1 /* 2 ** 3 ** Copyright 2012, 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 INCLUDING_FROM_AUDIOFLINGER_H 19 #error This header file should only be included from AudioFlinger.h 20 #endif 21 22 class ThreadBase : public Thread { 23 public: 24 25 #include "TrackBase.h" 26 27 enum type_t { 28 MIXER, // Thread class is MixerThread 29 DIRECT, // Thread class is DirectOutputThread 30 DUPLICATING, // Thread class is DuplicatingThread 31 RECORD, // Thread class is RecordThread 32 OFFLOAD // Thread class is OffloadThread 33 }; 34 35 static const char *threadTypeToString(type_t type); 36 37 ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 38 audio_devices_t outDevice, audio_devices_t inDevice, type_t type, 39 bool systemReady); 40 virtual ~ThreadBase(); 41 42 virtual status_t readyToRun(); 43 44 void dumpBase(int fd, const Vector<String16>& args); 45 void dumpEffectChains(int fd, const Vector<String16>& args); 46 47 void clearPowerManager(); 48 49 // base for record and playback 50 enum { 51 CFG_EVENT_IO, 52 CFG_EVENT_PRIO, 53 CFG_EVENT_SET_PARAMETER, 54 CFG_EVENT_CREATE_AUDIO_PATCH, 55 CFG_EVENT_RELEASE_AUDIO_PATCH, 56 }; 57 58 class ConfigEventData: public RefBase { 59 public: ~ConfigEventData()60 virtual ~ConfigEventData() {} 61 62 virtual void dump(char *buffer, size_t size) = 0; 63 protected: ConfigEventData()64 ConfigEventData() {} 65 }; 66 67 // Config event sequence by client if status needed (e.g binder thread calling setParameters()): 68 // 1. create SetParameterConfigEvent. This sets mWaitStatus in config event 69 // 2. Lock mLock 70 // 3. Call sendConfigEvent_l(): Append to mConfigEvents and mWaitWorkCV.signal 71 // 4. sendConfigEvent_l() reads status from event->mStatus; 72 // 5. sendConfigEvent_l() returns status 73 // 6. Unlock 74 // 75 // Parameter sequence by server: threadLoop calling processConfigEvents_l(): 76 // 1. Lock mLock 77 // 2. If there is an entry in mConfigEvents proceed ... 78 // 3. Read first entry in mConfigEvents 79 // 4. Remove first entry from mConfigEvents 80 // 5. Process 81 // 6. Set event->mStatus 82 // 7. event->mCond.signal 83 // 8. Unlock 84 85 class ConfigEvent: public RefBase { 86 public: ~ConfigEvent()87 virtual ~ConfigEvent() {} 88 dump(char * buffer,size_t size)89 void dump(char *buffer, size_t size) { mData->dump(buffer, size); } 90 91 const int mType; // event type e.g. CFG_EVENT_IO 92 Mutex mLock; // mutex associated with mCond 93 Condition mCond; // condition for status return 94 status_t mStatus; // status communicated to sender 95 bool mWaitStatus; // true if sender is waiting for status 96 bool mRequiresSystemReady; // true if must wait for system ready to enter event queue 97 sp<ConfigEventData> mData; // event specific parameter data 98 99 protected: 100 ConfigEvent(int type, bool requiresSystemReady = false) : mType(type)101 mType(type), mStatus(NO_ERROR), mWaitStatus(false), 102 mRequiresSystemReady(requiresSystemReady), mData(NULL) {} 103 }; 104 105 class IoConfigEventData : public ConfigEventData { 106 public: IoConfigEventData(audio_io_config_event event,pid_t pid)107 IoConfigEventData(audio_io_config_event event, pid_t pid) : 108 mEvent(event), mPid(pid) {} 109 dump(char * buffer,size_t size)110 virtual void dump(char *buffer, size_t size) { 111 snprintf(buffer, size, "IO event: event %d\n", mEvent); 112 } 113 114 const audio_io_config_event mEvent; 115 const pid_t mPid; 116 }; 117 118 class IoConfigEvent : public ConfigEvent { 119 public: IoConfigEvent(audio_io_config_event event,pid_t pid)120 IoConfigEvent(audio_io_config_event event, pid_t pid) : 121 ConfigEvent(CFG_EVENT_IO) { 122 mData = new IoConfigEventData(event, pid); 123 } ~IoConfigEvent()124 virtual ~IoConfigEvent() {} 125 }; 126 127 class PrioConfigEventData : public ConfigEventData { 128 public: PrioConfigEventData(pid_t pid,pid_t tid,int32_t prio)129 PrioConfigEventData(pid_t pid, pid_t tid, int32_t prio) : 130 mPid(pid), mTid(tid), mPrio(prio) {} 131 dump(char * buffer,size_t size)132 virtual void dump(char *buffer, size_t size) { 133 snprintf(buffer, size, "Prio event: pid %d, tid %d, prio %d\n", mPid, mTid, mPrio); 134 } 135 136 const pid_t mPid; 137 const pid_t mTid; 138 const int32_t mPrio; 139 }; 140 141 class PrioConfigEvent : public ConfigEvent { 142 public: PrioConfigEvent(pid_t pid,pid_t tid,int32_t prio)143 PrioConfigEvent(pid_t pid, pid_t tid, int32_t prio) : 144 ConfigEvent(CFG_EVENT_PRIO, true) { 145 mData = new PrioConfigEventData(pid, tid, prio); 146 } ~PrioConfigEvent()147 virtual ~PrioConfigEvent() {} 148 }; 149 150 class SetParameterConfigEventData : public ConfigEventData { 151 public: SetParameterConfigEventData(String8 keyValuePairs)152 SetParameterConfigEventData(String8 keyValuePairs) : 153 mKeyValuePairs(keyValuePairs) {} 154 dump(char * buffer,size_t size)155 virtual void dump(char *buffer, size_t size) { 156 snprintf(buffer, size, "KeyValue: %s\n", mKeyValuePairs.string()); 157 } 158 159 const String8 mKeyValuePairs; 160 }; 161 162 class SetParameterConfigEvent : public ConfigEvent { 163 public: SetParameterConfigEvent(String8 keyValuePairs)164 SetParameterConfigEvent(String8 keyValuePairs) : 165 ConfigEvent(CFG_EVENT_SET_PARAMETER) { 166 mData = new SetParameterConfigEventData(keyValuePairs); 167 mWaitStatus = true; 168 } ~SetParameterConfigEvent()169 virtual ~SetParameterConfigEvent() {} 170 }; 171 172 class CreateAudioPatchConfigEventData : public ConfigEventData { 173 public: CreateAudioPatchConfigEventData(const struct audio_patch patch,audio_patch_handle_t handle)174 CreateAudioPatchConfigEventData(const struct audio_patch patch, 175 audio_patch_handle_t handle) : 176 mPatch(patch), mHandle(handle) {} 177 dump(char * buffer,size_t size)178 virtual void dump(char *buffer, size_t size) { 179 snprintf(buffer, size, "Patch handle: %u\n", mHandle); 180 } 181 182 const struct audio_patch mPatch; 183 audio_patch_handle_t mHandle; 184 }; 185 186 class CreateAudioPatchConfigEvent : public ConfigEvent { 187 public: CreateAudioPatchConfigEvent(const struct audio_patch patch,audio_patch_handle_t handle)188 CreateAudioPatchConfigEvent(const struct audio_patch patch, 189 audio_patch_handle_t handle) : 190 ConfigEvent(CFG_EVENT_CREATE_AUDIO_PATCH) { 191 mData = new CreateAudioPatchConfigEventData(patch, handle); 192 mWaitStatus = true; 193 } ~CreateAudioPatchConfigEvent()194 virtual ~CreateAudioPatchConfigEvent() {} 195 }; 196 197 class ReleaseAudioPatchConfigEventData : public ConfigEventData { 198 public: ReleaseAudioPatchConfigEventData(const audio_patch_handle_t handle)199 ReleaseAudioPatchConfigEventData(const audio_patch_handle_t handle) : 200 mHandle(handle) {} 201 dump(char * buffer,size_t size)202 virtual void dump(char *buffer, size_t size) { 203 snprintf(buffer, size, "Patch handle: %u\n", mHandle); 204 } 205 206 audio_patch_handle_t mHandle; 207 }; 208 209 class ReleaseAudioPatchConfigEvent : public ConfigEvent { 210 public: ReleaseAudioPatchConfigEvent(const audio_patch_handle_t handle)211 ReleaseAudioPatchConfigEvent(const audio_patch_handle_t handle) : 212 ConfigEvent(CFG_EVENT_RELEASE_AUDIO_PATCH) { 213 mData = new ReleaseAudioPatchConfigEventData(handle); 214 mWaitStatus = true; 215 } ~ReleaseAudioPatchConfigEvent()216 virtual ~ReleaseAudioPatchConfigEvent() {} 217 }; 218 219 class PMDeathRecipient : public IBinder::DeathRecipient { 220 public: PMDeathRecipient(const wp<ThreadBase> & thread)221 PMDeathRecipient(const wp<ThreadBase>& thread) : mThread(thread) {} ~PMDeathRecipient()222 virtual ~PMDeathRecipient() {} 223 224 // IBinder::DeathRecipient 225 virtual void binderDied(const wp<IBinder>& who); 226 227 private: 228 PMDeathRecipient(const PMDeathRecipient&); 229 PMDeathRecipient& operator = (const PMDeathRecipient&); 230 231 wp<ThreadBase> mThread; 232 }; 233 234 virtual status_t initCheck() const = 0; 235 236 // static externally-visible type()237 type_t type() const { return mType; } isDuplicating()238 bool isDuplicating() const { return (mType == DUPLICATING); } 239 id()240 audio_io_handle_t id() const { return mId;} 241 242 // dynamic externally-visible sampleRate()243 uint32_t sampleRate() const { return mSampleRate; } channelMask()244 audio_channel_mask_t channelMask() const { return mChannelMask; } format()245 audio_format_t format() const { return mHALFormat; } channelCount()246 uint32_t channelCount() const { return mChannelCount; } 247 // Called by AudioFlinger::frameCount(audio_io_handle_t output) and effects, 248 // and returns the [normal mix] buffer's frame count. 249 virtual size_t frameCount() const = 0; 250 251 // Return's the HAL's frame count i.e. fast mixer buffer size. frameCountHAL()252 size_t frameCountHAL() const { return mFrameCount; } 253 frameSize()254 size_t frameSize() const { return mFrameSize; } 255 256 // Should be "virtual status_t requestExitAndWait()" and override same 257 // method in Thread, but Thread::requestExitAndWait() is not yet virtual. 258 void exit(); 259 virtual bool checkForNewParameter_l(const String8& keyValuePair, 260 status_t& status) = 0; 261 virtual status_t setParameters(const String8& keyValuePairs); 262 virtual String8 getParameters(const String8& keys) = 0; 263 virtual void ioConfigChanged(audio_io_config_event event, pid_t pid = 0) = 0; 264 // sendConfigEvent_l() must be called with ThreadBase::mLock held 265 // Can temporarily release the lock if waiting for a reply from 266 // processConfigEvents_l(). 267 status_t sendConfigEvent_l(sp<ConfigEvent>& event); 268 void sendIoConfigEvent(audio_io_config_event event, pid_t pid = 0); 269 void sendIoConfigEvent_l(audio_io_config_event event, pid_t pid = 0); 270 void sendPrioConfigEvent(pid_t pid, pid_t tid, int32_t prio); 271 void sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio); 272 status_t sendSetParameterConfigEvent_l(const String8& keyValuePair); 273 status_t sendCreateAudioPatchConfigEvent(const struct audio_patch *patch, 274 audio_patch_handle_t *handle); 275 status_t sendReleaseAudioPatchConfigEvent(audio_patch_handle_t handle); 276 void processConfigEvents_l(); 277 virtual void cacheParameters_l() = 0; 278 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 279 audio_patch_handle_t *handle) = 0; 280 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle) = 0; 281 virtual void getAudioPortConfig(struct audio_port_config *config) = 0; 282 283 284 // see note at declaration of mStandby, mOutDevice and mInDevice standby()285 bool standby() const { return mStandby; } outDevice()286 audio_devices_t outDevice() const { return mOutDevice; } inDevice()287 audio_devices_t inDevice() const { return mInDevice; } 288 289 virtual audio_stream_t* stream() const = 0; 290 291 sp<EffectHandle> createEffect_l( 292 const sp<AudioFlinger::Client>& client, 293 const sp<IEffectClient>& effectClient, 294 int32_t priority, 295 audio_session_t sessionId, 296 effect_descriptor_t *desc, 297 int *enabled, 298 status_t *status /*non-NULL*/); 299 300 // return values for hasAudioSession (bit field) 301 enum effect_state { 302 EFFECT_SESSION = 0x1, // the audio session corresponds to at least one 303 // effect 304 TRACK_SESSION = 0x2 // the audio session corresponds to at least one 305 // track 306 }; 307 308 // get effect chain corresponding to session Id. 309 sp<EffectChain> getEffectChain(audio_session_t sessionId); 310 // same as getEffectChain() but must be called with ThreadBase mutex locked 311 sp<EffectChain> getEffectChain_l(audio_session_t sessionId) const; 312 // add an effect chain to the chain list (mEffectChains) 313 virtual status_t addEffectChain_l(const sp<EffectChain>& chain) = 0; 314 // remove an effect chain from the chain list (mEffectChains) 315 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain) = 0; 316 // lock all effect chains Mutexes. Must be called before releasing the 317 // ThreadBase mutex before processing the mixer and effects. This guarantees the 318 // integrity of the chains during the process. 319 // Also sets the parameter 'effectChains' to current value of mEffectChains. 320 void lockEffectChains_l(Vector< sp<EffectChain> >& effectChains); 321 // unlock effect chains after process 322 void unlockEffectChains(const Vector< sp<EffectChain> >& effectChains); 323 // get a copy of mEffectChains vector getEffectChains_l()324 Vector< sp<EffectChain> > getEffectChains_l() const { return mEffectChains; }; 325 // set audio mode to all effect chains 326 void setMode(audio_mode_t mode); 327 // get effect module with corresponding ID on specified audio session 328 sp<AudioFlinger::EffectModule> getEffect(audio_session_t sessionId, int effectId); 329 sp<AudioFlinger::EffectModule> getEffect_l(audio_session_t sessionId, int effectId); 330 // add and effect module. Also creates the effect chain is none exists for 331 // the effects audio session 332 status_t addEffect_l(const sp< EffectModule>& effect); 333 // remove and effect module. Also removes the effect chain is this was the last 334 // effect 335 void removeEffect_l(const sp< EffectModule>& effect); 336 // detach all tracks connected to an auxiliary effect detachAuxEffect_l(int effectId __unused)337 virtual void detachAuxEffect_l(int effectId __unused) {} 338 // returns either EFFECT_SESSION if effects on this audio session exist in one 339 // chain, or TRACK_SESSION if tracks on this audio session exist, or both 340 virtual uint32_t hasAudioSession(audio_session_t sessionId) const = 0; 341 // the value returned by default implementation is not important as the 342 // strategy is only meaningful for PlaybackThread which implements this method getStrategyForSession_l(audio_session_t sessionId __unused)343 virtual uint32_t getStrategyForSession_l(audio_session_t sessionId __unused) 344 { return 0; } 345 346 // suspend or restore effect according to the type of effect passed. a NULL 347 // type pointer means suspend all effects in the session 348 void setEffectSuspended(const effect_uuid_t *type, 349 bool suspend, 350 audio_session_t sessionId = AUDIO_SESSION_OUTPUT_MIX); 351 // check if some effects must be suspended/restored when an effect is enabled 352 // or disabled 353 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 354 bool enabled, 355 audio_session_t sessionId = 356 AUDIO_SESSION_OUTPUT_MIX); 357 void checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 358 bool enabled, 359 audio_session_t sessionId = 360 AUDIO_SESSION_OUTPUT_MIX); 361 362 virtual status_t setSyncEvent(const sp<SyncEvent>& event) = 0; 363 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const = 0; 364 365 // Return a reference to a per-thread heap which can be used to allocate IMemory 366 // objects that will be read-only to client processes, read/write to mediaserver, 367 // and shared by all client processes of the thread. 368 // The heap is per-thread rather than common across all threads, because 369 // clients can't be trusted not to modify the offset of the IMemory they receive. 370 // If a thread does not have such a heap, this method returns 0. readOnlyHeap()371 virtual sp<MemoryDealer> readOnlyHeap() const { return 0; } 372 pipeMemory()373 virtual sp<IMemory> pipeMemory() const { return 0; } 374 375 void systemReady(); 376 377 mutable Mutex mLock; 378 379 protected: 380 381 // entry describing an effect being suspended in mSuspendedSessions keyed vector 382 class SuspendedSessionDesc : public RefBase { 383 public: SuspendedSessionDesc()384 SuspendedSessionDesc() : mRefCount(0) {} 385 386 int mRefCount; // number of active suspend requests 387 effect_uuid_t mType; // effect type UUID 388 }; 389 390 void acquireWakeLock(int uid = -1); 391 virtual void acquireWakeLock_l(int uid = -1); 392 void releaseWakeLock(); 393 void releaseWakeLock_l(); 394 void updateWakeLockUids(const SortedVector<int> &uids); 395 void updateWakeLockUids_l(const SortedVector<int> &uids); 396 void getPowerManager_l(); 397 void setEffectSuspended_l(const effect_uuid_t *type, 398 bool suspend, 399 audio_session_t sessionId); 400 // updated mSuspendedSessions when an effect suspended or restored 401 void updateSuspendedSessions_l(const effect_uuid_t *type, 402 bool suspend, 403 audio_session_t sessionId); 404 // check if some effects must be suspended when an effect chain is added 405 void checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain); 406 407 String16 getWakeLockTag(); 408 preExit()409 virtual void preExit() { } setMasterMono_l(bool mono __unused)410 virtual void setMasterMono_l(bool mono __unused) { } requireMonoBlend()411 virtual bool requireMonoBlend() { return false; } 412 413 friend class AudioFlinger; // for mEffectChains 414 415 const type_t mType; 416 417 // Used by parameters, config events, addTrack_l, exit 418 Condition mWaitWorkCV; 419 420 const sp<AudioFlinger> mAudioFlinger; 421 422 // updated by PlaybackThread::readOutputParameters_l() or 423 // RecordThread::readInputParameters_l() 424 uint32_t mSampleRate; 425 size_t mFrameCount; // output HAL, direct output, record 426 audio_channel_mask_t mChannelMask; 427 uint32_t mChannelCount; 428 size_t mFrameSize; 429 // not HAL frame size, this is for output sink (to pipe to fast mixer) 430 audio_format_t mFormat; // Source format for Recording and 431 // Sink format for Playback. 432 // Sink format may be different than 433 // HAL format if Fastmixer is used. 434 audio_format_t mHALFormat; 435 size_t mBufferSize; // HAL buffer size for read() or write() 436 437 Vector< sp<ConfigEvent> > mConfigEvents; 438 Vector< sp<ConfigEvent> > mPendingConfigEvents; // events awaiting system ready 439 440 // These fields are written and read by thread itself without lock or barrier, 441 // and read by other threads without lock or barrier via standby(), outDevice() 442 // and inDevice(). 443 // Because of the absence of a lock or barrier, any other thread that reads 444 // these fields must use the information in isolation, or be prepared to deal 445 // with possibility that it might be inconsistent with other information. 446 bool mStandby; // Whether thread is currently in standby. 447 audio_devices_t mOutDevice; // output device 448 audio_devices_t mInDevice; // input device 449 audio_devices_t mPrevOutDevice; // previous output device 450 audio_devices_t mPrevInDevice; // previous input device 451 struct audio_patch mPatch; 452 audio_source_t mAudioSource; 453 454 const audio_io_handle_t mId; 455 Vector< sp<EffectChain> > mEffectChains; 456 457 static const int kThreadNameLength = 16; // prctl(PR_SET_NAME) limit 458 char mThreadName[kThreadNameLength]; // guaranteed NUL-terminated 459 sp<IPowerManager> mPowerManager; 460 sp<IBinder> mWakeLockToken; 461 const sp<PMDeathRecipient> mDeathRecipient; 462 // list of suspended effects per session and per type. The first (outer) vector is 463 // keyed by session ID, the second (inner) by type UUID timeLow field 464 KeyedVector< audio_session_t, KeyedVector< int, sp<SuspendedSessionDesc> > > 465 mSuspendedSessions; 466 static const size_t kLogSize = 4 * 1024; 467 sp<NBLog::Writer> mNBLogWriter; 468 bool mSystemReady; 469 bool mNotifiedBatteryStart; 470 ExtendedTimestamp mTimestamp; 471 }; 472 473 // --- PlaybackThread --- 474 class PlaybackThread : public ThreadBase { 475 public: 476 477 #include "PlaybackTracks.h" 478 479 enum mixer_state { 480 MIXER_IDLE, // no active tracks 481 MIXER_TRACKS_ENABLED, // at least one active track, but no track has any data ready 482 MIXER_TRACKS_READY, // at least one active track, and at least one track has data 483 MIXER_DRAIN_TRACK, // drain currently playing track 484 MIXER_DRAIN_ALL, // fully drain the hardware 485 // standby mode does not have an enum value 486 // suspend by audio policy manager is orthogonal to mixer state 487 }; 488 489 // retry count before removing active track in case of underrun on offloaded thread: 490 // we need to make sure that AudioTrack client has enough time to send large buffers 491 //FIXME may be more appropriate if expressed in time units. Need to revise how underrun is 492 // handled for offloaded tracks 493 static const int8_t kMaxTrackRetriesOffload = 20; 494 static const int8_t kMaxTrackStartupRetriesOffload = 100; 495 static const int8_t kMaxTrackStopRetriesOffload = 2; 496 497 PlaybackThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 498 audio_io_handle_t id, audio_devices_t device, type_t type, bool systemReady); 499 virtual ~PlaybackThread(); 500 501 void dump(int fd, const Vector<String16>& args); 502 503 // Thread virtuals 504 virtual bool threadLoop(); 505 506 // RefBase 507 virtual void onFirstRef(); 508 509 protected: 510 // Code snippets that were lifted up out of threadLoop() 511 virtual void threadLoop_mix() = 0; 512 virtual void threadLoop_sleepTime() = 0; 513 virtual ssize_t threadLoop_write(); 514 virtual void threadLoop_drain(); 515 virtual void threadLoop_standby(); 516 virtual void threadLoop_exit(); 517 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 518 519 // prepareTracks_l reads and writes mActiveTracks, and returns 520 // the pending set of tracks to remove via Vector 'tracksToRemove'. The caller 521 // is responsible for clearing or destroying this Vector later on, when it 522 // is safe to do so. That will drop the final ref count and destroy the tracks. 523 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove) = 0; 524 void removeTracks_l(const Vector< sp<Track> >& tracksToRemove); 525 526 void writeCallback(); 527 void resetWriteBlocked(uint32_t sequence); 528 void drainCallback(); 529 void resetDraining(uint32_t sequence); 530 531 static int asyncCallback(stream_callback_event_t event, void *param, void *cookie); 532 533 virtual bool waitingAsyncCallback(); 534 virtual bool waitingAsyncCallback_l(); 535 virtual bool shouldStandby_l(); 536 virtual void onAddNewTrack_l(); 537 538 // ThreadBase virtuals 539 virtual void preExit(); 540 keepWakeLock()541 virtual bool keepWakeLock() const { return true; } 542 543 public: 544 initCheck()545 virtual status_t initCheck() const { return (mOutput == NULL) ? NO_INIT : NO_ERROR; } 546 547 // return estimated latency in milliseconds, as reported by HAL 548 uint32_t latency() const; 549 // same, but lock must already be held 550 uint32_t latency_l() const; 551 552 void setMasterVolume(float value); 553 void setMasterMute(bool muted); 554 555 void setStreamVolume(audio_stream_type_t stream, float value); 556 void setStreamMute(audio_stream_type_t stream, bool muted); 557 558 float streamVolume(audio_stream_type_t stream) const; 559 560 sp<Track> createTrack_l( 561 const sp<AudioFlinger::Client>& client, 562 audio_stream_type_t streamType, 563 uint32_t sampleRate, 564 audio_format_t format, 565 audio_channel_mask_t channelMask, 566 size_t *pFrameCount, 567 const sp<IMemory>& sharedBuffer, 568 audio_session_t sessionId, 569 IAudioFlinger::track_flags_t *flags, 570 pid_t tid, 571 int uid, 572 status_t *status /*non-NULL*/); 573 574 AudioStreamOut* getOutput() const; 575 AudioStreamOut* clearOutput(); 576 virtual audio_stream_t* stream() const; 577 578 // a very large number of suspend() will eventually wraparound, but unlikely suspend()579 void suspend() { (void) android_atomic_inc(&mSuspended); } restore()580 void restore() 581 { 582 // if restore() is done without suspend(), get back into 583 // range so that the next suspend() will operate correctly 584 if (android_atomic_dec(&mSuspended) <= 0) { 585 android_atomic_release_store(0, &mSuspended); 586 } 587 } isSuspended()588 bool isSuspended() const 589 { return android_atomic_acquire_load(&mSuspended) > 0; } 590 591 virtual String8 getParameters(const String8& keys); 592 virtual void ioConfigChanged(audio_io_config_event event, pid_t pid = 0); 593 status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames); 594 // FIXME rename mixBuffer() to sinkBuffer() and remove int16_t* dependency. 595 // Consider also removing and passing an explicit mMainBuffer initialization 596 // parameter to AF::PlaybackThread::Track::Track(). mixBuffer()597 int16_t *mixBuffer() const { 598 return reinterpret_cast<int16_t *>(mSinkBuffer); }; 599 600 virtual void detachAuxEffect_l(int effectId); 601 status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track> track, 602 int EffectId); 603 status_t attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track> track, 604 int EffectId); 605 606 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 607 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 608 virtual uint32_t hasAudioSession(audio_session_t sessionId) const; 609 virtual uint32_t getStrategyForSession_l(audio_session_t sessionId); 610 611 612 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 613 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 614 615 // called with AudioFlinger lock held 616 bool invalidateTracks_l(audio_stream_type_t streamType); 617 virtual void invalidateTracks(audio_stream_type_t streamType); 618 frameCount()619 virtual size_t frameCount() const { return mNormalFrameCount; } 620 621 status_t getTimestamp_l(AudioTimestamp& timestamp); 622 623 void addPatchTrack(const sp<PatchTrack>& track); 624 void deletePatchTrack(const sp<PatchTrack>& track); 625 626 virtual void getAudioPortConfig(struct audio_port_config *config); 627 628 protected: 629 // updated by readOutputParameters_l() 630 size_t mNormalFrameCount; // normal mixer and effects 631 632 bool mThreadThrottle; // throttle the thread processing 633 uint32_t mThreadThrottleTimeMs; // throttle time for MIXER threads 634 uint32_t mThreadThrottleEndMs; // notify once per throttling 635 uint32_t mHalfBufferMs; // half the buffer size in milliseconds 636 637 void* mSinkBuffer; // frame size aligned sink buffer 638 639 // TODO: 640 // Rearrange the buffer info into a struct/class with 641 // clear, copy, construction, destruction methods. 642 // 643 // mSinkBuffer also has associated with it: 644 // 645 // mSinkBufferSize: Sink Buffer Size 646 // mFormat: Sink Buffer Format 647 648 // Mixer Buffer (mMixerBuffer*) 649 // 650 // In the case of floating point or multichannel data, which is not in the 651 // sink format, it is required to accumulate in a higher precision or greater channel count 652 // buffer before downmixing or data conversion to the sink buffer. 653 654 // Set to "true" to enable the Mixer Buffer otherwise mixer output goes to sink buffer. 655 bool mMixerBufferEnabled; 656 657 // Storage, 32 byte aligned (may make this alignment a requirement later). 658 // Due to constraints on mNormalFrameCount, the buffer size is a multiple of 16 frames. 659 void* mMixerBuffer; 660 661 // Size of mMixerBuffer in bytes: mNormalFrameCount * #channels * sampsize. 662 size_t mMixerBufferSize; 663 664 // The audio format of mMixerBuffer. Set to AUDIO_FORMAT_PCM_(FLOAT|16_BIT) only. 665 audio_format_t mMixerBufferFormat; 666 667 // An internal flag set to true by MixerThread::prepareTracks_l() 668 // when mMixerBuffer contains valid data after mixing. 669 bool mMixerBufferValid; 670 671 // Effects Buffer (mEffectsBuffer*) 672 // 673 // In the case of effects data, which is not in the sink format, 674 // it is required to accumulate in a different buffer before data conversion 675 // to the sink buffer. 676 677 // Set to "true" to enable the Effects Buffer otherwise effects output goes to sink buffer. 678 bool mEffectBufferEnabled; 679 680 // Storage, 32 byte aligned (may make this alignment a requirement later). 681 // Due to constraints on mNormalFrameCount, the buffer size is a multiple of 16 frames. 682 void* mEffectBuffer; 683 684 // Size of mEffectsBuffer in bytes: mNormalFrameCount * #channels * sampsize. 685 size_t mEffectBufferSize; 686 687 // The audio format of mEffectsBuffer. Set to AUDIO_FORMAT_PCM_16_BIT only. 688 audio_format_t mEffectBufferFormat; 689 690 // An internal flag set to true by MixerThread::prepareTracks_l() 691 // when mEffectsBuffer contains valid data after mixing. 692 // 693 // When this is set, all mixer data is routed into the effects buffer 694 // for any processing (including output processing). 695 bool mEffectBufferValid; 696 697 // suspend count, > 0 means suspended. While suspended, the thread continues to pull from 698 // tracks and mix, but doesn't write to HAL. A2DP and SCO HAL implementations can't handle 699 // concurrent use of both of them, so Audio Policy Service suspends one of the threads to 700 // workaround that restriction. 701 // 'volatile' means accessed via atomic operations and no lock. 702 volatile int32_t mSuspended; 703 704 int64_t mBytesWritten; 705 int64_t mFramesWritten; // not reset on standby 706 private: 707 // mMasterMute is in both PlaybackThread and in AudioFlinger. When a 708 // PlaybackThread needs to find out if master-muted, it checks it's local 709 // copy rather than the one in AudioFlinger. This optimization saves a lock. 710 bool mMasterMute; setMasterMute_l(bool muted)711 void setMasterMute_l(bool muted) { mMasterMute = muted; } 712 protected: 713 SortedVector< wp<Track> > mActiveTracks; // FIXME check if this could be sp<> 714 SortedVector<int> mWakeLockUids; 715 int mActiveTracksGeneration; 716 wp<Track> mLatestActiveTrack; // latest track added to mActiveTracks 717 718 // Allocate a track name for a given channel mask. 719 // Returns name >= 0 if successful, -1 on failure. 720 virtual int getTrackName_l(audio_channel_mask_t channelMask, 721 audio_format_t format, audio_session_t sessionId) = 0; 722 virtual void deleteTrackName_l(int name) = 0; 723 724 // Time to sleep between cycles when: 725 virtual uint32_t activeSleepTimeUs() const; // mixer state MIXER_TRACKS_ENABLED 726 virtual uint32_t idleSleepTimeUs() const = 0; // mixer state MIXER_IDLE 727 virtual uint32_t suspendSleepTimeUs() const = 0; // audio policy manager suspended us 728 // No sleep when mixer state == MIXER_TRACKS_READY; relies on audio HAL stream->write() 729 // No sleep in standby mode; waits on a condition 730 731 // Code snippets that are temporarily lifted up out of threadLoop() until the merge 732 void checkSilentMode_l(); 733 734 // Non-trivial for DUPLICATING only saveOutputTracks()735 virtual void saveOutputTracks() { } clearOutputTracks()736 virtual void clearOutputTracks() { } 737 738 // Cache various calculated values, at threadLoop() entry and after a parameter change 739 virtual void cacheParameters_l(); 740 741 virtual uint32_t correctLatency_l(uint32_t latency) const; 742 743 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 744 audio_patch_handle_t *handle); 745 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 746 usesHwAvSync()747 bool usesHwAvSync() const { return (mType == DIRECT) && (mOutput != NULL) 748 && mHwSupportsPause 749 && (mOutput->flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC); } 750 751 private: 752 753 friend class AudioFlinger; // for numerous 754 755 PlaybackThread& operator = (const PlaybackThread&); 756 757 status_t addTrack_l(const sp<Track>& track); 758 bool destroyTrack_l(const sp<Track>& track); 759 void removeTrack_l(const sp<Track>& track); 760 void broadcast_l(); 761 762 void readOutputParameters_l(); 763 764 virtual void dumpInternals(int fd, const Vector<String16>& args); 765 void dumpTracks(int fd, const Vector<String16>& args); 766 767 SortedVector< sp<Track> > mTracks; 768 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 769 AudioStreamOut *mOutput; 770 771 float mMasterVolume; 772 nsecs_t mLastWriteTime; 773 int mNumWrites; 774 int mNumDelayedWrites; 775 bool mInWrite; 776 777 // FIXME rename these former local variables of threadLoop to standard "m" names 778 nsecs_t mStandbyTimeNs; 779 size_t mSinkBufferSize; 780 781 // cached copies of activeSleepTimeUs() and idleSleepTimeUs() made by cacheParameters_l() 782 uint32_t mActiveSleepTimeUs; 783 uint32_t mIdleSleepTimeUs; 784 785 uint32_t mSleepTimeUs; 786 787 // mixer status returned by prepareTracks_l() 788 mixer_state mMixerStatus; // current cycle 789 // previous cycle when in prepareTracks_l() 790 mixer_state mMixerStatusIgnoringFastTracks; 791 // FIXME or a separate ready state per track 792 793 // FIXME move these declarations into the specific sub-class that needs them 794 // MIXER only 795 uint32_t sleepTimeShift; 796 797 // same as AudioFlinger::mStandbyTimeInNsecs except for DIRECT which uses a shorter value 798 nsecs_t mStandbyDelayNs; 799 800 // MIXER only 801 nsecs_t maxPeriod; 802 803 // DUPLICATING only 804 uint32_t writeFrames; 805 806 size_t mBytesRemaining; 807 size_t mCurrentWriteLength; 808 bool mUseAsyncWrite; 809 // mWriteAckSequence contains current write sequence on bits 31-1. The write sequence is 810 // incremented each time a write(), a flush() or a standby() occurs. 811 // Bit 0 is set when a write blocks and indicates a callback is expected. 812 // Bit 0 is reset by the async callback thread calling resetWriteBlocked(). Out of sequence 813 // callbacks are ignored. 814 uint32_t mWriteAckSequence; 815 // mDrainSequence contains current drain sequence on bits 31-1. The drain sequence is 816 // incremented each time a drain is requested or a flush() or standby() occurs. 817 // Bit 0 is set when the drain() command is called at the HAL and indicates a callback is 818 // expected. 819 // Bit 0 is reset by the async callback thread calling resetDraining(). Out of sequence 820 // callbacks are ignored. 821 uint32_t mDrainSequence; 822 // A condition that must be evaluated by prepareTrack_l() has changed and we must not wait 823 // for async write callback in the thread loop before evaluating it 824 bool mSignalPending; 825 sp<AsyncCallbackThread> mCallbackThread; 826 827 private: 828 // The HAL output sink is treated as non-blocking, but current implementation is blocking 829 sp<NBAIO_Sink> mOutputSink; 830 // If a fast mixer is present, the blocking pipe sink, otherwise clear 831 sp<NBAIO_Sink> mPipeSink; 832 // The current sink for the normal mixer to write it's (sub)mix, mOutputSink or mPipeSink 833 sp<NBAIO_Sink> mNormalSink; 834 #ifdef TEE_SINK 835 // For dumpsys 836 sp<NBAIO_Sink> mTeeSink; 837 sp<NBAIO_Source> mTeeSource; 838 #endif 839 uint32_t mScreenState; // cached copy of gScreenState 840 static const size_t kFastMixerLogSize = 4 * 1024; 841 sp<NBLog::Writer> mFastMixerNBLogWriter; 842 public: 843 virtual bool hasFastMixer() const = 0; getFastTrackUnderruns(size_t fastIndex __unused)844 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex __unused) const 845 { FastTrackUnderruns dummy; return dummy; } 846 847 protected: 848 // accessed by both binder threads and within threadLoop(), lock on mutex needed 849 unsigned mFastTrackAvailMask; // bit i set if fast track [i] is available 850 bool mHwSupportsPause; 851 bool mHwPaused; 852 bool mFlushPending; 853 }; 854 855 class MixerThread : public PlaybackThread { 856 public: 857 MixerThread(const sp<AudioFlinger>& audioFlinger, 858 AudioStreamOut* output, 859 audio_io_handle_t id, 860 audio_devices_t device, 861 bool systemReady, 862 type_t type = MIXER); 863 virtual ~MixerThread(); 864 865 // Thread virtuals 866 867 virtual bool checkForNewParameter_l(const String8& keyValuePair, 868 status_t& status); 869 virtual void dumpInternals(int fd, const Vector<String16>& args); 870 871 protected: 872 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 873 virtual int getTrackName_l(audio_channel_mask_t channelMask, 874 audio_format_t format, audio_session_t sessionId); 875 virtual void deleteTrackName_l(int name); 876 virtual uint32_t idleSleepTimeUs() const; 877 virtual uint32_t suspendSleepTimeUs() const; 878 virtual void cacheParameters_l(); 879 880 virtual void acquireWakeLock_l(int uid = -1) { 881 PlaybackThread::acquireWakeLock_l(uid); 882 if (hasFastMixer()) { 883 mFastMixer->setBoottimeOffset( 884 mTimestamp.mTimebaseOffset[ExtendedTimestamp::TIMEBASE_BOOTTIME]); 885 } 886 } 887 888 // threadLoop snippets 889 virtual ssize_t threadLoop_write(); 890 virtual void threadLoop_standby(); 891 virtual void threadLoop_mix(); 892 virtual void threadLoop_sleepTime(); 893 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 894 virtual uint32_t correctLatency_l(uint32_t latency) const; 895 896 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 897 audio_patch_handle_t *handle); 898 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 899 900 AudioMixer* mAudioMixer; // normal mixer 901 private: 902 // one-time initialization, no locks required 903 sp<FastMixer> mFastMixer; // non-0 if there is also a fast mixer 904 sp<AudioWatchdog> mAudioWatchdog; // non-0 if there is an audio watchdog thread 905 906 // contents are not guaranteed to be consistent, no locks required 907 FastMixerDumpState mFastMixerDumpState; 908 #ifdef STATE_QUEUE_DUMP 909 StateQueueObserverDump mStateQueueObserverDump; 910 StateQueueMutatorDump mStateQueueMutatorDump; 911 #endif 912 AudioWatchdogDump mAudioWatchdogDump; 913 914 // accessible only within the threadLoop(), no locks required 915 // mFastMixer->sq() // for mutating and pushing state 916 int32_t mFastMixerFutex; // for cold idle 917 918 std::atomic_bool mMasterMono; 919 public: hasFastMixer()920 virtual bool hasFastMixer() const { return mFastMixer != 0; } getFastTrackUnderruns(size_t fastIndex)921 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const { 922 ALOG_ASSERT(fastIndex < FastMixerState::sMaxFastTracks); 923 return mFastMixerDumpState.mTracks[fastIndex].mUnderruns; 924 } 925 926 protected: setMasterMono_l(bool mono)927 virtual void setMasterMono_l(bool mono) { 928 mMasterMono.store(mono); 929 if (mFastMixer != nullptr) { /* hasFastMixer() */ 930 mFastMixer->setMasterMono(mMasterMono); 931 } 932 } 933 // the FastMixer performs mono blend if it exists. 934 // Blending with limiter is not idempotent, 935 // and blending without limiter is idempotent but inefficient to do twice. requireMonoBlend()936 virtual bool requireMonoBlend() { return mMasterMono.load() && !hasFastMixer(); } 937 }; 938 939 class DirectOutputThread : public PlaybackThread { 940 public: 941 942 DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 943 audio_io_handle_t id, audio_devices_t device, bool systemReady); 944 virtual ~DirectOutputThread(); 945 946 // Thread virtuals 947 948 virtual bool checkForNewParameter_l(const String8& keyValuePair, 949 status_t& status); 950 virtual void flushHw_l(); 951 952 protected: 953 virtual int getTrackName_l(audio_channel_mask_t channelMask, 954 audio_format_t format, audio_session_t sessionId); 955 virtual void deleteTrackName_l(int name); 956 virtual uint32_t activeSleepTimeUs() const; 957 virtual uint32_t idleSleepTimeUs() const; 958 virtual uint32_t suspendSleepTimeUs() const; 959 virtual void cacheParameters_l(); 960 961 // threadLoop snippets 962 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 963 virtual void threadLoop_mix(); 964 virtual void threadLoop_sleepTime(); 965 virtual void threadLoop_exit(); 966 virtual bool shouldStandby_l(); 967 968 virtual void onAddNewTrack_l(); 969 970 // volumes last sent to audio HAL with stream->set_volume() 971 float mLeftVolFloat; 972 float mRightVolFloat; 973 974 DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 975 audio_io_handle_t id, uint32_t device, ThreadBase::type_t type, 976 bool systemReady); 977 void processVolume_l(Track *track, bool lastTrack); 978 979 // prepareTracks_l() tells threadLoop_mix() the name of the single active track 980 sp<Track> mActiveTrack; 981 982 wp<Track> mPreviousTrack; // used to detect track switch 983 984 public: hasFastMixer()985 virtual bool hasFastMixer() const { return false; } 986 }; 987 988 class OffloadThread : public DirectOutputThread { 989 public: 990 991 OffloadThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 992 audio_io_handle_t id, uint32_t device, bool systemReady); ~OffloadThread()993 virtual ~OffloadThread() {}; 994 virtual void flushHw_l(); 995 996 protected: 997 // threadLoop snippets 998 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 999 virtual void threadLoop_exit(); 1000 1001 virtual bool waitingAsyncCallback(); 1002 virtual bool waitingAsyncCallback_l(); 1003 virtual void invalidateTracks(audio_stream_type_t streamType); 1004 keepWakeLock()1005 virtual bool keepWakeLock() const { return mKeepWakeLock; } 1006 1007 private: 1008 size_t mPausedWriteLength; // length in bytes of write interrupted by pause 1009 size_t mPausedBytesRemaining; // bytes still waiting in mixbuffer after resume 1010 bool mKeepWakeLock; // keep wake lock while waiting for write callback 1011 }; 1012 1013 class AsyncCallbackThread : public Thread { 1014 public: 1015 1016 AsyncCallbackThread(const wp<PlaybackThread>& playbackThread); 1017 1018 virtual ~AsyncCallbackThread(); 1019 1020 // Thread virtuals 1021 virtual bool threadLoop(); 1022 1023 // RefBase 1024 virtual void onFirstRef(); 1025 1026 void exit(); 1027 void setWriteBlocked(uint32_t sequence); 1028 void resetWriteBlocked(); 1029 void setDraining(uint32_t sequence); 1030 void resetDraining(); 1031 1032 private: 1033 const wp<PlaybackThread> mPlaybackThread; 1034 // mWriteAckSequence corresponds to the last write sequence passed by the offload thread via 1035 // setWriteBlocked(). The sequence is shifted one bit to the left and the lsb is used 1036 // to indicate that the callback has been received via resetWriteBlocked() 1037 uint32_t mWriteAckSequence; 1038 // mDrainSequence corresponds to the last drain sequence passed by the offload thread via 1039 // setDraining(). The sequence is shifted one bit to the left and the lsb is used 1040 // to indicate that the callback has been received via resetDraining() 1041 uint32_t mDrainSequence; 1042 Condition mWaitWorkCV; 1043 Mutex mLock; 1044 }; 1045 1046 class DuplicatingThread : public MixerThread { 1047 public: 1048 DuplicatingThread(const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread, 1049 audio_io_handle_t id, bool systemReady); 1050 virtual ~DuplicatingThread(); 1051 1052 // Thread virtuals 1053 void addOutputTrack(MixerThread* thread); 1054 void removeOutputTrack(MixerThread* thread); waitTimeMs()1055 uint32_t waitTimeMs() const { return mWaitTimeMs; } 1056 protected: 1057 virtual uint32_t activeSleepTimeUs() const; 1058 1059 private: 1060 bool outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks); 1061 protected: 1062 // threadLoop snippets 1063 virtual void threadLoop_mix(); 1064 virtual void threadLoop_sleepTime(); 1065 virtual ssize_t threadLoop_write(); 1066 virtual void threadLoop_standby(); 1067 virtual void cacheParameters_l(); 1068 1069 private: 1070 // called from threadLoop, addOutputTrack, removeOutputTrack 1071 virtual void updateWaitTime_l(); 1072 protected: 1073 virtual void saveOutputTracks(); 1074 virtual void clearOutputTracks(); 1075 private: 1076 1077 uint32_t mWaitTimeMs; 1078 SortedVector < sp<OutputTrack> > outputTracks; 1079 SortedVector < sp<OutputTrack> > mOutputTracks; 1080 public: hasFastMixer()1081 virtual bool hasFastMixer() const { return false; } 1082 }; 1083 1084 1085 // record thread 1086 class RecordThread : public ThreadBase 1087 { 1088 public: 1089 1090 class RecordTrack; 1091 1092 /* The ResamplerBufferProvider is used to retrieve recorded input data from the 1093 * RecordThread. It maintains local state on the relative position of the read 1094 * position of the RecordTrack compared with the RecordThread. 1095 */ 1096 class ResamplerBufferProvider : public AudioBufferProvider 1097 { 1098 public: ResamplerBufferProvider(RecordTrack * recordTrack)1099 ResamplerBufferProvider(RecordTrack* recordTrack) : 1100 mRecordTrack(recordTrack), 1101 mRsmpInUnrel(0), mRsmpInFront(0) { } ~ResamplerBufferProvider()1102 virtual ~ResamplerBufferProvider() { } 1103 1104 // called to set the ResamplerBufferProvider to head of the RecordThread data buffer, 1105 // skipping any previous data read from the hal. 1106 virtual void reset(); 1107 1108 /* Synchronizes RecordTrack position with the RecordThread. 1109 * Calculates available frames and handle overruns if the RecordThread 1110 * has advanced faster than the ResamplerBufferProvider has retrieved data. 1111 * TODO: why not do this for every getNextBuffer? 1112 * 1113 * Parameters 1114 * framesAvailable: pointer to optional output size_t to store record track 1115 * frames available. 1116 * hasOverrun: pointer to optional boolean, returns true if track has overrun. 1117 */ 1118 1119 virtual void sync(size_t *framesAvailable = NULL, bool *hasOverrun = NULL); 1120 1121 // AudioBufferProvider interface 1122 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer); 1123 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 1124 private: 1125 RecordTrack * const mRecordTrack; 1126 size_t mRsmpInUnrel; // unreleased frames remaining from 1127 // most recent getNextBuffer 1128 // for debug only 1129 int32_t mRsmpInFront; // next available frame 1130 // rolling counter that is never cleared 1131 }; 1132 1133 /* The RecordBufferConverter is used for format, channel, and sample rate 1134 * conversion for a RecordTrack. 1135 * 1136 * TODO: Self contained, so move to a separate file later. 1137 * 1138 * RecordBufferConverter uses the convert() method rather than exposing a 1139 * buffer provider interface; this is to save a memory copy. 1140 */ 1141 class RecordBufferConverter 1142 { 1143 public: 1144 RecordBufferConverter( 1145 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 1146 uint32_t srcSampleRate, 1147 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 1148 uint32_t dstSampleRate); 1149 1150 ~RecordBufferConverter(); 1151 1152 /* Converts input data from an AudioBufferProvider by format, channelMask, 1153 * and sampleRate to a destination buffer. 1154 * 1155 * Parameters 1156 * dst: buffer to place the converted data. 1157 * provider: buffer provider to obtain source data. 1158 * frames: number of frames to convert 1159 * 1160 * Returns the number of frames converted. 1161 */ 1162 size_t convert(void *dst, AudioBufferProvider *provider, size_t frames); 1163 1164 // returns NO_ERROR if constructor was successful initCheck()1165 status_t initCheck() const { 1166 // mSrcChannelMask set on successful updateParameters 1167 return mSrcChannelMask != AUDIO_CHANNEL_INVALID ? NO_ERROR : NO_INIT; 1168 } 1169 1170 // allows dynamic reconfigure of all parameters 1171 status_t updateParameters( 1172 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 1173 uint32_t srcSampleRate, 1174 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 1175 uint32_t dstSampleRate); 1176 1177 // called to reset resampler buffers on record track discontinuity reset()1178 void reset() { 1179 if (mResampler != NULL) { 1180 mResampler->reset(); 1181 } 1182 } 1183 1184 private: 1185 // format conversion when not using resampler 1186 void convertNoResampler(void *dst, const void *src, size_t frames); 1187 1188 // format conversion when using resampler; modifies src in-place 1189 void convertResampler(void *dst, /*not-a-const*/ void *src, size_t frames); 1190 1191 // user provided information 1192 audio_channel_mask_t mSrcChannelMask; 1193 audio_format_t mSrcFormat; 1194 uint32_t mSrcSampleRate; 1195 audio_channel_mask_t mDstChannelMask; 1196 audio_format_t mDstFormat; 1197 uint32_t mDstSampleRate; 1198 1199 // derived information 1200 uint32_t mSrcChannelCount; 1201 uint32_t mDstChannelCount; 1202 size_t mDstFrameSize; 1203 1204 // format conversion buffer 1205 void *mBuf; 1206 size_t mBufFrames; 1207 size_t mBufFrameSize; 1208 1209 // resampler info 1210 AudioResampler *mResampler; 1211 1212 bool mIsLegacyDownmix; // legacy stereo to mono conversion needed 1213 bool mIsLegacyUpmix; // legacy mono to stereo conversion needed 1214 bool mRequiresFloat; // data processing requires float (e.g. resampler) 1215 PassthruBufferProvider *mInputConverterProvider; // converts input to float 1216 int8_t mIdxAry[sizeof(uint32_t) * 8]; // used for channel mask conversion 1217 }; 1218 1219 #include "RecordTracks.h" 1220 1221 RecordThread(const sp<AudioFlinger>& audioFlinger, 1222 AudioStreamIn *input, 1223 audio_io_handle_t id, 1224 audio_devices_t outDevice, 1225 audio_devices_t inDevice, 1226 bool systemReady 1227 #ifdef TEE_SINK 1228 , const sp<NBAIO_Sink>& teeSink 1229 #endif 1230 ); 1231 virtual ~RecordThread(); 1232 1233 // no addTrack_l ? 1234 void destroyTrack_l(const sp<RecordTrack>& track); 1235 void removeTrack_l(const sp<RecordTrack>& track); 1236 1237 void dumpInternals(int fd, const Vector<String16>& args); 1238 void dumpTracks(int fd, const Vector<String16>& args); 1239 1240 // Thread virtuals 1241 virtual bool threadLoop(); 1242 1243 // RefBase 1244 virtual void onFirstRef(); 1245 initCheck()1246 virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; } 1247 readOnlyHeap()1248 virtual sp<MemoryDealer> readOnlyHeap() const { return mReadOnlyHeap; } 1249 pipeMemory()1250 virtual sp<IMemory> pipeMemory() const { return mPipeMemory; } 1251 1252 sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l( 1253 const sp<AudioFlinger::Client>& client, 1254 uint32_t sampleRate, 1255 audio_format_t format, 1256 audio_channel_mask_t channelMask, 1257 size_t *pFrameCount, 1258 audio_session_t sessionId, 1259 size_t *notificationFrames, 1260 int uid, 1261 IAudioFlinger::track_flags_t *flags, 1262 pid_t tid, 1263 status_t *status /*non-NULL*/); 1264 1265 status_t start(RecordTrack* recordTrack, 1266 AudioSystem::sync_event_t event, 1267 audio_session_t triggerSession); 1268 1269 // ask the thread to stop the specified track, and 1270 // return true if the caller should then do it's part of the stopping process 1271 bool stop(RecordTrack* recordTrack); 1272 1273 void dump(int fd, const Vector<String16>& args); 1274 AudioStreamIn* clearInput(); 1275 virtual audio_stream_t* stream() const; 1276 1277 1278 virtual bool checkForNewParameter_l(const String8& keyValuePair, 1279 status_t& status); cacheParameters_l()1280 virtual void cacheParameters_l() {} 1281 virtual String8 getParameters(const String8& keys); 1282 virtual void ioConfigChanged(audio_io_config_event event, pid_t pid = 0); 1283 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 1284 audio_patch_handle_t *handle); 1285 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 1286 1287 void addPatchRecord(const sp<PatchRecord>& record); 1288 void deletePatchRecord(const sp<PatchRecord>& record); 1289 1290 void readInputParameters_l(); 1291 virtual uint32_t getInputFramesLost(); 1292 1293 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1294 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1295 virtual uint32_t hasAudioSession(audio_session_t sessionId) const; 1296 1297 // Return the set of unique session IDs across all tracks. 1298 // The keys are the session IDs, and the associated values are meaningless. 1299 // FIXME replace by Set [and implement Bag/Multiset for other uses]. 1300 KeyedVector<audio_session_t, bool> sessionIds() const; 1301 1302 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 1303 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 1304 1305 static void syncStartEventCallback(const wp<SyncEvent>& event); 1306 frameCount()1307 virtual size_t frameCount() const { return mFrameCount; } hasFastCapture()1308 bool hasFastCapture() const { return mFastCapture != 0; } 1309 virtual void getAudioPortConfig(struct audio_port_config *config); 1310 1311 private: 1312 // Enter standby if not already in standby, and set mStandby flag 1313 void standbyIfNotAlreadyInStandby(); 1314 1315 // Call the HAL standby method unconditionally, and don't change mStandby flag 1316 void inputStandBy(); 1317 1318 AudioStreamIn *mInput; 1319 SortedVector < sp<RecordTrack> > mTracks; 1320 // mActiveTracks has dual roles: it indicates the current active track(s), and 1321 // is used together with mStartStopCond to indicate start()/stop() progress 1322 SortedVector< sp<RecordTrack> > mActiveTracks; 1323 // generation counter for mActiveTracks 1324 int mActiveTracksGen; 1325 Condition mStartStopCond; 1326 1327 // resampler converts input at HAL Hz to output at AudioRecord client Hz 1328 void *mRsmpInBuffer; // 1329 size_t mRsmpInFrames; // size of resampler input in frames 1330 size_t mRsmpInFramesP2;// size rounded up to a power-of-2 1331 1332 // rolling index that is never cleared 1333 int32_t mRsmpInRear; // last filled frame + 1 1334 1335 // For dumpsys 1336 const sp<NBAIO_Sink> mTeeSink; 1337 1338 const sp<MemoryDealer> mReadOnlyHeap; 1339 1340 // one-time initialization, no locks required 1341 sp<FastCapture> mFastCapture; // non-0 if there is also 1342 // a fast capture 1343 1344 // FIXME audio watchdog thread 1345 1346 // contents are not guaranteed to be consistent, no locks required 1347 FastCaptureDumpState mFastCaptureDumpState; 1348 #ifdef STATE_QUEUE_DUMP 1349 // FIXME StateQueue observer and mutator dump fields 1350 #endif 1351 // FIXME audio watchdog dump 1352 1353 // accessible only within the threadLoop(), no locks required 1354 // mFastCapture->sq() // for mutating and pushing state 1355 int32_t mFastCaptureFutex; // for cold idle 1356 1357 // The HAL input source is treated as non-blocking, 1358 // but current implementation is blocking 1359 sp<NBAIO_Source> mInputSource; 1360 // The source for the normal capture thread to read from: mInputSource or mPipeSource 1361 sp<NBAIO_Source> mNormalSource; 1362 // If a fast capture is present, the non-blocking pipe sink written to by fast capture, 1363 // otherwise clear 1364 sp<NBAIO_Sink> mPipeSink; 1365 // If a fast capture is present, the non-blocking pipe source read by normal thread, 1366 // otherwise clear 1367 sp<NBAIO_Source> mPipeSource; 1368 // Depth of pipe from fast capture to normal thread and fast clients, always power of 2 1369 size_t mPipeFramesP2; 1370 // If a fast capture is present, the Pipe as IMemory, otherwise clear 1371 sp<IMemory> mPipeMemory; 1372 1373 static const size_t kFastCaptureLogSize = 4 * 1024; 1374 sp<NBLog::Writer> mFastCaptureNBLogWriter; 1375 1376 bool mFastTrackAvail; // true if fast track available 1377 }; 1378