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 FAST_SESSION = 0x4 // the audio session corresponds to at least one 307 // fast track 308 }; 309 310 // get effect chain corresponding to session Id. 311 sp<EffectChain> getEffectChain(audio_session_t sessionId); 312 // same as getEffectChain() but must be called with ThreadBase mutex locked 313 sp<EffectChain> getEffectChain_l(audio_session_t sessionId) const; 314 // add an effect chain to the chain list (mEffectChains) 315 virtual status_t addEffectChain_l(const sp<EffectChain>& chain) = 0; 316 // remove an effect chain from the chain list (mEffectChains) 317 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain) = 0; 318 // lock all effect chains Mutexes. Must be called before releasing the 319 // ThreadBase mutex before processing the mixer and effects. This guarantees the 320 // integrity of the chains during the process. 321 // Also sets the parameter 'effectChains' to current value of mEffectChains. 322 void lockEffectChains_l(Vector< sp<EffectChain> >& effectChains); 323 // unlock effect chains after process 324 void unlockEffectChains(const Vector< sp<EffectChain> >& effectChains); 325 // get a copy of mEffectChains vector getEffectChains_l()326 Vector< sp<EffectChain> > getEffectChains_l() const { return mEffectChains; }; 327 // set audio mode to all effect chains 328 void setMode(audio_mode_t mode); 329 // get effect module with corresponding ID on specified audio session 330 sp<AudioFlinger::EffectModule> getEffect(audio_session_t sessionId, int effectId); 331 sp<AudioFlinger::EffectModule> getEffect_l(audio_session_t sessionId, int effectId); 332 // add and effect module. Also creates the effect chain is none exists for 333 // the effects audio session 334 status_t addEffect_l(const sp< EffectModule>& effect); 335 // remove and effect module. Also removes the effect chain is this was the last 336 // effect 337 void removeEffect_l(const sp< EffectModule>& effect); 338 // detach all tracks connected to an auxiliary effect detachAuxEffect_l(int effectId __unused)339 virtual void detachAuxEffect_l(int effectId __unused) {} 340 // returns a combination of: 341 // - EFFECT_SESSION if effects on this audio session exist in one chain 342 // - TRACK_SESSION if tracks on this audio session exist 343 // - FAST_SESSION if fast tracks on this audio session exist 344 virtual uint32_t hasAudioSession_l(audio_session_t sessionId) const = 0; hasAudioSession(audio_session_t sessionId)345 uint32_t hasAudioSession(audio_session_t sessionId) const { 346 Mutex::Autolock _l(mLock); 347 return hasAudioSession_l(sessionId); 348 } 349 350 // the value returned by default implementation is not important as the 351 // strategy is only meaningful for PlaybackThread which implements this method getStrategyForSession_l(audio_session_t sessionId __unused)352 virtual uint32_t getStrategyForSession_l(audio_session_t sessionId __unused) 353 { return 0; } 354 355 // suspend or restore effect according to the type of effect passed. a NULL 356 // type pointer means suspend all effects in the session 357 void setEffectSuspended(const effect_uuid_t *type, 358 bool suspend, 359 audio_session_t sessionId = AUDIO_SESSION_OUTPUT_MIX); 360 // check if some effects must be suspended/restored when an effect is enabled 361 // or disabled 362 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 363 bool enabled, 364 audio_session_t sessionId = 365 AUDIO_SESSION_OUTPUT_MIX); 366 void checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 367 bool enabled, 368 audio_session_t sessionId = 369 AUDIO_SESSION_OUTPUT_MIX); 370 371 virtual status_t setSyncEvent(const sp<SyncEvent>& event) = 0; 372 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const = 0; 373 374 // Return a reference to a per-thread heap which can be used to allocate IMemory 375 // objects that will be read-only to client processes, read/write to mediaserver, 376 // and shared by all client processes of the thread. 377 // The heap is per-thread rather than common across all threads, because 378 // clients can't be trusted not to modify the offset of the IMemory they receive. 379 // If a thread does not have such a heap, this method returns 0. readOnlyHeap()380 virtual sp<MemoryDealer> readOnlyHeap() const { return 0; } 381 pipeMemory()382 virtual sp<IMemory> pipeMemory() const { return 0; } 383 384 void systemReady(); 385 386 // checkEffectCompatibility_l() must be called with ThreadBase::mLock held 387 virtual status_t checkEffectCompatibility_l(const effect_descriptor_t *desc, 388 audio_session_t sessionId) = 0; 389 390 mutable Mutex mLock; 391 392 protected: 393 394 // entry describing an effect being suspended in mSuspendedSessions keyed vector 395 class SuspendedSessionDesc : public RefBase { 396 public: SuspendedSessionDesc()397 SuspendedSessionDesc() : mRefCount(0) {} 398 399 int mRefCount; // number of active suspend requests 400 effect_uuid_t mType; // effect type UUID 401 }; 402 403 void acquireWakeLock(int uid = -1); 404 virtual void acquireWakeLock_l(int uid = -1); 405 void releaseWakeLock(); 406 void releaseWakeLock_l(); 407 void updateWakeLockUids(const SortedVector<int> &uids); 408 void updateWakeLockUids_l(const SortedVector<int> &uids); 409 void getPowerManager_l(); 410 void setEffectSuspended_l(const effect_uuid_t *type, 411 bool suspend, 412 audio_session_t sessionId); 413 // updated mSuspendedSessions when an effect suspended or restored 414 void updateSuspendedSessions_l(const effect_uuid_t *type, 415 bool suspend, 416 audio_session_t sessionId); 417 // check if some effects must be suspended when an effect chain is added 418 void checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain); 419 420 String16 getWakeLockTag(); 421 preExit()422 virtual void preExit() { } setMasterMono_l(bool mono __unused)423 virtual void setMasterMono_l(bool mono __unused) { } requireMonoBlend()424 virtual bool requireMonoBlend() { return false; } 425 426 friend class AudioFlinger; // for mEffectChains 427 428 const type_t mType; 429 430 // Used by parameters, config events, addTrack_l, exit 431 Condition mWaitWorkCV; 432 433 const sp<AudioFlinger> mAudioFlinger; 434 435 // updated by PlaybackThread::readOutputParameters_l() or 436 // RecordThread::readInputParameters_l() 437 uint32_t mSampleRate; 438 size_t mFrameCount; // output HAL, direct output, record 439 audio_channel_mask_t mChannelMask; 440 uint32_t mChannelCount; 441 size_t mFrameSize; 442 // not HAL frame size, this is for output sink (to pipe to fast mixer) 443 audio_format_t mFormat; // Source format for Recording and 444 // Sink format for Playback. 445 // Sink format may be different than 446 // HAL format if Fastmixer is used. 447 audio_format_t mHALFormat; 448 size_t mBufferSize; // HAL buffer size for read() or write() 449 450 Vector< sp<ConfigEvent> > mConfigEvents; 451 Vector< sp<ConfigEvent> > mPendingConfigEvents; // events awaiting system ready 452 453 // These fields are written and read by thread itself without lock or barrier, 454 // and read by other threads without lock or barrier via standby(), outDevice() 455 // and inDevice(). 456 // Because of the absence of a lock or barrier, any other thread that reads 457 // these fields must use the information in isolation, or be prepared to deal 458 // with possibility that it might be inconsistent with other information. 459 bool mStandby; // Whether thread is currently in standby. 460 audio_devices_t mOutDevice; // output device 461 audio_devices_t mInDevice; // input device 462 audio_devices_t mPrevOutDevice; // previous output device 463 audio_devices_t mPrevInDevice; // previous input device 464 struct audio_patch mPatch; 465 audio_source_t mAudioSource; 466 467 const audio_io_handle_t mId; 468 Vector< sp<EffectChain> > mEffectChains; 469 470 static const int kThreadNameLength = 16; // prctl(PR_SET_NAME) limit 471 char mThreadName[kThreadNameLength]; // guaranteed NUL-terminated 472 sp<IPowerManager> mPowerManager; 473 sp<IBinder> mWakeLockToken; 474 const sp<PMDeathRecipient> mDeathRecipient; 475 // list of suspended effects per session and per type. The first (outer) vector is 476 // keyed by session ID, the second (inner) by type UUID timeLow field 477 KeyedVector< audio_session_t, KeyedVector< int, sp<SuspendedSessionDesc> > > 478 mSuspendedSessions; 479 static const size_t kLogSize = 4 * 1024; 480 sp<NBLog::Writer> mNBLogWriter; 481 bool mSystemReady; 482 bool mNotifiedBatteryStart; 483 ExtendedTimestamp mTimestamp; 484 }; 485 486 // --- PlaybackThread --- 487 class PlaybackThread : public ThreadBase { 488 public: 489 490 #include "PlaybackTracks.h" 491 492 enum mixer_state { 493 MIXER_IDLE, // no active tracks 494 MIXER_TRACKS_ENABLED, // at least one active track, but no track has any data ready 495 MIXER_TRACKS_READY, // at least one active track, and at least one track has data 496 MIXER_DRAIN_TRACK, // drain currently playing track 497 MIXER_DRAIN_ALL, // fully drain the hardware 498 // standby mode does not have an enum value 499 // suspend by audio policy manager is orthogonal to mixer state 500 }; 501 502 // retry count before removing active track in case of underrun on offloaded thread: 503 // we need to make sure that AudioTrack client has enough time to send large buffers 504 //FIXME may be more appropriate if expressed in time units. Need to revise how underrun is 505 // handled for offloaded tracks 506 static const int8_t kMaxTrackRetriesOffload = 20; 507 static const int8_t kMaxTrackStartupRetriesOffload = 100; 508 static const int8_t kMaxTrackStopRetriesOffload = 2; 509 510 PlaybackThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 511 audio_io_handle_t id, audio_devices_t device, type_t type, bool systemReady); 512 virtual ~PlaybackThread(); 513 514 void dump(int fd, const Vector<String16>& args); 515 516 // Thread virtuals 517 virtual bool threadLoop(); 518 519 // RefBase 520 virtual void onFirstRef(); 521 522 virtual status_t checkEffectCompatibility_l(const effect_descriptor_t *desc, 523 audio_session_t sessionId); 524 525 protected: 526 // Code snippets that were lifted up out of threadLoop() 527 virtual void threadLoop_mix() = 0; 528 virtual void threadLoop_sleepTime() = 0; 529 virtual ssize_t threadLoop_write(); 530 virtual void threadLoop_drain(); 531 virtual void threadLoop_standby(); 532 virtual void threadLoop_exit(); 533 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 534 535 // prepareTracks_l reads and writes mActiveTracks, and returns 536 // the pending set of tracks to remove via Vector 'tracksToRemove'. The caller 537 // is responsible for clearing or destroying this Vector later on, when it 538 // is safe to do so. That will drop the final ref count and destroy the tracks. 539 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove) = 0; 540 void removeTracks_l(const Vector< sp<Track> >& tracksToRemove); 541 542 void writeCallback(); 543 void resetWriteBlocked(uint32_t sequence); 544 void drainCallback(); 545 void resetDraining(uint32_t sequence); 546 void errorCallback(); 547 548 static int asyncCallback(stream_callback_event_t event, void *param, void *cookie); 549 550 virtual bool waitingAsyncCallback(); 551 virtual bool waitingAsyncCallback_l(); 552 virtual bool shouldStandby_l(); 553 virtual void onAddNewTrack_l(); 554 void onAsyncError(); // error reported by AsyncCallbackThread 555 556 // ThreadBase virtuals 557 virtual void preExit(); 558 keepWakeLock()559 virtual bool keepWakeLock() const { return true; } 560 561 public: 562 initCheck()563 virtual status_t initCheck() const { return (mOutput == NULL) ? NO_INIT : NO_ERROR; } 564 565 // return estimated latency in milliseconds, as reported by HAL 566 uint32_t latency() const; 567 // same, but lock must already be held 568 uint32_t latency_l() const; 569 570 void setMasterVolume(float value); 571 void setMasterMute(bool muted); 572 573 void setStreamVolume(audio_stream_type_t stream, float value); 574 void setStreamMute(audio_stream_type_t stream, bool muted); 575 576 float streamVolume(audio_stream_type_t stream) const; 577 578 sp<Track> createTrack_l( 579 const sp<AudioFlinger::Client>& client, 580 audio_stream_type_t streamType, 581 uint32_t sampleRate, 582 audio_format_t format, 583 audio_channel_mask_t channelMask, 584 size_t *pFrameCount, 585 const sp<IMemory>& sharedBuffer, 586 audio_session_t sessionId, 587 audio_output_flags_t *flags, 588 pid_t tid, 589 int uid, 590 status_t *status /*non-NULL*/); 591 592 AudioStreamOut* getOutput() const; 593 AudioStreamOut* clearOutput(); 594 virtual audio_stream_t* stream() const; 595 596 // a very large number of suspend() will eventually wraparound, but unlikely suspend()597 void suspend() { (void) android_atomic_inc(&mSuspended); } restore()598 void restore() 599 { 600 // if restore() is done without suspend(), get back into 601 // range so that the next suspend() will operate correctly 602 if (android_atomic_dec(&mSuspended) <= 0) { 603 android_atomic_release_store(0, &mSuspended); 604 } 605 } isSuspended()606 bool isSuspended() const 607 { return android_atomic_acquire_load(&mSuspended) > 0; } 608 609 virtual String8 getParameters(const String8& keys); 610 virtual void ioConfigChanged(audio_io_config_event event, pid_t pid = 0); 611 status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames); 612 // FIXME rename mixBuffer() to sinkBuffer() and remove int16_t* dependency. 613 // Consider also removing and passing an explicit mMainBuffer initialization 614 // parameter to AF::PlaybackThread::Track::Track(). mixBuffer()615 int16_t *mixBuffer() const { 616 return reinterpret_cast<int16_t *>(mSinkBuffer); }; 617 618 virtual void detachAuxEffect_l(int effectId); 619 status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track> track, 620 int EffectId); 621 status_t attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track> track, 622 int EffectId); 623 624 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 625 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 626 virtual uint32_t hasAudioSession_l(audio_session_t sessionId) const; 627 virtual uint32_t getStrategyForSession_l(audio_session_t sessionId); 628 629 630 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 631 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 632 633 // called with AudioFlinger lock held 634 bool invalidateTracks_l(audio_stream_type_t streamType); 635 virtual void invalidateTracks(audio_stream_type_t streamType); 636 frameCount()637 virtual size_t frameCount() const { return mNormalFrameCount; } 638 639 status_t getTimestamp_l(AudioTimestamp& timestamp); 640 641 void addPatchTrack(const sp<PatchTrack>& track); 642 void deletePatchTrack(const sp<PatchTrack>& track); 643 644 virtual void getAudioPortConfig(struct audio_port_config *config); 645 646 protected: 647 // updated by readOutputParameters_l() 648 size_t mNormalFrameCount; // normal mixer and effects 649 650 bool mThreadThrottle; // throttle the thread processing 651 uint32_t mThreadThrottleTimeMs; // throttle time for MIXER threads 652 uint32_t mThreadThrottleEndMs; // notify once per throttling 653 uint32_t mHalfBufferMs; // half the buffer size in milliseconds 654 655 void* mSinkBuffer; // frame size aligned sink buffer 656 657 // TODO: 658 // Rearrange the buffer info into a struct/class with 659 // clear, copy, construction, destruction methods. 660 // 661 // mSinkBuffer also has associated with it: 662 // 663 // mSinkBufferSize: Sink Buffer Size 664 // mFormat: Sink Buffer Format 665 666 // Mixer Buffer (mMixerBuffer*) 667 // 668 // In the case of floating point or multichannel data, which is not in the 669 // sink format, it is required to accumulate in a higher precision or greater channel count 670 // buffer before downmixing or data conversion to the sink buffer. 671 672 // Set to "true" to enable the Mixer Buffer otherwise mixer output goes to sink buffer. 673 bool mMixerBufferEnabled; 674 675 // Storage, 32 byte aligned (may make this alignment a requirement later). 676 // Due to constraints on mNormalFrameCount, the buffer size is a multiple of 16 frames. 677 void* mMixerBuffer; 678 679 // Size of mMixerBuffer in bytes: mNormalFrameCount * #channels * sampsize. 680 size_t mMixerBufferSize; 681 682 // The audio format of mMixerBuffer. Set to AUDIO_FORMAT_PCM_(FLOAT|16_BIT) only. 683 audio_format_t mMixerBufferFormat; 684 685 // An internal flag set to true by MixerThread::prepareTracks_l() 686 // when mMixerBuffer contains valid data after mixing. 687 bool mMixerBufferValid; 688 689 // Effects Buffer (mEffectsBuffer*) 690 // 691 // In the case of effects data, which is not in the sink format, 692 // it is required to accumulate in a different buffer before data conversion 693 // to the sink buffer. 694 695 // Set to "true" to enable the Effects Buffer otherwise effects output goes to sink buffer. 696 bool mEffectBufferEnabled; 697 698 // Storage, 32 byte aligned (may make this alignment a requirement later). 699 // Due to constraints on mNormalFrameCount, the buffer size is a multiple of 16 frames. 700 void* mEffectBuffer; 701 702 // Size of mEffectsBuffer in bytes: mNormalFrameCount * #channels * sampsize. 703 size_t mEffectBufferSize; 704 705 // The audio format of mEffectsBuffer. Set to AUDIO_FORMAT_PCM_16_BIT only. 706 audio_format_t mEffectBufferFormat; 707 708 // An internal flag set to true by MixerThread::prepareTracks_l() 709 // when mEffectsBuffer contains valid data after mixing. 710 // 711 // When this is set, all mixer data is routed into the effects buffer 712 // for any processing (including output processing). 713 bool mEffectBufferValid; 714 715 // suspend count, > 0 means suspended. While suspended, the thread continues to pull from 716 // tracks and mix, but doesn't write to HAL. A2DP and SCO HAL implementations can't handle 717 // concurrent use of both of them, so Audio Policy Service suspends one of the threads to 718 // workaround that restriction. 719 // 'volatile' means accessed via atomic operations and no lock. 720 volatile int32_t mSuspended; 721 722 int64_t mBytesWritten; 723 int64_t mFramesWritten; // not reset on standby 724 int64_t mSuspendedFrames; // not reset on standby 725 private: 726 // mMasterMute is in both PlaybackThread and in AudioFlinger. When a 727 // PlaybackThread needs to find out if master-muted, it checks it's local 728 // copy rather than the one in AudioFlinger. This optimization saves a lock. 729 bool mMasterMute; setMasterMute_l(bool muted)730 void setMasterMute_l(bool muted) { mMasterMute = muted; } 731 protected: 732 SortedVector< wp<Track> > mActiveTracks; // FIXME check if this could be sp<> 733 SortedVector<int> mWakeLockUids; 734 int mActiveTracksGeneration; 735 wp<Track> mLatestActiveTrack; // latest track added to mActiveTracks 736 737 // Allocate a track name for a given channel mask. 738 // Returns name >= 0 if successful, -1 on failure. 739 virtual int getTrackName_l(audio_channel_mask_t channelMask, 740 audio_format_t format, audio_session_t sessionId) = 0; 741 virtual void deleteTrackName_l(int name) = 0; 742 743 // Time to sleep between cycles when: 744 virtual uint32_t activeSleepTimeUs() const; // mixer state MIXER_TRACKS_ENABLED 745 virtual uint32_t idleSleepTimeUs() const = 0; // mixer state MIXER_IDLE 746 virtual uint32_t suspendSleepTimeUs() const = 0; // audio policy manager suspended us 747 // No sleep when mixer state == MIXER_TRACKS_READY; relies on audio HAL stream->write() 748 // No sleep in standby mode; waits on a condition 749 750 // Code snippets that are temporarily lifted up out of threadLoop() until the merge 751 void checkSilentMode_l(); 752 753 // Non-trivial for DUPLICATING only saveOutputTracks()754 virtual void saveOutputTracks() { } clearOutputTracks()755 virtual void clearOutputTracks() { } 756 757 // Cache various calculated values, at threadLoop() entry and after a parameter change 758 virtual void cacheParameters_l(); 759 760 virtual uint32_t correctLatency_l(uint32_t latency) const; 761 762 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 763 audio_patch_handle_t *handle); 764 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 765 usesHwAvSync()766 bool usesHwAvSync() const { return (mType == DIRECT) && (mOutput != NULL) 767 && mHwSupportsPause 768 && (mOutput->flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC); } 769 770 private: 771 772 friend class AudioFlinger; // for numerous 773 774 PlaybackThread& operator = (const PlaybackThread&); 775 776 status_t addTrack_l(const sp<Track>& track); 777 bool destroyTrack_l(const sp<Track>& track); 778 void removeTrack_l(const sp<Track>& track); 779 void broadcast_l(); 780 781 void readOutputParameters_l(); 782 783 virtual void dumpInternals(int fd, const Vector<String16>& args); 784 void dumpTracks(int fd, const Vector<String16>& args); 785 786 SortedVector< sp<Track> > mTracks; 787 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 788 AudioStreamOut *mOutput; 789 790 float mMasterVolume; 791 nsecs_t mLastWriteTime; 792 int mNumWrites; 793 int mNumDelayedWrites; 794 bool mInWrite; 795 796 // FIXME rename these former local variables of threadLoop to standard "m" names 797 nsecs_t mStandbyTimeNs; 798 size_t mSinkBufferSize; 799 800 // cached copies of activeSleepTimeUs() and idleSleepTimeUs() made by cacheParameters_l() 801 uint32_t mActiveSleepTimeUs; 802 uint32_t mIdleSleepTimeUs; 803 804 uint32_t mSleepTimeUs; 805 806 // mixer status returned by prepareTracks_l() 807 mixer_state mMixerStatus; // current cycle 808 // previous cycle when in prepareTracks_l() 809 mixer_state mMixerStatusIgnoringFastTracks; 810 // FIXME or a separate ready state per track 811 812 // FIXME move these declarations into the specific sub-class that needs them 813 // MIXER only 814 uint32_t sleepTimeShift; 815 816 // same as AudioFlinger::mStandbyTimeInNsecs except for DIRECT which uses a shorter value 817 nsecs_t mStandbyDelayNs; 818 819 // MIXER only 820 nsecs_t maxPeriod; 821 822 // DUPLICATING only 823 uint32_t writeFrames; 824 825 size_t mBytesRemaining; 826 size_t mCurrentWriteLength; 827 bool mUseAsyncWrite; 828 // mWriteAckSequence contains current write sequence on bits 31-1. The write sequence is 829 // incremented each time a write(), a flush() or a standby() occurs. 830 // Bit 0 is set when a write blocks and indicates a callback is expected. 831 // Bit 0 is reset by the async callback thread calling resetWriteBlocked(). Out of sequence 832 // callbacks are ignored. 833 uint32_t mWriteAckSequence; 834 // mDrainSequence contains current drain sequence on bits 31-1. The drain sequence is 835 // incremented each time a drain is requested or a flush() or standby() occurs. 836 // Bit 0 is set when the drain() command is called at the HAL and indicates a callback is 837 // expected. 838 // Bit 0 is reset by the async callback thread calling resetDraining(). Out of sequence 839 // callbacks are ignored. 840 uint32_t mDrainSequence; 841 // A condition that must be evaluated by prepareTrack_l() has changed and we must not wait 842 // for async write callback in the thread loop before evaluating it 843 bool mSignalPending; 844 sp<AsyncCallbackThread> mCallbackThread; 845 846 private: 847 // The HAL output sink is treated as non-blocking, but current implementation is blocking 848 sp<NBAIO_Sink> mOutputSink; 849 // If a fast mixer is present, the blocking pipe sink, otherwise clear 850 sp<NBAIO_Sink> mPipeSink; 851 // The current sink for the normal mixer to write it's (sub)mix, mOutputSink or mPipeSink 852 sp<NBAIO_Sink> mNormalSink; 853 #ifdef TEE_SINK 854 // For dumpsys 855 sp<NBAIO_Sink> mTeeSink; 856 sp<NBAIO_Source> mTeeSource; 857 #endif 858 uint32_t mScreenState; // cached copy of gScreenState 859 static const size_t kFastMixerLogSize = 4 * 1024; 860 sp<NBLog::Writer> mFastMixerNBLogWriter; 861 public: 862 virtual bool hasFastMixer() const = 0; getFastTrackUnderruns(size_t fastIndex __unused)863 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex __unused) const 864 { FastTrackUnderruns dummy; return dummy; } 865 866 protected: 867 // accessed by both binder threads and within threadLoop(), lock on mutex needed 868 unsigned mFastTrackAvailMask; // bit i set if fast track [i] is available 869 bool mHwSupportsPause; 870 bool mHwPaused; 871 bool mFlushPending; 872 }; 873 874 class MixerThread : public PlaybackThread { 875 public: 876 MixerThread(const sp<AudioFlinger>& audioFlinger, 877 AudioStreamOut* output, 878 audio_io_handle_t id, 879 audio_devices_t device, 880 bool systemReady, 881 type_t type = MIXER); 882 virtual ~MixerThread(); 883 884 // Thread virtuals 885 886 virtual bool checkForNewParameter_l(const String8& keyValuePair, 887 status_t& status); 888 virtual void dumpInternals(int fd, const Vector<String16>& args); 889 890 protected: 891 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 892 virtual int getTrackName_l(audio_channel_mask_t channelMask, 893 audio_format_t format, audio_session_t sessionId); 894 virtual void deleteTrackName_l(int name); 895 virtual uint32_t idleSleepTimeUs() const; 896 virtual uint32_t suspendSleepTimeUs() const; 897 virtual void cacheParameters_l(); 898 899 virtual void acquireWakeLock_l(int uid = -1) { 900 PlaybackThread::acquireWakeLock_l(uid); 901 if (hasFastMixer()) { 902 mFastMixer->setBoottimeOffset( 903 mTimestamp.mTimebaseOffset[ExtendedTimestamp::TIMEBASE_BOOTTIME]); 904 } 905 } 906 907 // threadLoop snippets 908 virtual ssize_t threadLoop_write(); 909 virtual void threadLoop_standby(); 910 virtual void threadLoop_mix(); 911 virtual void threadLoop_sleepTime(); 912 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 913 virtual uint32_t correctLatency_l(uint32_t latency) const; 914 915 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 916 audio_patch_handle_t *handle); 917 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 918 919 AudioMixer* mAudioMixer; // normal mixer 920 private: 921 // one-time initialization, no locks required 922 sp<FastMixer> mFastMixer; // non-0 if there is also a fast mixer 923 sp<AudioWatchdog> mAudioWatchdog; // non-0 if there is an audio watchdog thread 924 925 // contents are not guaranteed to be consistent, no locks required 926 FastMixerDumpState mFastMixerDumpState; 927 #ifdef STATE_QUEUE_DUMP 928 StateQueueObserverDump mStateQueueObserverDump; 929 StateQueueMutatorDump mStateQueueMutatorDump; 930 #endif 931 AudioWatchdogDump mAudioWatchdogDump; 932 933 // accessible only within the threadLoop(), no locks required 934 // mFastMixer->sq() // for mutating and pushing state 935 int32_t mFastMixerFutex; // for cold idle 936 937 std::atomic_bool mMasterMono; 938 public: hasFastMixer()939 virtual bool hasFastMixer() const { return mFastMixer != 0; } getFastTrackUnderruns(size_t fastIndex)940 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const { 941 ALOG_ASSERT(fastIndex < FastMixerState::sMaxFastTracks); 942 return mFastMixerDumpState.mTracks[fastIndex].mUnderruns; 943 } 944 945 protected: setMasterMono_l(bool mono)946 virtual void setMasterMono_l(bool mono) { 947 mMasterMono.store(mono); 948 if (mFastMixer != nullptr) { /* hasFastMixer() */ 949 mFastMixer->setMasterMono(mMasterMono); 950 } 951 } 952 // the FastMixer performs mono blend if it exists. 953 // Blending with limiter is not idempotent, 954 // and blending without limiter is idempotent but inefficient to do twice. requireMonoBlend()955 virtual bool requireMonoBlend() { return mMasterMono.load() && !hasFastMixer(); } 956 }; 957 958 class DirectOutputThread : public PlaybackThread { 959 public: 960 961 DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 962 audio_io_handle_t id, audio_devices_t device, bool systemReady); 963 virtual ~DirectOutputThread(); 964 965 // Thread virtuals 966 967 virtual bool checkForNewParameter_l(const String8& keyValuePair, 968 status_t& status); 969 virtual void flushHw_l(); 970 971 protected: 972 virtual int getTrackName_l(audio_channel_mask_t channelMask, 973 audio_format_t format, audio_session_t sessionId); 974 virtual void deleteTrackName_l(int name); 975 virtual uint32_t activeSleepTimeUs() const; 976 virtual uint32_t idleSleepTimeUs() const; 977 virtual uint32_t suspendSleepTimeUs() const; 978 virtual void cacheParameters_l(); 979 980 // threadLoop snippets 981 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 982 virtual void threadLoop_mix(); 983 virtual void threadLoop_sleepTime(); 984 virtual void threadLoop_exit(); 985 virtual bool shouldStandby_l(); 986 987 virtual void onAddNewTrack_l(); 988 989 // volumes last sent to audio HAL with stream->set_volume() 990 float mLeftVolFloat; 991 float mRightVolFloat; 992 993 DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 994 audio_io_handle_t id, uint32_t device, ThreadBase::type_t type, 995 bool systemReady); 996 void processVolume_l(Track *track, bool lastTrack); 997 998 // prepareTracks_l() tells threadLoop_mix() the name of the single active track 999 sp<Track> mActiveTrack; 1000 1001 wp<Track> mPreviousTrack; // used to detect track switch 1002 1003 public: hasFastMixer()1004 virtual bool hasFastMixer() const { return false; } 1005 }; 1006 1007 class OffloadThread : public DirectOutputThread { 1008 public: 1009 1010 OffloadThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1011 audio_io_handle_t id, uint32_t device, bool systemReady); ~OffloadThread()1012 virtual ~OffloadThread() {}; 1013 virtual void flushHw_l(); 1014 1015 protected: 1016 // threadLoop snippets 1017 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1018 virtual void threadLoop_exit(); 1019 1020 virtual bool waitingAsyncCallback(); 1021 virtual bool waitingAsyncCallback_l(); 1022 virtual void invalidateTracks(audio_stream_type_t streamType); 1023 keepWakeLock()1024 virtual bool keepWakeLock() const { return (mKeepWakeLock || (mDrainSequence & 1)); } 1025 1026 private: 1027 size_t mPausedWriteLength; // length in bytes of write interrupted by pause 1028 size_t mPausedBytesRemaining; // bytes still waiting in mixbuffer after resume 1029 bool mKeepWakeLock; // keep wake lock while waiting for write callback 1030 uint64_t mOffloadUnderrunPosition; // Current frame position for offloaded playback 1031 // used and valid only during underrun. ~0 if 1032 // no underrun has occurred during playback and 1033 // is not reset on standby. 1034 }; 1035 1036 class AsyncCallbackThread : public Thread { 1037 public: 1038 1039 AsyncCallbackThread(const wp<PlaybackThread>& playbackThread); 1040 1041 virtual ~AsyncCallbackThread(); 1042 1043 // Thread virtuals 1044 virtual bool threadLoop(); 1045 1046 // RefBase 1047 virtual void onFirstRef(); 1048 1049 void exit(); 1050 void setWriteBlocked(uint32_t sequence); 1051 void resetWriteBlocked(); 1052 void setDraining(uint32_t sequence); 1053 void resetDraining(); 1054 void setAsyncError(); 1055 1056 private: 1057 const wp<PlaybackThread> mPlaybackThread; 1058 // mWriteAckSequence corresponds to the last write sequence passed by the offload thread via 1059 // setWriteBlocked(). The sequence is shifted one bit to the left and the lsb is used 1060 // to indicate that the callback has been received via resetWriteBlocked() 1061 uint32_t mWriteAckSequence; 1062 // mDrainSequence corresponds to the last drain sequence passed by the offload thread via 1063 // setDraining(). The sequence is shifted one bit to the left and the lsb is used 1064 // to indicate that the callback has been received via resetDraining() 1065 uint32_t mDrainSequence; 1066 Condition mWaitWorkCV; 1067 Mutex mLock; 1068 bool mAsyncError; 1069 }; 1070 1071 class DuplicatingThread : public MixerThread { 1072 public: 1073 DuplicatingThread(const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread, 1074 audio_io_handle_t id, bool systemReady); 1075 virtual ~DuplicatingThread(); 1076 1077 // Thread virtuals 1078 void addOutputTrack(MixerThread* thread); 1079 void removeOutputTrack(MixerThread* thread); waitTimeMs()1080 uint32_t waitTimeMs() const { return mWaitTimeMs; } 1081 protected: 1082 virtual uint32_t activeSleepTimeUs() const; 1083 1084 private: 1085 bool outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks); 1086 protected: 1087 // threadLoop snippets 1088 virtual void threadLoop_mix(); 1089 virtual void threadLoop_sleepTime(); 1090 virtual ssize_t threadLoop_write(); 1091 virtual void threadLoop_standby(); 1092 virtual void cacheParameters_l(); 1093 1094 private: 1095 // called from threadLoop, addOutputTrack, removeOutputTrack 1096 virtual void updateWaitTime_l(); 1097 protected: 1098 virtual void saveOutputTracks(); 1099 virtual void clearOutputTracks(); 1100 private: 1101 1102 uint32_t mWaitTimeMs; 1103 SortedVector < sp<OutputTrack> > outputTracks; 1104 SortedVector < sp<OutputTrack> > mOutputTracks; 1105 public: hasFastMixer()1106 virtual bool hasFastMixer() const { return false; } 1107 }; 1108 1109 1110 // record thread 1111 class RecordThread : public ThreadBase 1112 { 1113 public: 1114 1115 class RecordTrack; 1116 1117 /* The ResamplerBufferProvider is used to retrieve recorded input data from the 1118 * RecordThread. It maintains local state on the relative position of the read 1119 * position of the RecordTrack compared with the RecordThread. 1120 */ 1121 class ResamplerBufferProvider : public AudioBufferProvider 1122 { 1123 public: ResamplerBufferProvider(RecordTrack * recordTrack)1124 ResamplerBufferProvider(RecordTrack* recordTrack) : 1125 mRecordTrack(recordTrack), 1126 mRsmpInUnrel(0), mRsmpInFront(0) { } ~ResamplerBufferProvider()1127 virtual ~ResamplerBufferProvider() { } 1128 1129 // called to set the ResamplerBufferProvider to head of the RecordThread data buffer, 1130 // skipping any previous data read from the hal. 1131 virtual void reset(); 1132 1133 /* Synchronizes RecordTrack position with the RecordThread. 1134 * Calculates available frames and handle overruns if the RecordThread 1135 * has advanced faster than the ResamplerBufferProvider has retrieved data. 1136 * TODO: why not do this for every getNextBuffer? 1137 * 1138 * Parameters 1139 * framesAvailable: pointer to optional output size_t to store record track 1140 * frames available. 1141 * hasOverrun: pointer to optional boolean, returns true if track has overrun. 1142 */ 1143 1144 virtual void sync(size_t *framesAvailable = NULL, bool *hasOverrun = NULL); 1145 1146 // AudioBufferProvider interface 1147 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer); 1148 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 1149 private: 1150 RecordTrack * const mRecordTrack; 1151 size_t mRsmpInUnrel; // unreleased frames remaining from 1152 // most recent getNextBuffer 1153 // for debug only 1154 int32_t mRsmpInFront; // next available frame 1155 // rolling counter that is never cleared 1156 }; 1157 1158 /* The RecordBufferConverter is used for format, channel, and sample rate 1159 * conversion for a RecordTrack. 1160 * 1161 * TODO: Self contained, so move to a separate file later. 1162 * 1163 * RecordBufferConverter uses the convert() method rather than exposing a 1164 * buffer provider interface; this is to save a memory copy. 1165 */ 1166 class RecordBufferConverter 1167 { 1168 public: 1169 RecordBufferConverter( 1170 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 1171 uint32_t srcSampleRate, 1172 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 1173 uint32_t dstSampleRate); 1174 1175 ~RecordBufferConverter(); 1176 1177 /* Converts input data from an AudioBufferProvider by format, channelMask, 1178 * and sampleRate to a destination buffer. 1179 * 1180 * Parameters 1181 * dst: buffer to place the converted data. 1182 * provider: buffer provider to obtain source data. 1183 * frames: number of frames to convert 1184 * 1185 * Returns the number of frames converted. 1186 */ 1187 size_t convert(void *dst, AudioBufferProvider *provider, size_t frames); 1188 1189 // returns NO_ERROR if constructor was successful initCheck()1190 status_t initCheck() const { 1191 // mSrcChannelMask set on successful updateParameters 1192 return mSrcChannelMask != AUDIO_CHANNEL_INVALID ? NO_ERROR : NO_INIT; 1193 } 1194 1195 // allows dynamic reconfigure of all parameters 1196 status_t updateParameters( 1197 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 1198 uint32_t srcSampleRate, 1199 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 1200 uint32_t dstSampleRate); 1201 1202 // called to reset resampler buffers on record track discontinuity reset()1203 void reset() { 1204 if (mResampler != NULL) { 1205 mResampler->reset(); 1206 } 1207 } 1208 1209 private: 1210 // format conversion when not using resampler 1211 void convertNoResampler(void *dst, const void *src, size_t frames); 1212 1213 // format conversion when using resampler; modifies src in-place 1214 void convertResampler(void *dst, /*not-a-const*/ void *src, size_t frames); 1215 1216 // user provided information 1217 audio_channel_mask_t mSrcChannelMask; 1218 audio_format_t mSrcFormat; 1219 uint32_t mSrcSampleRate; 1220 audio_channel_mask_t mDstChannelMask; 1221 audio_format_t mDstFormat; 1222 uint32_t mDstSampleRate; 1223 1224 // derived information 1225 uint32_t mSrcChannelCount; 1226 uint32_t mDstChannelCount; 1227 size_t mDstFrameSize; 1228 1229 // format conversion buffer 1230 void *mBuf; 1231 size_t mBufFrames; 1232 size_t mBufFrameSize; 1233 1234 // resampler info 1235 AudioResampler *mResampler; 1236 1237 bool mIsLegacyDownmix; // legacy stereo to mono conversion needed 1238 bool mIsLegacyUpmix; // legacy mono to stereo conversion needed 1239 bool mRequiresFloat; // data processing requires float (e.g. resampler) 1240 PassthruBufferProvider *mInputConverterProvider; // converts input to float 1241 int8_t mIdxAry[sizeof(uint32_t) * 8]; // used for channel mask conversion 1242 }; 1243 1244 #include "RecordTracks.h" 1245 1246 RecordThread(const sp<AudioFlinger>& audioFlinger, 1247 AudioStreamIn *input, 1248 audio_io_handle_t id, 1249 audio_devices_t outDevice, 1250 audio_devices_t inDevice, 1251 bool systemReady 1252 #ifdef TEE_SINK 1253 , const sp<NBAIO_Sink>& teeSink 1254 #endif 1255 ); 1256 virtual ~RecordThread(); 1257 1258 // no addTrack_l ? 1259 void destroyTrack_l(const sp<RecordTrack>& track); 1260 void removeTrack_l(const sp<RecordTrack>& track); 1261 1262 void dumpInternals(int fd, const Vector<String16>& args); 1263 void dumpTracks(int fd, const Vector<String16>& args); 1264 1265 // Thread virtuals 1266 virtual bool threadLoop(); 1267 1268 // RefBase 1269 virtual void onFirstRef(); 1270 initCheck()1271 virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; } 1272 readOnlyHeap()1273 virtual sp<MemoryDealer> readOnlyHeap() const { return mReadOnlyHeap; } 1274 pipeMemory()1275 virtual sp<IMemory> pipeMemory() const { return mPipeMemory; } 1276 1277 sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l( 1278 const sp<AudioFlinger::Client>& client, 1279 uint32_t sampleRate, 1280 audio_format_t format, 1281 audio_channel_mask_t channelMask, 1282 size_t *pFrameCount, 1283 audio_session_t sessionId, 1284 size_t *notificationFrames, 1285 int uid, 1286 audio_input_flags_t *flags, 1287 pid_t tid, 1288 status_t *status /*non-NULL*/); 1289 1290 status_t start(RecordTrack* recordTrack, 1291 AudioSystem::sync_event_t event, 1292 audio_session_t triggerSession); 1293 1294 // ask the thread to stop the specified track, and 1295 // return true if the caller should then do it's part of the stopping process 1296 bool stop(RecordTrack* recordTrack); 1297 1298 void dump(int fd, const Vector<String16>& args); 1299 AudioStreamIn* clearInput(); 1300 virtual audio_stream_t* stream() const; 1301 1302 1303 virtual bool checkForNewParameter_l(const String8& keyValuePair, 1304 status_t& status); cacheParameters_l()1305 virtual void cacheParameters_l() {} 1306 virtual String8 getParameters(const String8& keys); 1307 virtual void ioConfigChanged(audio_io_config_event event, pid_t pid = 0); 1308 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 1309 audio_patch_handle_t *handle); 1310 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 1311 1312 void addPatchRecord(const sp<PatchRecord>& record); 1313 void deletePatchRecord(const sp<PatchRecord>& record); 1314 1315 void readInputParameters_l(); 1316 virtual uint32_t getInputFramesLost(); 1317 1318 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1319 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1320 virtual uint32_t hasAudioSession_l(audio_session_t sessionId) const; 1321 1322 // Return the set of unique session IDs across all tracks. 1323 // The keys are the session IDs, and the associated values are meaningless. 1324 // FIXME replace by Set [and implement Bag/Multiset for other uses]. 1325 KeyedVector<audio_session_t, bool> sessionIds() const; 1326 1327 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 1328 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 1329 1330 static void syncStartEventCallback(const wp<SyncEvent>& event); 1331 frameCount()1332 virtual size_t frameCount() const { return mFrameCount; } hasFastCapture()1333 bool hasFastCapture() const { return mFastCapture != 0; } 1334 virtual void getAudioPortConfig(struct audio_port_config *config); 1335 1336 virtual status_t checkEffectCompatibility_l(const effect_descriptor_t *desc, 1337 audio_session_t sessionId); 1338 1339 private: 1340 // Enter standby if not already in standby, and set mStandby flag 1341 void standbyIfNotAlreadyInStandby(); 1342 1343 // Call the HAL standby method unconditionally, and don't change mStandby flag 1344 void inputStandBy(); 1345 1346 AudioStreamIn *mInput; 1347 SortedVector < sp<RecordTrack> > mTracks; 1348 // mActiveTracks has dual roles: it indicates the current active track(s), and 1349 // is used together with mStartStopCond to indicate start()/stop() progress 1350 SortedVector< sp<RecordTrack> > mActiveTracks; 1351 // generation counter for mActiveTracks 1352 int mActiveTracksGen; 1353 Condition mStartStopCond; 1354 1355 // resampler converts input at HAL Hz to output at AudioRecord client Hz 1356 void *mRsmpInBuffer; // 1357 size_t mRsmpInFrames; // size of resampler input in frames 1358 size_t mRsmpInFramesP2;// size rounded up to a power-of-2 1359 1360 // rolling index that is never cleared 1361 int32_t mRsmpInRear; // last filled frame + 1 1362 1363 // For dumpsys 1364 const sp<NBAIO_Sink> mTeeSink; 1365 1366 const sp<MemoryDealer> mReadOnlyHeap; 1367 1368 // one-time initialization, no locks required 1369 sp<FastCapture> mFastCapture; // non-0 if there is also 1370 // a fast capture 1371 1372 // FIXME audio watchdog thread 1373 1374 // contents are not guaranteed to be consistent, no locks required 1375 FastCaptureDumpState mFastCaptureDumpState; 1376 #ifdef STATE_QUEUE_DUMP 1377 // FIXME StateQueue observer and mutator dump fields 1378 #endif 1379 // FIXME audio watchdog dump 1380 1381 // accessible only within the threadLoop(), no locks required 1382 // mFastCapture->sq() // for mutating and pushing state 1383 int32_t mFastCaptureFutex; // for cold idle 1384 1385 // The HAL input source is treated as non-blocking, 1386 // but current implementation is blocking 1387 sp<NBAIO_Source> mInputSource; 1388 // The source for the normal capture thread to read from: mInputSource or mPipeSource 1389 sp<NBAIO_Source> mNormalSource; 1390 // If a fast capture is present, the non-blocking pipe sink written to by fast capture, 1391 // otherwise clear 1392 sp<NBAIO_Sink> mPipeSink; 1393 // If a fast capture is present, the non-blocking pipe source read by normal thread, 1394 // otherwise clear 1395 sp<NBAIO_Source> mPipeSource; 1396 // Depth of pipe from fast capture to normal thread and fast clients, always power of 2 1397 size_t mPipeFramesP2; 1398 // If a fast capture is present, the Pipe as IMemory, otherwise clear 1399 sp<IMemory> mPipeMemory; 1400 1401 static const size_t kFastCaptureLogSize = 4 * 1024; 1402 sp<NBLog::Writer> mFastCaptureNBLogWriter; 1403 1404 bool mFastTrackAvail; // true if fast track available 1405 }; 1406