlinger.h revision 810280460da5000785662f6c5b0c7ff3ee0a4cb3
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 RESUMING, 369 ACTIVE, 370 PAUSING, 371 PAUSED 372 }; 373 374 TrackBase(ThreadBase *thread, 375 const sp<Client>& client, 376 uint32_t sampleRate, 377 audio_format_t format, 378 uint32_t channelMask, 379 int frameCount, 380 const sp<IMemory>& sharedBuffer, 381 int sessionId); 382 virtual ~TrackBase(); 383 384 virtual status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 385 int triggerSession = 0) = 0; 386 virtual void stop() = 0; 387 sp<IMemory> getCblk() const { return mCblkMemory; } 388 audio_track_cblk_t* cblk() const { return mCblk; } 389 int sessionId() const { return mSessionId; } 390 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 391 392 protected: 393 TrackBase(const TrackBase&); 394 TrackBase& operator = (const TrackBase&); 395 396 // AudioBufferProvider interface 397 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) = 0; 398 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 399 400 // ExtendedAudioBufferProvider interface is only needed for Track, 401 // but putting it in TrackBase avoids the complexity of virtual inheritance 402 virtual size_t framesReady() const { return SIZE_MAX; } 403 404 audio_format_t format() const { 405 return mFormat; 406 } 407 408 int channelCount() const { return mChannelCount; } 409 410 uint32_t channelMask() const { return mChannelMask; } 411 412 int sampleRate() const; // FIXME inline after cblk sr moved 413 414 void* getBuffer(uint32_t offset, uint32_t frames) const; 415 416 bool isStopped() const { 417 return mState == STOPPED; 418 } 419 420 bool isTerminated() const { 421 return mState == TERMINATED; 422 } 423 424 bool step(); 425 void reset(); 426 427 const wp<ThreadBase> mThread; 428 /*const*/ sp<Client> mClient; // see explanation at ~TrackBase() why not const 429 sp<IMemory> mCblkMemory; 430 audio_track_cblk_t* mCblk; 431 void* mBuffer; 432 void* mBufferEnd; 433 uint32_t mFrameCount; 434 // we don't really need a lock for these 435 track_state mState; 436 const uint32_t mSampleRate; // initial sample rate only; for tracks which 437 // support dynamic rates, the current value is in control block 438 const audio_format_t mFormat; 439 bool mStepServerFailed; 440 const int mSessionId; 441 uint8_t mChannelCount; 442 uint32_t mChannelMask; 443 Vector < sp<SyncEvent> >mSyncEvents; 444 }; 445 446 class ConfigEvent { 447 public: 448 ConfigEvent() : mEvent(0), mParam(0) {} 449 450 int mEvent; 451 int mParam; 452 }; 453 454 class PMDeathRecipient : public IBinder::DeathRecipient { 455 public: 456 PMDeathRecipient(const wp<ThreadBase>& thread) : mThread(thread) {} 457 virtual ~PMDeathRecipient() {} 458 459 // IBinder::DeathRecipient 460 virtual void binderDied(const wp<IBinder>& who); 461 462 private: 463 PMDeathRecipient(const PMDeathRecipient&); 464 PMDeathRecipient& operator = (const PMDeathRecipient&); 465 466 wp<ThreadBase> mThread; 467 }; 468 469 virtual status_t initCheck() const = 0; 470 type_t type() const { return mType; } 471 uint32_t sampleRate() const { return mSampleRate; } 472 int channelCount() const { return mChannelCount; } 473 audio_format_t format() const { return mFormat; } 474 // Called by AudioFlinger::frameCount(audio_io_handle_t output) and effects, 475 // and returns the normal mix buffer's frame count. No API for HAL frame count. 476 size_t frameCount() const { return mNormalFrameCount; } 477 void wakeUp() { mWaitWorkCV.broadcast(); } 478 // Should be "virtual status_t requestExitAndWait()" and override same 479 // method in Thread, but Thread::requestExitAndWait() is not yet virtual. 480 void exit(); 481 virtual bool checkForNewParameters_l() = 0; 482 virtual status_t setParameters(const String8& keyValuePairs); 483 virtual String8 getParameters(const String8& keys) = 0; 484 virtual void audioConfigChanged_l(int event, int param = 0) = 0; 485 void sendConfigEvent(int event, int param = 0); 486 void sendConfigEvent_l(int event, int param = 0); 487 void processConfigEvents(); 488 audio_io_handle_t id() const { return mId;} 489 bool standby() const { return mStandby; } 490 uint32_t device() const { return mDevice; } 491 virtual audio_stream_t* stream() const = 0; 492 493 sp<EffectHandle> createEffect_l( 494 const sp<AudioFlinger::Client>& client, 495 const sp<IEffectClient>& effectClient, 496 int32_t priority, 497 int sessionId, 498 effect_descriptor_t *desc, 499 int *enabled, 500 status_t *status); 501 void disconnectEffect(const sp< EffectModule>& effect, 502 const wp<EffectHandle>& handle, 503 bool unpinIfLast); 504 505 // return values for hasAudioSession (bit field) 506 enum effect_state { 507 EFFECT_SESSION = 0x1, // the audio session corresponds to at least one 508 // effect 509 TRACK_SESSION = 0x2 // the audio session corresponds to at least one 510 // track 511 }; 512 513 // get effect chain corresponding to session Id. 514 sp<EffectChain> getEffectChain(int sessionId); 515 // same as getEffectChain() but must be called with ThreadBase mutex locked 516 sp<EffectChain> getEffectChain_l(int sessionId); 517 // add an effect chain to the chain list (mEffectChains) 518 virtual status_t addEffectChain_l(const sp<EffectChain>& chain) = 0; 519 // remove an effect chain from the chain list (mEffectChains) 520 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain) = 0; 521 // lock all effect chains Mutexes. Must be called before releasing the 522 // ThreadBase mutex before processing the mixer and effects. This guarantees the 523 // integrity of the chains during the process. 524 // Also sets the parameter 'effectChains' to current value of mEffectChains. 525 void lockEffectChains_l(Vector< sp<EffectChain> >& effectChains); 526 // unlock effect chains after process 527 void unlockEffectChains(const Vector< sp<EffectChain> >& effectChains); 528 // set audio mode to all effect chains 529 void setMode(audio_mode_t mode); 530 // get effect module with corresponding ID on specified audio session 531 sp<AudioFlinger::EffectModule> getEffect_l(int sessionId, int effectId); 532 // add and effect module. Also creates the effect chain is none exists for 533 // the effects audio session 534 status_t addEffect_l(const sp< EffectModule>& effect); 535 // remove and effect module. Also removes the effect chain is this was the last 536 // effect 537 void removeEffect_l(const sp< EffectModule>& effect); 538 // detach all tracks connected to an auxiliary effect 539 virtual void detachAuxEffect_l(int effectId) {} 540 // returns either EFFECT_SESSION if effects on this audio session exist in one 541 // chain, or TRACK_SESSION if tracks on this audio session exist, or both 542 virtual uint32_t hasAudioSession(int sessionId) = 0; 543 // the value returned by default implementation is not important as the 544 // strategy is only meaningful for PlaybackThread which implements this method 545 virtual uint32_t getStrategyForSession_l(int sessionId) { return 0; } 546 547 // suspend or restore effect according to the type of effect passed. a NULL 548 // type pointer means suspend all effects in the session 549 void setEffectSuspended(const effect_uuid_t *type, 550 bool suspend, 551 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 552 // check if some effects must be suspended/restored when an effect is enabled 553 // or disabled 554 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 555 bool enabled, 556 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 557 void checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 558 bool enabled, 559 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 560 561 virtual status_t setSyncEvent(const sp<SyncEvent>& event) = 0; 562 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) = 0; 563 564 565 mutable Mutex mLock; 566 567 protected: 568 569 // entry describing an effect being suspended in mSuspendedSessions keyed vector 570 class SuspendedSessionDesc : public RefBase { 571 public: 572 SuspendedSessionDesc() : mRefCount(0) {} 573 574 int mRefCount; // number of active suspend requests 575 effect_uuid_t mType; // effect type UUID 576 }; 577 578 void acquireWakeLock(); 579 void acquireWakeLock_l(); 580 void releaseWakeLock(); 581 void releaseWakeLock_l(); 582 void setEffectSuspended_l(const effect_uuid_t *type, 583 bool suspend, 584 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 585 // updated mSuspendedSessions when an effect suspended or restored 586 void updateSuspendedSessions_l(const effect_uuid_t *type, 587 bool suspend, 588 int sessionId); 589 // check if some effects must be suspended when an effect chain is added 590 void checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain); 591 592 friend class AudioFlinger; // for mEffectChains 593 594 const type_t mType; 595 596 // Used by parameters, config events, addTrack_l, exit 597 Condition mWaitWorkCV; 598 599 const sp<AudioFlinger> mAudioFlinger; 600 uint32_t mSampleRate; 601 size_t mFrameCount; // output HAL, direct output, record 602 size_t mNormalFrameCount; // normal mixer and effects 603 uint32_t mChannelMask; 604 uint16_t mChannelCount; 605 size_t mFrameSize; 606 audio_format_t mFormat; 607 608 // Parameter sequence by client: binder thread calling setParameters(): 609 // 1. Lock mLock 610 // 2. Append to mNewParameters 611 // 3. mWaitWorkCV.signal 612 // 4. mParamCond.waitRelative with timeout 613 // 5. read mParamStatus 614 // 6. mWaitWorkCV.signal 615 // 7. Unlock 616 // 617 // Parameter sequence by server: threadLoop calling checkForNewParameters_l(): 618 // 1. Lock mLock 619 // 2. If there is an entry in mNewParameters proceed ... 620 // 2. Read first entry in mNewParameters 621 // 3. Process 622 // 4. Remove first entry from mNewParameters 623 // 5. Set mParamStatus 624 // 6. mParamCond.signal 625 // 7. mWaitWorkCV.wait with timeout (this is to avoid overwriting mParamStatus) 626 // 8. Unlock 627 Condition mParamCond; 628 Vector<String8> mNewParameters; 629 status_t mParamStatus; 630 631 Vector<ConfigEvent> mConfigEvents; 632 bool mStandby; 633 const audio_io_handle_t mId; 634 Vector< sp<EffectChain> > mEffectChains; 635 uint32_t mDevice; // output device for PlaybackThread 636 // input + output devices for RecordThread 637 static const int kNameLength = 16; // prctl(PR_SET_NAME) limit 638 char mName[kNameLength]; 639 sp<IPowerManager> mPowerManager; 640 sp<IBinder> mWakeLockToken; 641 const sp<PMDeathRecipient> mDeathRecipient; 642 // list of suspended effects per session and per type. The first vector is 643 // keyed by session ID, the second by type UUID timeLow field 644 KeyedVector< int, KeyedVector< int, sp<SuspendedSessionDesc> > > mSuspendedSessions; 645 }; 646 647 struct stream_type_t { 648 stream_type_t() 649 : volume(1.0f), 650 mute(false) 651 { 652 } 653 float volume; 654 bool mute; 655 }; 656 657 // --- PlaybackThread --- 658 class PlaybackThread : public ThreadBase { 659 public: 660 661 enum mixer_state { 662 MIXER_IDLE, // no active tracks 663 MIXER_TRACKS_ENABLED, // at least one active track, but no track has any data ready 664 MIXER_TRACKS_READY // at least one active track, and at least one track has data 665 // standby mode does not have an enum value 666 // suspend by audio policy manager is orthogonal to mixer state 667 }; 668 669 // playback track 670 class Track : public TrackBase, public VolumeProvider { 671 public: 672 Track( PlaybackThread *thread, 673 const sp<Client>& client, 674 audio_stream_type_t streamType, 675 uint32_t sampleRate, 676 audio_format_t format, 677 uint32_t channelMask, 678 int frameCount, 679 const sp<IMemory>& sharedBuffer, 680 int sessionId, 681 IAudioFlinger::track_flags_t flags); 682 virtual ~Track(); 683 684 static void appendDumpHeader(String8& result); 685 void dump(char* buffer, size_t size); 686 virtual status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 687 int triggerSession = 0); 688 virtual void stop(); 689 void pause(); 690 691 void flush(); 692 void destroy(); 693 void mute(bool); 694 int name() const { 695 return mName; 696 } 697 698 audio_stream_type_t streamType() const { 699 return mStreamType; 700 } 701 status_t attachAuxEffect(int EffectId); 702 void setAuxBuffer(int EffectId, int32_t *buffer); 703 int32_t *auxBuffer() const { return mAuxBuffer; } 704 void setMainBuffer(int16_t *buffer) { mMainBuffer = buffer; } 705 int16_t *mainBuffer() const { return mMainBuffer; } 706 int auxEffectId() const { return mAuxEffectId; } 707 708 // implement FastMixerState::VolumeProvider interface 709 virtual uint32_t getVolumeLR(); 710 711 protected: 712 // for numerous 713 friend class PlaybackThread; 714 friend class MixerThread; 715 friend class DirectOutputThread; 716 717 Track(const Track&); 718 Track& operator = (const Track&); 719 720 // AudioBufferProvider interface 721 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts = kInvalidPTS); 722 // releaseBuffer() not overridden 723 724 virtual size_t framesReady() const; 725 726 bool isMuted() const { return mMute; } 727 bool isPausing() const { 728 return mState == PAUSING; 729 } 730 bool isPaused() const { 731 return mState == PAUSED; 732 } 733 bool isResuming() const { 734 return mState == RESUMING; 735 } 736 bool isReady() const; 737 void setPaused() { mState = PAUSED; } 738 void reset(); 739 740 bool isOutputTrack() const { 741 return (mStreamType == AUDIO_STREAM_CNT); 742 } 743 744 sp<IMemory> sharedBuffer() const { return mSharedBuffer; } 745 746 bool presentationComplete(size_t framesWritten, size_t audioHalFrames); 747 void triggerEvents(AudioSystem::sync_event_t type); 748 749 public: 750 virtual bool isTimedTrack() const { return false; } 751 bool isFastTrack() const { return (mFlags & IAudioFlinger::TRACK_FAST) != 0; } 752 protected: 753 754 // we don't really need a lock for these 755 volatile bool mMute; 756 // FILLED state is used for suppressing volume ramp at begin of playing 757 enum {FS_INVALID, FS_FILLING, FS_FILLED, FS_ACTIVE}; 758 mutable uint8_t mFillingUpStatus; 759 int8_t mRetryCount; 760 const sp<IMemory> mSharedBuffer; 761 bool mResetDone; 762 const audio_stream_type_t mStreamType; 763 int mName; // track name on the normal mixer, 764 // allocated statically at track creation time, 765 // and is even allocated (though unused) for fast tracks 766 int16_t *mMainBuffer; 767 int32_t *mAuxBuffer; 768 int mAuxEffectId; 769 bool mHasVolumeController; 770 size_t mPresentationCompleteFrames; // number of frames written to the audio HAL 771 // when this track will be fully rendered 772 private: 773 IAudioFlinger::track_flags_t mFlags; 774 775 // The following fields are only for fast tracks, and should be in a subclass 776 int mFastIndex; // index within FastMixerState::mFastTracks[]; 777 // either mFastIndex == -1 778 // or 0 < mFastIndex < FastMixerState::kMaxFast because 779 // index 0 is reserved for normal mixer's submix; 780 // index is allocated statically at track creation time 781 // but the slot is only used if track is active 782 uint32_t mObservedUnderruns; // Most recently observed value of 783 // mFastMixerDumpState.mTracks[mFastIndex].mUnderruns 784 uint32_t mUnderrunCount; // Counter of total number of underruns, never reset 785 volatile float mCachedVolume; // combined master volume and stream type volume; 786 // 'volatile' means accessed without lock or 787 // barrier, but is read/written atomically 788 }; // end of Track 789 790 class TimedTrack : public Track { 791 public: 792 static sp<TimedTrack> create(PlaybackThread *thread, 793 const sp<Client>& client, 794 audio_stream_type_t streamType, 795 uint32_t sampleRate, 796 audio_format_t format, 797 uint32_t channelMask, 798 int frameCount, 799 const sp<IMemory>& sharedBuffer, 800 int sessionId); 801 ~TimedTrack(); 802 803 class TimedBuffer { 804 public: 805 TimedBuffer(); 806 TimedBuffer(const sp<IMemory>& buffer, int64_t pts); 807 const sp<IMemory>& buffer() const { return mBuffer; } 808 int64_t pts() const { return mPTS; } 809 uint32_t position() const { return mPosition; } 810 void setPosition(uint32_t pos) { mPosition = pos; } 811 private: 812 sp<IMemory> mBuffer; 813 int64_t mPTS; 814 uint32_t mPosition; 815 }; 816 817 // Mixer facing methods. 818 virtual bool isTimedTrack() const { return true; } 819 virtual size_t framesReady() const; 820 821 // AudioBufferProvider interface 822 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, 823 int64_t pts); 824 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 825 826 // Client/App facing methods. 827 status_t allocateTimedBuffer(size_t size, 828 sp<IMemory>* buffer); 829 status_t queueTimedBuffer(const sp<IMemory>& buffer, 830 int64_t pts); 831 status_t setMediaTimeTransform(const LinearTransform& xform, 832 TimedAudioTrack::TargetTimeline target); 833 834 private: 835 TimedTrack(PlaybackThread *thread, 836 const sp<Client>& client, 837 audio_stream_type_t streamType, 838 uint32_t sampleRate, 839 audio_format_t format, 840 uint32_t channelMask, 841 int frameCount, 842 const sp<IMemory>& sharedBuffer, 843 int sessionId); 844 845 void timedYieldSamples_l(AudioBufferProvider::Buffer* buffer); 846 void timedYieldSilence_l(uint32_t numFrames, 847 AudioBufferProvider::Buffer* buffer); 848 void trimTimedBufferQueue_l(); 849 void trimTimedBufferQueueHead_l(const char* logTag); 850 void updateFramesPendingAfterTrim_l(const TimedBuffer& buf, 851 const char* logTag); 852 853 uint64_t mLocalTimeFreq; 854 LinearTransform mLocalTimeToSampleTransform; 855 LinearTransform mMediaTimeToSampleTransform; 856 sp<MemoryDealer> mTimedMemoryDealer; 857 858 Vector<TimedBuffer> mTimedBufferQueue; 859 bool mQueueHeadInFlight; 860 bool mTrimQueueHeadOnRelease; 861 uint32_t mFramesPendingInQueue; 862 863 uint8_t* mTimedSilenceBuffer; 864 uint32_t mTimedSilenceBufferSize; 865 mutable Mutex mTimedBufferQueueLock; 866 bool mTimedAudioOutputOnTime; 867 CCHelper mCCHelper; 868 869 Mutex mMediaTimeTransformLock; 870 LinearTransform mMediaTimeTransform; 871 bool mMediaTimeTransformValid; 872 TimedAudioTrack::TargetTimeline mMediaTimeTransformTarget; 873 }; 874 875 876 // playback track 877 class OutputTrack : public Track { 878 public: 879 880 class Buffer: public AudioBufferProvider::Buffer { 881 public: 882 int16_t *mBuffer; 883 }; 884 885 OutputTrack(PlaybackThread *thread, 886 DuplicatingThread *sourceThread, 887 uint32_t sampleRate, 888 audio_format_t format, 889 uint32_t channelMask, 890 int frameCount); 891 virtual ~OutputTrack(); 892 893 virtual status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 894 int triggerSession = 0); 895 virtual void stop(); 896 bool write(int16_t* data, uint32_t frames); 897 bool bufferQueueEmpty() const { return mBufferQueue.size() == 0; } 898 bool isActive() const { return mActive; } 899 const wp<ThreadBase>& thread() const { return mThread; } 900 901 private: 902 903 enum { 904 NO_MORE_BUFFERS = 0x80000001, // same in AudioTrack.h, ok to be different value 905 }; 906 907 status_t obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs); 908 void clearBufferQueue(); 909 910 // Maximum number of pending buffers allocated by OutputTrack::write() 911 static const uint8_t kMaxOverFlowBuffers = 10; 912 913 Vector < Buffer* > mBufferQueue; 914 AudioBufferProvider::Buffer mOutBuffer; 915 bool mActive; 916 DuplicatingThread* const mSourceThread; // for waitTimeMs() in write() 917 }; // end of OutputTrack 918 919 PlaybackThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 920 audio_io_handle_t id, uint32_t device, type_t type); 921 virtual ~PlaybackThread(); 922 923 status_t dump(int fd, const Vector<String16>& args); 924 925 // Thread virtuals 926 virtual status_t readyToRun(); 927 virtual bool threadLoop(); 928 929 // RefBase 930 virtual void onFirstRef(); 931 932protected: 933 // Code snippets that were lifted up out of threadLoop() 934 virtual void threadLoop_mix() = 0; 935 virtual void threadLoop_sleepTime() = 0; 936 virtual void threadLoop_write(); 937 virtual void threadLoop_standby(); 938 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) { } 939 940 // prepareTracks_l reads and writes mActiveTracks, and returns 941 // the pending set of tracks to remove via Vector 'tracksToRemove'. The caller 942 // is responsible for clearing or destroying this Vector later on, when it 943 // is safe to do so. That will drop the final ref count and destroy the tracks. 944 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove) = 0; 945 946public: 947 948 virtual status_t initCheck() const { return (mOutput == NULL) ? NO_INIT : NO_ERROR; } 949 950 // return estimated latency in milliseconds, as reported by HAL 951 uint32_t latency() const; 952 953 void setMasterVolume(float value); 954 void setMasterMute(bool muted); 955 956 void setStreamVolume(audio_stream_type_t stream, float value); 957 void setStreamMute(audio_stream_type_t stream, bool muted); 958 959 float streamVolume(audio_stream_type_t stream) const; 960 961 sp<Track> createTrack_l( 962 const sp<AudioFlinger::Client>& client, 963 audio_stream_type_t streamType, 964 uint32_t sampleRate, 965 audio_format_t format, 966 uint32_t channelMask, 967 int frameCount, 968 const sp<IMemory>& sharedBuffer, 969 int sessionId, 970 IAudioFlinger::track_flags_t flags, 971 pid_t tid, 972 status_t *status); 973 974 AudioStreamOut* getOutput() const; 975 AudioStreamOut* clearOutput(); 976 virtual audio_stream_t* stream() const; 977 978 void suspend() { mSuspended++; } 979 void restore() { if (mSuspended > 0) mSuspended--; } 980 bool isSuspended() const { return (mSuspended > 0); } 981 virtual String8 getParameters(const String8& keys); 982 virtual void audioConfigChanged_l(int event, int param = 0); 983 status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames); 984 int16_t *mixBuffer() const { return mMixBuffer; }; 985 986 virtual void detachAuxEffect_l(int effectId); 987 status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track> track, 988 int EffectId); 989 status_t attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track> track, 990 int EffectId); 991 992 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 993 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 994 virtual uint32_t hasAudioSession(int sessionId); 995 virtual uint32_t getStrategyForSession_l(int sessionId); 996 997 998 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 999 virtual bool isValidSyncEvent(const sp<SyncEvent>& event); 1000 1001 protected: 1002 int16_t* mMixBuffer; 1003 uint32_t mSuspended; // suspend count, > 0 means suspended 1004 int mBytesWritten; 1005 private: 1006 // mMasterMute is in both PlaybackThread and in AudioFlinger. When a 1007 // PlaybackThread needs to find out if master-muted, it checks it's local 1008 // copy rather than the one in AudioFlinger. This optimization saves a lock. 1009 bool mMasterMute; 1010 void setMasterMute_l(bool muted) { mMasterMute = muted; } 1011 protected: 1012 SortedVector< wp<Track> > mActiveTracks; // FIXME check if this could be sp<> 1013 1014 // Allocate a track name for a given channel mask. 1015 // Returns name >= 0 if successful, -1 on failure. 1016 virtual int getTrackName_l(audio_channel_mask_t channelMask) = 0; 1017 virtual void deleteTrackName_l(int name) = 0; 1018 1019 // Time to sleep between cycles when: 1020 virtual uint32_t activeSleepTimeUs() const; // mixer state MIXER_TRACKS_ENABLED 1021 virtual uint32_t idleSleepTimeUs() const = 0; // mixer state MIXER_IDLE 1022 virtual uint32_t suspendSleepTimeUs() const = 0; // audio policy manager suspended us 1023 // No sleep when mixer state == MIXER_TRACKS_READY; relies on audio HAL stream->write() 1024 // No sleep in standby mode; waits on a condition 1025 1026 // Code snippets that are temporarily lifted up out of threadLoop() until the merge 1027 void checkSilentMode_l(); 1028 1029 // Non-trivial for DUPLICATING only 1030 virtual void saveOutputTracks() { } 1031 virtual void clearOutputTracks() { } 1032 1033 // Cache various calculated values, at threadLoop() entry and after a parameter change 1034 virtual void cacheParameters_l(); 1035 1036 private: 1037 1038 friend class AudioFlinger; // for numerous 1039 1040 PlaybackThread(const Client&); 1041 PlaybackThread& operator = (const PlaybackThread&); 1042 1043 status_t addTrack_l(const sp<Track>& track); 1044 void destroyTrack_l(const sp<Track>& track); 1045 void removeTrack_l(const sp<Track>& track); 1046 1047 void readOutputParameters(); 1048 1049 virtual status_t dumpInternals(int fd, const Vector<String16>& args); 1050 status_t dumpTracks(int fd, const Vector<String16>& args); 1051 1052 SortedVector< sp<Track> > mTracks; 1053 // mStreamTypes[] uses 1 additional stream type internally for the OutputTrack used by DuplicatingThread 1054 stream_type_t mStreamTypes[AUDIO_STREAM_CNT + 1]; 1055 AudioStreamOut *mOutput; 1056 float mMasterVolume; 1057 nsecs_t mLastWriteTime; 1058 int mNumWrites; 1059 int mNumDelayedWrites; 1060 bool mInWrite; 1061 1062 // FIXME rename these former local variables of threadLoop to standard "m" names 1063 nsecs_t standbyTime; 1064 size_t mixBufferSize; 1065 1066 // cached copies of activeSleepTimeUs() and idleSleepTimeUs() made by cacheParameters_l() 1067 uint32_t activeSleepTime; 1068 uint32_t idleSleepTime; 1069 1070 uint32_t sleepTime; 1071 1072 // mixer status returned by prepareTracks_l() 1073 mixer_state mMixerStatus; // current cycle 1074 // previous cycle when in prepareTracks_l() 1075 mixer_state mMixerStatusIgnoringFastTracks; 1076 // FIXME or a separate ready state per track 1077 1078 // FIXME move these declarations into the specific sub-class that needs them 1079 // MIXER only 1080 bool longStandbyExit; 1081 uint32_t sleepTimeShift; 1082 1083 // same as AudioFlinger::mStandbyTimeInNsecs except for DIRECT which uses a shorter value 1084 nsecs_t standbyDelay; 1085 1086 // MIXER only 1087 nsecs_t maxPeriod; 1088 1089 // DUPLICATING only 1090 uint32_t writeFrames; 1091 1092 private: 1093 // The HAL output sink is treated as non-blocking, but current implementation is blocking 1094 sp<NBAIO_Sink> mOutputSink; 1095 // If a fast mixer is present, the blocking pipe sink, otherwise clear 1096 sp<NBAIO_Sink> mPipeSink; 1097 // The current sink for the normal mixer to write it's (sub)mix, mOutputSink or mPipeSink 1098 sp<NBAIO_Sink> mNormalSink; 1099 public: 1100 virtual bool hasFastMixer() const = 0; 1101 virtual uint32_t getFastTrackUnderruns(size_t fastIndex) const { return 0; } 1102 1103 protected: 1104 // accessed by both binder threads and within threadLoop(), lock on mutex needed 1105 unsigned mFastTrackAvailMask; // bit i set if fast track [i] is available 1106 1107 }; 1108 1109 class MixerThread : public PlaybackThread { 1110 public: 1111 MixerThread (const sp<AudioFlinger>& audioFlinger, 1112 AudioStreamOut* output, 1113 audio_io_handle_t id, 1114 uint32_t device, 1115 type_t type = MIXER); 1116 virtual ~MixerThread(); 1117 1118 // Thread virtuals 1119 1120 void invalidateTracks(audio_stream_type_t streamType); 1121 virtual bool checkForNewParameters_l(); 1122 virtual status_t dumpInternals(int fd, const Vector<String16>& args); 1123 1124 protected: 1125 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1126 virtual int getTrackName_l(audio_channel_mask_t channelMask); 1127 virtual void deleteTrackName_l(int name); 1128 virtual uint32_t idleSleepTimeUs() const; 1129 virtual uint32_t suspendSleepTimeUs() const; 1130 virtual void cacheParameters_l(); 1131 1132 // threadLoop snippets 1133 virtual void threadLoop_write(); 1134 virtual void threadLoop_standby(); 1135 virtual void threadLoop_mix(); 1136 virtual void threadLoop_sleepTime(); 1137 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 1138 1139 AudioMixer* mAudioMixer; // normal mixer 1140 private: 1141#ifdef SOAKER 1142 Thread* mSoaker; 1143#endif 1144 // one-time initialization, no locks required 1145 FastMixer* mFastMixer; // non-NULL if there is also a fast mixer 1146 1147 // contents are not guaranteed to be consistent, no locks required 1148 FastMixerDumpState mFastMixerDumpState; 1149 1150 // accessible only within the threadLoop(), no locks required 1151 // mFastMixer->sq() // for mutating and pushing state 1152 int32_t mFastMixerFutex; // for cold idle 1153 1154 public: 1155 virtual bool hasFastMixer() const { return mFastMixer != NULL; } 1156 virtual uint32_t getFastTrackUnderruns(size_t fastIndex) const { 1157 ALOG_ASSERT(0 < fastIndex && 1158 fastIndex < FastMixerState::kMaxFastTracks); 1159 return mFastMixerDumpState.mTracks[fastIndex].mUnderruns; 1160 } 1161 }; 1162 1163 class DirectOutputThread : public PlaybackThread { 1164 public: 1165 1166 DirectOutputThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1167 audio_io_handle_t id, uint32_t device); 1168 virtual ~DirectOutputThread(); 1169 1170 // Thread virtuals 1171 1172 virtual bool checkForNewParameters_l(); 1173 1174 protected: 1175 virtual int getTrackName_l(audio_channel_mask_t channelMask); 1176 virtual void deleteTrackName_l(int name); 1177 virtual uint32_t activeSleepTimeUs() const; 1178 virtual uint32_t idleSleepTimeUs() const; 1179 virtual uint32_t suspendSleepTimeUs() const; 1180 virtual void cacheParameters_l(); 1181 1182 // threadLoop snippets 1183 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1184 virtual void threadLoop_mix(); 1185 virtual void threadLoop_sleepTime(); 1186 1187 // volumes last sent to audio HAL with stream->set_volume() 1188 // FIXME use standard representation and names 1189 float mLeftVolFloat; 1190 float mRightVolFloat; 1191 uint16_t mLeftVolShort; 1192 uint16_t mRightVolShort; 1193 1194 // FIXME rename these former local variables of threadLoop to standard names 1195 // next 3 were local to the while !exitingPending loop 1196 bool rampVolume; 1197 uint16_t leftVol; 1198 uint16_t rightVol; 1199 1200private: 1201 // prepareTracks_l() tells threadLoop_mix() the name of the single active track 1202 sp<Track> mActiveTrack; 1203 public: 1204 virtual bool hasFastMixer() const { return false; } 1205 }; 1206 1207 class DuplicatingThread : public MixerThread { 1208 public: 1209 DuplicatingThread (const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread, 1210 audio_io_handle_t id); 1211 virtual ~DuplicatingThread(); 1212 1213 // Thread virtuals 1214 void addOutputTrack(MixerThread* thread); 1215 void removeOutputTrack(MixerThread* thread); 1216 uint32_t waitTimeMs() const { return mWaitTimeMs; } 1217 protected: 1218 virtual uint32_t activeSleepTimeUs() const; 1219 1220 private: 1221 bool outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks); 1222 protected: 1223 // threadLoop snippets 1224 virtual void threadLoop_mix(); 1225 virtual void threadLoop_sleepTime(); 1226 virtual void threadLoop_write(); 1227 virtual void threadLoop_standby(); 1228 virtual void cacheParameters_l(); 1229 1230 private: 1231 // called from threadLoop, addOutputTrack, removeOutputTrack 1232 virtual void updateWaitTime_l(); 1233 protected: 1234 virtual void saveOutputTracks(); 1235 virtual void clearOutputTracks(); 1236 private: 1237 1238 uint32_t mWaitTimeMs; 1239 SortedVector < sp<OutputTrack> > outputTracks; 1240 SortedVector < sp<OutputTrack> > mOutputTracks; 1241 public: 1242 virtual bool hasFastMixer() const { return false; } 1243 }; 1244 1245 PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const; 1246 MixerThread *checkMixerThread_l(audio_io_handle_t output) const; 1247 RecordThread *checkRecordThread_l(audio_io_handle_t input) const; 1248 // no range check, AudioFlinger::mLock held 1249 bool streamMute_l(audio_stream_type_t stream) const 1250 { return mStreamTypes[stream].mute; } 1251 // no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held 1252 float streamVolume_l(audio_stream_type_t stream) const 1253 { return mStreamTypes[stream].volume; } 1254 void audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2); 1255 1256 // allocate an audio_io_handle_t, session ID, or effect ID 1257 uint32_t nextUniqueId(); 1258 1259 status_t moveEffectChain_l(int sessionId, 1260 PlaybackThread *srcThread, 1261 PlaybackThread *dstThread, 1262 bool reRegister); 1263 // return thread associated with primary hardware device, or NULL 1264 PlaybackThread *primaryPlaybackThread_l() const; 1265 uint32_t primaryOutputDevice_l() const; 1266 1267 // server side of the client's IAudioTrack 1268 class TrackHandle : public android::BnAudioTrack { 1269 public: 1270 TrackHandle(const sp<PlaybackThread::Track>& track); 1271 virtual ~TrackHandle(); 1272 virtual sp<IMemory> getCblk() const; 1273 virtual status_t start(); 1274 virtual void stop(); 1275 virtual void flush(); 1276 virtual void mute(bool); 1277 virtual void pause(); 1278 virtual status_t attachAuxEffect(int effectId); 1279 virtual status_t allocateTimedBuffer(size_t size, 1280 sp<IMemory>* buffer); 1281 virtual status_t queueTimedBuffer(const sp<IMemory>& buffer, 1282 int64_t pts); 1283 virtual status_t setMediaTimeTransform(const LinearTransform& xform, 1284 int target); 1285 virtual status_t onTransact( 1286 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 1287 private: 1288 const sp<PlaybackThread::Track> mTrack; 1289 }; 1290 1291 void removeClient_l(pid_t pid); 1292 void removeNotificationClient(pid_t pid); 1293 1294 1295 // record thread 1296 class RecordThread : public ThreadBase, public AudioBufferProvider 1297 { 1298 public: 1299 1300 // record track 1301 class RecordTrack : public TrackBase { 1302 public: 1303 RecordTrack(RecordThread *thread, 1304 const sp<Client>& client, 1305 uint32_t sampleRate, 1306 audio_format_t format, 1307 uint32_t channelMask, 1308 int frameCount, 1309 int sessionId); 1310 virtual ~RecordTrack(); 1311 1312 virtual status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 1313 int triggerSession = 0); 1314 virtual void stop(); 1315 1316 bool overflow() { bool tmp = mOverflow; mOverflow = false; return tmp; } 1317 bool setOverflow() { bool tmp = mOverflow; mOverflow = true; return tmp; } 1318 1319 void dump(char* buffer, size_t size); 1320 1321 private: 1322 friend class AudioFlinger; // for mState 1323 1324 RecordTrack(const RecordTrack&); 1325 RecordTrack& operator = (const RecordTrack&); 1326 1327 // AudioBufferProvider interface 1328 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts = kInvalidPTS); 1329 // releaseBuffer() not overridden 1330 1331 bool mOverflow; 1332 }; 1333 1334 1335 RecordThread(const sp<AudioFlinger>& audioFlinger, 1336 AudioStreamIn *input, 1337 uint32_t sampleRate, 1338 uint32_t channels, 1339 audio_io_handle_t id, 1340 uint32_t device); 1341 virtual ~RecordThread(); 1342 1343 // Thread 1344 virtual bool threadLoop(); 1345 virtual status_t readyToRun(); 1346 1347 // RefBase 1348 virtual void onFirstRef(); 1349 1350 virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; } 1351 sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l( 1352 const sp<AudioFlinger::Client>& client, 1353 uint32_t sampleRate, 1354 audio_format_t format, 1355 int channelMask, 1356 int frameCount, 1357 int sessionId, 1358 status_t *status); 1359 1360 status_t start(RecordTrack* recordTrack, 1361 AudioSystem::sync_event_t event, 1362 int triggerSession); 1363 void stop(RecordTrack* recordTrack); 1364 status_t dump(int fd, const Vector<String16>& args); 1365 AudioStreamIn* getInput() const; 1366 AudioStreamIn* clearInput(); 1367 virtual audio_stream_t* stream() const; 1368 1369 // AudioBufferProvider interface 1370 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts); 1371 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 1372 1373 virtual bool checkForNewParameters_l(); 1374 virtual String8 getParameters(const String8& keys); 1375 virtual void audioConfigChanged_l(int event, int param = 0); 1376 void readInputParameters(); 1377 virtual unsigned int getInputFramesLost(); 1378 1379 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1380 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1381 virtual uint32_t hasAudioSession(int sessionId); 1382 RecordTrack* track(); 1383 1384 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 1385 virtual bool isValidSyncEvent(const sp<SyncEvent>& event); 1386 1387 static void syncStartEventCallback(const wp<SyncEvent>& event); 1388 void handleSyncStartEvent(const sp<SyncEvent>& event); 1389 1390 private: 1391 void clearSyncStartEvent(); 1392 1393 RecordThread(); 1394 AudioStreamIn *mInput; 1395 RecordTrack* mTrack; 1396 sp<RecordTrack> mActiveTrack; 1397 Condition mStartStopCond; 1398 AudioResampler *mResampler; 1399 int32_t *mRsmpOutBuffer; 1400 int16_t *mRsmpInBuffer; 1401 size_t mRsmpInIndex; 1402 size_t mInputBytes; 1403 const int mReqChannelCount; 1404 const uint32_t mReqSampleRate; 1405 ssize_t mBytesRead; 1406 // sync event triggering actual audio capture. Frames read before this event will 1407 // be dropped and therefore not read by the application. 1408 sp<SyncEvent> mSyncStartEvent; 1409 // number of captured frames to drop after the start sync event has been received. 1410 ssize_t mFramestoDrop; 1411 }; 1412 1413 // server side of the client's IAudioRecord 1414 class RecordHandle : public android::BnAudioRecord { 1415 public: 1416 RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack); 1417 virtual ~RecordHandle(); 1418 virtual sp<IMemory> getCblk() const; 1419 virtual status_t start(int event, int triggerSession); 1420 virtual void stop(); 1421 virtual status_t onTransact( 1422 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 1423 private: 1424 const sp<RecordThread::RecordTrack> mRecordTrack; 1425 }; 1426 1427 //--- Audio Effect Management 1428 1429 // EffectModule and EffectChain classes both have their own mutex to protect 1430 // state changes or resource modifications. Always respect the following order 1431 // if multiple mutexes must be acquired to avoid cross deadlock: 1432 // AudioFlinger -> ThreadBase -> EffectChain -> EffectModule 1433 1434 // The EffectModule class is a wrapper object controlling the effect engine implementation 1435 // in the effect library. It prevents concurrent calls to process() and command() functions 1436 // from different client threads. It keeps a list of EffectHandle objects corresponding 1437 // to all client applications using this effect and notifies applications of effect state, 1438 // control or parameter changes. It manages the activation state machine to send appropriate 1439 // reset, enable, disable commands to effect engine and provide volume 1440 // ramping when effects are activated/deactivated. 1441 // When controlling an auxiliary effect, the EffectModule also provides an input buffer used by 1442 // the attached track(s) to accumulate their auxiliary channel. 1443 class EffectModule: public RefBase { 1444 public: 1445 EffectModule(ThreadBase *thread, 1446 const wp<AudioFlinger::EffectChain>& chain, 1447 effect_descriptor_t *desc, 1448 int id, 1449 int sessionId); 1450 virtual ~EffectModule(); 1451 1452 enum effect_state { 1453 IDLE, 1454 RESTART, 1455 STARTING, 1456 ACTIVE, 1457 STOPPING, 1458 STOPPED, 1459 DESTROYED 1460 }; 1461 1462 int id() const { return mId; } 1463 void process(); 1464 void updateState(); 1465 status_t command(uint32_t cmdCode, 1466 uint32_t cmdSize, 1467 void *pCmdData, 1468 uint32_t *replySize, 1469 void *pReplyData); 1470 1471 void reset_l(); 1472 status_t configure(); 1473 status_t init(); 1474 effect_state state() const { 1475 return mState; 1476 } 1477 uint32_t status() { 1478 return mStatus; 1479 } 1480 int sessionId() const { 1481 return mSessionId; 1482 } 1483 status_t setEnabled(bool enabled); 1484 bool isEnabled() const; 1485 bool isProcessEnabled() const; 1486 1487 void setInBuffer(int16_t *buffer) { mConfig.inputCfg.buffer.s16 = buffer; } 1488 int16_t *inBuffer() { return mConfig.inputCfg.buffer.s16; } 1489 void setOutBuffer(int16_t *buffer) { mConfig.outputCfg.buffer.s16 = buffer; } 1490 int16_t *outBuffer() { return mConfig.outputCfg.buffer.s16; } 1491 void setChain(const wp<EffectChain>& chain) { mChain = chain; } 1492 void setThread(const wp<ThreadBase>& thread) { mThread = thread; } 1493 const wp<ThreadBase>& thread() { return mThread; } 1494 1495 status_t addHandle(const sp<EffectHandle>& handle); 1496 void disconnect(const wp<EffectHandle>& handle, bool unpinIfLast); 1497 size_t removeHandle (const wp<EffectHandle>& handle); 1498 1499 effect_descriptor_t& desc() { return mDescriptor; } 1500 wp<EffectChain>& chain() { return mChain; } 1501 1502 status_t setDevice(uint32_t device); 1503 status_t setVolume(uint32_t *left, uint32_t *right, bool controller); 1504 status_t setMode(audio_mode_t mode); 1505 status_t start(); 1506 status_t stop(); 1507 void setSuspended(bool suspended); 1508 bool suspended() const; 1509 1510 sp<EffectHandle> controlHandle(); 1511 1512 bool isPinned() const { return mPinned; } 1513 void unPin() { mPinned = false; } 1514 1515 status_t dump(int fd, const Vector<String16>& args); 1516 1517 protected: 1518 friend class AudioFlinger; // for mHandles 1519 bool mPinned; 1520 1521 // Maximum time allocated to effect engines to complete the turn off sequence 1522 static const uint32_t MAX_DISABLE_TIME_MS = 10000; 1523 1524 EffectModule(const EffectModule&); 1525 EffectModule& operator = (const EffectModule&); 1526 1527 status_t start_l(); 1528 status_t stop_l(); 1529 1530mutable Mutex mLock; // mutex for process, commands and handles list protection 1531 wp<ThreadBase> mThread; // parent thread 1532 wp<EffectChain> mChain; // parent effect chain 1533 int mId; // this instance unique ID 1534 int mSessionId; // audio session ID 1535 effect_descriptor_t mDescriptor;// effect descriptor received from effect engine 1536 effect_config_t mConfig; // input and output audio configuration 1537 effect_handle_t mEffectInterface; // Effect module C API 1538 status_t mStatus; // initialization status 1539 effect_state mState; // current activation state 1540 Vector< wp<EffectHandle> > mHandles; // list of client handles 1541 // First handle in mHandles has highest priority and controls the effect module 1542 uint32_t mMaxDisableWaitCnt; // maximum grace period before forcing an effect off after 1543 // sending disable command. 1544 uint32_t mDisableWaitCnt; // current process() calls count during disable period. 1545 bool mSuspended; // effect is suspended: temporarily disabled by framework 1546 }; 1547 1548 // The EffectHandle class implements the IEffect interface. It provides resources 1549 // to receive parameter updates, keeps track of effect control 1550 // ownership and state and has a pointer to the EffectModule object it is controlling. 1551 // There is one EffectHandle object for each application controlling (or using) 1552 // an effect module. 1553 // The EffectHandle is obtained by calling AudioFlinger::createEffect(). 1554 class EffectHandle: public android::BnEffect { 1555 public: 1556 1557 EffectHandle(const sp<EffectModule>& effect, 1558 const sp<AudioFlinger::Client>& client, 1559 const sp<IEffectClient>& effectClient, 1560 int32_t priority); 1561 virtual ~EffectHandle(); 1562 1563 // IEffect 1564 virtual status_t enable(); 1565 virtual status_t disable(); 1566 virtual status_t command(uint32_t cmdCode, 1567 uint32_t cmdSize, 1568 void *pCmdData, 1569 uint32_t *replySize, 1570 void *pReplyData); 1571 virtual void disconnect(); 1572 private: 1573 void disconnect(bool unpinIfLast); 1574 public: 1575 virtual sp<IMemory> getCblk() const { return mCblkMemory; } 1576 virtual status_t onTransact(uint32_t code, const Parcel& data, 1577 Parcel* reply, uint32_t flags); 1578 1579 1580 // Give or take control of effect module 1581 // - hasControl: true if control is given, false if removed 1582 // - signal: true client app should be signaled of change, false otherwise 1583 // - enabled: state of the effect when control is passed 1584 void setControl(bool hasControl, bool signal, bool enabled); 1585 void commandExecuted(uint32_t cmdCode, 1586 uint32_t cmdSize, 1587 void *pCmdData, 1588 uint32_t replySize, 1589 void *pReplyData); 1590 void setEnabled(bool enabled); 1591 bool enabled() const { return mEnabled; } 1592 1593 // Getters 1594 int id() const { return mEffect->id(); } 1595 int priority() const { return mPriority; } 1596 bool hasControl() const { return mHasControl; } 1597 sp<EffectModule> effect() const { return mEffect; } 1598 1599 void dump(char* buffer, size_t size); 1600 1601 protected: 1602 friend class AudioFlinger; // for mEffect, mHasControl, mEnabled 1603 EffectHandle(const EffectHandle&); 1604 EffectHandle& operator =(const EffectHandle&); 1605 1606 sp<EffectModule> mEffect; // pointer to controlled EffectModule 1607 sp<IEffectClient> mEffectClient; // callback interface for client notifications 1608 /*const*/ sp<Client> mClient; // client for shared memory allocation, see disconnect() 1609 sp<IMemory> mCblkMemory; // shared memory for control block 1610 effect_param_cblk_t* mCblk; // control block for deferred parameter setting via shared memory 1611 uint8_t* mBuffer; // pointer to parameter area in shared memory 1612 int mPriority; // client application priority to control the effect 1613 bool mHasControl; // true if this handle is controlling the effect 1614 bool mEnabled; // cached enable state: needed when the effect is 1615 // restored after being suspended 1616 }; 1617 1618 // the EffectChain class represents a group of effects associated to one audio session. 1619 // There can be any number of EffectChain objects per output mixer thread (PlaybackThread). 1620 // The EffecChain with session ID 0 contains global effects applied to the output mix. 1621 // Effects in this chain can be insert or auxiliary. Effects in other chains (attached to tracks) 1622 // are insert only. The EffectChain maintains an ordered list of effect module, the order corresponding 1623 // in the effect process order. When attached to a track (session ID != 0), it also provide it's own 1624 // input buffer used by the track as accumulation buffer. 1625 class EffectChain: public RefBase { 1626 public: 1627 EffectChain(const wp<ThreadBase>& wThread, int sessionId); 1628 EffectChain(ThreadBase *thread, int sessionId); 1629 virtual ~EffectChain(); 1630 1631 // special key used for an entry in mSuspendedEffects keyed vector 1632 // corresponding to a suspend all request. 1633 static const int kKeyForSuspendAll = 0; 1634 1635 // minimum duration during which we force calling effect process when last track on 1636 // a session is stopped or removed to allow effect tail to be rendered 1637 static const int kProcessTailDurationMs = 1000; 1638 1639 void process_l(); 1640 1641 void lock() { 1642 mLock.lock(); 1643 } 1644 void unlock() { 1645 mLock.unlock(); 1646 } 1647 1648 status_t addEffect_l(const sp<EffectModule>& handle); 1649 size_t removeEffect_l(const sp<EffectModule>& handle); 1650 1651 int sessionId() const { return mSessionId; } 1652 void setSessionId(int sessionId) { mSessionId = sessionId; } 1653 1654 sp<EffectModule> getEffectFromDesc_l(effect_descriptor_t *descriptor); 1655 sp<EffectModule> getEffectFromId_l(int id); 1656 sp<EffectModule> getEffectFromType_l(const effect_uuid_t *type); 1657 bool setVolume_l(uint32_t *left, uint32_t *right); 1658 void setDevice_l(uint32_t device); 1659 void setMode_l(audio_mode_t mode); 1660 1661 void setInBuffer(int16_t *buffer, bool ownsBuffer = false) { 1662 mInBuffer = buffer; 1663 mOwnInBuffer = ownsBuffer; 1664 } 1665 int16_t *inBuffer() const { 1666 return mInBuffer; 1667 } 1668 void setOutBuffer(int16_t *buffer) { 1669 mOutBuffer = buffer; 1670 } 1671 int16_t *outBuffer() const { 1672 return mOutBuffer; 1673 } 1674 1675 void incTrackCnt() { android_atomic_inc(&mTrackCnt); } 1676 void decTrackCnt() { android_atomic_dec(&mTrackCnt); } 1677 int32_t trackCnt() const { return mTrackCnt;} 1678 1679 void incActiveTrackCnt() { android_atomic_inc(&mActiveTrackCnt); 1680 mTailBufferCount = mMaxTailBuffers; } 1681 void decActiveTrackCnt() { android_atomic_dec(&mActiveTrackCnt); } 1682 int32_t activeTrackCnt() const { return mActiveTrackCnt;} 1683 1684 uint32_t strategy() const { return mStrategy; } 1685 void setStrategy(uint32_t strategy) 1686 { mStrategy = strategy; } 1687 1688 // suspend effect of the given type 1689 void setEffectSuspended_l(const effect_uuid_t *type, 1690 bool suspend); 1691 // suspend all eligible effects 1692 void setEffectSuspendedAll_l(bool suspend); 1693 // check if effects should be suspend or restored when a given effect is enable or disabled 1694 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1695 bool enabled); 1696 1697 status_t dump(int fd, const Vector<String16>& args); 1698 1699 protected: 1700 friend class AudioFlinger; // for mThread, mEffects 1701 EffectChain(const EffectChain&); 1702 EffectChain& operator =(const EffectChain&); 1703 1704 class SuspendedEffectDesc : public RefBase { 1705 public: 1706 SuspendedEffectDesc() : mRefCount(0) {} 1707 1708 int mRefCount; 1709 effect_uuid_t mType; 1710 wp<EffectModule> mEffect; 1711 }; 1712 1713 // get a list of effect modules to suspend when an effect of the type 1714 // passed is enabled. 1715 void getSuspendEligibleEffects(Vector< sp<EffectModule> > &effects); 1716 1717 // get an effect module if it is currently enable 1718 sp<EffectModule> getEffectIfEnabled(const effect_uuid_t *type); 1719 // true if the effect whose descriptor is passed can be suspended 1720 // OEMs can modify the rules implemented in this method to exclude specific effect 1721 // types or implementations from the suspend/restore mechanism. 1722 bool isEffectEligibleForSuspend(const effect_descriptor_t& desc); 1723 1724 wp<ThreadBase> mThread; // parent mixer thread 1725 Mutex mLock; // mutex protecting effect list 1726 Vector< sp<EffectModule> > mEffects; // list of effect modules 1727 int mSessionId; // audio session ID 1728 int16_t *mInBuffer; // chain input buffer 1729 int16_t *mOutBuffer; // chain output buffer 1730 volatile int32_t mActiveTrackCnt; // number of active tracks connected 1731 volatile int32_t mTrackCnt; // number of tracks connected 1732 int32_t mTailBufferCount; // current effect tail buffer count 1733 int32_t mMaxTailBuffers; // maximum effect tail buffers 1734 bool mOwnInBuffer; // true if the chain owns its input buffer 1735 int mVolumeCtrlIdx; // index of insert effect having control over volume 1736 uint32_t mLeftVolume; // previous volume on left channel 1737 uint32_t mRightVolume; // previous volume on right channel 1738 uint32_t mNewLeftVolume; // new volume on left channel 1739 uint32_t mNewRightVolume; // new volume on right channel 1740 uint32_t mStrategy; // strategy for this effect chain 1741 // mSuspendedEffects lists all effects currently suspended in the chain. 1742 // Use effect type UUID timelow field as key. There is no real risk of identical 1743 // timeLow fields among effect type UUIDs. 1744 // Updated by updateSuspendedSessions_l() only. 1745 KeyedVector< int, sp<SuspendedEffectDesc> > mSuspendedEffects; 1746 }; 1747 1748 // AudioStreamOut and AudioStreamIn are immutable, so their fields are const. 1749 // For emphasis, we could also make all pointers to them be "const *", 1750 // but that would clutter the code unnecessarily. 1751 1752 struct AudioStreamOut { 1753 audio_hw_device_t* const hwDev; 1754 audio_stream_out_t* const stream; 1755 1756 AudioStreamOut(audio_hw_device_t *dev, audio_stream_out_t *out) : 1757 hwDev(dev), stream(out) {} 1758 }; 1759 1760 struct AudioStreamIn { 1761 audio_hw_device_t* const hwDev; 1762 audio_stream_in_t* const stream; 1763 1764 AudioStreamIn(audio_hw_device_t *dev, audio_stream_in_t *in) : 1765 hwDev(dev), stream(in) {} 1766 }; 1767 1768 // for mAudioSessionRefs only 1769 struct AudioSessionRef { 1770 AudioSessionRef(int sessionid, pid_t pid) : 1771 mSessionid(sessionid), mPid(pid), mCnt(1) {} 1772 const int mSessionid; 1773 const pid_t mPid; 1774 int mCnt; 1775 }; 1776 1777 enum master_volume_support { 1778 // MVS_NONE: 1779 // Audio HAL has no support for master volume, either setting or 1780 // getting. All master volume control must be implemented in SW by the 1781 // AudioFlinger mixing core. 1782 MVS_NONE, 1783 1784 // MVS_SETONLY: 1785 // Audio HAL has support for setting master volume, but not for getting 1786 // master volume (original HAL design did not include a getter). 1787 // AudioFlinger needs to keep track of the last set master volume in 1788 // addition to needing to set an initial, default, master volume at HAL 1789 // load time. 1790 MVS_SETONLY, 1791 1792 // MVS_FULL: 1793 // Audio HAL has support both for setting and getting master volume. 1794 // AudioFlinger should send all set and get master volume requests 1795 // directly to the HAL. 1796 MVS_FULL, 1797 }; 1798 1799 class AudioHwDevice { 1800 public: 1801 AudioHwDevice(const char *moduleName, audio_hw_device_t *hwDevice) : 1802 mModuleName(strdup(moduleName)), mHwDevice(hwDevice){} 1803 ~AudioHwDevice() { free((void *)mModuleName); } 1804 1805 const char *moduleName() const { return mModuleName; } 1806 audio_hw_device_t *hwDevice() const { return mHwDevice; } 1807 private: 1808 const char * const mModuleName; 1809 audio_hw_device_t * const mHwDevice; 1810 }; 1811 1812 mutable Mutex mLock; 1813 1814 DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client() 1815 1816 mutable Mutex mHardwareLock; 1817 // NOTE: If both mLock and mHardwareLock mutexes must be held, 1818 // always take mLock before mHardwareLock 1819 1820 // These two fields are immutable after onFirstRef(), so no lock needed to access 1821 audio_hw_device_t* mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL 1822 DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*> mAudioHwDevs; 1823 1824 // for dump, indicates which hardware operation is currently in progress (but not stream ops) 1825 enum hardware_call_state { 1826 AUDIO_HW_IDLE = 0, // no operation in progress 1827 AUDIO_HW_INIT, // init_check 1828 AUDIO_HW_OUTPUT_OPEN, // open_output_stream 1829 AUDIO_HW_OUTPUT_CLOSE, // unused 1830 AUDIO_HW_INPUT_OPEN, // unused 1831 AUDIO_HW_INPUT_CLOSE, // unused 1832 AUDIO_HW_STANDBY, // unused 1833 AUDIO_HW_SET_MASTER_VOLUME, // set_master_volume 1834 AUDIO_HW_GET_ROUTING, // unused 1835 AUDIO_HW_SET_ROUTING, // unused 1836 AUDIO_HW_GET_MODE, // unused 1837 AUDIO_HW_SET_MODE, // set_mode 1838 AUDIO_HW_GET_MIC_MUTE, // get_mic_mute 1839 AUDIO_HW_SET_MIC_MUTE, // set_mic_mute 1840 AUDIO_HW_SET_VOICE_VOLUME, // set_voice_volume 1841 AUDIO_HW_SET_PARAMETER, // set_parameters 1842 AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size 1843 AUDIO_HW_GET_MASTER_VOLUME, // get_master_volume 1844 AUDIO_HW_GET_PARAMETER, // get_parameters 1845 }; 1846 1847 mutable hardware_call_state mHardwareStatus; // for dump only 1848 1849 1850 DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads; 1851 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 1852 1853 // both are protected by mLock 1854 float mMasterVolume; 1855 float mMasterVolumeSW; 1856 master_volume_support mMasterVolumeSupportLvl; 1857 bool mMasterMute; 1858 1859 DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads; 1860 1861 DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients; 1862 volatile int32_t mNextUniqueId; // updated by android_atomic_inc 1863 audio_mode_t mMode; 1864 bool mBtNrecIsOff; 1865 1866 // protected by mLock 1867 Vector<AudioSessionRef*> mAudioSessionRefs; 1868 1869 float masterVolume_l() const; 1870 float masterVolumeSW_l() const { return mMasterVolumeSW; } 1871 bool masterMute_l() const { return mMasterMute; } 1872 audio_module_handle_t loadHwModule_l(const char *name); 1873 1874 Vector < sp<SyncEvent> > mPendingSyncEvents; // sync events awaiting for a session 1875 // to be created 1876 1877private: 1878 sp<Client> registerPid_l(pid_t pid); // always returns non-0 1879 1880}; 1881 1882 1883// ---------------------------------------------------------------------------- 1884 1885}; // namespace android 1886 1887#endif // ANDROID_AUDIO_FLINGER_H 1888