Threads.h revision e8726fea8a53bf3474aa3c6deaf2f6c1f565e694
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 22class ThreadBase : public Thread { 23public: 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: 60 virtual ~ConfigEventData() {} 61 62 virtual void dump(char *buffer, size_t size) = 0; 63 protected: 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: 87 virtual ~ConfigEvent() {} 88 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) : 101 mType(type), mStatus(NO_ERROR), mWaitStatus(false), 102 mRequiresSystemReady(requiresSystemReady), mData(NULL) {} 103 }; 104 105 class IoConfigEventData : public ConfigEventData { 106 public: 107 IoConfigEventData(audio_io_config_event event) : 108 mEvent(event) {} 109 110 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 }; 116 117 class IoConfigEvent : public ConfigEvent { 118 public: 119 IoConfigEvent(audio_io_config_event event) : 120 ConfigEvent(CFG_EVENT_IO) { 121 mData = new IoConfigEventData(event); 122 } 123 virtual ~IoConfigEvent() {} 124 }; 125 126 class PrioConfigEventData : public ConfigEventData { 127 public: 128 PrioConfigEventData(pid_t pid, pid_t tid, int32_t prio) : 129 mPid(pid), mTid(tid), mPrio(prio) {} 130 131 virtual void dump(char *buffer, size_t size) { 132 snprintf(buffer, size, "Prio event: pid %d, tid %d, prio %d\n", mPid, mTid, mPrio); 133 } 134 135 const pid_t mPid; 136 const pid_t mTid; 137 const int32_t mPrio; 138 }; 139 140 class PrioConfigEvent : public ConfigEvent { 141 public: 142 PrioConfigEvent(pid_t pid, pid_t tid, int32_t prio) : 143 ConfigEvent(CFG_EVENT_PRIO, true) { 144 mData = new PrioConfigEventData(pid, tid, prio); 145 } 146 virtual ~PrioConfigEvent() {} 147 }; 148 149 class SetParameterConfigEventData : public ConfigEventData { 150 public: 151 SetParameterConfigEventData(String8 keyValuePairs) : 152 mKeyValuePairs(keyValuePairs) {} 153 154 virtual void dump(char *buffer, size_t size) { 155 snprintf(buffer, size, "KeyValue: %s\n", mKeyValuePairs.string()); 156 } 157 158 const String8 mKeyValuePairs; 159 }; 160 161 class SetParameterConfigEvent : public ConfigEvent { 162 public: 163 SetParameterConfigEvent(String8 keyValuePairs) : 164 ConfigEvent(CFG_EVENT_SET_PARAMETER) { 165 mData = new SetParameterConfigEventData(keyValuePairs); 166 mWaitStatus = true; 167 } 168 virtual ~SetParameterConfigEvent() {} 169 }; 170 171 class CreateAudioPatchConfigEventData : public ConfigEventData { 172 public: 173 CreateAudioPatchConfigEventData(const struct audio_patch patch, 174 audio_patch_handle_t handle) : 175 mPatch(patch), mHandle(handle) {} 176 177 virtual void dump(char *buffer, size_t size) { 178 snprintf(buffer, size, "Patch handle: %u\n", mHandle); 179 } 180 181 const struct audio_patch mPatch; 182 audio_patch_handle_t mHandle; 183 }; 184 185 class CreateAudioPatchConfigEvent : public ConfigEvent { 186 public: 187 CreateAudioPatchConfigEvent(const struct audio_patch patch, 188 audio_patch_handle_t handle) : 189 ConfigEvent(CFG_EVENT_CREATE_AUDIO_PATCH) { 190 mData = new CreateAudioPatchConfigEventData(patch, handle); 191 mWaitStatus = true; 192 } 193 virtual ~CreateAudioPatchConfigEvent() {} 194 }; 195 196 class ReleaseAudioPatchConfigEventData : public ConfigEventData { 197 public: 198 ReleaseAudioPatchConfigEventData(const audio_patch_handle_t handle) : 199 mHandle(handle) {} 200 201 virtual void dump(char *buffer, size_t size) { 202 snprintf(buffer, size, "Patch handle: %u\n", mHandle); 203 } 204 205 audio_patch_handle_t mHandle; 206 }; 207 208 class ReleaseAudioPatchConfigEvent : public ConfigEvent { 209 public: 210 ReleaseAudioPatchConfigEvent(const audio_patch_handle_t handle) : 211 ConfigEvent(CFG_EVENT_RELEASE_AUDIO_PATCH) { 212 mData = new ReleaseAudioPatchConfigEventData(handle); 213 mWaitStatus = true; 214 } 215 virtual ~ReleaseAudioPatchConfigEvent() {} 216 }; 217 218 class PMDeathRecipient : public IBinder::DeathRecipient { 219 public: 220 PMDeathRecipient(const wp<ThreadBase>& thread) : mThread(thread) {} 221 virtual ~PMDeathRecipient() {} 222 223 // IBinder::DeathRecipient 224 virtual void binderDied(const wp<IBinder>& who); 225 226 private: 227 PMDeathRecipient(const PMDeathRecipient&); 228 PMDeathRecipient& operator = (const PMDeathRecipient&); 229 230 wp<ThreadBase> mThread; 231 }; 232 233 virtual status_t initCheck() const = 0; 234 235 // static externally-visible 236 type_t type() const { return mType; } 237 bool isDuplicating() const { return (mType == DUPLICATING); } 238 239 audio_io_handle_t id() const { return mId;} 240 241 // dynamic externally-visible 242 uint32_t sampleRate() const { return mSampleRate; } 243 audio_channel_mask_t channelMask() const { return mChannelMask; } 244 audio_format_t format() const { return mHALFormat; } 245 uint32_t channelCount() const { return mChannelCount; } 246 // Called by AudioFlinger::frameCount(audio_io_handle_t output) and effects, 247 // and returns the [normal mix] buffer's frame count. 248 virtual size_t frameCount() const = 0; 249 size_t frameSize() const { return mFrameSize; } 250 251 // Should be "virtual status_t requestExitAndWait()" and override same 252 // method in Thread, but Thread::requestExitAndWait() is not yet virtual. 253 void exit(); 254 virtual bool checkForNewParameter_l(const String8& keyValuePair, 255 status_t& status) = 0; 256 virtual status_t setParameters(const String8& keyValuePairs); 257 virtual String8 getParameters(const String8& keys) = 0; 258 virtual void ioConfigChanged(audio_io_config_event event) = 0; 259 // sendConfigEvent_l() must be called with ThreadBase::mLock held 260 // Can temporarily release the lock if waiting for a reply from 261 // processConfigEvents_l(). 262 status_t sendConfigEvent_l(sp<ConfigEvent>& event); 263 void sendIoConfigEvent(audio_io_config_event event); 264 void sendIoConfigEvent_l(audio_io_config_event event); 265 void sendPrioConfigEvent(pid_t pid, pid_t tid, int32_t prio); 266 void sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio); 267 status_t sendSetParameterConfigEvent_l(const String8& keyValuePair); 268 status_t sendCreateAudioPatchConfigEvent(const struct audio_patch *patch, 269 audio_patch_handle_t *handle); 270 status_t sendReleaseAudioPatchConfigEvent(audio_patch_handle_t handle); 271 void processConfigEvents_l(); 272 virtual void cacheParameters_l() = 0; 273 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 274 audio_patch_handle_t *handle) = 0; 275 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle) = 0; 276 virtual void getAudioPortConfig(struct audio_port_config *config) = 0; 277 278 279 // see note at declaration of mStandby, mOutDevice and mInDevice 280 bool standby() const { return mStandby; } 281 audio_devices_t outDevice() const { return mOutDevice; } 282 audio_devices_t inDevice() const { return mInDevice; } 283 284 virtual audio_stream_t* stream() const = 0; 285 286 sp<EffectHandle> createEffect_l( 287 const sp<AudioFlinger::Client>& client, 288 const sp<IEffectClient>& effectClient, 289 int32_t priority, 290 int sessionId, 291 effect_descriptor_t *desc, 292 int *enabled, 293 status_t *status /*non-NULL*/); 294 295 // return values for hasAudioSession (bit field) 296 enum effect_state { 297 EFFECT_SESSION = 0x1, // the audio session corresponds to at least one 298 // effect 299 TRACK_SESSION = 0x2 // the audio session corresponds to at least one 300 // track 301 }; 302 303 // get effect chain corresponding to session Id. 304 sp<EffectChain> getEffectChain(int sessionId); 305 // same as getEffectChain() but must be called with ThreadBase mutex locked 306 sp<EffectChain> getEffectChain_l(int sessionId) const; 307 // add an effect chain to the chain list (mEffectChains) 308 virtual status_t addEffectChain_l(const sp<EffectChain>& chain) = 0; 309 // remove an effect chain from the chain list (mEffectChains) 310 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain) = 0; 311 // lock all effect chains Mutexes. Must be called before releasing the 312 // ThreadBase mutex before processing the mixer and effects. This guarantees the 313 // integrity of the chains during the process. 314 // Also sets the parameter 'effectChains' to current value of mEffectChains. 315 void lockEffectChains_l(Vector< sp<EffectChain> >& effectChains); 316 // unlock effect chains after process 317 void unlockEffectChains(const Vector< sp<EffectChain> >& effectChains); 318 // get a copy of mEffectChains vector 319 Vector< sp<EffectChain> > getEffectChains_l() const { return mEffectChains; }; 320 // set audio mode to all effect chains 321 void setMode(audio_mode_t mode); 322 // get effect module with corresponding ID on specified audio session 323 sp<AudioFlinger::EffectModule> getEffect(int sessionId, int effectId); 324 sp<AudioFlinger::EffectModule> getEffect_l(int sessionId, int effectId); 325 // add and effect module. Also creates the effect chain is none exists for 326 // the effects audio session 327 status_t addEffect_l(const sp< EffectModule>& effect); 328 // remove and effect module. Also removes the effect chain is this was the last 329 // effect 330 void removeEffect_l(const sp< EffectModule>& effect); 331 // detach all tracks connected to an auxiliary effect 332 virtual void detachAuxEffect_l(int effectId __unused) {} 333 // returns either EFFECT_SESSION if effects on this audio session exist in one 334 // chain, or TRACK_SESSION if tracks on this audio session exist, or both 335 virtual uint32_t hasAudioSession(int sessionId) const = 0; 336 // the value returned by default implementation is not important as the 337 // strategy is only meaningful for PlaybackThread which implements this method 338 virtual uint32_t getStrategyForSession_l(int sessionId __unused) { return 0; } 339 340 // suspend or restore effect according to the type of effect passed. a NULL 341 // type pointer means suspend all effects in the session 342 void setEffectSuspended(const effect_uuid_t *type, 343 bool suspend, 344 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 345 // check if some effects must be suspended/restored when an effect is enabled 346 // or disabled 347 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 348 bool enabled, 349 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 350 void checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 351 bool enabled, 352 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 353 354 virtual status_t setSyncEvent(const sp<SyncEvent>& event) = 0; 355 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const = 0; 356 357 // Return a reference to a per-thread heap which can be used to allocate IMemory 358 // objects that will be read-only to client processes, read/write to mediaserver, 359 // and shared by all client processes of the thread. 360 // The heap is per-thread rather than common across all threads, because 361 // clients can't be trusted not to modify the offset of the IMemory they receive. 362 // If a thread does not have such a heap, this method returns 0. 363 virtual sp<MemoryDealer> readOnlyHeap() const { return 0; } 364 365 virtual sp<IMemory> pipeMemory() const { return 0; } 366 367 void systemReady(); 368 369 mutable Mutex mLock; 370 371protected: 372 373 // entry describing an effect being suspended in mSuspendedSessions keyed vector 374 class SuspendedSessionDesc : public RefBase { 375 public: 376 SuspendedSessionDesc() : mRefCount(0) {} 377 378 int mRefCount; // number of active suspend requests 379 effect_uuid_t mType; // effect type UUID 380 }; 381 382 void acquireWakeLock(int uid = -1); 383 void acquireWakeLock_l(int uid = -1); 384 void releaseWakeLock(); 385 void releaseWakeLock_l(); 386 void updateWakeLockUids(const SortedVector<int> &uids); 387 void updateWakeLockUids_l(const SortedVector<int> &uids); 388 void getPowerManager_l(); 389 void setEffectSuspended_l(const effect_uuid_t *type, 390 bool suspend, 391 int sessionId); 392 // updated mSuspendedSessions when an effect suspended or restored 393 void updateSuspendedSessions_l(const effect_uuid_t *type, 394 bool suspend, 395 int sessionId); 396 // check if some effects must be suspended when an effect chain is added 397 void checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain); 398 399 String16 getWakeLockTag(); 400 401 virtual void preExit() { } 402 403 friend class AudioFlinger; // for mEffectChains 404 405 const type_t mType; 406 407 // Used by parameters, config events, addTrack_l, exit 408 Condition mWaitWorkCV; 409 410 const sp<AudioFlinger> mAudioFlinger; 411 412 // updated by PlaybackThread::readOutputParameters_l() or 413 // RecordThread::readInputParameters_l() 414 uint32_t mSampleRate; 415 size_t mFrameCount; // output HAL, direct output, record 416 audio_channel_mask_t mChannelMask; 417 uint32_t mChannelCount; 418 size_t mFrameSize; 419 // not HAL frame size, this is for output sink (to pipe to fast mixer) 420 audio_format_t mFormat; // Source format for Recording and 421 // Sink format for Playback. 422 // Sink format may be different than 423 // HAL format if Fastmixer is used. 424 audio_format_t mHALFormat; 425 size_t mBufferSize; // HAL buffer size for read() or write() 426 427 Vector< sp<ConfigEvent> > mConfigEvents; 428 Vector< sp<ConfigEvent> > mPendingConfigEvents; // events awaiting system ready 429 430 // These fields are written and read by thread itself without lock or barrier, 431 // and read by other threads without lock or barrier via standby(), outDevice() 432 // and inDevice(). 433 // Because of the absence of a lock or barrier, any other thread that reads 434 // these fields must use the information in isolation, or be prepared to deal 435 // with possibility that it might be inconsistent with other information. 436 bool mStandby; // Whether thread is currently in standby. 437 audio_devices_t mOutDevice; // output device 438 audio_devices_t mInDevice; // input device 439 audio_devices_t mPrevInDevice; // previous input device 440 struct audio_patch mPatch; 441 audio_source_t mAudioSource; 442 443 const audio_io_handle_t mId; 444 Vector< sp<EffectChain> > mEffectChains; 445 446 static const int kThreadNameLength = 16; // prctl(PR_SET_NAME) limit 447 char mThreadName[kThreadNameLength]; // guaranteed NUL-terminated 448 sp<IPowerManager> mPowerManager; 449 sp<IBinder> mWakeLockToken; 450 const sp<PMDeathRecipient> mDeathRecipient; 451 // list of suspended effects per session and per type. The first vector is 452 // keyed by session ID, the second by type UUID timeLow field 453 KeyedVector< int, KeyedVector< int, sp<SuspendedSessionDesc> > > 454 mSuspendedSessions; 455 static const size_t kLogSize = 4 * 1024; 456 sp<NBLog::Writer> mNBLogWriter; 457 bool mSystemReady; 458}; 459 460// --- PlaybackThread --- 461class PlaybackThread : public ThreadBase { 462public: 463 464#include "PlaybackTracks.h" 465 466 enum mixer_state { 467 MIXER_IDLE, // no active tracks 468 MIXER_TRACKS_ENABLED, // at least one active track, but no track has any data ready 469 MIXER_TRACKS_READY, // at least one active track, and at least one track has data 470 MIXER_DRAIN_TRACK, // drain currently playing track 471 MIXER_DRAIN_ALL, // fully drain the hardware 472 // standby mode does not have an enum value 473 // suspend by audio policy manager is orthogonal to mixer state 474 }; 475 476 // retry count before removing active track in case of underrun on offloaded thread: 477 // we need to make sure that AudioTrack client has enough time to send large buffers 478//FIXME may be more appropriate if expressed in time units. Need to revise how underrun is handled 479 // for offloaded tracks 480 static const int8_t kMaxTrackRetriesOffload = 20; 481 482 PlaybackThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 483 audio_io_handle_t id, audio_devices_t device, type_t type, bool systemReady); 484 virtual ~PlaybackThread(); 485 486 void dump(int fd, const Vector<String16>& args); 487 488 // Thread virtuals 489 virtual bool threadLoop(); 490 491 // RefBase 492 virtual void onFirstRef(); 493 494protected: 495 // Code snippets that were lifted up out of threadLoop() 496 virtual void threadLoop_mix() = 0; 497 virtual void threadLoop_sleepTime() = 0; 498 virtual ssize_t threadLoop_write(); 499 virtual void threadLoop_drain(); 500 virtual void threadLoop_standby(); 501 virtual void threadLoop_exit(); 502 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 503 504 // prepareTracks_l reads and writes mActiveTracks, and returns 505 // the pending set of tracks to remove via Vector 'tracksToRemove'. The caller 506 // is responsible for clearing or destroying this Vector later on, when it 507 // is safe to do so. That will drop the final ref count and destroy the tracks. 508 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove) = 0; 509 void removeTracks_l(const Vector< sp<Track> >& tracksToRemove); 510 511 void writeCallback(); 512 void resetWriteBlocked(uint32_t sequence); 513 void drainCallback(); 514 void resetDraining(uint32_t sequence); 515 516 static int asyncCallback(stream_callback_event_t event, void *param, void *cookie); 517 518 virtual bool waitingAsyncCallback(); 519 virtual bool waitingAsyncCallback_l(); 520 virtual bool shouldStandby_l(); 521 virtual void onAddNewTrack_l(); 522 523 // ThreadBase virtuals 524 virtual void preExit(); 525 526public: 527 528 virtual status_t initCheck() const { return (mOutput == NULL) ? NO_INIT : NO_ERROR; } 529 530 // return estimated latency in milliseconds, as reported by HAL 531 uint32_t latency() const; 532 // same, but lock must already be held 533 uint32_t latency_l() const; 534 535 void setMasterVolume(float value); 536 void setMasterMute(bool muted); 537 538 void setStreamVolume(audio_stream_type_t stream, float value); 539 void setStreamMute(audio_stream_type_t stream, bool muted); 540 541 float streamVolume(audio_stream_type_t stream) const; 542 543 sp<Track> createTrack_l( 544 const sp<AudioFlinger::Client>& client, 545 audio_stream_type_t streamType, 546 uint32_t sampleRate, 547 audio_format_t format, 548 audio_channel_mask_t channelMask, 549 size_t *pFrameCount, 550 const sp<IMemory>& sharedBuffer, 551 int sessionId, 552 IAudioFlinger::track_flags_t *flags, 553 pid_t tid, 554 int uid, 555 status_t *status /*non-NULL*/); 556 557 AudioStreamOut* getOutput() const; 558 AudioStreamOut* clearOutput(); 559 virtual audio_stream_t* stream() const; 560 561 // a very large number of suspend() will eventually wraparound, but unlikely 562 void suspend() { (void) android_atomic_inc(&mSuspended); } 563 void restore() 564 { 565 // if restore() is done without suspend(), get back into 566 // range so that the next suspend() will operate correctly 567 if (android_atomic_dec(&mSuspended) <= 0) { 568 android_atomic_release_store(0, &mSuspended); 569 } 570 } 571 bool isSuspended() const 572 { return android_atomic_acquire_load(&mSuspended) > 0; } 573 574 virtual String8 getParameters(const String8& keys); 575 virtual void ioConfigChanged(audio_io_config_event event); 576 status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames); 577 // FIXME rename mixBuffer() to sinkBuffer() and remove int16_t* dependency. 578 // Consider also removing and passing an explicit mMainBuffer initialization 579 // parameter to AF::PlaybackThread::Track::Track(). 580 int16_t *mixBuffer() const { 581 return reinterpret_cast<int16_t *>(mSinkBuffer); }; 582 583 virtual void detachAuxEffect_l(int effectId); 584 status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track> track, 585 int EffectId); 586 status_t attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track> track, 587 int EffectId); 588 589 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 590 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 591 virtual uint32_t hasAudioSession(int sessionId) const; 592 virtual uint32_t getStrategyForSession_l(int sessionId); 593 594 595 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 596 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 597 598 // called with AudioFlinger lock held 599 void invalidateTracks(audio_stream_type_t streamType); 600 601 virtual size_t frameCount() const { return mNormalFrameCount; } 602 603 // Return's the HAL's frame count i.e. fast mixer buffer size. 604 size_t frameCountHAL() const { return mFrameCount; } 605 606 status_t getTimestamp_l(AudioTimestamp& timestamp); 607 608 void addPatchTrack(const sp<PatchTrack>& track); 609 void deletePatchTrack(const sp<PatchTrack>& track); 610 611 virtual void getAudioPortConfig(struct audio_port_config *config); 612 613protected: 614 // updated by readOutputParameters_l() 615 size_t mNormalFrameCount; // normal mixer and effects 616 617 bool mThreadThrottle; // throttle the thread processing 618 uint32_t mThreadThrottleTimeMs; // throttle time for MIXER threads 619 uint32_t mThreadThrottleEndMs; // notify once per throttling 620 uint32_t mHalfBufferMs; // half the buffer size in milliseconds 621 622 void* mSinkBuffer; // frame size aligned sink buffer 623 624 // TODO: 625 // Rearrange the buffer info into a struct/class with 626 // clear, copy, construction, destruction methods. 627 // 628 // mSinkBuffer also has associated with it: 629 // 630 // mSinkBufferSize: Sink Buffer Size 631 // mFormat: Sink Buffer Format 632 633 // Mixer Buffer (mMixerBuffer*) 634 // 635 // In the case of floating point or multichannel data, which is not in the 636 // sink format, it is required to accumulate in a higher precision or greater channel count 637 // buffer before downmixing or data conversion to the sink buffer. 638 639 // Set to "true" to enable the Mixer Buffer otherwise mixer output goes to sink buffer. 640 bool mMixerBufferEnabled; 641 642 // Storage, 32 byte aligned (may make this alignment a requirement later). 643 // Due to constraints on mNormalFrameCount, the buffer size is a multiple of 16 frames. 644 void* mMixerBuffer; 645 646 // Size of mMixerBuffer in bytes: mNormalFrameCount * #channels * sampsize. 647 size_t mMixerBufferSize; 648 649 // The audio format of mMixerBuffer. Set to AUDIO_FORMAT_PCM_(FLOAT|16_BIT) only. 650 audio_format_t mMixerBufferFormat; 651 652 // An internal flag set to true by MixerThread::prepareTracks_l() 653 // when mMixerBuffer contains valid data after mixing. 654 bool mMixerBufferValid; 655 656 // Effects Buffer (mEffectsBuffer*) 657 // 658 // In the case of effects data, which is not in the sink format, 659 // it is required to accumulate in a different buffer before data conversion 660 // to the sink buffer. 661 662 // Set to "true" to enable the Effects Buffer otherwise effects output goes to sink buffer. 663 bool mEffectBufferEnabled; 664 665 // Storage, 32 byte aligned (may make this alignment a requirement later). 666 // Due to constraints on mNormalFrameCount, the buffer size is a multiple of 16 frames. 667 void* mEffectBuffer; 668 669 // Size of mEffectsBuffer in bytes: mNormalFrameCount * #channels * sampsize. 670 size_t mEffectBufferSize; 671 672 // The audio format of mEffectsBuffer. Set to AUDIO_FORMAT_PCM_16_BIT only. 673 audio_format_t mEffectBufferFormat; 674 675 // An internal flag set to true by MixerThread::prepareTracks_l() 676 // when mEffectsBuffer contains valid data after mixing. 677 // 678 // When this is set, all mixer data is routed into the effects buffer 679 // for any processing (including output processing). 680 bool mEffectBufferValid; 681 682 // suspend count, > 0 means suspended. While suspended, the thread continues to pull from 683 // tracks and mix, but doesn't write to HAL. A2DP and SCO HAL implementations can't handle 684 // concurrent use of both of them, so Audio Policy Service suspends one of the threads to 685 // workaround that restriction. 686 // 'volatile' means accessed via atomic operations and no lock. 687 volatile int32_t mSuspended; 688 689 // FIXME overflows every 6+ hours at 44.1 kHz stereo 16-bit samples 690 // mFramesWritten would be better, or 64-bit even better 691 size_t mBytesWritten; 692private: 693 // mMasterMute is in both PlaybackThread and in AudioFlinger. When a 694 // PlaybackThread needs to find out if master-muted, it checks it's local 695 // copy rather than the one in AudioFlinger. This optimization saves a lock. 696 bool mMasterMute; 697 void setMasterMute_l(bool muted) { mMasterMute = muted; } 698protected: 699 SortedVector< wp<Track> > mActiveTracks; // FIXME check if this could be sp<> 700 SortedVector<int> mWakeLockUids; 701 int mActiveTracksGeneration; 702 wp<Track> mLatestActiveTrack; // latest track added to mActiveTracks 703 704 // Allocate a track name for a given channel mask. 705 // Returns name >= 0 if successful, -1 on failure. 706 virtual int getTrackName_l(audio_channel_mask_t channelMask, 707 audio_format_t format, int sessionId) = 0; 708 virtual void deleteTrackName_l(int name) = 0; 709 710 // Time to sleep between cycles when: 711 virtual uint32_t activeSleepTimeUs() const; // mixer state MIXER_TRACKS_ENABLED 712 virtual uint32_t idleSleepTimeUs() const = 0; // mixer state MIXER_IDLE 713 virtual uint32_t suspendSleepTimeUs() const = 0; // audio policy manager suspended us 714 // No sleep when mixer state == MIXER_TRACKS_READY; relies on audio HAL stream->write() 715 // No sleep in standby mode; waits on a condition 716 717 // Code snippets that are temporarily lifted up out of threadLoop() until the merge 718 void checkSilentMode_l(); 719 720 // Non-trivial for DUPLICATING only 721 virtual void saveOutputTracks() { } 722 virtual void clearOutputTracks() { } 723 724 // Cache various calculated values, at threadLoop() entry and after a parameter change 725 virtual void cacheParameters_l(); 726 727 virtual uint32_t correctLatency_l(uint32_t latency) const; 728 729 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 730 audio_patch_handle_t *handle); 731 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 732 733 bool usesHwAvSync() const { return (mType == DIRECT) && (mOutput != NULL) 734 && mHwSupportsPause 735 && (mOutput->flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC); } 736 737private: 738 739 friend class AudioFlinger; // for numerous 740 741 PlaybackThread& operator = (const PlaybackThread&); 742 743 status_t addTrack_l(const sp<Track>& track); 744 bool destroyTrack_l(const sp<Track>& track); 745 void removeTrack_l(const sp<Track>& track); 746 void broadcast_l(); 747 748 void readOutputParameters_l(); 749 750 virtual void dumpInternals(int fd, const Vector<String16>& args); 751 void dumpTracks(int fd, const Vector<String16>& args); 752 753 SortedVector< sp<Track> > mTracks; 754 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 755 AudioStreamOut *mOutput; 756 757 float mMasterVolume; 758 nsecs_t mLastWriteTime; 759 int mNumWrites; 760 int mNumDelayedWrites; 761 bool mInWrite; 762 763 // FIXME rename these former local variables of threadLoop to standard "m" names 764 nsecs_t mStandbyTimeNs; 765 size_t mSinkBufferSize; 766 767 // cached copies of activeSleepTimeUs() and idleSleepTimeUs() made by cacheParameters_l() 768 uint32_t mActiveSleepTimeUs; 769 uint32_t mIdleSleepTimeUs; 770 771 uint32_t mSleepTimeUs; 772 773 // mixer status returned by prepareTracks_l() 774 mixer_state mMixerStatus; // current cycle 775 // previous cycle when in prepareTracks_l() 776 mixer_state mMixerStatusIgnoringFastTracks; 777 // FIXME or a separate ready state per track 778 779 // FIXME move these declarations into the specific sub-class that needs them 780 // MIXER only 781 uint32_t sleepTimeShift; 782 783 // same as AudioFlinger::mStandbyTimeInNsecs except for DIRECT which uses a shorter value 784 nsecs_t mStandbyDelayNs; 785 786 // MIXER only 787 nsecs_t maxPeriod; 788 789 // DUPLICATING only 790 uint32_t writeFrames; 791 792 size_t mBytesRemaining; 793 size_t mCurrentWriteLength; 794 bool mUseAsyncWrite; 795 // mWriteAckSequence contains current write sequence on bits 31-1. The write sequence is 796 // incremented each time a write(), a flush() or a standby() occurs. 797 // Bit 0 is set when a write blocks and indicates a callback is expected. 798 // Bit 0 is reset by the async callback thread calling resetWriteBlocked(). Out of sequence 799 // callbacks are ignored. 800 uint32_t mWriteAckSequence; 801 // mDrainSequence contains current drain sequence on bits 31-1. The drain sequence is 802 // incremented each time a drain is requested or a flush() or standby() occurs. 803 // Bit 0 is set when the drain() command is called at the HAL and indicates a callback is 804 // expected. 805 // Bit 0 is reset by the async callback thread calling resetDraining(). Out of sequence 806 // callbacks are ignored. 807 uint32_t mDrainSequence; 808 // A condition that must be evaluated by prepareTrack_l() has changed and we must not wait 809 // for async write callback in the thread loop before evaluating it 810 bool mSignalPending; 811 sp<AsyncCallbackThread> mCallbackThread; 812 813private: 814 // The HAL output sink is treated as non-blocking, but current implementation is blocking 815 sp<NBAIO_Sink> mOutputSink; 816 // If a fast mixer is present, the blocking pipe sink, otherwise clear 817 sp<NBAIO_Sink> mPipeSink; 818 // The current sink for the normal mixer to write it's (sub)mix, mOutputSink or mPipeSink 819 sp<NBAIO_Sink> mNormalSink; 820#ifdef TEE_SINK 821 // For dumpsys 822 sp<NBAIO_Sink> mTeeSink; 823 sp<NBAIO_Source> mTeeSource; 824#endif 825 uint32_t mScreenState; // cached copy of gScreenState 826 static const size_t kFastMixerLogSize = 4 * 1024; 827 sp<NBLog::Writer> mFastMixerNBLogWriter; 828public: 829 virtual bool hasFastMixer() const = 0; 830 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex __unused) const 831 { FastTrackUnderruns dummy; return dummy; } 832 833protected: 834 // accessed by both binder threads and within threadLoop(), lock on mutex needed 835 unsigned mFastTrackAvailMask; // bit i set if fast track [i] is available 836 bool mHwSupportsPause; 837 bool mHwPaused; 838 bool mFlushPending; 839private: 840 // timestamp latch: 841 // D input is written by threadLoop_write while mutex is unlocked, and read while locked 842 // Q output is written while locked, and read while locked 843 struct { 844 AudioTimestamp mTimestamp; 845 uint32_t mUnpresentedFrames; 846 KeyedVector<Track *, uint32_t> mFramesReleased; 847 } mLatchD, mLatchQ; 848 bool mLatchDValid; // true means mLatchD is valid 849 // (except for mFramesReleased which is filled in later), 850 // and clock it into latch at next opportunity 851 bool mLatchQValid; // true means mLatchQ is valid 852}; 853 854class MixerThread : public PlaybackThread { 855public: 856 MixerThread(const sp<AudioFlinger>& audioFlinger, 857 AudioStreamOut* output, 858 audio_io_handle_t id, 859 audio_devices_t device, 860 bool systemReady, 861 type_t type = MIXER); 862 virtual ~MixerThread(); 863 864 // Thread virtuals 865 866 virtual bool checkForNewParameter_l(const String8& keyValuePair, 867 status_t& status); 868 virtual void dumpInternals(int fd, const Vector<String16>& args); 869 870protected: 871 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 872 virtual int getTrackName_l(audio_channel_mask_t channelMask, 873 audio_format_t format, int sessionId); 874 virtual void deleteTrackName_l(int name); 875 virtual uint32_t idleSleepTimeUs() const; 876 virtual uint32_t suspendSleepTimeUs() const; 877 virtual void cacheParameters_l(); 878 879 // threadLoop snippets 880 virtual ssize_t threadLoop_write(); 881 virtual void threadLoop_standby(); 882 virtual void threadLoop_mix(); 883 virtual void threadLoop_sleepTime(); 884 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 885 virtual uint32_t correctLatency_l(uint32_t latency) const; 886 887 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 888 audio_patch_handle_t *handle); 889 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 890 891 AudioMixer* mAudioMixer; // normal mixer 892private: 893 // one-time initialization, no locks required 894 sp<FastMixer> mFastMixer; // non-0 if there is also a fast mixer 895 sp<AudioWatchdog> mAudioWatchdog; // non-0 if there is an audio watchdog thread 896 897 // contents are not guaranteed to be consistent, no locks required 898 FastMixerDumpState mFastMixerDumpState; 899#ifdef STATE_QUEUE_DUMP 900 StateQueueObserverDump mStateQueueObserverDump; 901 StateQueueMutatorDump mStateQueueMutatorDump; 902#endif 903 AudioWatchdogDump mAudioWatchdogDump; 904 905 // accessible only within the threadLoop(), no locks required 906 // mFastMixer->sq() // for mutating and pushing state 907 int32_t mFastMixerFutex; // for cold idle 908 909public: 910 virtual bool hasFastMixer() const { return mFastMixer != 0; } 911 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const { 912 ALOG_ASSERT(fastIndex < FastMixerState::kMaxFastTracks); 913 return mFastMixerDumpState.mTracks[fastIndex].mUnderruns; 914 } 915 916}; 917 918class DirectOutputThread : public PlaybackThread { 919public: 920 921 DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 922 audio_io_handle_t id, audio_devices_t device, bool systemReady); 923 virtual ~DirectOutputThread(); 924 925 // Thread virtuals 926 927 virtual bool checkForNewParameter_l(const String8& keyValuePair, 928 status_t& status); 929 virtual void flushHw_l(); 930 931protected: 932 virtual int getTrackName_l(audio_channel_mask_t channelMask, 933 audio_format_t format, int sessionId); 934 virtual void deleteTrackName_l(int name); 935 virtual uint32_t activeSleepTimeUs() const; 936 virtual uint32_t idleSleepTimeUs() const; 937 virtual uint32_t suspendSleepTimeUs() const; 938 virtual void cacheParameters_l(); 939 940 // threadLoop snippets 941 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 942 virtual void threadLoop_mix(); 943 virtual void threadLoop_sleepTime(); 944 virtual void threadLoop_exit(); 945 virtual bool shouldStandby_l(); 946 947 virtual void onAddNewTrack_l(); 948 949 // volumes last sent to audio HAL with stream->set_volume() 950 float mLeftVolFloat; 951 float mRightVolFloat; 952 953 DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 954 audio_io_handle_t id, uint32_t device, ThreadBase::type_t type, 955 bool systemReady); 956 void processVolume_l(Track *track, bool lastTrack); 957 958 // prepareTracks_l() tells threadLoop_mix() the name of the single active track 959 sp<Track> mActiveTrack; 960 961 wp<Track> mPreviousTrack; // used to detect track switch 962 963public: 964 virtual bool hasFastMixer() const { return false; } 965}; 966 967class OffloadThread : public DirectOutputThread { 968public: 969 970 OffloadThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 971 audio_io_handle_t id, uint32_t device, bool systemReady); 972 virtual ~OffloadThread() {}; 973 virtual void flushHw_l(); 974 975protected: 976 // threadLoop snippets 977 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 978 virtual void threadLoop_exit(); 979 980 virtual bool waitingAsyncCallback(); 981 virtual bool waitingAsyncCallback_l(); 982 983private: 984 size_t mPausedWriteLength; // length in bytes of write interrupted by pause 985 size_t mPausedBytesRemaining; // bytes still waiting in mixbuffer after resume 986}; 987 988class AsyncCallbackThread : public Thread { 989public: 990 991 AsyncCallbackThread(const wp<PlaybackThread>& playbackThread); 992 993 virtual ~AsyncCallbackThread(); 994 995 // Thread virtuals 996 virtual bool threadLoop(); 997 998 // RefBase 999 virtual void onFirstRef(); 1000 1001 void exit(); 1002 void setWriteBlocked(uint32_t sequence); 1003 void resetWriteBlocked(); 1004 void setDraining(uint32_t sequence); 1005 void resetDraining(); 1006 1007private: 1008 const wp<PlaybackThread> mPlaybackThread; 1009 // mWriteAckSequence corresponds to the last write sequence passed by the offload thread via 1010 // setWriteBlocked(). The sequence is shifted one bit to the left and the lsb is used 1011 // to indicate that the callback has been received via resetWriteBlocked() 1012 uint32_t mWriteAckSequence; 1013 // mDrainSequence corresponds to the last drain sequence passed by the offload thread via 1014 // setDraining(). The sequence is shifted one bit to the left and the lsb is used 1015 // to indicate that the callback has been received via resetDraining() 1016 uint32_t mDrainSequence; 1017 Condition mWaitWorkCV; 1018 Mutex mLock; 1019}; 1020 1021class DuplicatingThread : public MixerThread { 1022public: 1023 DuplicatingThread(const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread, 1024 audio_io_handle_t id, bool systemReady); 1025 virtual ~DuplicatingThread(); 1026 1027 // Thread virtuals 1028 void addOutputTrack(MixerThread* thread); 1029 void removeOutputTrack(MixerThread* thread); 1030 uint32_t waitTimeMs() const { return mWaitTimeMs; } 1031protected: 1032 virtual uint32_t activeSleepTimeUs() const; 1033 1034private: 1035 bool outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks); 1036protected: 1037 // threadLoop snippets 1038 virtual void threadLoop_mix(); 1039 virtual void threadLoop_sleepTime(); 1040 virtual ssize_t threadLoop_write(); 1041 virtual void threadLoop_standby(); 1042 virtual void cacheParameters_l(); 1043 1044private: 1045 // called from threadLoop, addOutputTrack, removeOutputTrack 1046 virtual void updateWaitTime_l(); 1047protected: 1048 virtual void saveOutputTracks(); 1049 virtual void clearOutputTracks(); 1050private: 1051 1052 uint32_t mWaitTimeMs; 1053 SortedVector < sp<OutputTrack> > outputTracks; 1054 SortedVector < sp<OutputTrack> > mOutputTracks; 1055public: 1056 virtual bool hasFastMixer() const { return false; } 1057}; 1058 1059 1060// record thread 1061class RecordThread : public ThreadBase 1062{ 1063public: 1064 1065 class RecordTrack; 1066 1067 /* The ResamplerBufferProvider is used to retrieve recorded input data from the 1068 * RecordThread. It maintains local state on the relative position of the read 1069 * position of the RecordTrack compared with the RecordThread. 1070 */ 1071 class ResamplerBufferProvider : public AudioBufferProvider 1072 { 1073 public: 1074 ResamplerBufferProvider(RecordTrack* recordTrack) : 1075 mRecordTrack(recordTrack), 1076 mRsmpInUnrel(0), mRsmpInFront(0) { } 1077 virtual ~ResamplerBufferProvider() { } 1078 1079 // called to set the ResamplerBufferProvider to head of the RecordThread data buffer, 1080 // skipping any previous data read from the hal. 1081 virtual void reset(); 1082 1083 /* Synchronizes RecordTrack position with the RecordThread. 1084 * Calculates available frames and handle overruns if the RecordThread 1085 * has advanced faster than the ResamplerBufferProvider has retrieved data. 1086 * TODO: why not do this for every getNextBuffer? 1087 * 1088 * Parameters 1089 * framesAvailable: pointer to optional output size_t to store record track 1090 * frames available. 1091 * hasOverrun: pointer to optional boolean, returns true if track has overrun. 1092 */ 1093 1094 virtual void sync(size_t *framesAvailable = NULL, bool *hasOverrun = NULL); 1095 1096 // AudioBufferProvider interface 1097 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts); 1098 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 1099 private: 1100 RecordTrack * const mRecordTrack; 1101 size_t mRsmpInUnrel; // unreleased frames remaining from 1102 // most recent getNextBuffer 1103 // for debug only 1104 int32_t mRsmpInFront; // next available frame 1105 // rolling counter that is never cleared 1106 }; 1107 1108 /* The RecordBufferConverter is used for format, channel, and sample rate 1109 * conversion for a RecordTrack. 1110 * 1111 * TODO: Self contained, so move to a separate file later. 1112 * 1113 * RecordBufferConverter uses the convert() method rather than exposing a 1114 * buffer provider interface; this is to save a memory copy. 1115 */ 1116 class RecordBufferConverter 1117 { 1118 public: 1119 RecordBufferConverter( 1120 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 1121 uint32_t srcSampleRate, 1122 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 1123 uint32_t dstSampleRate); 1124 1125 ~RecordBufferConverter(); 1126 1127 /* Converts input data from an AudioBufferProvider by format, channelMask, 1128 * and sampleRate to a destination buffer. 1129 * 1130 * Parameters 1131 * dst: buffer to place the converted data. 1132 * provider: buffer provider to obtain source data. 1133 * frames: number of frames to convert 1134 * 1135 * Returns the number of frames converted. 1136 */ 1137 size_t convert(void *dst, AudioBufferProvider *provider, size_t frames); 1138 1139 // returns NO_ERROR if constructor was successful 1140 status_t initCheck() const { 1141 // mSrcChannelMask set on successful updateParameters 1142 return mSrcChannelMask != AUDIO_CHANNEL_INVALID ? NO_ERROR : NO_INIT; 1143 } 1144 1145 // allows dynamic reconfigure of all parameters 1146 status_t updateParameters( 1147 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 1148 uint32_t srcSampleRate, 1149 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 1150 uint32_t dstSampleRate); 1151 1152 // called to reset resampler buffers on record track discontinuity 1153 void reset() { 1154 if (mResampler != NULL) { 1155 mResampler->reset(); 1156 } 1157 } 1158 1159 private: 1160 // format conversion when not using resampler 1161 void convertNoResampler(void *dst, const void *src, size_t frames); 1162 1163 // format conversion when using resampler; modifies src in-place 1164 void convertResampler(void *dst, /*not-a-const*/ void *src, size_t frames); 1165 1166 // user provided information 1167 audio_channel_mask_t mSrcChannelMask; 1168 audio_format_t mSrcFormat; 1169 uint32_t mSrcSampleRate; 1170 audio_channel_mask_t mDstChannelMask; 1171 audio_format_t mDstFormat; 1172 uint32_t mDstSampleRate; 1173 1174 // derived information 1175 uint32_t mSrcChannelCount; 1176 uint32_t mDstChannelCount; 1177 size_t mDstFrameSize; 1178 1179 // format conversion buffer 1180 void *mBuf; 1181 size_t mBufFrames; 1182 size_t mBufFrameSize; 1183 1184 // resampler info 1185 AudioResampler *mResampler; 1186 1187 bool mIsLegacyDownmix; // legacy stereo to mono conversion needed 1188 bool mIsLegacyUpmix; // legacy mono to stereo conversion needed 1189 bool mRequiresFloat; // data processing requires float (e.g. resampler) 1190 PassthruBufferProvider *mInputConverterProvider; // converts input to float 1191 int8_t mIdxAry[sizeof(uint32_t) * 8]; // used for channel mask conversion 1192 }; 1193 1194#include "RecordTracks.h" 1195 1196 RecordThread(const sp<AudioFlinger>& audioFlinger, 1197 AudioStreamIn *input, 1198 audio_io_handle_t id, 1199 audio_devices_t outDevice, 1200 audio_devices_t inDevice, 1201 bool systemReady 1202#ifdef TEE_SINK 1203 , const sp<NBAIO_Sink>& teeSink 1204#endif 1205 ); 1206 virtual ~RecordThread(); 1207 1208 // no addTrack_l ? 1209 void destroyTrack_l(const sp<RecordTrack>& track); 1210 void removeTrack_l(const sp<RecordTrack>& track); 1211 1212 void dumpInternals(int fd, const Vector<String16>& args); 1213 void dumpTracks(int fd, const Vector<String16>& args); 1214 1215 // Thread virtuals 1216 virtual bool threadLoop(); 1217 1218 // RefBase 1219 virtual void onFirstRef(); 1220 1221 virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; } 1222 1223 virtual sp<MemoryDealer> readOnlyHeap() const { return mReadOnlyHeap; } 1224 1225 virtual sp<IMemory> pipeMemory() const { return mPipeMemory; } 1226 1227 sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l( 1228 const sp<AudioFlinger::Client>& client, 1229 uint32_t sampleRate, 1230 audio_format_t format, 1231 audio_channel_mask_t channelMask, 1232 size_t *pFrameCount, 1233 int sessionId, 1234 size_t *notificationFrames, 1235 int uid, 1236 IAudioFlinger::track_flags_t *flags, 1237 pid_t tid, 1238 status_t *status /*non-NULL*/); 1239 1240 status_t start(RecordTrack* recordTrack, 1241 AudioSystem::sync_event_t event, 1242 int triggerSession); 1243 1244 // ask the thread to stop the specified track, and 1245 // return true if the caller should then do it's part of the stopping process 1246 bool stop(RecordTrack* recordTrack); 1247 1248 void dump(int fd, const Vector<String16>& args); 1249 AudioStreamIn* clearInput(); 1250 virtual audio_stream_t* stream() const; 1251 1252 1253 virtual bool checkForNewParameter_l(const String8& keyValuePair, 1254 status_t& status); 1255 virtual void cacheParameters_l() {} 1256 virtual String8 getParameters(const String8& keys); 1257 virtual void ioConfigChanged(audio_io_config_event event); 1258 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 1259 audio_patch_handle_t *handle); 1260 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 1261 1262 void addPatchRecord(const sp<PatchRecord>& record); 1263 void deletePatchRecord(const sp<PatchRecord>& record); 1264 1265 void readInputParameters_l(); 1266 virtual uint32_t getInputFramesLost(); 1267 1268 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1269 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1270 virtual uint32_t hasAudioSession(int sessionId) const; 1271 1272 // Return the set of unique session IDs across all tracks. 1273 // The keys are the session IDs, and the associated values are meaningless. 1274 // FIXME replace by Set [and implement Bag/Multiset for other uses]. 1275 KeyedVector<int, bool> sessionIds() const; 1276 1277 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 1278 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 1279 1280 static void syncStartEventCallback(const wp<SyncEvent>& event); 1281 1282 virtual size_t frameCount() const { return mFrameCount; } 1283 bool hasFastCapture() const { return mFastCapture != 0; } 1284 virtual void getAudioPortConfig(struct audio_port_config *config); 1285 1286private: 1287 // Enter standby if not already in standby, and set mStandby flag 1288 void standbyIfNotAlreadyInStandby(); 1289 1290 // Call the HAL standby method unconditionally, and don't change mStandby flag 1291 void inputStandBy(); 1292 1293 AudioStreamIn *mInput; 1294 SortedVector < sp<RecordTrack> > mTracks; 1295 // mActiveTracks has dual roles: it indicates the current active track(s), and 1296 // is used together with mStartStopCond to indicate start()/stop() progress 1297 SortedVector< sp<RecordTrack> > mActiveTracks; 1298 // generation counter for mActiveTracks 1299 int mActiveTracksGen; 1300 Condition mStartStopCond; 1301 1302 // resampler converts input at HAL Hz to output at AudioRecord client Hz 1303 void *mRsmpInBuffer; // 1304 size_t mRsmpInFrames; // size of resampler input in frames 1305 size_t mRsmpInFramesP2;// size rounded up to a power-of-2 1306 1307 // rolling index that is never cleared 1308 int32_t mRsmpInRear; // last filled frame + 1 1309 1310 // For dumpsys 1311 const sp<NBAIO_Sink> mTeeSink; 1312 1313 const sp<MemoryDealer> mReadOnlyHeap; 1314 1315 // one-time initialization, no locks required 1316 sp<FastCapture> mFastCapture; // non-0 if there is also 1317 // a fast capture 1318 1319 // FIXME audio watchdog thread 1320 1321 // contents are not guaranteed to be consistent, no locks required 1322 FastCaptureDumpState mFastCaptureDumpState; 1323#ifdef STATE_QUEUE_DUMP 1324 // FIXME StateQueue observer and mutator dump fields 1325#endif 1326 // FIXME audio watchdog dump 1327 1328 // accessible only within the threadLoop(), no locks required 1329 // mFastCapture->sq() // for mutating and pushing state 1330 int32_t mFastCaptureFutex; // for cold idle 1331 1332 // The HAL input source is treated as non-blocking, 1333 // but current implementation is blocking 1334 sp<NBAIO_Source> mInputSource; 1335 // The source for the normal capture thread to read from: mInputSource or mPipeSource 1336 sp<NBAIO_Source> mNormalSource; 1337 // If a fast capture is present, the non-blocking pipe sink written to by fast capture, 1338 // otherwise clear 1339 sp<NBAIO_Sink> mPipeSink; 1340 // If a fast capture is present, the non-blocking pipe source read by normal thread, 1341 // otherwise clear 1342 sp<NBAIO_Source> mPipeSource; 1343 // Depth of pipe from fast capture to normal thread and fast clients, always power of 2 1344 size_t mPipeFramesP2; 1345 // If a fast capture is present, the Pipe as IMemory, otherwise clear 1346 sp<IMemory> mPipeMemory; 1347 1348 static const size_t kFastCaptureLogSize = 4 * 1024; 1349 sp<NBLog::Writer> mFastCaptureNBLogWriter; 1350 1351 bool mFastTrackAvail; // true if fast track available 1352}; 1353