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; frameSize()250 size_t frameSize() const { return mFrameSize; } 251 252 // Should be "virtual status_t requestExitAndWait()" and override same 253 // method in Thread, but Thread::requestExitAndWait() is not yet virtual. 254 void exit(); 255 virtual bool checkForNewParameter_l(const String8& keyValuePair, 256 status_t& status) = 0; 257 virtual status_t setParameters(const String8& keyValuePairs); 258 virtual String8 getParameters(const String8& keys) = 0; 259 virtual void ioConfigChanged(audio_io_config_event event, pid_t pid = 0) = 0; 260 // sendConfigEvent_l() must be called with ThreadBase::mLock held 261 // Can temporarily release the lock if waiting for a reply from 262 // processConfigEvents_l(). 263 status_t sendConfigEvent_l(sp<ConfigEvent>& event); 264 void sendIoConfigEvent(audio_io_config_event event, pid_t pid = 0); 265 void sendIoConfigEvent_l(audio_io_config_event event, pid_t pid = 0); 266 void sendPrioConfigEvent(pid_t pid, pid_t tid, int32_t prio); 267 void sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio); 268 status_t sendSetParameterConfigEvent_l(const String8& keyValuePair); 269 status_t sendCreateAudioPatchConfigEvent(const struct audio_patch *patch, 270 audio_patch_handle_t *handle); 271 status_t sendReleaseAudioPatchConfigEvent(audio_patch_handle_t handle); 272 void processConfigEvents_l(); 273 virtual void cacheParameters_l() = 0; 274 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 275 audio_patch_handle_t *handle) = 0; 276 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle) = 0; 277 virtual void getAudioPortConfig(struct audio_port_config *config) = 0; 278 279 280 // see note at declaration of mStandby, mOutDevice and mInDevice standby()281 bool standby() const { return mStandby; } outDevice()282 audio_devices_t outDevice() const { return mOutDevice; } inDevice()283 audio_devices_t inDevice() const { return mInDevice; } 284 285 virtual audio_stream_t* stream() const = 0; 286 287 sp<EffectHandle> createEffect_l( 288 const sp<AudioFlinger::Client>& client, 289 const sp<IEffectClient>& effectClient, 290 int32_t priority, 291 int sessionId, 292 effect_descriptor_t *desc, 293 int *enabled, 294 status_t *status /*non-NULL*/); 295 296 // return values for hasAudioSession (bit field) 297 enum effect_state { 298 EFFECT_SESSION = 0x1, // the audio session corresponds to at least one 299 // effect 300 TRACK_SESSION = 0x2 // the audio session corresponds to at least one 301 // track 302 }; 303 304 // get effect chain corresponding to session Id. 305 sp<EffectChain> getEffectChain(int sessionId); 306 // same as getEffectChain() but must be called with ThreadBase mutex locked 307 sp<EffectChain> getEffectChain_l(int sessionId) const; 308 // add an effect chain to the chain list (mEffectChains) 309 virtual status_t addEffectChain_l(const sp<EffectChain>& chain) = 0; 310 // remove an effect chain from the chain list (mEffectChains) 311 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain) = 0; 312 // lock all effect chains Mutexes. Must be called before releasing the 313 // ThreadBase mutex before processing the mixer and effects. This guarantees the 314 // integrity of the chains during the process. 315 // Also sets the parameter 'effectChains' to current value of mEffectChains. 316 void lockEffectChains_l(Vector< sp<EffectChain> >& effectChains); 317 // unlock effect chains after process 318 void unlockEffectChains(const Vector< sp<EffectChain> >& effectChains); 319 // get a copy of mEffectChains vector getEffectChains_l()320 Vector< sp<EffectChain> > getEffectChains_l() const { return mEffectChains; }; 321 // set audio mode to all effect chains 322 void setMode(audio_mode_t mode); 323 // get effect module with corresponding ID on specified audio session 324 sp<AudioFlinger::EffectModule> getEffect(int sessionId, int effectId); 325 sp<AudioFlinger::EffectModule> getEffect_l(int sessionId, int effectId); 326 // add and effect module. Also creates the effect chain is none exists for 327 // the effects audio session 328 status_t addEffect_l(const sp< EffectModule>& effect); 329 // remove and effect module. Also removes the effect chain is this was the last 330 // effect 331 void removeEffect_l(const sp< EffectModule>& effect); 332 // detach all tracks connected to an auxiliary effect detachAuxEffect_l(int effectId __unused)333 virtual void detachAuxEffect_l(int effectId __unused) {} 334 // returns either EFFECT_SESSION if effects on this audio session exist in one 335 // chain, or TRACK_SESSION if tracks on this audio session exist, or both 336 virtual uint32_t hasAudioSession(int sessionId) const = 0; 337 // the value returned by default implementation is not important as the 338 // strategy is only meaningful for PlaybackThread which implements this method getStrategyForSession_l(int sessionId __unused)339 virtual uint32_t getStrategyForSession_l(int sessionId __unused) { return 0; } 340 341 // suspend or restore effect according to the type of effect passed. a NULL 342 // type pointer means suspend all effects in the session 343 void setEffectSuspended(const effect_uuid_t *type, 344 bool suspend, 345 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 346 // check if some effects must be suspended/restored when an effect is enabled 347 // or disabled 348 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 349 bool enabled, 350 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 351 void checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 352 bool enabled, 353 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 354 355 virtual status_t setSyncEvent(const sp<SyncEvent>& event) = 0; 356 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const = 0; 357 358 // Return a reference to a per-thread heap which can be used to allocate IMemory 359 // objects that will be read-only to client processes, read/write to mediaserver, 360 // and shared by all client processes of the thread. 361 // The heap is per-thread rather than common across all threads, because 362 // clients can't be trusted not to modify the offset of the IMemory they receive. 363 // If a thread does not have such a heap, this method returns 0. readOnlyHeap()364 virtual sp<MemoryDealer> readOnlyHeap() const { return 0; } 365 pipeMemory()366 virtual sp<IMemory> pipeMemory() const { return 0; } 367 368 void systemReady(); 369 370 mutable Mutex mLock; 371 372 protected: 373 374 // entry describing an effect being suspended in mSuspendedSessions keyed vector 375 class SuspendedSessionDesc : public RefBase { 376 public: SuspendedSessionDesc()377 SuspendedSessionDesc() : mRefCount(0) {} 378 379 int mRefCount; // number of active suspend requests 380 effect_uuid_t mType; // effect type UUID 381 }; 382 383 void acquireWakeLock(int uid = -1); 384 void acquireWakeLock_l(int uid = -1); 385 void releaseWakeLock(); 386 void releaseWakeLock_l(); 387 void updateWakeLockUids(const SortedVector<int> &uids); 388 void updateWakeLockUids_l(const SortedVector<int> &uids); 389 void getPowerManager_l(); 390 void setEffectSuspended_l(const effect_uuid_t *type, 391 bool suspend, 392 int sessionId); 393 // updated mSuspendedSessions when an effect suspended or restored 394 void updateSuspendedSessions_l(const effect_uuid_t *type, 395 bool suspend, 396 int sessionId); 397 // check if some effects must be suspended when an effect chain is added 398 void checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain); 399 400 String16 getWakeLockTag(); 401 preExit()402 virtual void preExit() { } 403 404 friend class AudioFlinger; // for mEffectChains 405 406 const type_t mType; 407 408 // Used by parameters, config events, addTrack_l, exit 409 Condition mWaitWorkCV; 410 411 const sp<AudioFlinger> mAudioFlinger; 412 413 // updated by PlaybackThread::readOutputParameters_l() or 414 // RecordThread::readInputParameters_l() 415 uint32_t mSampleRate; 416 size_t mFrameCount; // output HAL, direct output, record 417 audio_channel_mask_t mChannelMask; 418 uint32_t mChannelCount; 419 size_t mFrameSize; 420 // not HAL frame size, this is for output sink (to pipe to fast mixer) 421 audio_format_t mFormat; // Source format for Recording and 422 // Sink format for Playback. 423 // Sink format may be different than 424 // HAL format if Fastmixer is used. 425 audio_format_t mHALFormat; 426 size_t mBufferSize; // HAL buffer size for read() or write() 427 428 Vector< sp<ConfigEvent> > mConfigEvents; 429 Vector< sp<ConfigEvent> > mPendingConfigEvents; // events awaiting system ready 430 431 // These fields are written and read by thread itself without lock or barrier, 432 // and read by other threads without lock or barrier via standby(), outDevice() 433 // and inDevice(). 434 // Because of the absence of a lock or barrier, any other thread that reads 435 // these fields must use the information in isolation, or be prepared to deal 436 // with possibility that it might be inconsistent with other information. 437 bool mStandby; // Whether thread is currently in standby. 438 audio_devices_t mOutDevice; // output device 439 audio_devices_t mInDevice; // input device 440 audio_devices_t mPrevOutDevice; // previous output device 441 audio_devices_t mPrevInDevice; // previous input device 442 struct audio_patch mPatch; 443 audio_source_t mAudioSource; 444 445 const audio_io_handle_t mId; 446 Vector< sp<EffectChain> > mEffectChains; 447 448 static const int kThreadNameLength = 16; // prctl(PR_SET_NAME) limit 449 char mThreadName[kThreadNameLength]; // guaranteed NUL-terminated 450 sp<IPowerManager> mPowerManager; 451 sp<IBinder> mWakeLockToken; 452 const sp<PMDeathRecipient> mDeathRecipient; 453 // list of suspended effects per session and per type. The first vector is 454 // keyed by session ID, the second by type UUID timeLow field 455 KeyedVector< int, KeyedVector< int, sp<SuspendedSessionDesc> > > 456 mSuspendedSessions; 457 static const size_t kLogSize = 4 * 1024; 458 sp<NBLog::Writer> mNBLogWriter; 459 bool mSystemReady; 460 }; 461 462 // --- PlaybackThread --- 463 class PlaybackThread : public ThreadBase { 464 public: 465 466 #include "PlaybackTracks.h" 467 468 enum mixer_state { 469 MIXER_IDLE, // no active tracks 470 MIXER_TRACKS_ENABLED, // at least one active track, but no track has any data ready 471 MIXER_TRACKS_READY, // at least one active track, and at least one track has data 472 MIXER_DRAIN_TRACK, // drain currently playing track 473 MIXER_DRAIN_ALL, // fully drain the hardware 474 // standby mode does not have an enum value 475 // suspend by audio policy manager is orthogonal to mixer state 476 }; 477 478 // retry count before removing active track in case of underrun on offloaded thread: 479 // we need to make sure that AudioTrack client has enough time to send large buffers 480 //FIXME may be more appropriate if expressed in time units. Need to revise how underrun is handled 481 // for offloaded tracks 482 static const int8_t kMaxTrackRetriesOffload = 20; 483 484 PlaybackThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 485 audio_io_handle_t id, audio_devices_t device, type_t type, bool systemReady); 486 virtual ~PlaybackThread(); 487 488 void dump(int fd, const Vector<String16>& args); 489 490 // Thread virtuals 491 virtual bool threadLoop(); 492 493 // RefBase 494 virtual void onFirstRef(); 495 496 protected: 497 // Code snippets that were lifted up out of threadLoop() 498 virtual void threadLoop_mix() = 0; 499 virtual void threadLoop_sleepTime() = 0; 500 virtual ssize_t threadLoop_write(); 501 virtual void threadLoop_drain(); 502 virtual void threadLoop_standby(); 503 virtual void threadLoop_exit(); 504 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 505 506 // prepareTracks_l reads and writes mActiveTracks, and returns 507 // the pending set of tracks to remove via Vector 'tracksToRemove'. The caller 508 // is responsible for clearing or destroying this Vector later on, when it 509 // is safe to do so. That will drop the final ref count and destroy the tracks. 510 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove) = 0; 511 void removeTracks_l(const Vector< sp<Track> >& tracksToRemove); 512 513 void writeCallback(); 514 void resetWriteBlocked(uint32_t sequence); 515 void drainCallback(); 516 void resetDraining(uint32_t sequence); 517 518 static int asyncCallback(stream_callback_event_t event, void *param, void *cookie); 519 520 virtual bool waitingAsyncCallback(); 521 virtual bool waitingAsyncCallback_l(); 522 virtual bool shouldStandby_l(); 523 virtual void onAddNewTrack_l(); 524 525 // ThreadBase virtuals 526 virtual void preExit(); 527 528 public: 529 initCheck()530 virtual status_t initCheck() const { return (mOutput == NULL) ? NO_INIT : NO_ERROR; } 531 532 // return estimated latency in milliseconds, as reported by HAL 533 uint32_t latency() const; 534 // same, but lock must already be held 535 uint32_t latency_l() const; 536 537 void setMasterVolume(float value); 538 void setMasterMute(bool muted); 539 540 void setStreamVolume(audio_stream_type_t stream, float value); 541 void setStreamMute(audio_stream_type_t stream, bool muted); 542 543 float streamVolume(audio_stream_type_t stream) const; 544 545 sp<Track> createTrack_l( 546 const sp<AudioFlinger::Client>& client, 547 audio_stream_type_t streamType, 548 uint32_t sampleRate, 549 audio_format_t format, 550 audio_channel_mask_t channelMask, 551 size_t *pFrameCount, 552 const sp<IMemory>& sharedBuffer, 553 int sessionId, 554 IAudioFlinger::track_flags_t *flags, 555 pid_t tid, 556 int uid, 557 status_t *status /*non-NULL*/); 558 559 AudioStreamOut* getOutput() const; 560 AudioStreamOut* clearOutput(); 561 virtual audio_stream_t* stream() const; 562 563 // a very large number of suspend() will eventually wraparound, but unlikely suspend()564 void suspend() { (void) android_atomic_inc(&mSuspended); } restore()565 void restore() 566 { 567 // if restore() is done without suspend(), get back into 568 // range so that the next suspend() will operate correctly 569 if (android_atomic_dec(&mSuspended) <= 0) { 570 android_atomic_release_store(0, &mSuspended); 571 } 572 } isSuspended()573 bool isSuspended() const 574 { return android_atomic_acquire_load(&mSuspended) > 0; } 575 576 virtual String8 getParameters(const String8& keys); 577 virtual void ioConfigChanged(audio_io_config_event event, pid_t pid = 0); 578 status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames); 579 // FIXME rename mixBuffer() to sinkBuffer() and remove int16_t* dependency. 580 // Consider also removing and passing an explicit mMainBuffer initialization 581 // parameter to AF::PlaybackThread::Track::Track(). mixBuffer()582 int16_t *mixBuffer() const { 583 return reinterpret_cast<int16_t *>(mSinkBuffer); }; 584 585 virtual void detachAuxEffect_l(int effectId); 586 status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track> track, 587 int EffectId); 588 status_t attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track> track, 589 int EffectId); 590 591 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 592 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 593 virtual uint32_t hasAudioSession(int sessionId) const; 594 virtual uint32_t getStrategyForSession_l(int sessionId); 595 596 597 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 598 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 599 600 // called with AudioFlinger lock held 601 void invalidateTracks(audio_stream_type_t streamType); 602 frameCount()603 virtual size_t frameCount() const { return mNormalFrameCount; } 604 605 // Return's the HAL's frame count i.e. fast mixer buffer size. frameCountHAL()606 size_t frameCountHAL() const { return mFrameCount; } 607 608 status_t getTimestamp_l(AudioTimestamp& timestamp); 609 610 void addPatchTrack(const sp<PatchTrack>& track); 611 void deletePatchTrack(const sp<PatchTrack>& track); 612 613 virtual void getAudioPortConfig(struct audio_port_config *config); 614 615 protected: 616 // updated by readOutputParameters_l() 617 size_t mNormalFrameCount; // normal mixer and effects 618 619 bool mThreadThrottle; // throttle the thread processing 620 uint32_t mThreadThrottleTimeMs; // throttle time for MIXER threads 621 uint32_t mThreadThrottleEndMs; // notify once per throttling 622 uint32_t mHalfBufferMs; // half the buffer size in milliseconds 623 624 void* mSinkBuffer; // frame size aligned sink buffer 625 626 // TODO: 627 // Rearrange the buffer info into a struct/class with 628 // clear, copy, construction, destruction methods. 629 // 630 // mSinkBuffer also has associated with it: 631 // 632 // mSinkBufferSize: Sink Buffer Size 633 // mFormat: Sink Buffer Format 634 635 // Mixer Buffer (mMixerBuffer*) 636 // 637 // In the case of floating point or multichannel data, which is not in the 638 // sink format, it is required to accumulate in a higher precision or greater channel count 639 // buffer before downmixing or data conversion to the sink buffer. 640 641 // Set to "true" to enable the Mixer Buffer otherwise mixer output goes to sink buffer. 642 bool mMixerBufferEnabled; 643 644 // Storage, 32 byte aligned (may make this alignment a requirement later). 645 // Due to constraints on mNormalFrameCount, the buffer size is a multiple of 16 frames. 646 void* mMixerBuffer; 647 648 // Size of mMixerBuffer in bytes: mNormalFrameCount * #channels * sampsize. 649 size_t mMixerBufferSize; 650 651 // The audio format of mMixerBuffer. Set to AUDIO_FORMAT_PCM_(FLOAT|16_BIT) only. 652 audio_format_t mMixerBufferFormat; 653 654 // An internal flag set to true by MixerThread::prepareTracks_l() 655 // when mMixerBuffer contains valid data after mixing. 656 bool mMixerBufferValid; 657 658 // Effects Buffer (mEffectsBuffer*) 659 // 660 // In the case of effects data, which is not in the sink format, 661 // it is required to accumulate in a different buffer before data conversion 662 // to the sink buffer. 663 664 // Set to "true" to enable the Effects Buffer otherwise effects output goes to sink buffer. 665 bool mEffectBufferEnabled; 666 667 // Storage, 32 byte aligned (may make this alignment a requirement later). 668 // Due to constraints on mNormalFrameCount, the buffer size is a multiple of 16 frames. 669 void* mEffectBuffer; 670 671 // Size of mEffectsBuffer in bytes: mNormalFrameCount * #channels * sampsize. 672 size_t mEffectBufferSize; 673 674 // The audio format of mEffectsBuffer. Set to AUDIO_FORMAT_PCM_16_BIT only. 675 audio_format_t mEffectBufferFormat; 676 677 // An internal flag set to true by MixerThread::prepareTracks_l() 678 // when mEffectsBuffer contains valid data after mixing. 679 // 680 // When this is set, all mixer data is routed into the effects buffer 681 // for any processing (including output processing). 682 bool mEffectBufferValid; 683 684 // suspend count, > 0 means suspended. While suspended, the thread continues to pull from 685 // tracks and mix, but doesn't write to HAL. A2DP and SCO HAL implementations can't handle 686 // concurrent use of both of them, so Audio Policy Service suspends one of the threads to 687 // workaround that restriction. 688 // 'volatile' means accessed via atomic operations and no lock. 689 volatile int32_t mSuspended; 690 691 // FIXME overflows every 6+ hours at 44.1 kHz stereo 16-bit samples 692 // mFramesWritten would be better, or 64-bit even better 693 size_t mBytesWritten; 694 private: 695 // mMasterMute is in both PlaybackThread and in AudioFlinger. When a 696 // PlaybackThread needs to find out if master-muted, it checks it's local 697 // copy rather than the one in AudioFlinger. This optimization saves a lock. 698 bool mMasterMute; setMasterMute_l(bool muted)699 void setMasterMute_l(bool muted) { mMasterMute = muted; } 700 protected: 701 SortedVector< wp<Track> > mActiveTracks; // FIXME check if this could be sp<> 702 SortedVector<int> mWakeLockUids; 703 int mActiveTracksGeneration; 704 wp<Track> mLatestActiveTrack; // latest track added to mActiveTracks 705 706 // Allocate a track name for a given channel mask. 707 // Returns name >= 0 if successful, -1 on failure. 708 virtual int getTrackName_l(audio_channel_mask_t channelMask, 709 audio_format_t format, int sessionId) = 0; 710 virtual void deleteTrackName_l(int name) = 0; 711 712 // Time to sleep between cycles when: 713 virtual uint32_t activeSleepTimeUs() const; // mixer state MIXER_TRACKS_ENABLED 714 virtual uint32_t idleSleepTimeUs() const = 0; // mixer state MIXER_IDLE 715 virtual uint32_t suspendSleepTimeUs() const = 0; // audio policy manager suspended us 716 // No sleep when mixer state == MIXER_TRACKS_READY; relies on audio HAL stream->write() 717 // No sleep in standby mode; waits on a condition 718 719 // Code snippets that are temporarily lifted up out of threadLoop() until the merge 720 void checkSilentMode_l(); 721 722 // Non-trivial for DUPLICATING only saveOutputTracks()723 virtual void saveOutputTracks() { } clearOutputTracks()724 virtual void clearOutputTracks() { } 725 726 // Cache various calculated values, at threadLoop() entry and after a parameter change 727 virtual void cacheParameters_l(); 728 729 virtual uint32_t correctLatency_l(uint32_t latency) const; 730 731 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 732 audio_patch_handle_t *handle); 733 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 734 usesHwAvSync()735 bool usesHwAvSync() const { return (mType == DIRECT) && (mOutput != NULL) 736 && mHwSupportsPause 737 && (mOutput->flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC); } 738 739 private: 740 741 friend class AudioFlinger; // for numerous 742 743 PlaybackThread& operator = (const PlaybackThread&); 744 745 status_t addTrack_l(const sp<Track>& track); 746 bool destroyTrack_l(const sp<Track>& track); 747 void removeTrack_l(const sp<Track>& track); 748 void broadcast_l(); 749 750 void readOutputParameters_l(); 751 752 virtual void dumpInternals(int fd, const Vector<String16>& args); 753 void dumpTracks(int fd, const Vector<String16>& args); 754 755 SortedVector< sp<Track> > mTracks; 756 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 757 AudioStreamOut *mOutput; 758 759 float mMasterVolume; 760 nsecs_t mLastWriteTime; 761 int mNumWrites; 762 int mNumDelayedWrites; 763 bool mInWrite; 764 765 // FIXME rename these former local variables of threadLoop to standard "m" names 766 nsecs_t mStandbyTimeNs; 767 size_t mSinkBufferSize; 768 769 // cached copies of activeSleepTimeUs() and idleSleepTimeUs() made by cacheParameters_l() 770 uint32_t mActiveSleepTimeUs; 771 uint32_t mIdleSleepTimeUs; 772 773 uint32_t mSleepTimeUs; 774 775 // mixer status returned by prepareTracks_l() 776 mixer_state mMixerStatus; // current cycle 777 // previous cycle when in prepareTracks_l() 778 mixer_state mMixerStatusIgnoringFastTracks; 779 // FIXME or a separate ready state per track 780 781 // FIXME move these declarations into the specific sub-class that needs them 782 // MIXER only 783 uint32_t sleepTimeShift; 784 785 // same as AudioFlinger::mStandbyTimeInNsecs except for DIRECT which uses a shorter value 786 nsecs_t mStandbyDelayNs; 787 788 // MIXER only 789 nsecs_t maxPeriod; 790 791 // DUPLICATING only 792 uint32_t writeFrames; 793 794 size_t mBytesRemaining; 795 size_t mCurrentWriteLength; 796 bool mUseAsyncWrite; 797 // mWriteAckSequence contains current write sequence on bits 31-1. The write sequence is 798 // incremented each time a write(), a flush() or a standby() occurs. 799 // Bit 0 is set when a write blocks and indicates a callback is expected. 800 // Bit 0 is reset by the async callback thread calling resetWriteBlocked(). Out of sequence 801 // callbacks are ignored. 802 uint32_t mWriteAckSequence; 803 // mDrainSequence contains current drain sequence on bits 31-1. The drain sequence is 804 // incremented each time a drain is requested or a flush() or standby() occurs. 805 // Bit 0 is set when the drain() command is called at the HAL and indicates a callback is 806 // expected. 807 // Bit 0 is reset by the async callback thread calling resetDraining(). Out of sequence 808 // callbacks are ignored. 809 uint32_t mDrainSequence; 810 // A condition that must be evaluated by prepareTrack_l() has changed and we must not wait 811 // for async write callback in the thread loop before evaluating it 812 bool mSignalPending; 813 sp<AsyncCallbackThread> mCallbackThread; 814 815 private: 816 // The HAL output sink is treated as non-blocking, but current implementation is blocking 817 sp<NBAIO_Sink> mOutputSink; 818 // If a fast mixer is present, the blocking pipe sink, otherwise clear 819 sp<NBAIO_Sink> mPipeSink; 820 // The current sink for the normal mixer to write it's (sub)mix, mOutputSink or mPipeSink 821 sp<NBAIO_Sink> mNormalSink; 822 #ifdef TEE_SINK 823 // For dumpsys 824 sp<NBAIO_Sink> mTeeSink; 825 sp<NBAIO_Source> mTeeSource; 826 #endif 827 uint32_t mScreenState; // cached copy of gScreenState 828 static const size_t kFastMixerLogSize = 4 * 1024; 829 sp<NBLog::Writer> mFastMixerNBLogWriter; 830 public: 831 virtual bool hasFastMixer() const = 0; getFastTrackUnderruns(size_t fastIndex __unused)832 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex __unused) const 833 { FastTrackUnderruns dummy; return dummy; } 834 835 protected: 836 // accessed by both binder threads and within threadLoop(), lock on mutex needed 837 unsigned mFastTrackAvailMask; // bit i set if fast track [i] is available 838 bool mHwSupportsPause; 839 bool mHwPaused; 840 bool mFlushPending; 841 private: 842 // timestamp latch: 843 // D input is written by threadLoop_write while mutex is unlocked, and read while locked 844 // Q output is written while locked, and read while locked 845 struct { 846 AudioTimestamp mTimestamp; 847 uint32_t mUnpresentedFrames; 848 KeyedVector<Track *, uint32_t> mFramesReleased; 849 } mLatchD, mLatchQ; 850 bool mLatchDValid; // true means mLatchD is valid 851 // (except for mFramesReleased which is filled in later), 852 // and clock it into latch at next opportunity 853 bool mLatchQValid; // true means mLatchQ is valid 854 }; 855 856 class MixerThread : public PlaybackThread { 857 public: 858 MixerThread(const sp<AudioFlinger>& audioFlinger, 859 AudioStreamOut* output, 860 audio_io_handle_t id, 861 audio_devices_t device, 862 bool systemReady, 863 type_t type = MIXER); 864 virtual ~MixerThread(); 865 866 // Thread virtuals 867 868 virtual bool checkForNewParameter_l(const String8& keyValuePair, 869 status_t& status); 870 virtual void dumpInternals(int fd, const Vector<String16>& args); 871 872 protected: 873 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 874 virtual int getTrackName_l(audio_channel_mask_t channelMask, 875 audio_format_t format, int sessionId); 876 virtual void deleteTrackName_l(int name); 877 virtual uint32_t idleSleepTimeUs() const; 878 virtual uint32_t suspendSleepTimeUs() const; 879 virtual void cacheParameters_l(); 880 881 // threadLoop snippets 882 virtual ssize_t threadLoop_write(); 883 virtual void threadLoop_standby(); 884 virtual void threadLoop_mix(); 885 virtual void threadLoop_sleepTime(); 886 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 887 virtual uint32_t correctLatency_l(uint32_t latency) const; 888 889 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 890 audio_patch_handle_t *handle); 891 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 892 893 AudioMixer* mAudioMixer; // normal mixer 894 private: 895 // one-time initialization, no locks required 896 sp<FastMixer> mFastMixer; // non-0 if there is also a fast mixer 897 sp<AudioWatchdog> mAudioWatchdog; // non-0 if there is an audio watchdog thread 898 899 // contents are not guaranteed to be consistent, no locks required 900 FastMixerDumpState mFastMixerDumpState; 901 #ifdef STATE_QUEUE_DUMP 902 StateQueueObserverDump mStateQueueObserverDump; 903 StateQueueMutatorDump mStateQueueMutatorDump; 904 #endif 905 AudioWatchdogDump mAudioWatchdogDump; 906 907 // accessible only within the threadLoop(), no locks required 908 // mFastMixer->sq() // for mutating and pushing state 909 int32_t mFastMixerFutex; // for cold idle 910 911 public: hasFastMixer()912 virtual bool hasFastMixer() const { return mFastMixer != 0; } getFastTrackUnderruns(size_t fastIndex)913 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const { 914 ALOG_ASSERT(fastIndex < FastMixerState::kMaxFastTracks); 915 return mFastMixerDumpState.mTracks[fastIndex].mUnderruns; 916 } 917 918 }; 919 920 class DirectOutputThread : public PlaybackThread { 921 public: 922 923 DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 924 audio_io_handle_t id, audio_devices_t device, bool systemReady); 925 virtual ~DirectOutputThread(); 926 927 // Thread virtuals 928 929 virtual bool checkForNewParameter_l(const String8& keyValuePair, 930 status_t& status); 931 virtual void flushHw_l(); 932 933 protected: 934 virtual int getTrackName_l(audio_channel_mask_t channelMask, 935 audio_format_t format, int sessionId); 936 virtual void deleteTrackName_l(int name); 937 virtual uint32_t activeSleepTimeUs() const; 938 virtual uint32_t idleSleepTimeUs() const; 939 virtual uint32_t suspendSleepTimeUs() const; 940 virtual void cacheParameters_l(); 941 942 // threadLoop snippets 943 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 944 virtual void threadLoop_mix(); 945 virtual void threadLoop_sleepTime(); 946 virtual void threadLoop_exit(); 947 virtual bool shouldStandby_l(); 948 949 virtual void onAddNewTrack_l(); 950 951 // volumes last sent to audio HAL with stream->set_volume() 952 float mLeftVolFloat; 953 float mRightVolFloat; 954 955 DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 956 audio_io_handle_t id, uint32_t device, ThreadBase::type_t type, 957 bool systemReady); 958 void processVolume_l(Track *track, bool lastTrack); 959 960 // prepareTracks_l() tells threadLoop_mix() the name of the single active track 961 sp<Track> mActiveTrack; 962 963 wp<Track> mPreviousTrack; // used to detect track switch 964 965 public: hasFastMixer()966 virtual bool hasFastMixer() const { return false; } 967 }; 968 969 class OffloadThread : public DirectOutputThread { 970 public: 971 972 OffloadThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 973 audio_io_handle_t id, uint32_t device, bool systemReady); ~OffloadThread()974 virtual ~OffloadThread() {}; 975 virtual void flushHw_l(); 976 977 protected: 978 // threadLoop snippets 979 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 980 virtual void threadLoop_exit(); 981 982 virtual bool waitingAsyncCallback(); 983 virtual bool waitingAsyncCallback_l(); 984 985 private: 986 size_t mPausedWriteLength; // length in bytes of write interrupted by pause 987 size_t mPausedBytesRemaining; // bytes still waiting in mixbuffer after resume 988 }; 989 990 class AsyncCallbackThread : public Thread { 991 public: 992 993 AsyncCallbackThread(const wp<PlaybackThread>& playbackThread); 994 995 virtual ~AsyncCallbackThread(); 996 997 // Thread virtuals 998 virtual bool threadLoop(); 999 1000 // RefBase 1001 virtual void onFirstRef(); 1002 1003 void exit(); 1004 void setWriteBlocked(uint32_t sequence); 1005 void resetWriteBlocked(); 1006 void setDraining(uint32_t sequence); 1007 void resetDraining(); 1008 1009 private: 1010 const wp<PlaybackThread> mPlaybackThread; 1011 // mWriteAckSequence corresponds to the last write sequence passed by the offload thread via 1012 // setWriteBlocked(). The sequence is shifted one bit to the left and the lsb is used 1013 // to indicate that the callback has been received via resetWriteBlocked() 1014 uint32_t mWriteAckSequence; 1015 // mDrainSequence corresponds to the last drain sequence passed by the offload thread via 1016 // setDraining(). The sequence is shifted one bit to the left and the lsb is used 1017 // to indicate that the callback has been received via resetDraining() 1018 uint32_t mDrainSequence; 1019 Condition mWaitWorkCV; 1020 Mutex mLock; 1021 }; 1022 1023 class DuplicatingThread : public MixerThread { 1024 public: 1025 DuplicatingThread(const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread, 1026 audio_io_handle_t id, bool systemReady); 1027 virtual ~DuplicatingThread(); 1028 1029 // Thread virtuals 1030 void addOutputTrack(MixerThread* thread); 1031 void removeOutputTrack(MixerThread* thread); waitTimeMs()1032 uint32_t waitTimeMs() const { return mWaitTimeMs; } 1033 protected: 1034 virtual uint32_t activeSleepTimeUs() const; 1035 1036 private: 1037 bool outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks); 1038 protected: 1039 // threadLoop snippets 1040 virtual void threadLoop_mix(); 1041 virtual void threadLoop_sleepTime(); 1042 virtual ssize_t threadLoop_write(); 1043 virtual void threadLoop_standby(); 1044 virtual void cacheParameters_l(); 1045 1046 private: 1047 // called from threadLoop, addOutputTrack, removeOutputTrack 1048 virtual void updateWaitTime_l(); 1049 protected: 1050 virtual void saveOutputTracks(); 1051 virtual void clearOutputTracks(); 1052 private: 1053 1054 uint32_t mWaitTimeMs; 1055 SortedVector < sp<OutputTrack> > outputTracks; 1056 SortedVector < sp<OutputTrack> > mOutputTracks; 1057 public: hasFastMixer()1058 virtual bool hasFastMixer() const { return false; } 1059 }; 1060 1061 1062 // record thread 1063 class RecordThread : public ThreadBase 1064 { 1065 public: 1066 1067 class RecordTrack; 1068 1069 /* The ResamplerBufferProvider is used to retrieve recorded input data from the 1070 * RecordThread. It maintains local state on the relative position of the read 1071 * position of the RecordTrack compared with the RecordThread. 1072 */ 1073 class ResamplerBufferProvider : public AudioBufferProvider 1074 { 1075 public: ResamplerBufferProvider(RecordTrack * recordTrack)1076 ResamplerBufferProvider(RecordTrack* recordTrack) : 1077 mRecordTrack(recordTrack), 1078 mRsmpInUnrel(0), mRsmpInFront(0) { } ~ResamplerBufferProvider()1079 virtual ~ResamplerBufferProvider() { } 1080 1081 // called to set the ResamplerBufferProvider to head of the RecordThread data buffer, 1082 // skipping any previous data read from the hal. 1083 virtual void reset(); 1084 1085 /* Synchronizes RecordTrack position with the RecordThread. 1086 * Calculates available frames and handle overruns if the RecordThread 1087 * has advanced faster than the ResamplerBufferProvider has retrieved data. 1088 * TODO: why not do this for every getNextBuffer? 1089 * 1090 * Parameters 1091 * framesAvailable: pointer to optional output size_t to store record track 1092 * frames available. 1093 * hasOverrun: pointer to optional boolean, returns true if track has overrun. 1094 */ 1095 1096 virtual void sync(size_t *framesAvailable = NULL, bool *hasOverrun = NULL); 1097 1098 // AudioBufferProvider interface 1099 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts); 1100 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 1101 private: 1102 RecordTrack * const mRecordTrack; 1103 size_t mRsmpInUnrel; // unreleased frames remaining from 1104 // most recent getNextBuffer 1105 // for debug only 1106 int32_t mRsmpInFront; // next available frame 1107 // rolling counter that is never cleared 1108 }; 1109 1110 /* The RecordBufferConverter is used for format, channel, and sample rate 1111 * conversion for a RecordTrack. 1112 * 1113 * TODO: Self contained, so move to a separate file later. 1114 * 1115 * RecordBufferConverter uses the convert() method rather than exposing a 1116 * buffer provider interface; this is to save a memory copy. 1117 */ 1118 class RecordBufferConverter 1119 { 1120 public: 1121 RecordBufferConverter( 1122 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 1123 uint32_t srcSampleRate, 1124 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 1125 uint32_t dstSampleRate); 1126 1127 ~RecordBufferConverter(); 1128 1129 /* Converts input data from an AudioBufferProvider by format, channelMask, 1130 * and sampleRate to a destination buffer. 1131 * 1132 * Parameters 1133 * dst: buffer to place the converted data. 1134 * provider: buffer provider to obtain source data. 1135 * frames: number of frames to convert 1136 * 1137 * Returns the number of frames converted. 1138 */ 1139 size_t convert(void *dst, AudioBufferProvider *provider, size_t frames); 1140 1141 // returns NO_ERROR if constructor was successful initCheck()1142 status_t initCheck() const { 1143 // mSrcChannelMask set on successful updateParameters 1144 return mSrcChannelMask != AUDIO_CHANNEL_INVALID ? NO_ERROR : NO_INIT; 1145 } 1146 1147 // allows dynamic reconfigure of all parameters 1148 status_t updateParameters( 1149 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 1150 uint32_t srcSampleRate, 1151 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 1152 uint32_t dstSampleRate); 1153 1154 // called to reset resampler buffers on record track discontinuity reset()1155 void reset() { 1156 if (mResampler != NULL) { 1157 mResampler->reset(); 1158 } 1159 } 1160 1161 private: 1162 // format conversion when not using resampler 1163 void convertNoResampler(void *dst, const void *src, size_t frames); 1164 1165 // format conversion when using resampler; modifies src in-place 1166 void convertResampler(void *dst, /*not-a-const*/ void *src, size_t frames); 1167 1168 // user provided information 1169 audio_channel_mask_t mSrcChannelMask; 1170 audio_format_t mSrcFormat; 1171 uint32_t mSrcSampleRate; 1172 audio_channel_mask_t mDstChannelMask; 1173 audio_format_t mDstFormat; 1174 uint32_t mDstSampleRate; 1175 1176 // derived information 1177 uint32_t mSrcChannelCount; 1178 uint32_t mDstChannelCount; 1179 size_t mDstFrameSize; 1180 1181 // format conversion buffer 1182 void *mBuf; 1183 size_t mBufFrames; 1184 size_t mBufFrameSize; 1185 1186 // resampler info 1187 AudioResampler *mResampler; 1188 1189 bool mIsLegacyDownmix; // legacy stereo to mono conversion needed 1190 bool mIsLegacyUpmix; // legacy mono to stereo conversion needed 1191 bool mRequiresFloat; // data processing requires float (e.g. resampler) 1192 PassthruBufferProvider *mInputConverterProvider; // converts input to float 1193 int8_t mIdxAry[sizeof(uint32_t) * 8]; // used for channel mask conversion 1194 }; 1195 1196 #include "RecordTracks.h" 1197 1198 RecordThread(const sp<AudioFlinger>& audioFlinger, 1199 AudioStreamIn *input, 1200 audio_io_handle_t id, 1201 audio_devices_t outDevice, 1202 audio_devices_t inDevice, 1203 bool systemReady 1204 #ifdef TEE_SINK 1205 , const sp<NBAIO_Sink>& teeSink 1206 #endif 1207 ); 1208 virtual ~RecordThread(); 1209 1210 // no addTrack_l ? 1211 void destroyTrack_l(const sp<RecordTrack>& track); 1212 void removeTrack_l(const sp<RecordTrack>& track); 1213 1214 void dumpInternals(int fd, const Vector<String16>& args); 1215 void dumpTracks(int fd, const Vector<String16>& args); 1216 1217 // Thread virtuals 1218 virtual bool threadLoop(); 1219 1220 // RefBase 1221 virtual void onFirstRef(); 1222 initCheck()1223 virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; } 1224 readOnlyHeap()1225 virtual sp<MemoryDealer> readOnlyHeap() const { return mReadOnlyHeap; } 1226 pipeMemory()1227 virtual sp<IMemory> pipeMemory() const { return mPipeMemory; } 1228 1229 sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l( 1230 const sp<AudioFlinger::Client>& client, 1231 uint32_t sampleRate, 1232 audio_format_t format, 1233 audio_channel_mask_t channelMask, 1234 size_t *pFrameCount, 1235 int sessionId, 1236 size_t *notificationFrames, 1237 int uid, 1238 IAudioFlinger::track_flags_t *flags, 1239 pid_t tid, 1240 status_t *status /*non-NULL*/); 1241 1242 status_t start(RecordTrack* recordTrack, 1243 AudioSystem::sync_event_t event, 1244 int triggerSession); 1245 1246 // ask the thread to stop the specified track, and 1247 // return true if the caller should then do it's part of the stopping process 1248 bool stop(RecordTrack* recordTrack); 1249 1250 void dump(int fd, const Vector<String16>& args); 1251 AudioStreamIn* clearInput(); 1252 virtual audio_stream_t* stream() const; 1253 1254 1255 virtual bool checkForNewParameter_l(const String8& keyValuePair, 1256 status_t& status); cacheParameters_l()1257 virtual void cacheParameters_l() {} 1258 virtual String8 getParameters(const String8& keys); 1259 virtual void ioConfigChanged(audio_io_config_event event, pid_t pid = 0); 1260 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 1261 audio_patch_handle_t *handle); 1262 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 1263 1264 void addPatchRecord(const sp<PatchRecord>& record); 1265 void deletePatchRecord(const sp<PatchRecord>& record); 1266 1267 void readInputParameters_l(); 1268 virtual uint32_t getInputFramesLost(); 1269 1270 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1271 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1272 virtual uint32_t hasAudioSession(int sessionId) const; 1273 1274 // Return the set of unique session IDs across all tracks. 1275 // The keys are the session IDs, and the associated values are meaningless. 1276 // FIXME replace by Set [and implement Bag/Multiset for other uses]. 1277 KeyedVector<int, bool> sessionIds() const; 1278 1279 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 1280 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 1281 1282 static void syncStartEventCallback(const wp<SyncEvent>& event); 1283 frameCount()1284 virtual size_t frameCount() const { return mFrameCount; } hasFastCapture()1285 bool hasFastCapture() const { return mFastCapture != 0; } 1286 virtual void getAudioPortConfig(struct audio_port_config *config); 1287 1288 private: 1289 // Enter standby if not already in standby, and set mStandby flag 1290 void standbyIfNotAlreadyInStandby(); 1291 1292 // Call the HAL standby method unconditionally, and don't change mStandby flag 1293 void inputStandBy(); 1294 1295 AudioStreamIn *mInput; 1296 SortedVector < sp<RecordTrack> > mTracks; 1297 // mActiveTracks has dual roles: it indicates the current active track(s), and 1298 // is used together with mStartStopCond to indicate start()/stop() progress 1299 SortedVector< sp<RecordTrack> > mActiveTracks; 1300 // generation counter for mActiveTracks 1301 int mActiveTracksGen; 1302 Condition mStartStopCond; 1303 1304 // resampler converts input at HAL Hz to output at AudioRecord client Hz 1305 void *mRsmpInBuffer; // 1306 size_t mRsmpInFrames; // size of resampler input in frames 1307 size_t mRsmpInFramesP2;// size rounded up to a power-of-2 1308 1309 // rolling index that is never cleared 1310 int32_t mRsmpInRear; // last filled frame + 1 1311 1312 // For dumpsys 1313 const sp<NBAIO_Sink> mTeeSink; 1314 1315 const sp<MemoryDealer> mReadOnlyHeap; 1316 1317 // one-time initialization, no locks required 1318 sp<FastCapture> mFastCapture; // non-0 if there is also 1319 // a fast capture 1320 1321 // FIXME audio watchdog thread 1322 1323 // contents are not guaranteed to be consistent, no locks required 1324 FastCaptureDumpState mFastCaptureDumpState; 1325 #ifdef STATE_QUEUE_DUMP 1326 // FIXME StateQueue observer and mutator dump fields 1327 #endif 1328 // FIXME audio watchdog dump 1329 1330 // accessible only within the threadLoop(), no locks required 1331 // mFastCapture->sq() // for mutating and pushing state 1332 int32_t mFastCaptureFutex; // for cold idle 1333 1334 // The HAL input source is treated as non-blocking, 1335 // but current implementation is blocking 1336 sp<NBAIO_Source> mInputSource; 1337 // The source for the normal capture thread to read from: mInputSource or mPipeSource 1338 sp<NBAIO_Source> mNormalSource; 1339 // If a fast capture is present, the non-blocking pipe sink written to by fast capture, 1340 // otherwise clear 1341 sp<NBAIO_Sink> mPipeSink; 1342 // If a fast capture is present, the non-blocking pipe source read by normal thread, 1343 // otherwise clear 1344 sp<NBAIO_Source> mPipeSource; 1345 // Depth of pipe from fast capture to normal thread and fast clients, always power of 2 1346 size_t mPipeFramesP2; 1347 // If a fast capture is present, the Pipe as IMemory, otherwise clear 1348 sp<IMemory> mPipeMemory; 1349 1350 static const size_t kFastCaptureLogSize = 4 * 1024; 1351 sp<NBLog::Writer> mFastCaptureNBLogWriter; 1352 1353 bool mFastTrackAvail; // true if fast track available 1354 }; 1355