AudioFlinger.h revision 1d491ff06f4b9c90ff24fe953b90d0843eaf1c04
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 pid_t tid, 112 int *sessionId, 113 status_t *status); 114 115 virtual uint32_t sampleRate(audio_io_handle_t output) const; 116 virtual int channelCount(audio_io_handle_t output) const; 117 virtual audio_format_t format(audio_io_handle_t output) const; 118 virtual size_t frameCount(audio_io_handle_t output) const; 119 virtual uint32_t latency(audio_io_handle_t output) const; 120 121 virtual status_t setMasterVolume(float value); 122 virtual status_t setMasterMute(bool muted); 123 124 virtual float masterVolume() const; 125 virtual float masterVolumeSW() const; 126 virtual bool masterMute() const; 127 128 virtual status_t setStreamVolume(audio_stream_type_t stream, float value, 129 audio_io_handle_t output); 130 virtual status_t setStreamMute(audio_stream_type_t stream, bool muted); 131 132 virtual float streamVolume(audio_stream_type_t stream, 133 audio_io_handle_t output) const; 134 virtual bool streamMute(audio_stream_type_t stream) const; 135 136 virtual status_t setMode(audio_mode_t mode); 137 138 virtual status_t setMicMute(bool state); 139 virtual bool getMicMute() const; 140 141 virtual status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs); 142 virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) const; 143 144 virtual void registerClient(const sp<IAudioFlingerClient>& client); 145 146 virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, 147 audio_channel_mask_t channelMask) const; 148 149 virtual audio_io_handle_t openOutput(audio_module_handle_t module, 150 audio_devices_t *pDevices, 151 uint32_t *pSamplingRate, 152 audio_format_t *pFormat, 153 audio_channel_mask_t *pChannelMask, 154 uint32_t *pLatencyMs, 155 audio_output_flags_t flags); 156 157 virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1, 158 audio_io_handle_t output2); 159 160 virtual status_t closeOutput(audio_io_handle_t output); 161 162 virtual status_t suspendOutput(audio_io_handle_t output); 163 164 virtual status_t restoreOutput(audio_io_handle_t output); 165 166 virtual audio_io_handle_t openInput(audio_module_handle_t module, 167 audio_devices_t *pDevices, 168 uint32_t *pSamplingRate, 169 audio_format_t *pFormat, 170 audio_channel_mask_t *pChannelMask); 171 172 virtual status_t closeInput(audio_io_handle_t input); 173 174 virtual status_t setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output); 175 176 virtual status_t setVoiceVolume(float volume); 177 178 virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 179 audio_io_handle_t output) const; 180 181 virtual unsigned int getInputFramesLost(audio_io_handle_t ioHandle) const; 182 183 virtual int newAudioSessionId(); 184 185 virtual void acquireAudioSessionId(int audioSession); 186 187 virtual void releaseAudioSessionId(int audioSession); 188 189 virtual status_t queryNumberEffects(uint32_t *numEffects) const; 190 191 virtual status_t queryEffect(uint32_t index, effect_descriptor_t *descriptor) const; 192 193 virtual status_t getEffectDescriptor(const effect_uuid_t *pUuid, 194 effect_descriptor_t *descriptor) const; 195 196 virtual sp<IEffect> createEffect(pid_t pid, 197 effect_descriptor_t *pDesc, 198 const sp<IEffectClient>& effectClient, 199 int32_t priority, 200 audio_io_handle_t io, 201 int sessionId, 202 status_t *status, 203 int *id, 204 int *enabled); 205 206 virtual status_t moveEffects(int sessionId, audio_io_handle_t srcOutput, 207 audio_io_handle_t dstOutput); 208 209 virtual audio_module_handle_t loadHwModule(const char *name); 210 211 virtual status_t onTransact( 212 uint32_t code, 213 const Parcel& data, 214 Parcel* reply, 215 uint32_t flags); 216 217 // end of IAudioFlinger interface 218 219 class SyncEvent; 220 221 typedef void (*sync_event_callback_t)(const wp<SyncEvent>& event) ; 222 223 class SyncEvent : public RefBase { 224 public: 225 SyncEvent(AudioSystem::sync_event_t type, 226 int triggerSession, 227 int listenerSession, 228 sync_event_callback_t callBack, 229 void *cookie) 230 : mType(type), mTriggerSession(triggerSession), mListenerSession(listenerSession), 231 mCallback(callBack), mCookie(cookie) 232 {} 233 234 virtual ~SyncEvent() {} 235 236 void trigger() { Mutex::Autolock _l(mLock); if (mCallback) mCallback(this); } 237 bool isCancelled() { Mutex::Autolock _l(mLock); return (mCallback == NULL); } 238 void cancel() {Mutex::Autolock _l(mLock); mCallback = NULL; } 239 AudioSystem::sync_event_t type() const { return mType; } 240 int triggerSession() const { return mTriggerSession; } 241 int listenerSession() const { return mListenerSession; } 242 void *cookie() const { return mCookie; } 243 244 private: 245 const AudioSystem::sync_event_t mType; 246 const int mTriggerSession; 247 const int mListenerSession; 248 sync_event_callback_t mCallback; 249 void * const mCookie; 250 Mutex mLock; 251 }; 252 253 sp<SyncEvent> createSyncEvent(AudioSystem::sync_event_t type, 254 int triggerSession, 255 int listenerSession, 256 sync_event_callback_t callBack, 257 void *cookie); 258 259private: 260 audio_mode_t getMode() const { return mMode; } 261 262 bool btNrecIsOff() const { return mBtNrecIsOff; } 263 264 AudioFlinger(); 265 virtual ~AudioFlinger(); 266 267 // call in any IAudioFlinger method that accesses mPrimaryHardwareDev 268 status_t initCheck() const { return mPrimaryHardwareDev == NULL ? NO_INIT : NO_ERROR; } 269 270 // RefBase 271 virtual void onFirstRef(); 272 273 audio_hw_device_t* findSuitableHwDev_l(audio_module_handle_t module, audio_devices_t devices); 274 void purgeStaleEffects_l(); 275 276 // standby delay for MIXER and DUPLICATING playback threads is read from property 277 // ro.audio.flinger_standbytime_ms or defaults to kDefaultStandbyTimeInNsecs 278 static nsecs_t mStandbyTimeInNsecs; 279 280 // Internal dump utilities. 281 void dumpPermissionDenial(int fd, const Vector<String16>& args); 282 void dumpClients(int fd, const Vector<String16>& args); 283 void dumpInternals(int fd, const Vector<String16>& args); 284 285 // --- Client --- 286 class Client : public RefBase { 287 public: 288 Client(const sp<AudioFlinger>& audioFlinger, pid_t pid); 289 virtual ~Client(); 290 sp<MemoryDealer> heap() const; 291 pid_t pid() const { return mPid; } 292 sp<AudioFlinger> audioFlinger() const { return mAudioFlinger; } 293 294 bool reserveTimedTrack(); 295 void releaseTimedTrack(); 296 297 private: 298 Client(const Client&); 299 Client& operator = (const Client&); 300 const sp<AudioFlinger> mAudioFlinger; 301 const sp<MemoryDealer> mMemoryDealer; 302 const pid_t mPid; 303 304 Mutex mTimedTrackLock; 305 int mTimedTrackCount; 306 }; 307 308 // --- Notification Client --- 309 class NotificationClient : public IBinder::DeathRecipient { 310 public: 311 NotificationClient(const sp<AudioFlinger>& audioFlinger, 312 const sp<IAudioFlingerClient>& client, 313 pid_t pid); 314 virtual ~NotificationClient(); 315 316 sp<IAudioFlingerClient> audioFlingerClient() const { return mAudioFlingerClient; } 317 318 // IBinder::DeathRecipient 319 virtual void binderDied(const wp<IBinder>& who); 320 321 private: 322 NotificationClient(const NotificationClient&); 323 NotificationClient& operator = (const NotificationClient&); 324 325 const sp<AudioFlinger> mAudioFlinger; 326 const pid_t mPid; 327 const sp<IAudioFlingerClient> mAudioFlingerClient; 328 }; 329 330 class TrackHandle; 331 class RecordHandle; 332 class RecordThread; 333 class PlaybackThread; 334 class MixerThread; 335 class DirectOutputThread; 336 class DuplicatingThread; 337 class Track; 338 class RecordTrack; 339 class EffectModule; 340 class EffectHandle; 341 class EffectChain; 342 struct AudioStreamOut; 343 struct AudioStreamIn; 344 345 class ThreadBase : public Thread { 346 public: 347 348 enum type_t { 349 MIXER, // Thread class is MixerThread 350 DIRECT, // Thread class is DirectOutputThread 351 DUPLICATING, // Thread class is DuplicatingThread 352 RECORD // Thread class is RecordThread 353 }; 354 355 ThreadBase (const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, audio_devices_t device, type_t type); 356 virtual ~ThreadBase(); 357 358 void dumpBase(int fd, const Vector<String16>& args); 359 void dumpEffectChains(int fd, const Vector<String16>& args); 360 361 void clearPowerManager(); 362 363 // base for record and playback 364 class TrackBase : public ExtendedAudioBufferProvider, public RefBase { 365 366 public: 367 enum track_state { 368 IDLE, 369 TERMINATED, 370 FLUSHED, 371 STOPPED, 372 // next 2 states are currently used for fast tracks only 373 STOPPING_1, // waiting for first underrun 374 STOPPING_2, // waiting for presentation complete 375 RESUMING, 376 ACTIVE, 377 PAUSING, 378 PAUSED 379 }; 380 381 TrackBase(ThreadBase *thread, 382 const sp<Client>& client, 383 uint32_t sampleRate, 384 audio_format_t format, 385 audio_channel_mask_t channelMask, 386 int frameCount, 387 const sp<IMemory>& sharedBuffer, 388 int sessionId); 389 virtual ~TrackBase(); 390 391 virtual status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 392 int triggerSession = 0) = 0; 393 virtual void stop() = 0; 394 sp<IMemory> getCblk() const { return mCblkMemory; } 395 audio_track_cblk_t* cblk() const { return mCblk; } 396 int sessionId() const { return mSessionId; } 397 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 398 399 protected: 400 TrackBase(const TrackBase&); 401 TrackBase& operator = (const TrackBase&); 402 403 // AudioBufferProvider interface 404 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) = 0; 405 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 406 407 // ExtendedAudioBufferProvider interface is only needed for Track, 408 // but putting it in TrackBase avoids the complexity of virtual inheritance 409 virtual size_t framesReady() const { return SIZE_MAX; } 410 411 audio_format_t format() const { 412 return mFormat; 413 } 414 415 int channelCount() const { return mChannelCount; } 416 417 audio_channel_mask_t channelMask() const { return mChannelMask; } 418 419 int sampleRate() const; // FIXME inline after cblk sr moved 420 421 // Return a pointer to the start of a contiguous slice of the track buffer. 422 // Parameter 'offset' is the requested start position, expressed in 423 // monotonically increasing frame units relative to the track epoch. 424 // Parameter 'frames' is the requested length, also in frame units. 425 // Always returns non-NULL. It is the caller's responsibility to 426 // verify that this will be successful; the result of calling this 427 // function with invalid 'offset' or 'frames' is undefined. 428 void* getBuffer(uint32_t offset, uint32_t frames) const; 429 430 bool isStopped() const { 431 return (mState == STOPPED || mState == FLUSHED); 432 } 433 434 // for fast tracks only 435 bool isStopping() const { 436 return mState == STOPPING_1 || mState == STOPPING_2; 437 } 438 bool isStopping_1() const { 439 return mState == STOPPING_1; 440 } 441 bool isStopping_2() const { 442 return mState == STOPPING_2; 443 } 444 445 bool isTerminated() const { 446 return mState == TERMINATED; 447 } 448 449 bool step(); 450 void reset(); 451 452 const wp<ThreadBase> mThread; 453 /*const*/ sp<Client> mClient; // see explanation at ~TrackBase() why not const 454 sp<IMemory> mCblkMemory; 455 audio_track_cblk_t* mCblk; 456 void* mBuffer; 457 void* mBufferEnd; 458 uint32_t mFrameCount; 459 // we don't really need a lock for these 460 track_state mState; 461 const uint32_t mSampleRate; // initial sample rate only; for tracks which 462 // support dynamic rates, the current value is in control block 463 const audio_format_t mFormat; 464 bool mStepServerFailed; 465 const int mSessionId; 466 uint8_t mChannelCount; 467 audio_channel_mask_t mChannelMask; 468 Vector < sp<SyncEvent> >mSyncEvents; 469 }; 470 471 class ConfigEvent { 472 public: 473 ConfigEvent() : mEvent(0), mParam(0) {} 474 475 int mEvent; 476 int mParam; 477 }; 478 479 class PMDeathRecipient : public IBinder::DeathRecipient { 480 public: 481 PMDeathRecipient(const wp<ThreadBase>& thread) : mThread(thread) {} 482 virtual ~PMDeathRecipient() {} 483 484 // IBinder::DeathRecipient 485 virtual void binderDied(const wp<IBinder>& who); 486 487 private: 488 PMDeathRecipient(const PMDeathRecipient&); 489 PMDeathRecipient& operator = (const PMDeathRecipient&); 490 491 wp<ThreadBase> mThread; 492 }; 493 494 virtual status_t initCheck() const = 0; 495 496 // static externally-visible 497 type_t type() const { return mType; } 498 audio_io_handle_t id() const { return mId;} 499 500 // dynamic externally-visible 501 uint32_t sampleRate() const { return mSampleRate; } 502 int channelCount() const { return mChannelCount; } 503 audio_channel_mask_t channelMask() const { return mChannelMask; } 504 audio_format_t format() const { return mFormat; } 505 // Called by AudioFlinger::frameCount(audio_io_handle_t output) and effects, 506 // and returns the normal mix buffer's frame count. No API for HAL frame count. 507 size_t frameCount() const { return mNormalFrameCount; } 508 509 // Should be "virtual status_t requestExitAndWait()" and override same 510 // method in Thread, but Thread::requestExitAndWait() is not yet virtual. 511 void exit(); 512 virtual bool checkForNewParameters_l() = 0; 513 virtual status_t setParameters(const String8& keyValuePairs); 514 virtual String8 getParameters(const String8& keys) = 0; 515 virtual void audioConfigChanged_l(int event, int param = 0) = 0; 516 void sendConfigEvent(int event, int param = 0); 517 void sendConfigEvent_l(int event, int param = 0); 518 void processConfigEvents(); 519 520 // see note at declaration of mStandby and mDevice 521 bool standby() const { return mStandby; } 522 audio_devices_t device() const { return mDevice; } 523 524 virtual audio_stream_t* stream() const = 0; 525 526 sp<EffectHandle> createEffect_l( 527 const sp<AudioFlinger::Client>& client, 528 const sp<IEffectClient>& effectClient, 529 int32_t priority, 530 int sessionId, 531 effect_descriptor_t *desc, 532 int *enabled, 533 status_t *status); 534 void disconnectEffect(const sp< EffectModule>& effect, 535 EffectHandle *handle, 536 bool unpinIfLast); 537 538 // return values for hasAudioSession (bit field) 539 enum effect_state { 540 EFFECT_SESSION = 0x1, // the audio session corresponds to at least one 541 // effect 542 TRACK_SESSION = 0x2 // the audio session corresponds to at least one 543 // track 544 }; 545 546 // get effect chain corresponding to session Id. 547 sp<EffectChain> getEffectChain(int sessionId); 548 // same as getEffectChain() but must be called with ThreadBase mutex locked 549 sp<EffectChain> getEffectChain_l(int sessionId); 550 // add an effect chain to the chain list (mEffectChains) 551 virtual status_t addEffectChain_l(const sp<EffectChain>& chain) = 0; 552 // remove an effect chain from the chain list (mEffectChains) 553 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain) = 0; 554 // lock all effect chains Mutexes. Must be called before releasing the 555 // ThreadBase mutex before processing the mixer and effects. This guarantees the 556 // integrity of the chains during the process. 557 // Also sets the parameter 'effectChains' to current value of mEffectChains. 558 void lockEffectChains_l(Vector< sp<EffectChain> >& effectChains); 559 // unlock effect chains after process 560 void unlockEffectChains(const Vector< sp<EffectChain> >& effectChains); 561 // set audio mode to all effect chains 562 void setMode(audio_mode_t mode); 563 // get effect module with corresponding ID on specified audio session 564 sp<AudioFlinger::EffectModule> getEffect(int sessionId, int effectId); 565 sp<AudioFlinger::EffectModule> getEffect_l(int sessionId, int effectId); 566 // add and effect module. Also creates the effect chain is none exists for 567 // the effects audio session 568 status_t addEffect_l(const sp< EffectModule>& effect); 569 // remove and effect module. Also removes the effect chain is this was the last 570 // effect 571 void removeEffect_l(const sp< EffectModule>& effect); 572 // detach all tracks connected to an auxiliary effect 573 virtual void detachAuxEffect_l(int effectId) {} 574 // returns either EFFECT_SESSION if effects on this audio session exist in one 575 // chain, or TRACK_SESSION if tracks on this audio session exist, or both 576 virtual uint32_t hasAudioSession(int sessionId) = 0; 577 // the value returned by default implementation is not important as the 578 // strategy is only meaningful for PlaybackThread which implements this method 579 virtual uint32_t getStrategyForSession_l(int sessionId) { return 0; } 580 581 // suspend or restore effect according to the type of effect passed. a NULL 582 // type pointer means suspend all effects in the session 583 void setEffectSuspended(const effect_uuid_t *type, 584 bool suspend, 585 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 586 // check if some effects must be suspended/restored when an effect is enabled 587 // or disabled 588 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 589 bool enabled, 590 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 591 void checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 592 bool enabled, 593 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 594 595 virtual status_t setSyncEvent(const sp<SyncEvent>& event) = 0; 596 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) = 0; 597 598 599 mutable Mutex mLock; 600 601 protected: 602 603 // entry describing an effect being suspended in mSuspendedSessions keyed vector 604 class SuspendedSessionDesc : public RefBase { 605 public: 606 SuspendedSessionDesc() : mRefCount(0) {} 607 608 int mRefCount; // number of active suspend requests 609 effect_uuid_t mType; // effect type UUID 610 }; 611 612 void acquireWakeLock(); 613 void acquireWakeLock_l(); 614 void releaseWakeLock(); 615 void releaseWakeLock_l(); 616 void setEffectSuspended_l(const effect_uuid_t *type, 617 bool suspend, 618 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 619 // updated mSuspendedSessions when an effect suspended or restored 620 void updateSuspendedSessions_l(const effect_uuid_t *type, 621 bool suspend, 622 int sessionId); 623 // check if some effects must be suspended when an effect chain is added 624 void checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain); 625 626 friend class AudioFlinger; // for mEffectChains 627 628 const type_t mType; 629 630 // Used by parameters, config events, addTrack_l, exit 631 Condition mWaitWorkCV; 632 633 const sp<AudioFlinger> mAudioFlinger; 634 uint32_t mSampleRate; 635 size_t mFrameCount; // output HAL, direct output, record 636 size_t mNormalFrameCount; // normal mixer and effects 637 audio_channel_mask_t mChannelMask; 638 uint16_t mChannelCount; 639 size_t mFrameSize; 640 audio_format_t mFormat; 641 642 // Parameter sequence by client: binder thread calling setParameters(): 643 // 1. Lock mLock 644 // 2. Append to mNewParameters 645 // 3. mWaitWorkCV.signal 646 // 4. mParamCond.waitRelative with timeout 647 // 5. read mParamStatus 648 // 6. mWaitWorkCV.signal 649 // 7. Unlock 650 // 651 // Parameter sequence by server: threadLoop calling checkForNewParameters_l(): 652 // 1. Lock mLock 653 // 2. If there is an entry in mNewParameters proceed ... 654 // 2. Read first entry in mNewParameters 655 // 3. Process 656 // 4. Remove first entry from mNewParameters 657 // 5. Set mParamStatus 658 // 6. mParamCond.signal 659 // 7. mWaitWorkCV.wait with timeout (this is to avoid overwriting mParamStatus) 660 // 8. Unlock 661 Condition mParamCond; 662 Vector<String8> mNewParameters; 663 status_t mParamStatus; 664 665 Vector<ConfigEvent> mConfigEvents; 666 667 // These fields are written and read by thread itself without lock or barrier, 668 // and read by other threads without lock or barrier via standby() and device(). 669 // Because of the absence of a lock or barrier, any other thread that reads 670 // these fields must use the information in isolation, or be prepared to deal 671 // with possibility that it might be inconsistent with other information. 672 bool mStandby; // Whether thread is currently in standby. 673 audio_devices_t mDevice; // output device for PlaybackThread 674 // input + output devices for RecordThread 675 676 const audio_io_handle_t mId; 677 Vector< sp<EffectChain> > mEffectChains; 678 679 static const int kNameLength = 16; // prctl(PR_SET_NAME) limit 680 char mName[kNameLength]; 681 sp<IPowerManager> mPowerManager; 682 sp<IBinder> mWakeLockToken; 683 const sp<PMDeathRecipient> mDeathRecipient; 684 // list of suspended effects per session and per type. The first vector is 685 // keyed by session ID, the second by type UUID timeLow field 686 KeyedVector< int, KeyedVector< int, sp<SuspendedSessionDesc> > > mSuspendedSessions; 687 }; 688 689 struct stream_type_t { 690 stream_type_t() 691 : volume(1.0f), 692 mute(false) 693 { 694 } 695 float volume; 696 bool mute; 697 }; 698 699 // --- PlaybackThread --- 700 class PlaybackThread : public ThreadBase { 701 public: 702 703 enum mixer_state { 704 MIXER_IDLE, // no active tracks 705 MIXER_TRACKS_ENABLED, // at least one active track, but no track has any data ready 706 MIXER_TRACKS_READY // at least one active track, and at least one track has data 707 // standby mode does not have an enum value 708 // suspend by audio policy manager is orthogonal to mixer state 709 }; 710 711 // playback track 712 class Track : public TrackBase, public VolumeProvider { 713 public: 714 Track( PlaybackThread *thread, 715 const sp<Client>& client, 716 audio_stream_type_t streamType, 717 uint32_t sampleRate, 718 audio_format_t format, 719 audio_channel_mask_t channelMask, 720 int frameCount, 721 const sp<IMemory>& sharedBuffer, 722 int sessionId, 723 IAudioFlinger::track_flags_t flags); 724 virtual ~Track(); 725 726 static void appendDumpHeader(String8& result); 727 void dump(char* buffer, size_t size); 728 virtual status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 729 int triggerSession = 0); 730 virtual void stop(); 731 void pause(); 732 733 void flush(); 734 void destroy(); 735 void mute(bool); 736 int name() const { return mName; } 737 738 audio_stream_type_t streamType() const { 739 return mStreamType; 740 } 741 status_t attachAuxEffect(int EffectId); 742 void setAuxBuffer(int EffectId, int32_t *buffer); 743 int32_t *auxBuffer() const { return mAuxBuffer; } 744 void setMainBuffer(int16_t *buffer) { mMainBuffer = buffer; } 745 int16_t *mainBuffer() const { return mMainBuffer; } 746 int auxEffectId() const { return mAuxEffectId; } 747 748 // implement FastMixerState::VolumeProvider interface 749 virtual uint32_t getVolumeLR(); 750 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 751 752 protected: 753 // for numerous 754 friend class PlaybackThread; 755 friend class MixerThread; 756 friend class DirectOutputThread; 757 758 Track(const Track&); 759 Track& operator = (const Track&); 760 761 // AudioBufferProvider interface 762 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts = kInvalidPTS); 763 // releaseBuffer() not overridden 764 765 virtual size_t framesReady() const; 766 767 bool isMuted() const { return mMute; } 768 bool isPausing() const { 769 return mState == PAUSING; 770 } 771 bool isPaused() const { 772 return mState == PAUSED; 773 } 774 bool isResuming() const { 775 return mState == RESUMING; 776 } 777 bool isReady() const; 778 void setPaused() { mState = PAUSED; } 779 void reset(); 780 781 bool isOutputTrack() const { 782 return (mStreamType == AUDIO_STREAM_CNT); 783 } 784 785 sp<IMemory> sharedBuffer() const { return mSharedBuffer; } 786 787 bool presentationComplete(size_t framesWritten, size_t audioHalFrames); 788 789 public: 790 void triggerEvents(AudioSystem::sync_event_t type); 791 virtual bool isTimedTrack() const { return false; } 792 bool isFastTrack() const { return (mFlags & IAudioFlinger::TRACK_FAST) != 0; } 793 794 protected: 795 796 // written by Track::mute() called by binder thread(s), without a mutex or barrier. 797 // read by Track::isMuted() called by playback thread, also without a mutex or barrier. 798 // The lack of mutex or barrier is safe because the mute status is only used by itself. 799 bool mMute; 800 801 // FILLED state is used for suppressing volume ramp at begin of playing 802 enum {FS_INVALID, FS_FILLING, FS_FILLED, FS_ACTIVE}; 803 mutable uint8_t mFillingUpStatus; 804 int8_t mRetryCount; 805 const sp<IMemory> mSharedBuffer; 806 bool mResetDone; 807 const audio_stream_type_t mStreamType; 808 int mName; // track name on the normal mixer, 809 // allocated statically at track creation time, 810 // and is even allocated (though unused) for fast tracks 811 // FIXME don't allocate track name for fast tracks 812 int16_t *mMainBuffer; 813 int32_t *mAuxBuffer; 814 int mAuxEffectId; 815 bool mHasVolumeController; 816 size_t mPresentationCompleteFrames; // number of frames written to the audio HAL 817 // when this track will be fully rendered 818 private: 819 IAudioFlinger::track_flags_t mFlags; 820 821 // The following fields are only for fast tracks, and should be in a subclass 822 int mFastIndex; // index within FastMixerState::mFastTracks[]; 823 // either mFastIndex == -1 if not isFastTrack() 824 // or 0 < mFastIndex < FastMixerState::kMaxFast because 825 // index 0 is reserved for normal mixer's submix; 826 // index is allocated statically at track creation time 827 // but the slot is only used if track is active 828 FastTrackUnderruns mObservedUnderruns; // Most recently observed value of 829 // mFastMixerDumpState.mTracks[mFastIndex].mUnderruns 830 uint32_t mUnderrunCount; // Counter of total number of underruns, never reset 831 volatile float mCachedVolume; // combined master volume and stream type volume; 832 // 'volatile' means accessed without lock or 833 // barrier, but is read/written atomically 834 }; // end of Track 835 836 class TimedTrack : public Track { 837 public: 838 static sp<TimedTrack> create(PlaybackThread *thread, 839 const sp<Client>& client, 840 audio_stream_type_t streamType, 841 uint32_t sampleRate, 842 audio_format_t format, 843 audio_channel_mask_t channelMask, 844 int frameCount, 845 const sp<IMemory>& sharedBuffer, 846 int sessionId); 847 virtual ~TimedTrack(); 848 849 class TimedBuffer { 850 public: 851 TimedBuffer(); 852 TimedBuffer(const sp<IMemory>& buffer, int64_t pts); 853 const sp<IMemory>& buffer() const { return mBuffer; } 854 int64_t pts() const { return mPTS; } 855 uint32_t position() const { return mPosition; } 856 void setPosition(uint32_t pos) { mPosition = pos; } 857 private: 858 sp<IMemory> mBuffer; 859 int64_t mPTS; 860 uint32_t mPosition; 861 }; 862 863 // Mixer facing methods. 864 virtual bool isTimedTrack() const { return true; } 865 virtual size_t framesReady() const; 866 867 // AudioBufferProvider interface 868 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, 869 int64_t pts); 870 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 871 872 // Client/App facing methods. 873 status_t allocateTimedBuffer(size_t size, 874 sp<IMemory>* buffer); 875 status_t queueTimedBuffer(const sp<IMemory>& buffer, 876 int64_t pts); 877 status_t setMediaTimeTransform(const LinearTransform& xform, 878 TimedAudioTrack::TargetTimeline target); 879 880 private: 881 TimedTrack(PlaybackThread *thread, 882 const sp<Client>& client, 883 audio_stream_type_t streamType, 884 uint32_t sampleRate, 885 audio_format_t format, 886 audio_channel_mask_t channelMask, 887 int frameCount, 888 const sp<IMemory>& sharedBuffer, 889 int sessionId); 890 891 void timedYieldSamples_l(AudioBufferProvider::Buffer* buffer); 892 void timedYieldSilence_l(uint32_t numFrames, 893 AudioBufferProvider::Buffer* buffer); 894 void trimTimedBufferQueue_l(); 895 void trimTimedBufferQueueHead_l(const char* logTag); 896 void updateFramesPendingAfterTrim_l(const TimedBuffer& buf, 897 const char* logTag); 898 899 uint64_t mLocalTimeFreq; 900 LinearTransform mLocalTimeToSampleTransform; 901 LinearTransform mMediaTimeToSampleTransform; 902 sp<MemoryDealer> mTimedMemoryDealer; 903 904 Vector<TimedBuffer> mTimedBufferQueue; 905 bool mQueueHeadInFlight; 906 bool mTrimQueueHeadOnRelease; 907 uint32_t mFramesPendingInQueue; 908 909 uint8_t* mTimedSilenceBuffer; 910 uint32_t mTimedSilenceBufferSize; 911 mutable Mutex mTimedBufferQueueLock; 912 bool mTimedAudioOutputOnTime; 913 CCHelper mCCHelper; 914 915 Mutex mMediaTimeTransformLock; 916 LinearTransform mMediaTimeTransform; 917 bool mMediaTimeTransformValid; 918 TimedAudioTrack::TargetTimeline mMediaTimeTransformTarget; 919 }; 920 921 922 // playback track 923 class OutputTrack : public Track { 924 public: 925 926 class Buffer: public AudioBufferProvider::Buffer { 927 public: 928 int16_t *mBuffer; 929 }; 930 931 OutputTrack(PlaybackThread *thread, 932 DuplicatingThread *sourceThread, 933 uint32_t sampleRate, 934 audio_format_t format, 935 audio_channel_mask_t channelMask, 936 int frameCount); 937 virtual ~OutputTrack(); 938 939 virtual status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 940 int triggerSession = 0); 941 virtual void stop(); 942 bool write(int16_t* data, uint32_t frames); 943 bool bufferQueueEmpty() const { return mBufferQueue.size() == 0; } 944 bool isActive() const { return mActive; } 945 const wp<ThreadBase>& thread() const { return mThread; } 946 947 private: 948 949 enum { 950 NO_MORE_BUFFERS = 0x80000001, // same in AudioTrack.h, ok to be different value 951 }; 952 953 status_t obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs); 954 void clearBufferQueue(); 955 956 // Maximum number of pending buffers allocated by OutputTrack::write() 957 static const uint8_t kMaxOverFlowBuffers = 10; 958 959 Vector < Buffer* > mBufferQueue; 960 AudioBufferProvider::Buffer mOutBuffer; 961 bool mActive; 962 DuplicatingThread* const mSourceThread; // for waitTimeMs() in write() 963 }; // end of OutputTrack 964 965 PlaybackThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 966 audio_io_handle_t id, audio_devices_t device, type_t type); 967 virtual ~PlaybackThread(); 968 969 void dump(int fd, const Vector<String16>& args); 970 971 // Thread virtuals 972 virtual status_t readyToRun(); 973 virtual bool threadLoop(); 974 975 // RefBase 976 virtual void onFirstRef(); 977 978protected: 979 // Code snippets that were lifted up out of threadLoop() 980 virtual void threadLoop_mix() = 0; 981 virtual void threadLoop_sleepTime() = 0; 982 virtual void threadLoop_write(); 983 virtual void threadLoop_standby(); 984 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 985 986 // prepareTracks_l reads and writes mActiveTracks, and returns 987 // the pending set of tracks to remove via Vector 'tracksToRemove'. The caller 988 // is responsible for clearing or destroying this Vector later on, when it 989 // is safe to do so. That will drop the final ref count and destroy the tracks. 990 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove) = 0; 991 992public: 993 994 virtual status_t initCheck() const { return (mOutput == NULL) ? NO_INIT : NO_ERROR; } 995 996 // return estimated latency in milliseconds, as reported by HAL 997 uint32_t latency() const; 998 // same, but lock must already be held 999 uint32_t latency_l() const; 1000 1001 void setMasterVolume(float value); 1002 void setMasterMute(bool muted); 1003 1004 void setStreamVolume(audio_stream_type_t stream, float value); 1005 void setStreamMute(audio_stream_type_t stream, bool muted); 1006 1007 float streamVolume(audio_stream_type_t stream) const; 1008 1009 sp<Track> createTrack_l( 1010 const sp<AudioFlinger::Client>& client, 1011 audio_stream_type_t streamType, 1012 uint32_t sampleRate, 1013 audio_format_t format, 1014 audio_channel_mask_t channelMask, 1015 int frameCount, 1016 const sp<IMemory>& sharedBuffer, 1017 int sessionId, 1018 IAudioFlinger::track_flags_t flags, 1019 pid_t tid, 1020 status_t *status); 1021 1022 AudioStreamOut* getOutput() const; 1023 AudioStreamOut* clearOutput(); 1024 virtual audio_stream_t* stream() const; 1025 1026 // a very large number of suspend() will eventually wraparound, but unlikely 1027 void suspend() { (void) android_atomic_inc(&mSuspended); } 1028 void restore() 1029 { 1030 // if restore() is done without suspend(), get back into 1031 // range so that the next suspend() will operate correctly 1032 if (android_atomic_dec(&mSuspended) <= 0) { 1033 android_atomic_release_store(0, &mSuspended); 1034 } 1035 } 1036 bool isSuspended() const 1037 { return android_atomic_acquire_load(&mSuspended) > 0; } 1038 1039 virtual String8 getParameters(const String8& keys); 1040 virtual void audioConfigChanged_l(int event, int param = 0); 1041 status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames); 1042 int16_t *mixBuffer() const { return mMixBuffer; }; 1043 1044 virtual void detachAuxEffect_l(int effectId); 1045 status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track> track, 1046 int EffectId); 1047 status_t attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track> track, 1048 int EffectId); 1049 1050 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1051 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1052 virtual uint32_t hasAudioSession(int sessionId); 1053 virtual uint32_t getStrategyForSession_l(int sessionId); 1054 1055 1056 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 1057 virtual bool isValidSyncEvent(const sp<SyncEvent>& event); 1058 void invalidateTracks(audio_stream_type_t streamType); 1059 1060 1061 protected: 1062 int16_t* mMixBuffer; 1063 1064 // suspend count, > 0 means suspended. While suspended, the thread continues to pull from 1065 // tracks and mix, but doesn't write to HAL. A2DP and SCO HAL implementations can't handle 1066 // concurrent use of both of them, so Audio Policy Service suspends one of the threads to 1067 // workaround that restriction. 1068 // 'volatile' means accessed via atomic operations and no lock. 1069 volatile int32_t mSuspended; 1070 1071 int mBytesWritten; 1072 private: 1073 // mMasterMute is in both PlaybackThread and in AudioFlinger. When a 1074 // PlaybackThread needs to find out if master-muted, it checks it's local 1075 // copy rather than the one in AudioFlinger. This optimization saves a lock. 1076 bool mMasterMute; 1077 void setMasterMute_l(bool muted) { mMasterMute = muted; } 1078 protected: 1079 SortedVector< wp<Track> > mActiveTracks; // FIXME check if this could be sp<> 1080 1081 // Allocate a track name for a given channel mask. 1082 // Returns name >= 0 if successful, -1 on failure. 1083 virtual int getTrackName_l(audio_channel_mask_t channelMask) = 0; 1084 virtual void deleteTrackName_l(int name) = 0; 1085 1086 // Time to sleep between cycles when: 1087 virtual uint32_t activeSleepTimeUs() const; // mixer state MIXER_TRACKS_ENABLED 1088 virtual uint32_t idleSleepTimeUs() const = 0; // mixer state MIXER_IDLE 1089 virtual uint32_t suspendSleepTimeUs() const = 0; // audio policy manager suspended us 1090 // No sleep when mixer state == MIXER_TRACKS_READY; relies on audio HAL stream->write() 1091 // No sleep in standby mode; waits on a condition 1092 1093 // Code snippets that are temporarily lifted up out of threadLoop() until the merge 1094 void checkSilentMode_l(); 1095 1096 // Non-trivial for DUPLICATING only 1097 virtual void saveOutputTracks() { } 1098 virtual void clearOutputTracks() { } 1099 1100 // Cache various calculated values, at threadLoop() entry and after a parameter change 1101 virtual void cacheParameters_l(); 1102 1103 virtual uint32_t correctLatency(uint32_t latency) const; 1104 1105 private: 1106 1107 friend class AudioFlinger; // for numerous 1108 1109 PlaybackThread(const Client&); 1110 PlaybackThread& operator = (const PlaybackThread&); 1111 1112 status_t addTrack_l(const sp<Track>& track); 1113 void destroyTrack_l(const sp<Track>& track); 1114 void removeTrack_l(const sp<Track>& track); 1115 1116 void readOutputParameters(); 1117 1118 virtual void dumpInternals(int fd, const Vector<String16>& args); 1119 void dumpTracks(int fd, const Vector<String16>& args); 1120 1121 SortedVector< sp<Track> > mTracks; 1122 // mStreamTypes[] uses 1 additional stream type internally for the OutputTrack used by DuplicatingThread 1123 stream_type_t mStreamTypes[AUDIO_STREAM_CNT + 1]; 1124 AudioStreamOut *mOutput; 1125 1126 float mMasterVolume; 1127 nsecs_t mLastWriteTime; 1128 int mNumWrites; 1129 int mNumDelayedWrites; 1130 bool mInWrite; 1131 1132 // FIXME rename these former local variables of threadLoop to standard "m" names 1133 nsecs_t standbyTime; 1134 size_t mixBufferSize; 1135 1136 // cached copies of activeSleepTimeUs() and idleSleepTimeUs() made by cacheParameters_l() 1137 uint32_t activeSleepTime; 1138 uint32_t idleSleepTime; 1139 1140 uint32_t sleepTime; 1141 1142 // mixer status returned by prepareTracks_l() 1143 mixer_state mMixerStatus; // current cycle 1144 // previous cycle when in prepareTracks_l() 1145 mixer_state mMixerStatusIgnoringFastTracks; 1146 // FIXME or a separate ready state per track 1147 1148 // FIXME move these declarations into the specific sub-class that needs them 1149 // MIXER only 1150 uint32_t sleepTimeShift; 1151 1152 // same as AudioFlinger::mStandbyTimeInNsecs except for DIRECT which uses a shorter value 1153 nsecs_t standbyDelay; 1154 1155 // MIXER only 1156 nsecs_t maxPeriod; 1157 1158 // DUPLICATING only 1159 uint32_t writeFrames; 1160 1161 private: 1162 // The HAL output sink is treated as non-blocking, but current implementation is blocking 1163 sp<NBAIO_Sink> mOutputSink; 1164 // If a fast mixer is present, the blocking pipe sink, otherwise clear 1165 sp<NBAIO_Sink> mPipeSink; 1166 // The current sink for the normal mixer to write it's (sub)mix, mOutputSink or mPipeSink 1167 sp<NBAIO_Sink> mNormalSink; 1168 // For dumpsys 1169 sp<NBAIO_Sink> mTeeSink; 1170 sp<NBAIO_Source> mTeeSource; 1171 uint32_t mScreenState; // cached copy of gScreenState 1172 public: 1173 virtual bool hasFastMixer() const = 0; 1174 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const 1175 { FastTrackUnderruns dummy; return dummy; } 1176 1177 protected: 1178 // accessed by both binder threads and within threadLoop(), lock on mutex needed 1179 unsigned mFastTrackAvailMask; // bit i set if fast track [i] is available 1180 1181 }; 1182 1183 class MixerThread : public PlaybackThread { 1184 public: 1185 MixerThread (const sp<AudioFlinger>& audioFlinger, 1186 AudioStreamOut* output, 1187 audio_io_handle_t id, 1188 audio_devices_t device, 1189 type_t type = MIXER); 1190 virtual ~MixerThread(); 1191 1192 // Thread virtuals 1193 1194 virtual bool checkForNewParameters_l(); 1195 virtual void dumpInternals(int fd, const Vector<String16>& args); 1196 1197 protected: 1198 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1199 virtual int getTrackName_l(audio_channel_mask_t channelMask); 1200 virtual void deleteTrackName_l(int name); 1201 virtual uint32_t idleSleepTimeUs() const; 1202 virtual uint32_t suspendSleepTimeUs() const; 1203 virtual void cacheParameters_l(); 1204 1205 // threadLoop snippets 1206 virtual void threadLoop_write(); 1207 virtual void threadLoop_standby(); 1208 virtual void threadLoop_mix(); 1209 virtual void threadLoop_sleepTime(); 1210 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 1211 virtual uint32_t correctLatency(uint32_t latency) const; 1212 1213 AudioMixer* mAudioMixer; // normal mixer 1214 private: 1215 // one-time initialization, no locks required 1216 FastMixer* mFastMixer; // non-NULL if there is also a fast mixer 1217 sp<AudioWatchdog> mAudioWatchdog; // non-0 if there is an audio watchdog thread 1218 1219 // contents are not guaranteed to be consistent, no locks required 1220 FastMixerDumpState mFastMixerDumpState; 1221#ifdef STATE_QUEUE_DUMP 1222 StateQueueObserverDump mStateQueueObserverDump; 1223 StateQueueMutatorDump mStateQueueMutatorDump; 1224#endif 1225 AudioWatchdogDump mAudioWatchdogDump; 1226 1227 // accessible only within the threadLoop(), no locks required 1228 // mFastMixer->sq() // for mutating and pushing state 1229 int32_t mFastMixerFutex; // for cold idle 1230 1231 public: 1232 virtual bool hasFastMixer() const { return mFastMixer != NULL; } 1233 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const { 1234 ALOG_ASSERT(fastIndex < FastMixerState::kMaxFastTracks); 1235 return mFastMixerDumpState.mTracks[fastIndex].mUnderruns; 1236 } 1237 }; 1238 1239 class DirectOutputThread : public PlaybackThread { 1240 public: 1241 1242 DirectOutputThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1243 audio_io_handle_t id, audio_devices_t device); 1244 virtual ~DirectOutputThread(); 1245 1246 // Thread virtuals 1247 1248 virtual bool checkForNewParameters_l(); 1249 1250 protected: 1251 virtual int getTrackName_l(audio_channel_mask_t channelMask); 1252 virtual void deleteTrackName_l(int name); 1253 virtual uint32_t activeSleepTimeUs() const; 1254 virtual uint32_t idleSleepTimeUs() const; 1255 virtual uint32_t suspendSleepTimeUs() const; 1256 virtual void cacheParameters_l(); 1257 1258 // threadLoop snippets 1259 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1260 virtual void threadLoop_mix(); 1261 virtual void threadLoop_sleepTime(); 1262 1263 // volumes last sent to audio HAL with stream->set_volume() 1264 float mLeftVolFloat; 1265 float mRightVolFloat; 1266 1267private: 1268 // prepareTracks_l() tells threadLoop_mix() the name of the single active track 1269 sp<Track> mActiveTrack; 1270 public: 1271 virtual bool hasFastMixer() const { return false; } 1272 }; 1273 1274 class DuplicatingThread : public MixerThread { 1275 public: 1276 DuplicatingThread (const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread, 1277 audio_io_handle_t id); 1278 virtual ~DuplicatingThread(); 1279 1280 // Thread virtuals 1281 void addOutputTrack(MixerThread* thread); 1282 void removeOutputTrack(MixerThread* thread); 1283 uint32_t waitTimeMs() const { return mWaitTimeMs; } 1284 protected: 1285 virtual uint32_t activeSleepTimeUs() const; 1286 1287 private: 1288 bool outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks); 1289 protected: 1290 // threadLoop snippets 1291 virtual void threadLoop_mix(); 1292 virtual void threadLoop_sleepTime(); 1293 virtual void threadLoop_write(); 1294 virtual void threadLoop_standby(); 1295 virtual void cacheParameters_l(); 1296 1297 private: 1298 // called from threadLoop, addOutputTrack, removeOutputTrack 1299 virtual void updateWaitTime_l(); 1300 protected: 1301 virtual void saveOutputTracks(); 1302 virtual void clearOutputTracks(); 1303 private: 1304 1305 uint32_t mWaitTimeMs; 1306 SortedVector < sp<OutputTrack> > outputTracks; 1307 SortedVector < sp<OutputTrack> > mOutputTracks; 1308 public: 1309 virtual bool hasFastMixer() const { return false; } 1310 }; 1311 1312 PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const; 1313 MixerThread *checkMixerThread_l(audio_io_handle_t output) const; 1314 RecordThread *checkRecordThread_l(audio_io_handle_t input) const; 1315 // no range check, AudioFlinger::mLock held 1316 bool streamMute_l(audio_stream_type_t stream) const 1317 { return mStreamTypes[stream].mute; } 1318 // no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held 1319 float streamVolume_l(audio_stream_type_t stream) const 1320 { return mStreamTypes[stream].volume; } 1321 void audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2); 1322 1323 // allocate an audio_io_handle_t, session ID, or effect ID 1324 uint32_t nextUniqueId(); 1325 1326 status_t moveEffectChain_l(int sessionId, 1327 PlaybackThread *srcThread, 1328 PlaybackThread *dstThread, 1329 bool reRegister); 1330 // return thread associated with primary hardware device, or NULL 1331 PlaybackThread *primaryPlaybackThread_l() const; 1332 audio_devices_t primaryOutputDevice_l() const; 1333 1334 sp<PlaybackThread> getEffectThread_l(int sessionId, int EffectId); 1335 1336 // server side of the client's IAudioTrack 1337 class TrackHandle : public android::BnAudioTrack { 1338 public: 1339 TrackHandle(const sp<PlaybackThread::Track>& track); 1340 virtual ~TrackHandle(); 1341 virtual sp<IMemory> getCblk() const; 1342 virtual status_t start(); 1343 virtual void stop(); 1344 virtual void flush(); 1345 virtual void mute(bool); 1346 virtual void pause(); 1347 virtual status_t attachAuxEffect(int effectId); 1348 virtual status_t allocateTimedBuffer(size_t size, 1349 sp<IMemory>* buffer); 1350 virtual status_t queueTimedBuffer(const sp<IMemory>& buffer, 1351 int64_t pts); 1352 virtual status_t setMediaTimeTransform(const LinearTransform& xform, 1353 int target); 1354 virtual status_t onTransact( 1355 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 1356 private: 1357 const sp<PlaybackThread::Track> mTrack; 1358 }; 1359 1360 void removeClient_l(pid_t pid); 1361 void removeNotificationClient(pid_t pid); 1362 1363 1364 // record thread 1365 class RecordThread : public ThreadBase, public AudioBufferProvider 1366 { 1367 public: 1368 1369 // record track 1370 class RecordTrack : public TrackBase { 1371 public: 1372 RecordTrack(RecordThread *thread, 1373 const sp<Client>& client, 1374 uint32_t sampleRate, 1375 audio_format_t format, 1376 audio_channel_mask_t channelMask, 1377 int frameCount, 1378 int sessionId); 1379 virtual ~RecordTrack(); 1380 1381 virtual status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 1382 int triggerSession = 0); 1383 virtual void stop(); 1384 1385 // clear the buffer overflow flag 1386 void clearOverflow() { mOverflow = false; } 1387 // set the buffer overflow flag and return previous value 1388 bool setOverflow() { bool tmp = mOverflow; mOverflow = true; return tmp; } 1389 1390 void dump(char* buffer, size_t size); 1391 1392 private: 1393 friend class AudioFlinger; // for mState 1394 1395 RecordTrack(const RecordTrack&); 1396 RecordTrack& operator = (const RecordTrack&); 1397 1398 // AudioBufferProvider interface 1399 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts = kInvalidPTS); 1400 // releaseBuffer() not overridden 1401 1402 bool mOverflow; // overflow on most recent attempt to fill client buffer 1403 }; 1404 1405 RecordThread(const sp<AudioFlinger>& audioFlinger, 1406 AudioStreamIn *input, 1407 uint32_t sampleRate, 1408 audio_channel_mask_t channelMask, 1409 audio_io_handle_t id, 1410 audio_devices_t device); 1411 virtual ~RecordThread(); 1412 1413 // Thread 1414 virtual bool threadLoop(); 1415 virtual status_t readyToRun(); 1416 1417 // RefBase 1418 virtual void onFirstRef(); 1419 1420 virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; } 1421 sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l( 1422 const sp<AudioFlinger::Client>& client, 1423 uint32_t sampleRate, 1424 audio_format_t format, 1425 audio_channel_mask_t channelMask, 1426 int frameCount, 1427 int sessionId, 1428 IAudioFlinger::track_flags_t flags, 1429 pid_t tid, 1430 status_t *status); 1431 1432 status_t start(RecordTrack* recordTrack, 1433 AudioSystem::sync_event_t event, 1434 int triggerSession); 1435 1436 // ask the thread to stop the specified track, and 1437 // return true if the caller should then do it's part of the stopping process 1438 bool stop_l(RecordTrack* recordTrack); 1439 1440 void dump(int fd, const Vector<String16>& args); 1441 AudioStreamIn* clearInput(); 1442 virtual audio_stream_t* stream() const; 1443 1444 // AudioBufferProvider interface 1445 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts); 1446 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 1447 1448 virtual bool checkForNewParameters_l(); 1449 virtual String8 getParameters(const String8& keys); 1450 virtual void audioConfigChanged_l(int event, int param = 0); 1451 void readInputParameters(); 1452 virtual unsigned int getInputFramesLost(); 1453 1454 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1455 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1456 virtual uint32_t hasAudioSession(int sessionId); 1457 RecordTrack* track(); 1458 1459 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 1460 virtual bool isValidSyncEvent(const sp<SyncEvent>& event); 1461 1462 static void syncStartEventCallback(const wp<SyncEvent>& event); 1463 void handleSyncStartEvent(const sp<SyncEvent>& event); 1464 1465 private: 1466 void clearSyncStartEvent(); 1467 1468 AudioStreamIn *mInput; 1469 RecordTrack* mTrack; 1470 sp<RecordTrack> mActiveTrack; 1471 Condition mStartStopCond; 1472 AudioResampler *mResampler; 1473 int32_t *mRsmpOutBuffer; 1474 int16_t *mRsmpInBuffer; 1475 size_t mRsmpInIndex; 1476 size_t mInputBytes; 1477 const int mReqChannelCount; 1478 const uint32_t mReqSampleRate; 1479 ssize_t mBytesRead; 1480 // sync event triggering actual audio capture. Frames read before this event will 1481 // be dropped and therefore not read by the application. 1482 sp<SyncEvent> mSyncStartEvent; 1483 // number of captured frames to drop after the start sync event has been received. 1484 // when < 0, maximum frames to drop before starting capture even if sync event is 1485 // not received 1486 ssize_t mFramestoDrop; 1487 }; 1488 1489 // server side of the client's IAudioRecord 1490 class RecordHandle : public android::BnAudioRecord { 1491 public: 1492 RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack); 1493 virtual ~RecordHandle(); 1494 virtual sp<IMemory> getCblk() const; 1495 virtual status_t start(int event, int triggerSession); 1496 virtual void stop(); 1497 virtual status_t onTransact( 1498 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 1499 private: 1500 const sp<RecordThread::RecordTrack> mRecordTrack; 1501 1502 // for use from destructor 1503 void stop_nonvirtual(); 1504 }; 1505 1506 //--- Audio Effect Management 1507 1508 // EffectModule and EffectChain classes both have their own mutex to protect 1509 // state changes or resource modifications. Always respect the following order 1510 // if multiple mutexes must be acquired to avoid cross deadlock: 1511 // AudioFlinger -> ThreadBase -> EffectChain -> EffectModule 1512 1513 // The EffectModule class is a wrapper object controlling the effect engine implementation 1514 // in the effect library. It prevents concurrent calls to process() and command() functions 1515 // from different client threads. It keeps a list of EffectHandle objects corresponding 1516 // to all client applications using this effect and notifies applications of effect state, 1517 // control or parameter changes. It manages the activation state machine to send appropriate 1518 // reset, enable, disable commands to effect engine and provide volume 1519 // ramping when effects are activated/deactivated. 1520 // When controlling an auxiliary effect, the EffectModule also provides an input buffer used by 1521 // the attached track(s) to accumulate their auxiliary channel. 1522 class EffectModule: public RefBase { 1523 public: 1524 EffectModule(ThreadBase *thread, 1525 const wp<AudioFlinger::EffectChain>& chain, 1526 effect_descriptor_t *desc, 1527 int id, 1528 int sessionId); 1529 virtual ~EffectModule(); 1530 1531 enum effect_state { 1532 IDLE, 1533 RESTART, 1534 STARTING, 1535 ACTIVE, 1536 STOPPING, 1537 STOPPED, 1538 DESTROYED 1539 }; 1540 1541 int id() const { return mId; } 1542 void process(); 1543 void updateState(); 1544 status_t command(uint32_t cmdCode, 1545 uint32_t cmdSize, 1546 void *pCmdData, 1547 uint32_t *replySize, 1548 void *pReplyData); 1549 1550 void reset_l(); 1551 status_t configure(); 1552 status_t init(); 1553 effect_state state() const { 1554 return mState; 1555 } 1556 uint32_t status() { 1557 return mStatus; 1558 } 1559 int sessionId() const { 1560 return mSessionId; 1561 } 1562 status_t setEnabled(bool enabled); 1563 status_t setEnabled_l(bool enabled); 1564 bool isEnabled() const; 1565 bool isProcessEnabled() const; 1566 1567 void setInBuffer(int16_t *buffer) { mConfig.inputCfg.buffer.s16 = buffer; } 1568 int16_t *inBuffer() { return mConfig.inputCfg.buffer.s16; } 1569 void setOutBuffer(int16_t *buffer) { mConfig.outputCfg.buffer.s16 = buffer; } 1570 int16_t *outBuffer() { return mConfig.outputCfg.buffer.s16; } 1571 void setChain(const wp<EffectChain>& chain) { mChain = chain; } 1572 void setThread(const wp<ThreadBase>& thread) { mThread = thread; } 1573 const wp<ThreadBase>& thread() { return mThread; } 1574 1575 status_t addHandle(EffectHandle *handle); 1576 size_t disconnect(EffectHandle *handle, bool unpinIfLast); 1577 size_t removeHandle(EffectHandle *handle); 1578 1579 effect_descriptor_t& desc() { return mDescriptor; } 1580 wp<EffectChain>& chain() { return mChain; } 1581 1582 status_t setDevice(audio_devices_t device); 1583 status_t setVolume(uint32_t *left, uint32_t *right, bool controller); 1584 status_t setMode(audio_mode_t mode); 1585 status_t start(); 1586 status_t stop(); 1587 void setSuspended(bool suspended); 1588 bool suspended() const; 1589 1590 EffectHandle* controlHandle_l(); 1591 1592 bool isPinned() const { return mPinned; } 1593 void unPin() { mPinned = false; } 1594 bool purgeHandles(); 1595 void lock() { mLock.lock(); } 1596 void unlock() { mLock.unlock(); } 1597 1598 void dump(int fd, const Vector<String16>& args); 1599 1600 protected: 1601 friend class AudioFlinger; // for mHandles 1602 bool mPinned; 1603 1604 // Maximum time allocated to effect engines to complete the turn off sequence 1605 static const uint32_t MAX_DISABLE_TIME_MS = 10000; 1606 1607 EffectModule(const EffectModule&); 1608 EffectModule& operator = (const EffectModule&); 1609 1610 status_t start_l(); 1611 status_t stop_l(); 1612 1613mutable Mutex mLock; // mutex for process, commands and handles list protection 1614 wp<ThreadBase> mThread; // parent thread 1615 wp<EffectChain> mChain; // parent effect chain 1616 const int mId; // this instance unique ID 1617 const int mSessionId; // audio session ID 1618 effect_descriptor_t mDescriptor;// effect descriptor received from effect engine 1619 effect_config_t mConfig; // input and output audio configuration 1620 effect_handle_t mEffectInterface; // Effect module C API 1621 status_t mStatus; // initialization status 1622 effect_state mState; // current activation state 1623 Vector<EffectHandle *> mHandles; // list of client handles 1624 // First handle in mHandles has highest priority and controls the effect module 1625 uint32_t mMaxDisableWaitCnt; // maximum grace period before forcing an effect off after 1626 // sending disable command. 1627 uint32_t mDisableWaitCnt; // current process() calls count during disable period. 1628 bool mSuspended; // effect is suspended: temporarily disabled by framework 1629 }; 1630 1631 // The EffectHandle class implements the IEffect interface. It provides resources 1632 // to receive parameter updates, keeps track of effect control 1633 // ownership and state and has a pointer to the EffectModule object it is controlling. 1634 // There is one EffectHandle object for each application controlling (or using) 1635 // an effect module. 1636 // The EffectHandle is obtained by calling AudioFlinger::createEffect(). 1637 class EffectHandle: public android::BnEffect { 1638 public: 1639 1640 EffectHandle(const sp<EffectModule>& effect, 1641 const sp<AudioFlinger::Client>& client, 1642 const sp<IEffectClient>& effectClient, 1643 int32_t priority); 1644 virtual ~EffectHandle(); 1645 1646 // IEffect 1647 virtual status_t enable(); 1648 virtual status_t disable(); 1649 virtual status_t command(uint32_t cmdCode, 1650 uint32_t cmdSize, 1651 void *pCmdData, 1652 uint32_t *replySize, 1653 void *pReplyData); 1654 virtual void disconnect(); 1655 private: 1656 void disconnect(bool unpinIfLast); 1657 public: 1658 virtual sp<IMemory> getCblk() const { return mCblkMemory; } 1659 virtual status_t onTransact(uint32_t code, const Parcel& data, 1660 Parcel* reply, uint32_t flags); 1661 1662 1663 // Give or take control of effect module 1664 // - hasControl: true if control is given, false if removed 1665 // - signal: true client app should be signaled of change, false otherwise 1666 // - enabled: state of the effect when control is passed 1667 void setControl(bool hasControl, bool signal, bool enabled); 1668 void commandExecuted(uint32_t cmdCode, 1669 uint32_t cmdSize, 1670 void *pCmdData, 1671 uint32_t replySize, 1672 void *pReplyData); 1673 void setEnabled(bool enabled); 1674 bool enabled() const { return mEnabled; } 1675 1676 // Getters 1677 int id() const { return mEffect->id(); } 1678 int priority() const { return mPriority; } 1679 bool hasControl() const { return mHasControl; } 1680 sp<EffectModule> effect() const { return mEffect; } 1681 // destroyed_l() must be called with the associated EffectModule mLock held 1682 bool destroyed_l() const { return mDestroyed; } 1683 1684 void dump(char* buffer, size_t size); 1685 1686 protected: 1687 friend class AudioFlinger; // for mEffect, mHasControl, mEnabled 1688 EffectHandle(const EffectHandle&); 1689 EffectHandle& operator =(const EffectHandle&); 1690 1691 sp<EffectModule> mEffect; // pointer to controlled EffectModule 1692 sp<IEffectClient> mEffectClient; // callback interface for client notifications 1693 /*const*/ sp<Client> mClient; // client for shared memory allocation, see disconnect() 1694 sp<IMemory> mCblkMemory; // shared memory for control block 1695 effect_param_cblk_t* mCblk; // control block for deferred parameter setting via shared memory 1696 uint8_t* mBuffer; // pointer to parameter area in shared memory 1697 int mPriority; // client application priority to control the effect 1698 bool mHasControl; // true if this handle is controlling the effect 1699 bool mEnabled; // cached enable state: needed when the effect is 1700 // restored after being suspended 1701 bool mDestroyed; // Set to true by destructor. Access with EffectModule 1702 // mLock held 1703 }; 1704 1705 // the EffectChain class represents a group of effects associated to one audio session. 1706 // There can be any number of EffectChain objects per output mixer thread (PlaybackThread). 1707 // The EffecChain with session ID 0 contains global effects applied to the output mix. 1708 // Effects in this chain can be insert or auxiliary. Effects in other chains (attached to tracks) 1709 // are insert only. The EffectChain maintains an ordered list of effect module, the order corresponding 1710 // in the effect process order. When attached to a track (session ID != 0), it also provide it's own 1711 // input buffer used by the track as accumulation buffer. 1712 class EffectChain: public RefBase { 1713 public: 1714 EffectChain(const wp<ThreadBase>& wThread, int sessionId); 1715 EffectChain(ThreadBase *thread, int sessionId); 1716 virtual ~EffectChain(); 1717 1718 // special key used for an entry in mSuspendedEffects keyed vector 1719 // corresponding to a suspend all request. 1720 static const int kKeyForSuspendAll = 0; 1721 1722 // minimum duration during which we force calling effect process when last track on 1723 // a session is stopped or removed to allow effect tail to be rendered 1724 static const int kProcessTailDurationMs = 1000; 1725 1726 void process_l(); 1727 1728 void lock() { 1729 mLock.lock(); 1730 } 1731 void unlock() { 1732 mLock.unlock(); 1733 } 1734 1735 status_t addEffect_l(const sp<EffectModule>& handle); 1736 size_t removeEffect_l(const sp<EffectModule>& handle); 1737 1738 int sessionId() const { return mSessionId; } 1739 void setSessionId(int sessionId) { mSessionId = sessionId; } 1740 1741 sp<EffectModule> getEffectFromDesc_l(effect_descriptor_t *descriptor); 1742 sp<EffectModule> getEffectFromId_l(int id); 1743 sp<EffectModule> getEffectFromType_l(const effect_uuid_t *type); 1744 bool setVolume_l(uint32_t *left, uint32_t *right); 1745 void setDevice_l(audio_devices_t device); 1746 void setMode_l(audio_mode_t mode); 1747 1748 void setInBuffer(int16_t *buffer, bool ownsBuffer = false) { 1749 mInBuffer = buffer; 1750 mOwnInBuffer = ownsBuffer; 1751 } 1752 int16_t *inBuffer() const { 1753 return mInBuffer; 1754 } 1755 void setOutBuffer(int16_t *buffer) { 1756 mOutBuffer = buffer; 1757 } 1758 int16_t *outBuffer() const { 1759 return mOutBuffer; 1760 } 1761 1762 void incTrackCnt() { android_atomic_inc(&mTrackCnt); } 1763 void decTrackCnt() { android_atomic_dec(&mTrackCnt); } 1764 int32_t trackCnt() const { return android_atomic_acquire_load(&mTrackCnt); } 1765 1766 void incActiveTrackCnt() { android_atomic_inc(&mActiveTrackCnt); 1767 mTailBufferCount = mMaxTailBuffers; } 1768 void decActiveTrackCnt() { android_atomic_dec(&mActiveTrackCnt); } 1769 int32_t activeTrackCnt() const { return android_atomic_acquire_load(&mActiveTrackCnt); } 1770 1771 uint32_t strategy() const { return mStrategy; } 1772 void setStrategy(uint32_t strategy) 1773 { mStrategy = strategy; } 1774 1775 // suspend effect of the given type 1776 void setEffectSuspended_l(const effect_uuid_t *type, 1777 bool suspend); 1778 // suspend all eligible effects 1779 void setEffectSuspendedAll_l(bool suspend); 1780 // check if effects should be suspend or restored when a given effect is enable or disabled 1781 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1782 bool enabled); 1783 1784 void clearInputBuffer(); 1785 1786 void dump(int fd, const Vector<String16>& args); 1787 1788 protected: 1789 friend class AudioFlinger; // for mThread, mEffects 1790 EffectChain(const EffectChain&); 1791 EffectChain& operator =(const EffectChain&); 1792 1793 class SuspendedEffectDesc : public RefBase { 1794 public: 1795 SuspendedEffectDesc() : mRefCount(0) {} 1796 1797 int mRefCount; 1798 effect_uuid_t mType; 1799 wp<EffectModule> mEffect; 1800 }; 1801 1802 // get a list of effect modules to suspend when an effect of the type 1803 // passed is enabled. 1804 void getSuspendEligibleEffects(Vector< sp<EffectModule> > &effects); 1805 1806 // get an effect module if it is currently enable 1807 sp<EffectModule> getEffectIfEnabled(const effect_uuid_t *type); 1808 // true if the effect whose descriptor is passed can be suspended 1809 // OEMs can modify the rules implemented in this method to exclude specific effect 1810 // types or implementations from the suspend/restore mechanism. 1811 bool isEffectEligibleForSuspend(const effect_descriptor_t& desc); 1812 1813 void clearInputBuffer_l(sp<ThreadBase> thread); 1814 1815 wp<ThreadBase> mThread; // parent mixer thread 1816 Mutex mLock; // mutex protecting effect list 1817 Vector< sp<EffectModule> > mEffects; // list of effect modules 1818 int mSessionId; // audio session ID 1819 int16_t *mInBuffer; // chain input buffer 1820 int16_t *mOutBuffer; // chain output buffer 1821 1822 // 'volatile' here means these are accessed with atomic operations instead of mutex 1823 volatile int32_t mActiveTrackCnt; // number of active tracks connected 1824 volatile int32_t mTrackCnt; // number of tracks connected 1825 1826 int32_t mTailBufferCount; // current effect tail buffer count 1827 int32_t mMaxTailBuffers; // maximum effect tail buffers 1828 bool mOwnInBuffer; // true if the chain owns its input buffer 1829 int mVolumeCtrlIdx; // index of insert effect having control over volume 1830 uint32_t mLeftVolume; // previous volume on left channel 1831 uint32_t mRightVolume; // previous volume on right channel 1832 uint32_t mNewLeftVolume; // new volume on left channel 1833 uint32_t mNewRightVolume; // new volume on right channel 1834 uint32_t mStrategy; // strategy for this effect chain 1835 // mSuspendedEffects lists all effects currently suspended in the chain. 1836 // Use effect type UUID timelow field as key. There is no real risk of identical 1837 // timeLow fields among effect type UUIDs. 1838 // Updated by updateSuspendedSessions_l() only. 1839 KeyedVector< int, sp<SuspendedEffectDesc> > mSuspendedEffects; 1840 }; 1841 1842 // AudioStreamOut and AudioStreamIn are immutable, so their fields are const. 1843 // For emphasis, we could also make all pointers to them be "const *", 1844 // but that would clutter the code unnecessarily. 1845 1846 struct AudioStreamOut { 1847 audio_hw_device_t* const hwDev; 1848 audio_stream_out_t* const stream; 1849 1850 AudioStreamOut(audio_hw_device_t *dev, audio_stream_out_t *out) : 1851 hwDev(dev), stream(out) {} 1852 }; 1853 1854 struct AudioStreamIn { 1855 audio_hw_device_t* const hwDev; 1856 audio_stream_in_t* const stream; 1857 1858 AudioStreamIn(audio_hw_device_t *dev, audio_stream_in_t *in) : 1859 hwDev(dev), stream(in) {} 1860 }; 1861 1862 // for mAudioSessionRefs only 1863 struct AudioSessionRef { 1864 AudioSessionRef(int sessionid, pid_t pid) : 1865 mSessionid(sessionid), mPid(pid), mCnt(1) {} 1866 const int mSessionid; 1867 const pid_t mPid; 1868 int mCnt; 1869 }; 1870 1871 enum master_volume_support { 1872 // MVS_NONE: 1873 // Audio HAL has no support for master volume, either setting or 1874 // getting. All master volume control must be implemented in SW by the 1875 // AudioFlinger mixing core. 1876 MVS_NONE, 1877 1878 // MVS_SETONLY: 1879 // Audio HAL has support for setting master volume, but not for getting 1880 // master volume (original HAL design did not include a getter). 1881 // AudioFlinger needs to keep track of the last set master volume in 1882 // addition to needing to set an initial, default, master volume at HAL 1883 // load time. 1884 MVS_SETONLY, 1885 1886 // MVS_FULL: 1887 // Audio HAL has support both for setting and getting master volume. 1888 // AudioFlinger should send all set and get master volume requests 1889 // directly to the HAL. 1890 MVS_FULL, 1891 }; 1892 1893 class AudioHwDevice { 1894 public: 1895 AudioHwDevice(const char *moduleName, audio_hw_device_t *hwDevice) : 1896 mModuleName(strdup(moduleName)), mHwDevice(hwDevice){} 1897 /*virtual*/ ~AudioHwDevice() { free((void *)mModuleName); } 1898 1899 const char *moduleName() const { return mModuleName; } 1900 audio_hw_device_t *hwDevice() const { return mHwDevice; } 1901 private: 1902 const char * const mModuleName; 1903 audio_hw_device_t * const mHwDevice; 1904 }; 1905 1906 mutable Mutex mLock; 1907 1908 DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client() 1909 1910 mutable Mutex mHardwareLock; 1911 // NOTE: If both mLock and mHardwareLock mutexes must be held, 1912 // always take mLock before mHardwareLock 1913 1914 // These two fields are immutable after onFirstRef(), so no lock needed to access 1915 audio_hw_device_t* mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL 1916 DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*> mAudioHwDevs; 1917 1918 // for dump, indicates which hardware operation is currently in progress (but not stream ops) 1919 enum hardware_call_state { 1920 AUDIO_HW_IDLE = 0, // no operation in progress 1921 AUDIO_HW_INIT, // init_check 1922 AUDIO_HW_OUTPUT_OPEN, // open_output_stream 1923 AUDIO_HW_OUTPUT_CLOSE, // unused 1924 AUDIO_HW_INPUT_OPEN, // unused 1925 AUDIO_HW_INPUT_CLOSE, // unused 1926 AUDIO_HW_STANDBY, // unused 1927 AUDIO_HW_SET_MASTER_VOLUME, // set_master_volume 1928 AUDIO_HW_GET_ROUTING, // unused 1929 AUDIO_HW_SET_ROUTING, // unused 1930 AUDIO_HW_GET_MODE, // unused 1931 AUDIO_HW_SET_MODE, // set_mode 1932 AUDIO_HW_GET_MIC_MUTE, // get_mic_mute 1933 AUDIO_HW_SET_MIC_MUTE, // set_mic_mute 1934 AUDIO_HW_SET_VOICE_VOLUME, // set_voice_volume 1935 AUDIO_HW_SET_PARAMETER, // set_parameters 1936 AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size 1937 AUDIO_HW_GET_MASTER_VOLUME, // get_master_volume 1938 AUDIO_HW_GET_PARAMETER, // get_parameters 1939 }; 1940 1941 mutable hardware_call_state mHardwareStatus; // for dump only 1942 1943 1944 DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads; 1945 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 1946 1947 // both are protected by mLock 1948 float mMasterVolume; 1949 float mMasterVolumeSW; 1950 master_volume_support mMasterVolumeSupportLvl; 1951 bool mMasterMute; 1952 1953 DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads; 1954 1955 DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients; 1956 volatile int32_t mNextUniqueId; // updated by android_atomic_inc 1957 audio_mode_t mMode; 1958 bool mBtNrecIsOff; 1959 1960 // protected by mLock 1961 Vector<AudioSessionRef*> mAudioSessionRefs; 1962 1963 float masterVolume_l() const; 1964 float masterVolumeSW_l() const { return mMasterVolumeSW; } 1965 bool masterMute_l() const { return mMasterMute; } 1966 audio_module_handle_t loadHwModule_l(const char *name); 1967 1968 Vector < sp<SyncEvent> > mPendingSyncEvents; // sync events awaiting for a session 1969 // to be created 1970 1971private: 1972 sp<Client> registerPid_l(pid_t pid); // always returns non-0 1973 1974 // for use from destructor 1975 status_t closeOutput_nonvirtual(audio_io_handle_t output); 1976 status_t closeInput_nonvirtual(audio_io_handle_t input); 1977}; 1978 1979 1980// ---------------------------------------------------------------------------- 1981 1982}; // namespace android 1983 1984#endif // ANDROID_AUDIO_FLINGER_H 1985