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 <media/AudioBufferProvider.h> 49 #include <media/ExtendedAudioBufferProvider.h> 50 #include "FastMixer.h" 51 #include <media/nbaio/NBAIO.h> 52 #include "AudioWatchdog.h" 53 54 #include <powermanager/IPowerManager.h> 55 56 namespace android { 57 58 class audio_track_cblk_t; 59 class effect_param_cblk_t; 60 class AudioMixer; 61 class AudioBuffer; 62 class AudioResampler; 63 class 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 76 static const nsecs_t kDefaultStandbyTimeInNsecs = seconds(3); 77 78 class AudioFlinger : 79 public BinderService<AudioFlinger>, 80 public BnAudioFlinger 81 { 82 friend class BinderService<AudioFlinger>; // for AudioFlinger() 83 public: 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 bool masterMute() const; 126 127 virtual status_t setStreamVolume(audio_stream_type_t stream, float value, 128 audio_io_handle_t output); 129 virtual status_t setStreamMute(audio_stream_type_t stream, bool muted); 130 131 virtual float streamVolume(audio_stream_type_t stream, 132 audio_io_handle_t output) const; 133 virtual bool streamMute(audio_stream_type_t stream) const; 134 135 virtual status_t setMode(audio_mode_t mode); 136 137 virtual status_t setMicMute(bool state); 138 virtual bool getMicMute() const; 139 140 virtual status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs); 141 virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) const; 142 143 virtual void registerClient(const sp<IAudioFlingerClient>& client); 144 145 virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, 146 audio_channel_mask_t channelMask) const; 147 148 virtual audio_io_handle_t openOutput(audio_module_handle_t module, 149 audio_devices_t *pDevices, 150 uint32_t *pSamplingRate, 151 audio_format_t *pFormat, 152 audio_channel_mask_t *pChannelMask, 153 uint32_t *pLatencyMs, 154 audio_output_flags_t flags); 155 156 virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1, 157 audio_io_handle_t output2); 158 159 virtual status_t closeOutput(audio_io_handle_t output); 160 161 virtual status_t suspendOutput(audio_io_handle_t output); 162 163 virtual status_t restoreOutput(audio_io_handle_t output); 164 165 virtual audio_io_handle_t openInput(audio_module_handle_t module, 166 audio_devices_t *pDevices, 167 uint32_t *pSamplingRate, 168 audio_format_t *pFormat, 169 audio_channel_mask_t *pChannelMask); 170 171 virtual status_t closeInput(audio_io_handle_t input); 172 173 virtual status_t setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output); 174 175 virtual status_t setVoiceVolume(float volume); 176 177 virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 178 audio_io_handle_t output) const; 179 180 virtual unsigned int getInputFramesLost(audio_io_handle_t ioHandle) const; 181 182 virtual int newAudioSessionId(); 183 184 virtual void acquireAudioSessionId(int audioSession); 185 186 virtual void releaseAudioSessionId(int audioSession); 187 188 virtual status_t queryNumberEffects(uint32_t *numEffects) const; 189 190 virtual status_t queryEffect(uint32_t index, effect_descriptor_t *descriptor) const; 191 192 virtual status_t getEffectDescriptor(const effect_uuid_t *pUuid, 193 effect_descriptor_t *descriptor) const; 194 195 virtual sp<IEffect> createEffect(pid_t pid, 196 effect_descriptor_t *pDesc, 197 const sp<IEffectClient>& effectClient, 198 int32_t priority, 199 audio_io_handle_t io, 200 int sessionId, 201 status_t *status, 202 int *id, 203 int *enabled); 204 205 virtual status_t moveEffects(int sessionId, audio_io_handle_t srcOutput, 206 audio_io_handle_t dstOutput); 207 208 virtual audio_module_handle_t loadHwModule(const char *name); 209 210 virtual int32_t getPrimaryOutputSamplingRate(); 211 virtual int32_t getPrimaryOutputFrameCount(); 212 213 virtual status_t onTransact( 214 uint32_t code, 215 const Parcel& data, 216 Parcel* reply, 217 uint32_t flags); 218 219 // end of IAudioFlinger interface 220 221 class SyncEvent; 222 223 typedef void (*sync_event_callback_t)(const wp<SyncEvent>& event) ; 224 225 class SyncEvent : public RefBase { 226 public: 227 SyncEvent(AudioSystem::sync_event_t type, 228 int triggerSession, 229 int listenerSession, 230 sync_event_callback_t callBack, 231 void *cookie) 232 : mType(type), mTriggerSession(triggerSession), mListenerSession(listenerSession), 233 mCallback(callBack), mCookie(cookie) 234 {} 235 236 virtual ~SyncEvent() {} 237 238 void trigger() { Mutex::Autolock _l(mLock); if (mCallback) mCallback(this); } 239 bool isCancelled() const { Mutex::Autolock _l(mLock); return (mCallback == NULL); } 240 void cancel() { Mutex::Autolock _l(mLock); mCallback = NULL; } 241 AudioSystem::sync_event_t type() const { return mType; } 242 int triggerSession() const { return mTriggerSession; } 243 int listenerSession() const { return mListenerSession; } 244 void *cookie() const { return mCookie; } 245 246 private: 247 const AudioSystem::sync_event_t mType; 248 const int mTriggerSession; 249 const int mListenerSession; 250 sync_event_callback_t mCallback; 251 void * const mCookie; 252 mutable Mutex mLock; 253 }; 254 255 sp<SyncEvent> createSyncEvent(AudioSystem::sync_event_t type, 256 int triggerSession, 257 int listenerSession, 258 sync_event_callback_t callBack, 259 void *cookie); 260 261 private: 262 class AudioHwDevice; // fwd declaration for findSuitableHwDev_l 263 264 audio_mode_t getMode() const { return mMode; } 265 266 bool btNrecIsOff() const { return mBtNrecIsOff; } 267 268 AudioFlinger(); 269 virtual ~AudioFlinger(); 270 271 // call in any IAudioFlinger method that accesses mPrimaryHardwareDev 272 status_t initCheck() const { return mPrimaryHardwareDev == NULL ? NO_INIT : NO_ERROR; } 273 274 // RefBase 275 virtual void onFirstRef(); 276 277 AudioHwDevice* findSuitableHwDev_l(audio_module_handle_t module, audio_devices_t devices); 278 void purgeStaleEffects_l(); 279 280 // standby delay for MIXER and DUPLICATING playback threads is read from property 281 // ro.audio.flinger_standbytime_ms or defaults to kDefaultStandbyTimeInNsecs 282 static nsecs_t mStandbyTimeInNsecs; 283 284 // Internal dump utilities. 285 void dumpPermissionDenial(int fd, const Vector<String16>& args); 286 void dumpClients(int fd, const Vector<String16>& args); 287 void dumpInternals(int fd, const Vector<String16>& args); 288 289 // --- Client --- 290 class Client : public RefBase { 291 public: 292 Client(const sp<AudioFlinger>& audioFlinger, pid_t pid); 293 virtual ~Client(); 294 sp<MemoryDealer> heap() const; 295 pid_t pid() const { return mPid; } 296 sp<AudioFlinger> audioFlinger() const { return mAudioFlinger; } 297 298 bool reserveTimedTrack(); 299 void releaseTimedTrack(); 300 301 private: 302 Client(const Client&); 303 Client& operator = (const Client&); 304 const sp<AudioFlinger> mAudioFlinger; 305 const sp<MemoryDealer> mMemoryDealer; 306 const pid_t mPid; 307 308 Mutex mTimedTrackLock; 309 int mTimedTrackCount; 310 }; 311 312 // --- Notification Client --- 313 class NotificationClient : public IBinder::DeathRecipient { 314 public: 315 NotificationClient(const sp<AudioFlinger>& audioFlinger, 316 const sp<IAudioFlingerClient>& client, 317 pid_t pid); 318 virtual ~NotificationClient(); 319 320 sp<IAudioFlingerClient> audioFlingerClient() const { return mAudioFlingerClient; } 321 322 // IBinder::DeathRecipient 323 virtual void binderDied(const wp<IBinder>& who); 324 325 private: 326 NotificationClient(const NotificationClient&); 327 NotificationClient& operator = (const NotificationClient&); 328 329 const sp<AudioFlinger> mAudioFlinger; 330 const pid_t mPid; 331 const sp<IAudioFlingerClient> mAudioFlingerClient; 332 }; 333 334 class TrackHandle; 335 class RecordHandle; 336 class RecordThread; 337 class PlaybackThread; 338 class MixerThread; 339 class DirectOutputThread; 340 class DuplicatingThread; 341 class Track; 342 class RecordTrack; 343 class EffectModule; 344 class EffectHandle; 345 class EffectChain; 346 struct AudioStreamOut; 347 struct AudioStreamIn; 348 349 class ThreadBase : public Thread { 350 public: 351 352 enum type_t { 353 MIXER, // Thread class is MixerThread 354 DIRECT, // Thread class is DirectOutputThread 355 DUPLICATING, // Thread class is DuplicatingThread 356 RECORD // Thread class is RecordThread 357 }; 358 359 ThreadBase (const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 360 audio_devices_t outDevice, audio_devices_t inDevice, type_t type); 361 virtual ~ThreadBase(); 362 363 void dumpBase(int fd, const Vector<String16>& args); 364 void dumpEffectChains(int fd, const Vector<String16>& args); 365 366 void clearPowerManager(); 367 368 // base for record and playback 369 class TrackBase : public ExtendedAudioBufferProvider, public RefBase { 370 371 public: 372 enum track_state { 373 IDLE, 374 TERMINATED, 375 FLUSHED, 376 STOPPED, 377 // next 2 states are currently used for fast tracks only 378 STOPPING_1, // waiting for first underrun 379 STOPPING_2, // waiting for presentation complete 380 RESUMING, 381 ACTIVE, 382 PAUSING, 383 PAUSED 384 }; 385 386 TrackBase(ThreadBase *thread, 387 const sp<Client>& client, 388 uint32_t sampleRate, 389 audio_format_t format, 390 audio_channel_mask_t channelMask, 391 int frameCount, 392 const sp<IMemory>& sharedBuffer, 393 int sessionId); 394 virtual ~TrackBase(); 395 396 virtual status_t start(AudioSystem::sync_event_t event, 397 int triggerSession) = 0; 398 virtual void stop() = 0; 399 sp<IMemory> getCblk() const { return mCblkMemory; } 400 audio_track_cblk_t* cblk() const { return mCblk; } 401 int sessionId() const { return mSessionId; } 402 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 403 404 protected: 405 TrackBase(const TrackBase&); 406 TrackBase& operator = (const TrackBase&); 407 408 // AudioBufferProvider interface 409 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) = 0; 410 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 411 412 // ExtendedAudioBufferProvider interface is only needed for Track, 413 // but putting it in TrackBase avoids the complexity of virtual inheritance 414 virtual size_t framesReady() const { return SIZE_MAX; } 415 416 audio_format_t format() const { 417 return mFormat; 418 } 419 420 int channelCount() const { return mChannelCount; } 421 422 audio_channel_mask_t channelMask() const { return mChannelMask; } 423 424 int sampleRate() const; // FIXME inline after cblk sr moved 425 426 // Return a pointer to the start of a contiguous slice of the track buffer. 427 // Parameter 'offset' is the requested start position, expressed in 428 // monotonically increasing frame units relative to the track epoch. 429 // Parameter 'frames' is the requested length, also in frame units. 430 // Always returns non-NULL. It is the caller's responsibility to 431 // verify that this will be successful; the result of calling this 432 // function with invalid 'offset' or 'frames' is undefined. 433 void* getBuffer(uint32_t offset, uint32_t frames) const; 434 435 bool isStopped() const { 436 return (mState == STOPPED || mState == FLUSHED); 437 } 438 439 // for fast tracks only 440 bool isStopping() const { 441 return mState == STOPPING_1 || mState == STOPPING_2; 442 } 443 bool isStopping_1() const { 444 return mState == STOPPING_1; 445 } 446 bool isStopping_2() const { 447 return mState == STOPPING_2; 448 } 449 450 bool isTerminated() const { 451 return mState == TERMINATED; 452 } 453 454 bool step(); 455 void reset(); 456 457 const wp<ThreadBase> mThread; 458 /*const*/ sp<Client> mClient; // see explanation at ~TrackBase() why not const 459 sp<IMemory> mCblkMemory; 460 audio_track_cblk_t* mCblk; 461 void* mBuffer; // start of track buffer, typically in shared memory 462 void* mBufferEnd; // &mBuffer[mFrameCount * frameSize], where frameSize 463 // is based on mChannelCount and 16-bit samples 464 uint32_t mFrameCount; 465 // we don't really need a lock for these 466 track_state mState; 467 const uint32_t mSampleRate; // initial sample rate only; for tracks which 468 // support dynamic rates, the current value is in control block 469 const audio_format_t mFormat; 470 bool mStepServerFailed; 471 const int mSessionId; 472 uint8_t mChannelCount; 473 audio_channel_mask_t mChannelMask; 474 Vector < sp<SyncEvent> >mSyncEvents; 475 }; 476 477 enum { 478 CFG_EVENT_IO, 479 CFG_EVENT_PRIO 480 }; 481 482 class ConfigEvent { 483 public: 484 ConfigEvent(int type) : mType(type) {} 485 virtual ~ConfigEvent() {} 486 487 int type() const { return mType; } 488 489 virtual void dump(char *buffer, size_t size) = 0; 490 491 private: 492 const int mType; 493 }; 494 495 class IoConfigEvent : public ConfigEvent { 496 public: 497 IoConfigEvent(int event, int param) : 498 ConfigEvent(CFG_EVENT_IO), mEvent(event), mParam(event) {} 499 virtual ~IoConfigEvent() {} 500 501 int event() const { return mEvent; } 502 int param() const { return mParam; } 503 504 virtual void dump(char *buffer, size_t size) { 505 snprintf(buffer, size, "IO event: event %d, param %d\n", mEvent, mParam); 506 } 507 508 private: 509 const int mEvent; 510 const int mParam; 511 }; 512 513 class PrioConfigEvent : public ConfigEvent { 514 public: 515 PrioConfigEvent(pid_t pid, pid_t tid, int32_t prio) : 516 ConfigEvent(CFG_EVENT_PRIO), mPid(pid), mTid(tid), mPrio(prio) {} 517 virtual ~PrioConfigEvent() {} 518 519 pid_t pid() const { return mPid; } 520 pid_t tid() const { return mTid; } 521 int32_t prio() const { return mPrio; } 522 523 virtual void dump(char *buffer, size_t size) { 524 snprintf(buffer, size, "Prio event: pid %d, tid %d, prio %d\n", mPid, mTid, mPrio); 525 } 526 527 private: 528 const pid_t mPid; 529 const pid_t mTid; 530 const int32_t mPrio; 531 }; 532 533 534 class PMDeathRecipient : public IBinder::DeathRecipient { 535 public: 536 PMDeathRecipient(const wp<ThreadBase>& thread) : mThread(thread) {} 537 virtual ~PMDeathRecipient() {} 538 539 // IBinder::DeathRecipient 540 virtual void binderDied(const wp<IBinder>& who); 541 542 private: 543 PMDeathRecipient(const PMDeathRecipient&); 544 PMDeathRecipient& operator = (const PMDeathRecipient&); 545 546 wp<ThreadBase> mThread; 547 }; 548 549 virtual status_t initCheck() const = 0; 550 551 // static externally-visible 552 type_t type() const { return mType; } 553 audio_io_handle_t id() const { return mId;} 554 555 // dynamic externally-visible 556 uint32_t sampleRate() const { return mSampleRate; } 557 int channelCount() const { return mChannelCount; } 558 audio_channel_mask_t channelMask() const { return mChannelMask; } 559 audio_format_t format() const { return mFormat; } 560 // Called by AudioFlinger::frameCount(audio_io_handle_t output) and effects, 561 // and returns the normal mix buffer's frame count. 562 size_t frameCount() const { return mNormalFrameCount; } 563 // Return's the HAL's frame count i.e. fast mixer buffer size. 564 size_t frameCountHAL() const { return mFrameCount; } 565 566 // Should be "virtual status_t requestExitAndWait()" and override same 567 // method in Thread, but Thread::requestExitAndWait() is not yet virtual. 568 void exit(); 569 virtual bool checkForNewParameters_l() = 0; 570 virtual status_t setParameters(const String8& keyValuePairs); 571 virtual String8 getParameters(const String8& keys) = 0; 572 virtual void audioConfigChanged_l(int event, int param = 0) = 0; 573 void sendIoConfigEvent(int event, int param = 0); 574 void sendIoConfigEvent_l(int event, int param = 0); 575 void sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio); 576 void processConfigEvents(); 577 578 // see note at declaration of mStandby, mOutDevice and mInDevice 579 bool standby() const { return mStandby; } 580 audio_devices_t outDevice() const { return mOutDevice; } 581 audio_devices_t inDevice() const { return mInDevice; } 582 583 virtual audio_stream_t* stream() const = 0; 584 585 sp<EffectHandle> createEffect_l( 586 const sp<AudioFlinger::Client>& client, 587 const sp<IEffectClient>& effectClient, 588 int32_t priority, 589 int sessionId, 590 effect_descriptor_t *desc, 591 int *enabled, 592 status_t *status); 593 void disconnectEffect(const sp< EffectModule>& effect, 594 EffectHandle *handle, 595 bool unpinIfLast); 596 597 // return values for hasAudioSession (bit field) 598 enum effect_state { 599 EFFECT_SESSION = 0x1, // the audio session corresponds to at least one 600 // effect 601 TRACK_SESSION = 0x2 // the audio session corresponds to at least one 602 // track 603 }; 604 605 // get effect chain corresponding to session Id. 606 sp<EffectChain> getEffectChain(int sessionId); 607 // same as getEffectChain() but must be called with ThreadBase mutex locked 608 sp<EffectChain> getEffectChain_l(int sessionId) const; 609 // add an effect chain to the chain list (mEffectChains) 610 virtual status_t addEffectChain_l(const sp<EffectChain>& chain) = 0; 611 // remove an effect chain from the chain list (mEffectChains) 612 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain) = 0; 613 // lock all effect chains Mutexes. Must be called before releasing the 614 // ThreadBase mutex before processing the mixer and effects. This guarantees the 615 // integrity of the chains during the process. 616 // Also sets the parameter 'effectChains' to current value of mEffectChains. 617 void lockEffectChains_l(Vector< sp<EffectChain> >& effectChains); 618 // unlock effect chains after process 619 void unlockEffectChains(const Vector< sp<EffectChain> >& effectChains); 620 // set audio mode to all effect chains 621 void setMode(audio_mode_t mode); 622 // get effect module with corresponding ID on specified audio session 623 sp<AudioFlinger::EffectModule> getEffect(int sessionId, int effectId); 624 sp<AudioFlinger::EffectModule> getEffect_l(int sessionId, int effectId); 625 // add and effect module. Also creates the effect chain is none exists for 626 // the effects audio session 627 status_t addEffect_l(const sp< EffectModule>& effect); 628 // remove and effect module. Also removes the effect chain is this was the last 629 // effect 630 void removeEffect_l(const sp< EffectModule>& effect); 631 // detach all tracks connected to an auxiliary effect 632 virtual void detachAuxEffect_l(int effectId) {} 633 // returns either EFFECT_SESSION if effects on this audio session exist in one 634 // chain, or TRACK_SESSION if tracks on this audio session exist, or both 635 virtual uint32_t hasAudioSession(int sessionId) const = 0; 636 // the value returned by default implementation is not important as the 637 // strategy is only meaningful for PlaybackThread which implements this method 638 virtual uint32_t getStrategyForSession_l(int sessionId) { return 0; } 639 640 // suspend or restore effect according to the type of effect passed. a NULL 641 // type pointer means suspend all effects in the session 642 void setEffectSuspended(const effect_uuid_t *type, 643 bool suspend, 644 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 645 // check if some effects must be suspended/restored when an effect is enabled 646 // or disabled 647 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 648 bool enabled, 649 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 650 void checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 651 bool enabled, 652 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 653 654 virtual status_t setSyncEvent(const sp<SyncEvent>& event) = 0; 655 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const = 0; 656 657 658 mutable Mutex mLock; 659 660 protected: 661 662 // entry describing an effect being suspended in mSuspendedSessions keyed vector 663 class SuspendedSessionDesc : public RefBase { 664 public: 665 SuspendedSessionDesc() : mRefCount(0) {} 666 667 int mRefCount; // number of active suspend requests 668 effect_uuid_t mType; // effect type UUID 669 }; 670 671 void acquireWakeLock(); 672 void acquireWakeLock_l(); 673 void releaseWakeLock(); 674 void releaseWakeLock_l(); 675 void setEffectSuspended_l(const effect_uuid_t *type, 676 bool suspend, 677 int sessionId); 678 // updated mSuspendedSessions when an effect suspended or restored 679 void updateSuspendedSessions_l(const effect_uuid_t *type, 680 bool suspend, 681 int sessionId); 682 // check if some effects must be suspended when an effect chain is added 683 void checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain); 684 685 virtual void preExit() { } 686 687 friend class AudioFlinger; // for mEffectChains 688 689 const type_t mType; 690 691 // Used by parameters, config events, addTrack_l, exit 692 Condition mWaitWorkCV; 693 694 const sp<AudioFlinger> mAudioFlinger; 695 uint32_t mSampleRate; 696 size_t mFrameCount; // output HAL, direct output, record 697 size_t mNormalFrameCount; // normal mixer and effects 698 audio_channel_mask_t mChannelMask; 699 uint16_t mChannelCount; 700 size_t mFrameSize; 701 audio_format_t mFormat; 702 703 // Parameter sequence by client: binder thread calling setParameters(): 704 // 1. Lock mLock 705 // 2. Append to mNewParameters 706 // 3. mWaitWorkCV.signal 707 // 4. mParamCond.waitRelative with timeout 708 // 5. read mParamStatus 709 // 6. mWaitWorkCV.signal 710 // 7. Unlock 711 // 712 // Parameter sequence by server: threadLoop calling checkForNewParameters_l(): 713 // 1. Lock mLock 714 // 2. If there is an entry in mNewParameters proceed ... 715 // 2. Read first entry in mNewParameters 716 // 3. Process 717 // 4. Remove first entry from mNewParameters 718 // 5. Set mParamStatus 719 // 6. mParamCond.signal 720 // 7. mWaitWorkCV.wait with timeout (this is to avoid overwriting mParamStatus) 721 // 8. Unlock 722 Condition mParamCond; 723 Vector<String8> mNewParameters; 724 status_t mParamStatus; 725 726 Vector<ConfigEvent *> mConfigEvents; 727 728 // These fields are written and read by thread itself without lock or barrier, 729 // and read by other threads without lock or barrier via standby() , outDevice() 730 // and inDevice(). 731 // Because of the absence of a lock or barrier, any other thread that reads 732 // these fields must use the information in isolation, or be prepared to deal 733 // with possibility that it might be inconsistent with other information. 734 bool mStandby; // Whether thread is currently in standby. 735 audio_devices_t mOutDevice; // output device 736 audio_devices_t mInDevice; // input device 737 audio_source_t mAudioSource; // (see audio.h, audio_source_t) 738 739 const audio_io_handle_t mId; 740 Vector< sp<EffectChain> > mEffectChains; 741 742 static const int kNameLength = 16; // prctl(PR_SET_NAME) limit 743 char mName[kNameLength]; 744 sp<IPowerManager> mPowerManager; 745 sp<IBinder> mWakeLockToken; 746 const sp<PMDeathRecipient> mDeathRecipient; 747 // list of suspended effects per session and per type. The first vector is 748 // keyed by session ID, the second by type UUID timeLow field 749 KeyedVector< int, KeyedVector< int, sp<SuspendedSessionDesc> > > mSuspendedSessions; 750 }; 751 752 struct stream_type_t { 753 stream_type_t() 754 : volume(1.0f), 755 mute(false) 756 { 757 } 758 float volume; 759 bool mute; 760 }; 761 762 // --- PlaybackThread --- 763 class PlaybackThread : public ThreadBase { 764 public: 765 766 enum mixer_state { 767 MIXER_IDLE, // no active tracks 768 MIXER_TRACKS_ENABLED, // at least one active track, but no track has any data ready 769 MIXER_TRACKS_READY // at least one active track, and at least one track has data 770 // standby mode does not have an enum value 771 // suspend by audio policy manager is orthogonal to mixer state 772 }; 773 774 // playback track 775 class Track : public TrackBase, public VolumeProvider { 776 public: 777 Track( PlaybackThread *thread, 778 const sp<Client>& client, 779 audio_stream_type_t streamType, 780 uint32_t sampleRate, 781 audio_format_t format, 782 audio_channel_mask_t channelMask, 783 int frameCount, 784 const sp<IMemory>& sharedBuffer, 785 int sessionId, 786 IAudioFlinger::track_flags_t flags); 787 virtual ~Track(); 788 789 static void appendDumpHeader(String8& result); 790 void dump(char* buffer, size_t size); 791 virtual status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 792 int triggerSession = 0); 793 virtual void stop(); 794 void pause(); 795 796 void flush(); 797 void destroy(); 798 void mute(bool); 799 int name() const { return mName; } 800 801 audio_stream_type_t streamType() const { 802 return mStreamType; 803 } 804 status_t attachAuxEffect(int EffectId); 805 void setAuxBuffer(int EffectId, int32_t *buffer); 806 int32_t *auxBuffer() const { return mAuxBuffer; } 807 void setMainBuffer(int16_t *buffer) { mMainBuffer = buffer; } 808 int16_t *mainBuffer() const { return mMainBuffer; } 809 int auxEffectId() const { return mAuxEffectId; } 810 811 // implement FastMixerState::VolumeProvider interface 812 virtual uint32_t getVolumeLR(); 813 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 814 815 protected: 816 // for numerous 817 friend class PlaybackThread; 818 friend class MixerThread; 819 friend class DirectOutputThread; 820 821 Track(const Track&); 822 Track& operator = (const Track&); 823 824 // AudioBufferProvider interface 825 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts = kInvalidPTS); 826 // releaseBuffer() not overridden 827 828 virtual size_t framesReady() const; 829 830 bool isMuted() const { return mMute; } 831 bool isPausing() const { 832 return mState == PAUSING; 833 } 834 bool isPaused() const { 835 return mState == PAUSED; 836 } 837 bool isResuming() const { 838 return mState == RESUMING; 839 } 840 bool isReady() const; 841 void setPaused() { mState = PAUSED; } 842 void reset(); 843 844 bool isOutputTrack() const { 845 return (mStreamType == AUDIO_STREAM_CNT); 846 } 847 848 sp<IMemory> sharedBuffer() const { return mSharedBuffer; } 849 850 bool presentationComplete(size_t framesWritten, size_t audioHalFrames); 851 852 public: 853 void triggerEvents(AudioSystem::sync_event_t type); 854 virtual bool isTimedTrack() const { return false; } 855 bool isFastTrack() const { return (mFlags & IAudioFlinger::TRACK_FAST) != 0; } 856 857 protected: 858 859 // written by Track::mute() called by binder thread(s), without a mutex or barrier. 860 // read by Track::isMuted() called by playback thread, also without a mutex or barrier. 861 // The lack of mutex or barrier is safe because the mute status is only used by itself. 862 bool mMute; 863 864 // FILLED state is used for suppressing volume ramp at begin of playing 865 enum {FS_INVALID, FS_FILLING, FS_FILLED, FS_ACTIVE}; 866 mutable uint8_t mFillingUpStatus; 867 int8_t mRetryCount; 868 const sp<IMemory> mSharedBuffer; 869 bool mResetDone; 870 const audio_stream_type_t mStreamType; 871 int mName; // track name on the normal mixer, 872 // allocated statically at track creation time, 873 // and is even allocated (though unused) for fast tracks 874 // FIXME don't allocate track name for fast tracks 875 int16_t *mMainBuffer; 876 int32_t *mAuxBuffer; 877 int mAuxEffectId; 878 bool mHasVolumeController; 879 size_t mPresentationCompleteFrames; // number of frames written to the audio HAL 880 // when this track will be fully rendered 881 private: 882 IAudioFlinger::track_flags_t mFlags; 883 884 // The following fields are only for fast tracks, and should be in a subclass 885 int mFastIndex; // index within FastMixerState::mFastTracks[]; 886 // either mFastIndex == -1 if not isFastTrack() 887 // or 0 < mFastIndex < FastMixerState::kMaxFast because 888 // index 0 is reserved for normal mixer's submix; 889 // index is allocated statically at track creation time 890 // but the slot is only used if track is active 891 FastTrackUnderruns mObservedUnderruns; // Most recently observed value of 892 // mFastMixerDumpState.mTracks[mFastIndex].mUnderruns 893 uint32_t mUnderrunCount; // Counter of total number of underruns, never reset 894 volatile float mCachedVolume; // combined master volume and stream type volume; 895 // 'volatile' means accessed without lock or 896 // barrier, but is read/written atomically 897 }; // end of Track 898 899 class TimedTrack : public Track { 900 public: 901 static sp<TimedTrack> create(PlaybackThread *thread, 902 const sp<Client>& client, 903 audio_stream_type_t streamType, 904 uint32_t sampleRate, 905 audio_format_t format, 906 audio_channel_mask_t channelMask, 907 int frameCount, 908 const sp<IMemory>& sharedBuffer, 909 int sessionId); 910 virtual ~TimedTrack(); 911 912 class TimedBuffer { 913 public: 914 TimedBuffer(); 915 TimedBuffer(const sp<IMemory>& buffer, int64_t pts); 916 const sp<IMemory>& buffer() const { return mBuffer; } 917 int64_t pts() const { return mPTS; } 918 uint32_t position() const { return mPosition; } 919 void setPosition(uint32_t pos) { mPosition = pos; } 920 private: 921 sp<IMemory> mBuffer; 922 int64_t mPTS; 923 uint32_t mPosition; 924 }; 925 926 // Mixer facing methods. 927 virtual bool isTimedTrack() const { return true; } 928 virtual size_t framesReady() const; 929 930 // AudioBufferProvider interface 931 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, 932 int64_t pts); 933 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 934 935 // Client/App facing methods. 936 status_t allocateTimedBuffer(size_t size, 937 sp<IMemory>* buffer); 938 status_t queueTimedBuffer(const sp<IMemory>& buffer, 939 int64_t pts); 940 status_t setMediaTimeTransform(const LinearTransform& xform, 941 TimedAudioTrack::TargetTimeline target); 942 943 private: 944 TimedTrack(PlaybackThread *thread, 945 const sp<Client>& client, 946 audio_stream_type_t streamType, 947 uint32_t sampleRate, 948 audio_format_t format, 949 audio_channel_mask_t channelMask, 950 int frameCount, 951 const sp<IMemory>& sharedBuffer, 952 int sessionId); 953 954 void timedYieldSamples_l(AudioBufferProvider::Buffer* buffer); 955 void timedYieldSilence_l(uint32_t numFrames, 956 AudioBufferProvider::Buffer* buffer); 957 void trimTimedBufferQueue_l(); 958 void trimTimedBufferQueueHead_l(const char* logTag); 959 void updateFramesPendingAfterTrim_l(const TimedBuffer& buf, 960 const char* logTag); 961 962 uint64_t mLocalTimeFreq; 963 LinearTransform mLocalTimeToSampleTransform; 964 LinearTransform mMediaTimeToSampleTransform; 965 sp<MemoryDealer> mTimedMemoryDealer; 966 967 Vector<TimedBuffer> mTimedBufferQueue; 968 bool mQueueHeadInFlight; 969 bool mTrimQueueHeadOnRelease; 970 uint32_t mFramesPendingInQueue; 971 972 uint8_t* mTimedSilenceBuffer; 973 uint32_t mTimedSilenceBufferSize; 974 mutable Mutex mTimedBufferQueueLock; 975 bool mTimedAudioOutputOnTime; 976 CCHelper mCCHelper; 977 978 Mutex mMediaTimeTransformLock; 979 LinearTransform mMediaTimeTransform; 980 bool mMediaTimeTransformValid; 981 TimedAudioTrack::TargetTimeline mMediaTimeTransformTarget; 982 }; 983 984 985 // playback track 986 class OutputTrack : public Track { 987 public: 988 989 class Buffer: public AudioBufferProvider::Buffer { 990 public: 991 int16_t *mBuffer; 992 }; 993 994 OutputTrack(PlaybackThread *thread, 995 DuplicatingThread *sourceThread, 996 uint32_t sampleRate, 997 audio_format_t format, 998 audio_channel_mask_t channelMask, 999 int frameCount); 1000 virtual ~OutputTrack(); 1001 1002 virtual status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 1003 int triggerSession = 0); 1004 virtual void stop(); 1005 bool write(int16_t* data, uint32_t frames); 1006 bool bufferQueueEmpty() const { return mBufferQueue.size() == 0; } 1007 bool isActive() const { return mActive; } 1008 const wp<ThreadBase>& thread() const { return mThread; } 1009 1010 private: 1011 1012 enum { 1013 NO_MORE_BUFFERS = 0x80000001, // same in AudioTrack.h, ok to be different value 1014 }; 1015 1016 status_t obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs); 1017 void clearBufferQueue(); 1018 1019 // Maximum number of pending buffers allocated by OutputTrack::write() 1020 static const uint8_t kMaxOverFlowBuffers = 10; 1021 1022 Vector < Buffer* > mBufferQueue; 1023 AudioBufferProvider::Buffer mOutBuffer; 1024 bool mActive; 1025 DuplicatingThread* const mSourceThread; // for waitTimeMs() in write() 1026 }; // end of OutputTrack 1027 1028 PlaybackThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1029 audio_io_handle_t id, audio_devices_t device, type_t type); 1030 virtual ~PlaybackThread(); 1031 1032 void dump(int fd, const Vector<String16>& args); 1033 1034 // Thread virtuals 1035 virtual status_t readyToRun(); 1036 virtual bool threadLoop(); 1037 1038 // RefBase 1039 virtual void onFirstRef(); 1040 1041 protected: 1042 // Code snippets that were lifted up out of threadLoop() 1043 virtual void threadLoop_mix() = 0; 1044 virtual void threadLoop_sleepTime() = 0; 1045 virtual void threadLoop_write(); 1046 virtual void threadLoop_standby(); 1047 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 1048 1049 // prepareTracks_l reads and writes mActiveTracks, and returns 1050 // the pending set of tracks to remove via Vector 'tracksToRemove'. The caller 1051 // is responsible for clearing or destroying this Vector later on, when it 1052 // is safe to do so. That will drop the final ref count and destroy the tracks. 1053 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove) = 0; 1054 1055 // ThreadBase virtuals 1056 virtual void preExit(); 1057 1058 public: 1059 1060 virtual status_t initCheck() const { return (mOutput == NULL) ? NO_INIT : NO_ERROR; } 1061 1062 // return estimated latency in milliseconds, as reported by HAL 1063 uint32_t latency() const; 1064 // same, but lock must already be held 1065 uint32_t latency_l() const; 1066 1067 void setMasterVolume(float value); 1068 void setMasterMute(bool muted); 1069 1070 void setStreamVolume(audio_stream_type_t stream, float value); 1071 void setStreamMute(audio_stream_type_t stream, bool muted); 1072 1073 float streamVolume(audio_stream_type_t stream) const; 1074 1075 sp<Track> createTrack_l( 1076 const sp<AudioFlinger::Client>& client, 1077 audio_stream_type_t streamType, 1078 uint32_t sampleRate, 1079 audio_format_t format, 1080 audio_channel_mask_t channelMask, 1081 int frameCount, 1082 const sp<IMemory>& sharedBuffer, 1083 int sessionId, 1084 IAudioFlinger::track_flags_t flags, 1085 pid_t tid, 1086 status_t *status); 1087 1088 AudioStreamOut* getOutput() const; 1089 AudioStreamOut* clearOutput(); 1090 virtual audio_stream_t* stream() const; 1091 1092 // a very large number of suspend() will eventually wraparound, but unlikely 1093 void suspend() { (void) android_atomic_inc(&mSuspended); } 1094 void restore() 1095 { 1096 // if restore() is done without suspend(), get back into 1097 // range so that the next suspend() will operate correctly 1098 if (android_atomic_dec(&mSuspended) <= 0) { 1099 android_atomic_release_store(0, &mSuspended); 1100 } 1101 } 1102 bool isSuspended() const 1103 { return android_atomic_acquire_load(&mSuspended) > 0; } 1104 1105 virtual String8 getParameters(const String8& keys); 1106 virtual void audioConfigChanged_l(int event, int param = 0); 1107 status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames); 1108 int16_t *mixBuffer() const { return mMixBuffer; }; 1109 1110 virtual void detachAuxEffect_l(int effectId); 1111 status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track> track, 1112 int EffectId); 1113 status_t attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track> track, 1114 int EffectId); 1115 1116 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1117 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1118 virtual uint32_t hasAudioSession(int sessionId) const; 1119 virtual uint32_t getStrategyForSession_l(int sessionId); 1120 1121 1122 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 1123 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 1124 void invalidateTracks(audio_stream_type_t streamType); 1125 1126 1127 protected: 1128 int16_t* mMixBuffer; 1129 1130 // suspend count, > 0 means suspended. While suspended, the thread continues to pull from 1131 // tracks and mix, but doesn't write to HAL. A2DP and SCO HAL implementations can't handle 1132 // concurrent use of both of them, so Audio Policy Service suspends one of the threads to 1133 // workaround that restriction. 1134 // 'volatile' means accessed via atomic operations and no lock. 1135 volatile int32_t mSuspended; 1136 1137 int mBytesWritten; 1138 private: 1139 // mMasterMute is in both PlaybackThread and in AudioFlinger. When a 1140 // PlaybackThread needs to find out if master-muted, it checks it's local 1141 // copy rather than the one in AudioFlinger. This optimization saves a lock. 1142 bool mMasterMute; 1143 void setMasterMute_l(bool muted) { mMasterMute = muted; } 1144 protected: 1145 SortedVector< wp<Track> > mActiveTracks; // FIXME check if this could be sp<> 1146 1147 // Allocate a track name for a given channel mask. 1148 // Returns name >= 0 if successful, -1 on failure. 1149 virtual int getTrackName_l(audio_channel_mask_t channelMask, int sessionId) = 0; 1150 virtual void deleteTrackName_l(int name) = 0; 1151 1152 // Time to sleep between cycles when: 1153 virtual uint32_t activeSleepTimeUs() const; // mixer state MIXER_TRACKS_ENABLED 1154 virtual uint32_t idleSleepTimeUs() const = 0; // mixer state MIXER_IDLE 1155 virtual uint32_t suspendSleepTimeUs() const = 0; // audio policy manager suspended us 1156 // No sleep when mixer state == MIXER_TRACKS_READY; relies on audio HAL stream->write() 1157 // No sleep in standby mode; waits on a condition 1158 1159 // Code snippets that are temporarily lifted up out of threadLoop() until the merge 1160 void checkSilentMode_l(); 1161 1162 // Non-trivial for DUPLICATING only 1163 virtual void saveOutputTracks() { } 1164 virtual void clearOutputTracks() { } 1165 1166 // Cache various calculated values, at threadLoop() entry and after a parameter change 1167 virtual void cacheParameters_l(); 1168 1169 virtual uint32_t correctLatency(uint32_t latency) const; 1170 1171 private: 1172 1173 friend class AudioFlinger; // for numerous 1174 1175 PlaybackThread(const Client&); 1176 PlaybackThread& operator = (const PlaybackThread&); 1177 1178 status_t addTrack_l(const sp<Track>& track); 1179 void destroyTrack_l(const sp<Track>& track); 1180 void removeTrack_l(const sp<Track>& track); 1181 1182 void readOutputParameters(); 1183 1184 virtual void dumpInternals(int fd, const Vector<String16>& args); 1185 void dumpTracks(int fd, const Vector<String16>& args); 1186 1187 SortedVector< sp<Track> > mTracks; 1188 // mStreamTypes[] uses 1 additional stream type internally for the OutputTrack used by DuplicatingThread 1189 stream_type_t mStreamTypes[AUDIO_STREAM_CNT + 1]; 1190 AudioStreamOut *mOutput; 1191 1192 float mMasterVolume; 1193 nsecs_t mLastWriteTime; 1194 int mNumWrites; 1195 int mNumDelayedWrites; 1196 bool mInWrite; 1197 1198 // FIXME rename these former local variables of threadLoop to standard "m" names 1199 nsecs_t standbyTime; 1200 size_t mixBufferSize; 1201 1202 // cached copies of activeSleepTimeUs() and idleSleepTimeUs() made by cacheParameters_l() 1203 uint32_t activeSleepTime; 1204 uint32_t idleSleepTime; 1205 1206 uint32_t sleepTime; 1207 1208 // mixer status returned by prepareTracks_l() 1209 mixer_state mMixerStatus; // current cycle 1210 // previous cycle when in prepareTracks_l() 1211 mixer_state mMixerStatusIgnoringFastTracks; 1212 // FIXME or a separate ready state per track 1213 1214 // FIXME move these declarations into the specific sub-class that needs them 1215 // MIXER only 1216 uint32_t sleepTimeShift; 1217 1218 // same as AudioFlinger::mStandbyTimeInNsecs except for DIRECT which uses a shorter value 1219 nsecs_t standbyDelay; 1220 1221 // MIXER only 1222 nsecs_t maxPeriod; 1223 1224 // DUPLICATING only 1225 uint32_t writeFrames; 1226 1227 private: 1228 // The HAL output sink is treated as non-blocking, but current implementation is blocking 1229 sp<NBAIO_Sink> mOutputSink; 1230 // If a fast mixer is present, the blocking pipe sink, otherwise clear 1231 sp<NBAIO_Sink> mPipeSink; 1232 // The current sink for the normal mixer to write it's (sub)mix, mOutputSink or mPipeSink 1233 sp<NBAIO_Sink> mNormalSink; 1234 // For dumpsys 1235 sp<NBAIO_Sink> mTeeSink; 1236 sp<NBAIO_Source> mTeeSource; 1237 uint32_t mScreenState; // cached copy of gScreenState 1238 public: 1239 virtual bool hasFastMixer() const = 0; 1240 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const 1241 { FastTrackUnderruns dummy; return dummy; } 1242 1243 protected: 1244 // accessed by both binder threads and within threadLoop(), lock on mutex needed 1245 unsigned mFastTrackAvailMask; // bit i set if fast track [i] is available 1246 1247 }; 1248 1249 class MixerThread : public PlaybackThread { 1250 public: 1251 MixerThread (const sp<AudioFlinger>& audioFlinger, 1252 AudioStreamOut* output, 1253 audio_io_handle_t id, 1254 audio_devices_t device, 1255 type_t type = MIXER); 1256 virtual ~MixerThread(); 1257 1258 // Thread virtuals 1259 1260 virtual bool checkForNewParameters_l(); 1261 virtual void dumpInternals(int fd, const Vector<String16>& args); 1262 1263 protected: 1264 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1265 virtual int getTrackName_l(audio_channel_mask_t channelMask, int sessionId); 1266 virtual void deleteTrackName_l(int name); 1267 virtual uint32_t idleSleepTimeUs() const; 1268 virtual uint32_t suspendSleepTimeUs() const; 1269 virtual void cacheParameters_l(); 1270 1271 // threadLoop snippets 1272 virtual void threadLoop_write(); 1273 virtual void threadLoop_standby(); 1274 virtual void threadLoop_mix(); 1275 virtual void threadLoop_sleepTime(); 1276 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 1277 virtual uint32_t correctLatency(uint32_t latency) const; 1278 1279 AudioMixer* mAudioMixer; // normal mixer 1280 private: 1281 // one-time initialization, no locks required 1282 FastMixer* mFastMixer; // non-NULL if there is also a fast mixer 1283 sp<AudioWatchdog> mAudioWatchdog; // non-0 if there is an audio watchdog thread 1284 1285 // contents are not guaranteed to be consistent, no locks required 1286 FastMixerDumpState mFastMixerDumpState; 1287 #ifdef STATE_QUEUE_DUMP 1288 StateQueueObserverDump mStateQueueObserverDump; 1289 StateQueueMutatorDump mStateQueueMutatorDump; 1290 #endif 1291 AudioWatchdogDump mAudioWatchdogDump; 1292 1293 // accessible only within the threadLoop(), no locks required 1294 // mFastMixer->sq() // for mutating and pushing state 1295 int32_t mFastMixerFutex; // for cold idle 1296 1297 public: 1298 virtual bool hasFastMixer() const { return mFastMixer != NULL; } 1299 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const { 1300 ALOG_ASSERT(fastIndex < FastMixerState::kMaxFastTracks); 1301 return mFastMixerDumpState.mTracks[fastIndex].mUnderruns; 1302 } 1303 }; 1304 1305 class DirectOutputThread : public PlaybackThread { 1306 public: 1307 1308 DirectOutputThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1309 audio_io_handle_t id, audio_devices_t device); 1310 virtual ~DirectOutputThread(); 1311 1312 // Thread virtuals 1313 1314 virtual bool checkForNewParameters_l(); 1315 1316 protected: 1317 virtual int getTrackName_l(audio_channel_mask_t channelMask, int sessionId); 1318 virtual void deleteTrackName_l(int name); 1319 virtual uint32_t activeSleepTimeUs() const; 1320 virtual uint32_t idleSleepTimeUs() const; 1321 virtual uint32_t suspendSleepTimeUs() const; 1322 virtual void cacheParameters_l(); 1323 1324 // threadLoop snippets 1325 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1326 virtual void threadLoop_mix(); 1327 virtual void threadLoop_sleepTime(); 1328 1329 // volumes last sent to audio HAL with stream->set_volume() 1330 float mLeftVolFloat; 1331 float mRightVolFloat; 1332 1333 private: 1334 // prepareTracks_l() tells threadLoop_mix() the name of the single active track 1335 sp<Track> mActiveTrack; 1336 public: 1337 virtual bool hasFastMixer() const { return false; } 1338 }; 1339 1340 class DuplicatingThread : public MixerThread { 1341 public: 1342 DuplicatingThread (const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread, 1343 audio_io_handle_t id); 1344 virtual ~DuplicatingThread(); 1345 1346 // Thread virtuals 1347 void addOutputTrack(MixerThread* thread); 1348 void removeOutputTrack(MixerThread* thread); 1349 uint32_t waitTimeMs() const { return mWaitTimeMs; } 1350 protected: 1351 virtual uint32_t activeSleepTimeUs() const; 1352 1353 private: 1354 bool outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks); 1355 protected: 1356 // threadLoop snippets 1357 virtual void threadLoop_mix(); 1358 virtual void threadLoop_sleepTime(); 1359 virtual void threadLoop_write(); 1360 virtual void threadLoop_standby(); 1361 virtual void cacheParameters_l(); 1362 1363 private: 1364 // called from threadLoop, addOutputTrack, removeOutputTrack 1365 virtual void updateWaitTime_l(); 1366 protected: 1367 virtual void saveOutputTracks(); 1368 virtual void clearOutputTracks(); 1369 private: 1370 1371 uint32_t mWaitTimeMs; 1372 SortedVector < sp<OutputTrack> > outputTracks; 1373 SortedVector < sp<OutputTrack> > mOutputTracks; 1374 public: 1375 virtual bool hasFastMixer() const { return false; } 1376 }; 1377 1378 PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const; 1379 MixerThread *checkMixerThread_l(audio_io_handle_t output) const; 1380 RecordThread *checkRecordThread_l(audio_io_handle_t input) const; 1381 // no range check, AudioFlinger::mLock held 1382 bool streamMute_l(audio_stream_type_t stream) const 1383 { return mStreamTypes[stream].mute; } 1384 // no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held 1385 float streamVolume_l(audio_stream_type_t stream) const 1386 { return mStreamTypes[stream].volume; } 1387 void audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2); 1388 1389 // allocate an audio_io_handle_t, session ID, or effect ID 1390 uint32_t nextUniqueId(); 1391 1392 status_t moveEffectChain_l(int sessionId, 1393 PlaybackThread *srcThread, 1394 PlaybackThread *dstThread, 1395 bool reRegister); 1396 // return thread associated with primary hardware device, or NULL 1397 PlaybackThread *primaryPlaybackThread_l() const; 1398 audio_devices_t primaryOutputDevice_l() const; 1399 1400 sp<PlaybackThread> getEffectThread_l(int sessionId, int EffectId); 1401 1402 // server side of the client's IAudioTrack 1403 class TrackHandle : public android::BnAudioTrack { 1404 public: 1405 TrackHandle(const sp<PlaybackThread::Track>& track); 1406 virtual ~TrackHandle(); 1407 virtual sp<IMemory> getCblk() const; 1408 virtual status_t start(); 1409 virtual void stop(); 1410 virtual void flush(); 1411 virtual void mute(bool); 1412 virtual void pause(); 1413 virtual status_t attachAuxEffect(int effectId); 1414 virtual status_t allocateTimedBuffer(size_t size, 1415 sp<IMemory>* buffer); 1416 virtual status_t queueTimedBuffer(const sp<IMemory>& buffer, 1417 int64_t pts); 1418 virtual status_t setMediaTimeTransform(const LinearTransform& xform, 1419 int target); 1420 virtual status_t onTransact( 1421 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 1422 private: 1423 const sp<PlaybackThread::Track> mTrack; 1424 }; 1425 1426 void removeClient_l(pid_t pid); 1427 void removeNotificationClient(pid_t pid); 1428 1429 1430 // record thread 1431 class RecordThread : public ThreadBase, public AudioBufferProvider 1432 // derives from AudioBufferProvider interface for use by resampler 1433 { 1434 public: 1435 1436 // record track 1437 class RecordTrack : public TrackBase { 1438 public: 1439 RecordTrack(RecordThread *thread, 1440 const sp<Client>& client, 1441 uint32_t sampleRate, 1442 audio_format_t format, 1443 audio_channel_mask_t channelMask, 1444 int frameCount, 1445 int sessionId); 1446 virtual ~RecordTrack(); 1447 1448 virtual status_t start(AudioSystem::sync_event_t event, int triggerSession); 1449 virtual void stop(); 1450 1451 void destroy(); 1452 1453 // clear the buffer overflow flag 1454 void clearOverflow() { mOverflow = false; } 1455 // set the buffer overflow flag and return previous value 1456 bool setOverflow() { bool tmp = mOverflow; mOverflow = true; return tmp; } 1457 1458 static void appendDumpHeader(String8& result); 1459 void dump(char* buffer, size_t size); 1460 1461 private: 1462 friend class AudioFlinger; // for mState 1463 1464 RecordTrack(const RecordTrack&); 1465 RecordTrack& operator = (const RecordTrack&); 1466 1467 // AudioBufferProvider interface 1468 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts = kInvalidPTS); 1469 // releaseBuffer() not overridden 1470 1471 bool mOverflow; // overflow on most recent attempt to fill client buffer 1472 }; 1473 1474 RecordThread(const sp<AudioFlinger>& audioFlinger, 1475 AudioStreamIn *input, 1476 uint32_t sampleRate, 1477 audio_channel_mask_t channelMask, 1478 audio_io_handle_t id, 1479 audio_devices_t device); 1480 virtual ~RecordThread(); 1481 1482 // no addTrack_l ? 1483 void destroyTrack_l(const sp<RecordTrack>& track); 1484 void removeTrack_l(const sp<RecordTrack>& track); 1485 1486 void dumpInternals(int fd, const Vector<String16>& args); 1487 void dumpTracks(int fd, const Vector<String16>& args); 1488 1489 // Thread virtuals 1490 virtual bool threadLoop(); 1491 virtual status_t readyToRun(); 1492 1493 // RefBase 1494 virtual void onFirstRef(); 1495 1496 virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; } 1497 sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l( 1498 const sp<AudioFlinger::Client>& client, 1499 uint32_t sampleRate, 1500 audio_format_t format, 1501 audio_channel_mask_t channelMask, 1502 int frameCount, 1503 int sessionId, 1504 IAudioFlinger::track_flags_t flags, 1505 pid_t tid, 1506 status_t *status); 1507 1508 status_t start(RecordTrack* recordTrack, 1509 AudioSystem::sync_event_t event, 1510 int triggerSession); 1511 1512 // ask the thread to stop the specified track, and 1513 // return true if the caller should then do it's part of the stopping process 1514 bool stop_l(RecordTrack* recordTrack); 1515 1516 void dump(int fd, const Vector<String16>& args); 1517 AudioStreamIn* clearInput(); 1518 virtual audio_stream_t* stream() const; 1519 1520 // AudioBufferProvider interface 1521 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts); 1522 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 1523 1524 virtual bool checkForNewParameters_l(); 1525 virtual String8 getParameters(const String8& keys); 1526 virtual void audioConfigChanged_l(int event, int param = 0); 1527 void readInputParameters(); 1528 virtual unsigned int getInputFramesLost(); 1529 1530 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1531 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1532 virtual uint32_t hasAudioSession(int sessionId) const; 1533 1534 // Return the set of unique session IDs across all tracks. 1535 // The keys are the session IDs, and the associated values are meaningless. 1536 // FIXME replace by Set [and implement Bag/Multiset for other uses]. 1537 KeyedVector<int, bool> sessionIds() const; 1538 1539 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 1540 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 1541 1542 static void syncStartEventCallback(const wp<SyncEvent>& event); 1543 void handleSyncStartEvent(const sp<SyncEvent>& event); 1544 1545 private: 1546 void clearSyncStartEvent(); 1547 1548 // Enter standby if not already in standby, and set mStandby flag 1549 void standby(); 1550 1551 // Call the HAL standby method unconditionally, and don't change mStandby flag 1552 void inputStandBy(); 1553 1554 AudioStreamIn *mInput; 1555 SortedVector < sp<RecordTrack> > mTracks; 1556 // mActiveTrack has dual roles: it indicates the current active track, and 1557 // is used together with mStartStopCond to indicate start()/stop() progress 1558 sp<RecordTrack> mActiveTrack; 1559 Condition mStartStopCond; 1560 AudioResampler *mResampler; 1561 int32_t *mRsmpOutBuffer; 1562 int16_t *mRsmpInBuffer; 1563 size_t mRsmpInIndex; 1564 size_t mInputBytes; 1565 const int mReqChannelCount; 1566 const uint32_t mReqSampleRate; 1567 ssize_t mBytesRead; 1568 // sync event triggering actual audio capture. Frames read before this event will 1569 // be dropped and therefore not read by the application. 1570 sp<SyncEvent> mSyncStartEvent; 1571 // number of captured frames to drop after the start sync event has been received. 1572 // when < 0, maximum frames to drop before starting capture even if sync event is 1573 // not received 1574 ssize_t mFramestoDrop; 1575 }; 1576 1577 // server side of the client's IAudioRecord 1578 class RecordHandle : public android::BnAudioRecord { 1579 public: 1580 RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack); 1581 virtual ~RecordHandle(); 1582 virtual sp<IMemory> getCblk() const; 1583 virtual status_t start(int /*AudioSystem::sync_event_t*/ event, int triggerSession); 1584 virtual void stop(); 1585 virtual status_t onTransact( 1586 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 1587 private: 1588 const sp<RecordThread::RecordTrack> mRecordTrack; 1589 1590 // for use from destructor 1591 void stop_nonvirtual(); 1592 }; 1593 1594 //--- Audio Effect Management 1595 1596 // EffectModule and EffectChain classes both have their own mutex to protect 1597 // state changes or resource modifications. Always respect the following order 1598 // if multiple mutexes must be acquired to avoid cross deadlock: 1599 // AudioFlinger -> ThreadBase -> EffectChain -> EffectModule 1600 1601 // The EffectModule class is a wrapper object controlling the effect engine implementation 1602 // in the effect library. It prevents concurrent calls to process() and command() functions 1603 // from different client threads. It keeps a list of EffectHandle objects corresponding 1604 // to all client applications using this effect and notifies applications of effect state, 1605 // control or parameter changes. It manages the activation state machine to send appropriate 1606 // reset, enable, disable commands to effect engine and provide volume 1607 // ramping when effects are activated/deactivated. 1608 // When controlling an auxiliary effect, the EffectModule also provides an input buffer used by 1609 // the attached track(s) to accumulate their auxiliary channel. 1610 class EffectModule: public RefBase { 1611 public: 1612 EffectModule(ThreadBase *thread, 1613 const wp<AudioFlinger::EffectChain>& chain, 1614 effect_descriptor_t *desc, 1615 int id, 1616 int sessionId); 1617 virtual ~EffectModule(); 1618 1619 enum effect_state { 1620 IDLE, 1621 RESTART, 1622 STARTING, 1623 ACTIVE, 1624 STOPPING, 1625 STOPPED, 1626 DESTROYED 1627 }; 1628 1629 int id() const { return mId; } 1630 void process(); 1631 void updateState(); 1632 status_t command(uint32_t cmdCode, 1633 uint32_t cmdSize, 1634 void *pCmdData, 1635 uint32_t *replySize, 1636 void *pReplyData); 1637 1638 void reset_l(); 1639 status_t configure(); 1640 status_t init(); 1641 effect_state state() const { 1642 return mState; 1643 } 1644 uint32_t status() { 1645 return mStatus; 1646 } 1647 int sessionId() const { 1648 return mSessionId; 1649 } 1650 status_t setEnabled(bool enabled); 1651 status_t setEnabled_l(bool enabled); 1652 bool isEnabled() const; 1653 bool isProcessEnabled() const; 1654 1655 void setInBuffer(int16_t *buffer) { mConfig.inputCfg.buffer.s16 = buffer; } 1656 int16_t *inBuffer() { return mConfig.inputCfg.buffer.s16; } 1657 void setOutBuffer(int16_t *buffer) { mConfig.outputCfg.buffer.s16 = buffer; } 1658 int16_t *outBuffer() { return mConfig.outputCfg.buffer.s16; } 1659 void setChain(const wp<EffectChain>& chain) { mChain = chain; } 1660 void setThread(const wp<ThreadBase>& thread) { mThread = thread; } 1661 const wp<ThreadBase>& thread() { return mThread; } 1662 1663 status_t addHandle(EffectHandle *handle); 1664 size_t disconnect(EffectHandle *handle, bool unpinIfLast); 1665 size_t removeHandle(EffectHandle *handle); 1666 1667 const effect_descriptor_t& desc() const { return mDescriptor; } 1668 wp<EffectChain>& chain() { return mChain; } 1669 1670 status_t setDevice(audio_devices_t device); 1671 status_t setVolume(uint32_t *left, uint32_t *right, bool controller); 1672 status_t setMode(audio_mode_t mode); 1673 status_t setAudioSource(audio_source_t source); 1674 status_t start(); 1675 status_t stop(); 1676 void setSuspended(bool suspended); 1677 bool suspended() const; 1678 1679 EffectHandle* controlHandle_l(); 1680 1681 bool isPinned() const { return mPinned; } 1682 void unPin() { mPinned = false; } 1683 bool purgeHandles(); 1684 void lock() { mLock.lock(); } 1685 void unlock() { mLock.unlock(); } 1686 1687 void dump(int fd, const Vector<String16>& args); 1688 1689 protected: 1690 friend class AudioFlinger; // for mHandles 1691 bool mPinned; 1692 1693 // Maximum time allocated to effect engines to complete the turn off sequence 1694 static const uint32_t MAX_DISABLE_TIME_MS = 10000; 1695 1696 EffectModule(const EffectModule&); 1697 EffectModule& operator = (const EffectModule&); 1698 1699 status_t start_l(); 1700 status_t stop_l(); 1701 1702 mutable Mutex mLock; // mutex for process, commands and handles list protection 1703 wp<ThreadBase> mThread; // parent thread 1704 wp<EffectChain> mChain; // parent effect chain 1705 const int mId; // this instance unique ID 1706 const int mSessionId; // audio session ID 1707 const effect_descriptor_t mDescriptor;// effect descriptor received from effect engine 1708 effect_config_t mConfig; // input and output audio configuration 1709 effect_handle_t mEffectInterface; // Effect module C API 1710 status_t mStatus; // initialization status 1711 effect_state mState; // current activation state 1712 Vector<EffectHandle *> mHandles; // list of client handles 1713 // First handle in mHandles has highest priority and controls the effect module 1714 uint32_t mMaxDisableWaitCnt; // maximum grace period before forcing an effect off after 1715 // sending disable command. 1716 uint32_t mDisableWaitCnt; // current process() calls count during disable period. 1717 bool mSuspended; // effect is suspended: temporarily disabled by framework 1718 }; 1719 1720 // The EffectHandle class implements the IEffect interface. It provides resources 1721 // to receive parameter updates, keeps track of effect control 1722 // ownership and state and has a pointer to the EffectModule object it is controlling. 1723 // There is one EffectHandle object for each application controlling (or using) 1724 // an effect module. 1725 // The EffectHandle is obtained by calling AudioFlinger::createEffect(). 1726 class EffectHandle: public android::BnEffect { 1727 public: 1728 1729 EffectHandle(const sp<EffectModule>& effect, 1730 const sp<AudioFlinger::Client>& client, 1731 const sp<IEffectClient>& effectClient, 1732 int32_t priority); 1733 virtual ~EffectHandle(); 1734 1735 // IEffect 1736 virtual status_t enable(); 1737 virtual status_t disable(); 1738 virtual status_t command(uint32_t cmdCode, 1739 uint32_t cmdSize, 1740 void *pCmdData, 1741 uint32_t *replySize, 1742 void *pReplyData); 1743 virtual void disconnect(); 1744 private: 1745 void disconnect(bool unpinIfLast); 1746 public: 1747 virtual sp<IMemory> getCblk() const { return mCblkMemory; } 1748 virtual status_t onTransact(uint32_t code, const Parcel& data, 1749 Parcel* reply, uint32_t flags); 1750 1751 1752 // Give or take control of effect module 1753 // - hasControl: true if control is given, false if removed 1754 // - signal: true client app should be signaled of change, false otherwise 1755 // - enabled: state of the effect when control is passed 1756 void setControl(bool hasControl, bool signal, bool enabled); 1757 void commandExecuted(uint32_t cmdCode, 1758 uint32_t cmdSize, 1759 void *pCmdData, 1760 uint32_t replySize, 1761 void *pReplyData); 1762 void setEnabled(bool enabled); 1763 bool enabled() const { return mEnabled; } 1764 1765 // Getters 1766 int id() const { return mEffect->id(); } 1767 int priority() const { return mPriority; } 1768 bool hasControl() const { return mHasControl; } 1769 sp<EffectModule> effect() const { return mEffect; } 1770 // destroyed_l() must be called with the associated EffectModule mLock held 1771 bool destroyed_l() const { return mDestroyed; } 1772 1773 void dump(char* buffer, size_t size); 1774 1775 protected: 1776 friend class AudioFlinger; // for mEffect, mHasControl, mEnabled 1777 EffectHandle(const EffectHandle&); 1778 EffectHandle& operator =(const EffectHandle&); 1779 1780 sp<EffectModule> mEffect; // pointer to controlled EffectModule 1781 sp<IEffectClient> mEffectClient; // callback interface for client notifications 1782 /*const*/ sp<Client> mClient; // client for shared memory allocation, see disconnect() 1783 sp<IMemory> mCblkMemory; // shared memory for control block 1784 effect_param_cblk_t* mCblk; // control block for deferred parameter setting via shared memory 1785 uint8_t* mBuffer; // pointer to parameter area in shared memory 1786 int mPriority; // client application priority to control the effect 1787 bool mHasControl; // true if this handle is controlling the effect 1788 bool mEnabled; // cached enable state: needed when the effect is 1789 // restored after being suspended 1790 bool mDestroyed; // Set to true by destructor. Access with EffectModule 1791 // mLock held 1792 }; 1793 1794 // the EffectChain class represents a group of effects associated to one audio session. 1795 // There can be any number of EffectChain objects per output mixer thread (PlaybackThread). 1796 // The EffecChain with session ID 0 contains global effects applied to the output mix. 1797 // Effects in this chain can be insert or auxiliary. Effects in other chains (attached to tracks) 1798 // are insert only. The EffectChain maintains an ordered list of effect module, the order corresponding 1799 // in the effect process order. When attached to a track (session ID != 0), it also provide it's own 1800 // input buffer used by the track as accumulation buffer. 1801 class EffectChain: public RefBase { 1802 public: 1803 EffectChain(const wp<ThreadBase>& wThread, int sessionId); 1804 EffectChain(ThreadBase *thread, int sessionId); 1805 virtual ~EffectChain(); 1806 1807 // special key used for an entry in mSuspendedEffects keyed vector 1808 // corresponding to a suspend all request. 1809 static const int kKeyForSuspendAll = 0; 1810 1811 // minimum duration during which we force calling effect process when last track on 1812 // a session is stopped or removed to allow effect tail to be rendered 1813 static const int kProcessTailDurationMs = 1000; 1814 1815 void process_l(); 1816 1817 void lock() { 1818 mLock.lock(); 1819 } 1820 void unlock() { 1821 mLock.unlock(); 1822 } 1823 1824 status_t addEffect_l(const sp<EffectModule>& handle); 1825 size_t removeEffect_l(const sp<EffectModule>& handle); 1826 1827 int sessionId() const { return mSessionId; } 1828 void setSessionId(int sessionId) { mSessionId = sessionId; } 1829 1830 sp<EffectModule> getEffectFromDesc_l(effect_descriptor_t *descriptor); 1831 sp<EffectModule> getEffectFromId_l(int id); 1832 sp<EffectModule> getEffectFromType_l(const effect_uuid_t *type); 1833 bool setVolume_l(uint32_t *left, uint32_t *right); 1834 void setDevice_l(audio_devices_t device); 1835 void setMode_l(audio_mode_t mode); 1836 void setAudioSource_l(audio_source_t source); 1837 1838 void setInBuffer(int16_t *buffer, bool ownsBuffer = false) { 1839 mInBuffer = buffer; 1840 mOwnInBuffer = ownsBuffer; 1841 } 1842 int16_t *inBuffer() const { 1843 return mInBuffer; 1844 } 1845 void setOutBuffer(int16_t *buffer) { 1846 mOutBuffer = buffer; 1847 } 1848 int16_t *outBuffer() const { 1849 return mOutBuffer; 1850 } 1851 1852 void incTrackCnt() { android_atomic_inc(&mTrackCnt); } 1853 void decTrackCnt() { android_atomic_dec(&mTrackCnt); } 1854 int32_t trackCnt() const { return android_atomic_acquire_load(&mTrackCnt); } 1855 1856 void incActiveTrackCnt() { android_atomic_inc(&mActiveTrackCnt); 1857 mTailBufferCount = mMaxTailBuffers; } 1858 void decActiveTrackCnt() { android_atomic_dec(&mActiveTrackCnt); } 1859 int32_t activeTrackCnt() const { return android_atomic_acquire_load(&mActiveTrackCnt); } 1860 1861 uint32_t strategy() const { return mStrategy; } 1862 void setStrategy(uint32_t strategy) 1863 { mStrategy = strategy; } 1864 1865 // suspend effect of the given type 1866 void setEffectSuspended_l(const effect_uuid_t *type, 1867 bool suspend); 1868 // suspend all eligible effects 1869 void setEffectSuspendedAll_l(bool suspend); 1870 // check if effects should be suspend or restored when a given effect is enable or disabled 1871 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1872 bool enabled); 1873 1874 void clearInputBuffer(); 1875 1876 void dump(int fd, const Vector<String16>& args); 1877 1878 protected: 1879 friend class AudioFlinger; // for mThread, mEffects 1880 EffectChain(const EffectChain&); 1881 EffectChain& operator =(const EffectChain&); 1882 1883 class SuspendedEffectDesc : public RefBase { 1884 public: 1885 SuspendedEffectDesc() : mRefCount(0) {} 1886 1887 int mRefCount; 1888 effect_uuid_t mType; 1889 wp<EffectModule> mEffect; 1890 }; 1891 1892 // get a list of effect modules to suspend when an effect of the type 1893 // passed is enabled. 1894 void getSuspendEligibleEffects(Vector< sp<EffectModule> > &effects); 1895 1896 // get an effect module if it is currently enable 1897 sp<EffectModule> getEffectIfEnabled(const effect_uuid_t *type); 1898 // true if the effect whose descriptor is passed can be suspended 1899 // OEMs can modify the rules implemented in this method to exclude specific effect 1900 // types or implementations from the suspend/restore mechanism. 1901 bool isEffectEligibleForSuspend(const effect_descriptor_t& desc); 1902 1903 void clearInputBuffer_l(sp<ThreadBase> thread); 1904 1905 wp<ThreadBase> mThread; // parent mixer thread 1906 Mutex mLock; // mutex protecting effect list 1907 Vector< sp<EffectModule> > mEffects; // list of effect modules 1908 int mSessionId; // audio session ID 1909 int16_t *mInBuffer; // chain input buffer 1910 int16_t *mOutBuffer; // chain output buffer 1911 1912 // 'volatile' here means these are accessed with atomic operations instead of mutex 1913 volatile int32_t mActiveTrackCnt; // number of active tracks connected 1914 volatile int32_t mTrackCnt; // number of tracks connected 1915 1916 int32_t mTailBufferCount; // current effect tail buffer count 1917 int32_t mMaxTailBuffers; // maximum effect tail buffers 1918 bool mOwnInBuffer; // true if the chain owns its input buffer 1919 int mVolumeCtrlIdx; // index of insert effect having control over volume 1920 uint32_t mLeftVolume; // previous volume on left channel 1921 uint32_t mRightVolume; // previous volume on right channel 1922 uint32_t mNewLeftVolume; // new volume on left channel 1923 uint32_t mNewRightVolume; // new volume on right channel 1924 uint32_t mStrategy; // strategy for this effect chain 1925 // mSuspendedEffects lists all effects currently suspended in the chain. 1926 // Use effect type UUID timelow field as key. There is no real risk of identical 1927 // timeLow fields among effect type UUIDs. 1928 // Updated by updateSuspendedSessions_l() only. 1929 KeyedVector< int, sp<SuspendedEffectDesc> > mSuspendedEffects; 1930 }; 1931 1932 class AudioHwDevice { 1933 public: 1934 enum Flags { 1935 AHWD_CAN_SET_MASTER_VOLUME = 0x1, 1936 AHWD_CAN_SET_MASTER_MUTE = 0x2, 1937 }; 1938 1939 AudioHwDevice(const char *moduleName, 1940 audio_hw_device_t *hwDevice, 1941 Flags flags) 1942 : mModuleName(strdup(moduleName)) 1943 , mHwDevice(hwDevice) 1944 , mFlags(flags) { } 1945 /*virtual*/ ~AudioHwDevice() { free((void *)mModuleName); } 1946 1947 bool canSetMasterVolume() const { 1948 return (0 != (mFlags & AHWD_CAN_SET_MASTER_VOLUME)); 1949 } 1950 1951 bool canSetMasterMute() const { 1952 return (0 != (mFlags & AHWD_CAN_SET_MASTER_MUTE)); 1953 } 1954 1955 const char *moduleName() const { return mModuleName; } 1956 audio_hw_device_t *hwDevice() const { return mHwDevice; } 1957 private: 1958 const char * const mModuleName; 1959 audio_hw_device_t * const mHwDevice; 1960 Flags mFlags; 1961 }; 1962 1963 // AudioStreamOut and AudioStreamIn are immutable, so their fields are const. 1964 // For emphasis, we could also make all pointers to them be "const *", 1965 // but that would clutter the code unnecessarily. 1966 1967 struct AudioStreamOut { 1968 AudioHwDevice* const audioHwDev; 1969 audio_stream_out_t* const stream; 1970 1971 audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); } 1972 1973 AudioStreamOut(AudioHwDevice *dev, audio_stream_out_t *out) : 1974 audioHwDev(dev), stream(out) {} 1975 }; 1976 1977 struct AudioStreamIn { 1978 AudioHwDevice* const audioHwDev; 1979 audio_stream_in_t* const stream; 1980 1981 audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); } 1982 1983 AudioStreamIn(AudioHwDevice *dev, audio_stream_in_t *in) : 1984 audioHwDev(dev), stream(in) {} 1985 }; 1986 1987 // for mAudioSessionRefs only 1988 struct AudioSessionRef { 1989 AudioSessionRef(int sessionid, pid_t pid) : 1990 mSessionid(sessionid), mPid(pid), mCnt(1) {} 1991 const int mSessionid; 1992 const pid_t mPid; 1993 int mCnt; 1994 }; 1995 1996 mutable Mutex mLock; 1997 1998 DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client() 1999 2000 mutable Mutex mHardwareLock; 2001 // NOTE: If both mLock and mHardwareLock mutexes must be held, 2002 // always take mLock before mHardwareLock 2003 2004 // These two fields are immutable after onFirstRef(), so no lock needed to access 2005 AudioHwDevice* mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL 2006 DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*> mAudioHwDevs; 2007 2008 // for dump, indicates which hardware operation is currently in progress (but not stream ops) 2009 enum hardware_call_state { 2010 AUDIO_HW_IDLE = 0, // no operation in progress 2011 AUDIO_HW_INIT, // init_check 2012 AUDIO_HW_OUTPUT_OPEN, // open_output_stream 2013 AUDIO_HW_OUTPUT_CLOSE, // unused 2014 AUDIO_HW_INPUT_OPEN, // unused 2015 AUDIO_HW_INPUT_CLOSE, // unused 2016 AUDIO_HW_STANDBY, // unused 2017 AUDIO_HW_SET_MASTER_VOLUME, // set_master_volume 2018 AUDIO_HW_GET_ROUTING, // unused 2019 AUDIO_HW_SET_ROUTING, // unused 2020 AUDIO_HW_GET_MODE, // unused 2021 AUDIO_HW_SET_MODE, // set_mode 2022 AUDIO_HW_GET_MIC_MUTE, // get_mic_mute 2023 AUDIO_HW_SET_MIC_MUTE, // set_mic_mute 2024 AUDIO_HW_SET_VOICE_VOLUME, // set_voice_volume 2025 AUDIO_HW_SET_PARAMETER, // set_parameters 2026 AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size 2027 AUDIO_HW_GET_MASTER_VOLUME, // get_master_volume 2028 AUDIO_HW_GET_PARAMETER, // get_parameters 2029 AUDIO_HW_SET_MASTER_MUTE, // set_master_mute 2030 AUDIO_HW_GET_MASTER_MUTE, // get_master_mute 2031 }; 2032 2033 mutable hardware_call_state mHardwareStatus; // for dump only 2034 2035 2036 DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads; 2037 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 2038 2039 // member variables below are protected by mLock 2040 float mMasterVolume; 2041 bool mMasterMute; 2042 // end of variables protected by mLock 2043 2044 DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads; 2045 2046 DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients; 2047 volatile int32_t mNextUniqueId; // updated by android_atomic_inc 2048 audio_mode_t mMode; 2049 bool mBtNrecIsOff; 2050 2051 // protected by mLock 2052 Vector<AudioSessionRef*> mAudioSessionRefs; 2053 2054 float masterVolume_l() const; 2055 bool masterMute_l() const; 2056 audio_module_handle_t loadHwModule_l(const char *name); 2057 2058 Vector < sp<SyncEvent> > mPendingSyncEvents; // sync events awaiting for a session 2059 // to be created 2060 2061 private: 2062 sp<Client> registerPid_l(pid_t pid); // always returns non-0 2063 2064 // for use from destructor 2065 status_t closeOutput_nonvirtual(audio_io_handle_t output); 2066 status_t closeInput_nonvirtual(audio_io_handle_t input); 2067 }; 2068 2069 2070 // ---------------------------------------------------------------------------- 2071 2072 }; // namespace android 2073 2074 #endif // ANDROID_AUDIO_FLINGER_H 2075
