Threads.h revision 2f366df67c31119bb6dd726becd32d14b18e6573
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 explicit 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, pid_t pid) : 108 mEvent(event), mPid(pid) {} 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 const pid_t mPid; 116 }; 117 118 class IoConfigEvent : public ConfigEvent { 119 public: 120 IoConfigEvent(audio_io_config_event event, pid_t pid) : 121 ConfigEvent(CFG_EVENT_IO) { 122 mData = new IoConfigEventData(event, pid); 123 } 124 virtual ~IoConfigEvent() {} 125 }; 126 127 class PrioConfigEventData : public ConfigEventData { 128 public: 129 PrioConfigEventData(pid_t pid, pid_t tid, int32_t prio) : 130 mPid(pid), mTid(tid), mPrio(prio) {} 131 132 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: 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 } 147 virtual ~PrioConfigEvent() {} 148 }; 149 150 class SetParameterConfigEventData : public ConfigEventData { 151 public: 152 explicit SetParameterConfigEventData(String8 keyValuePairs) : 153 mKeyValuePairs(keyValuePairs) {} 154 155 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: 164 explicit SetParameterConfigEvent(String8 keyValuePairs) : 165 ConfigEvent(CFG_EVENT_SET_PARAMETER) { 166 mData = new SetParameterConfigEventData(keyValuePairs); 167 mWaitStatus = true; 168 } 169 virtual ~SetParameterConfigEvent() {} 170 }; 171 172 class CreateAudioPatchConfigEventData : public ConfigEventData { 173 public: 174 CreateAudioPatchConfigEventData(const struct audio_patch patch, 175 audio_patch_handle_t handle) : 176 mPatch(patch), mHandle(handle) {} 177 178 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: 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 } 194 virtual ~CreateAudioPatchConfigEvent() {} 195 }; 196 197 class ReleaseAudioPatchConfigEventData : public ConfigEventData { 198 public: 199 explicit ReleaseAudioPatchConfigEventData(const audio_patch_handle_t handle) : 200 mHandle(handle) {} 201 202 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: 211 explicit ReleaseAudioPatchConfigEvent(const audio_patch_handle_t handle) : 212 ConfigEvent(CFG_EVENT_RELEASE_AUDIO_PATCH) { 213 mData = new ReleaseAudioPatchConfigEventData(handle); 214 mWaitStatus = true; 215 } 216 virtual ~ReleaseAudioPatchConfigEvent() {} 217 }; 218 219 class PMDeathRecipient : public IBinder::DeathRecipient { 220 public: 221 explicit PMDeathRecipient(const wp<ThreadBase>& thread) : mThread(thread) {} 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 237 type_t type() const { return mType; } 238 bool isDuplicating() const { return (mType == DUPLICATING); } 239 240 audio_io_handle_t id() const { return mId;} 241 242 // dynamic externally-visible 243 uint32_t sampleRate() const { return mSampleRate; } 244 audio_channel_mask_t channelMask() const { return mChannelMask; } 245 audio_format_t format() const { return mHALFormat; } 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. 252 size_t frameCountHAL() const { return mFrameCount; } 253 254 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 285 bool standby() const { return mStandby; } 286 audio_devices_t outDevice() const { return mOutDevice; } 287 audio_devices_t inDevice() const { return mInDevice; } 288 289 virtual sp<StreamHalInterface> 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 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 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; 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 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. 380 virtual sp<MemoryDealer> readOnlyHeap() const { return 0; } 381 382 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 392protected: 393 394 // entry describing an effect being suspended in mSuspendedSessions keyed vector 395 class SuspendedSessionDesc : public RefBase { 396 public: 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(); 404 virtual void acquireWakeLock_l(); 405 void releaseWakeLock(); 406 void releaseWakeLock_l(); 407 void updateWakeLockUids_l(const SortedVector<int> &uids); 408 void getPowerManager_l(); 409 void setEffectSuspended_l(const effect_uuid_t *type, 410 bool suspend, 411 audio_session_t sessionId); 412 // updated mSuspendedSessions when an effect suspended or restored 413 void updateSuspendedSessions_l(const effect_uuid_t *type, 414 bool suspend, 415 audio_session_t sessionId); 416 // check if some effects must be suspended when an effect chain is added 417 void checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain); 418 419 String16 getWakeLockTag(); 420 421 virtual void preExit() { } 422 virtual void setMasterMono_l(bool mono __unused) { } 423 virtual bool requireMonoBlend() { return false; } 424 425 friend class AudioFlinger; // for mEffectChains 426 427 const type_t mType; 428 429 // Used by parameters, config events, addTrack_l, exit 430 Condition mWaitWorkCV; 431 432 const sp<AudioFlinger> mAudioFlinger; 433 434 // updated by PlaybackThread::readOutputParameters_l() or 435 // RecordThread::readInputParameters_l() 436 uint32_t mSampleRate; 437 size_t mFrameCount; // output HAL, direct output, record 438 audio_channel_mask_t mChannelMask; 439 uint32_t mChannelCount; 440 size_t mFrameSize; 441 // not HAL frame size, this is for output sink (to pipe to fast mixer) 442 audio_format_t mFormat; // Source format for Recording and 443 // Sink format for Playback. 444 // Sink format may be different than 445 // HAL format if Fastmixer is used. 446 audio_format_t mHALFormat; 447 size_t mBufferSize; // HAL buffer size for read() or write() 448 449 Vector< sp<ConfigEvent> > mConfigEvents; 450 Vector< sp<ConfigEvent> > mPendingConfigEvents; // events awaiting system ready 451 452 // These fields are written and read by thread itself without lock or barrier, 453 // and read by other threads without lock or barrier via standby(), outDevice() 454 // and inDevice(). 455 // Because of the absence of a lock or barrier, any other thread that reads 456 // these fields must use the information in isolation, or be prepared to deal 457 // with possibility that it might be inconsistent with other information. 458 bool mStandby; // Whether thread is currently in standby. 459 audio_devices_t mOutDevice; // output device 460 audio_devices_t mInDevice; // input device 461 audio_devices_t mPrevOutDevice; // previous output device 462 audio_devices_t mPrevInDevice; // previous input device 463 struct audio_patch mPatch; 464 audio_source_t mAudioSource; 465 466 const audio_io_handle_t mId; 467 Vector< sp<EffectChain> > mEffectChains; 468 469 static const int kThreadNameLength = 16; // prctl(PR_SET_NAME) limit 470 char mThreadName[kThreadNameLength]; // guaranteed NUL-terminated 471 sp<IPowerManager> mPowerManager; 472 sp<IBinder> mWakeLockToken; 473 const sp<PMDeathRecipient> mDeathRecipient; 474 // list of suspended effects per session and per type. The first (outer) vector is 475 // keyed by session ID, the second (inner) by type UUID timeLow field 476 KeyedVector< audio_session_t, KeyedVector< int, sp<SuspendedSessionDesc> > > 477 mSuspendedSessions; 478 static const size_t kLogSize = 4 * 1024; 479 sp<NBLog::Writer> mNBLogWriter; 480 bool mSystemReady; 481 ExtendedTimestamp mTimestamp; 482 483 // ActiveTracks is a sorted vector of track type T representing the 484 // active tracks of threadLoop() to be considered by the locked prepare portion. 485 // ActiveTracks should be accessed with the ThreadBase lock held. 486 // 487 // During processing and I/O, the threadLoop does not hold the lock; 488 // hence it does not directly use ActiveTracks. Care should be taken 489 // to hold local strong references or defer removal of tracks 490 // if the threadLoop may still be accessing those tracks due to mix, etc. 491 // 492 // This class updates power information appropriately. 493 // 494 495 template <typename T> 496 class ActiveTracks { 497 public: 498 ActiveTracks() 499 : mActiveTracksGeneration(0) 500 , mLastActiveTracksGeneration(0) 501 { } 502 503 ~ActiveTracks() { 504 clear(); 505 } 506 // returns the last track added (even though it may have been 507 // subsequently removed from ActiveTracks). 508 // 509 // Used for DirectOutputThread to ensure a flush is called when transitioning 510 // to a new track (even though it may be on the same session). 511 // Used for OffloadThread to ensure that volume and mixer state is 512 // taken from the latest track added. 513 // 514 // The latest track is saved with a weak pointer to prevent keeping an 515 // otherwise useless track alive. Thus the function will return nullptr 516 // if the latest track has subsequently been removed and destroyed. 517 sp<T> getLatest() { 518 return mLatestActiveTrack.promote(); 519 } 520 521 // Updates ActiveTracks client uids to the thread wakelock. 522 void updateWakeLockUids(sp<ThreadBase> thread, bool force = false) { 523 if (mActiveTracksGeneration != mLastActiveTracksGeneration || force) { 524 thread->updateWakeLockUids_l(getWakeLockUids()); 525 mLastActiveTracksGeneration = mActiveTracksGeneration; 526 } 527 } 528 529 // SortedVector methods 530 ssize_t add(const sp<T> &track); 531 ssize_t remove(const sp<T> &track); 532 size_t size() const { 533 return mActiveTracks.size(); 534 } 535 ssize_t indexOf(const sp<T>& item) { 536 return mActiveTracks.indexOf(item); 537 } 538 sp<T> operator[](size_t index) const { 539 return mActiveTracks[index]; 540 } 541 typename SortedVector<sp<T>>::iterator begin() { 542 return mActiveTracks.begin(); 543 } 544 typename SortedVector<sp<T>>::iterator end() { 545 return mActiveTracks.end(); 546 } 547 void clear(); 548 549 private: 550 SortedVector<int> getWakeLockUids() { 551 SortedVector<int> wakeLockUids; 552 for (const sp<T> &track : mActiveTracks) { 553 wakeLockUids.add(track->uid()); 554 } 555 return wakeLockUids; // moved by underlying SharedBuffer 556 } 557 558 SortedVector<sp<T>> mActiveTracks; 559 int mActiveTracksGeneration; 560 int mLastActiveTracksGeneration; 561 wp<T> mLatestActiveTrack; // latest track added to ActiveTracks 562 }; 563}; 564 565// --- PlaybackThread --- 566class PlaybackThread : public ThreadBase, public StreamOutHalInterfaceCallback { 567public: 568 569#include "PlaybackTracks.h" 570 571 enum mixer_state { 572 MIXER_IDLE, // no active tracks 573 MIXER_TRACKS_ENABLED, // at least one active track, but no track has any data ready 574 MIXER_TRACKS_READY, // at least one active track, and at least one track has data 575 MIXER_DRAIN_TRACK, // drain currently playing track 576 MIXER_DRAIN_ALL, // fully drain the hardware 577 // standby mode does not have an enum value 578 // suspend by audio policy manager is orthogonal to mixer state 579 }; 580 581 // retry count before removing active track in case of underrun on offloaded thread: 582 // we need to make sure that AudioTrack client has enough time to send large buffers 583 //FIXME may be more appropriate if expressed in time units. Need to revise how underrun is 584 // handled for offloaded tracks 585 static const int8_t kMaxTrackRetriesOffload = 20; 586 static const int8_t kMaxTrackStartupRetriesOffload = 100; 587 static const int8_t kMaxTrackStopRetriesOffload = 2; 588 // 14 tracks max per client allows for 2 misbehaving application leaving 4 available tracks. 589 static const uint32_t kMaxTracksPerUid = 14; 590 591 PlaybackThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 592 audio_io_handle_t id, audio_devices_t device, type_t type, bool systemReady); 593 virtual ~PlaybackThread(); 594 595 void dump(int fd, const Vector<String16>& args); 596 597 // Thread virtuals 598 virtual bool threadLoop(); 599 600 // RefBase 601 virtual void onFirstRef(); 602 603 virtual status_t checkEffectCompatibility_l(const effect_descriptor_t *desc, 604 audio_session_t sessionId); 605 606protected: 607 // Code snippets that were lifted up out of threadLoop() 608 virtual void threadLoop_mix() = 0; 609 virtual void threadLoop_sleepTime() = 0; 610 virtual ssize_t threadLoop_write(); 611 virtual void threadLoop_drain(); 612 virtual void threadLoop_standby(); 613 virtual void threadLoop_exit(); 614 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 615 616 // prepareTracks_l reads and writes mActiveTracks, and returns 617 // the pending set of tracks to remove via Vector 'tracksToRemove'. The caller 618 // is responsible for clearing or destroying this Vector later on, when it 619 // is safe to do so. That will drop the final ref count and destroy the tracks. 620 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove) = 0; 621 void removeTracks_l(const Vector< sp<Track> >& tracksToRemove); 622 623 // StreamOutHalInterfaceCallback implementation 624 virtual void onWriteReady(); 625 virtual void onDrainReady(); 626 virtual void onError(); 627 628 void resetWriteBlocked(uint32_t sequence); 629 void resetDraining(uint32_t sequence); 630 631 virtual bool waitingAsyncCallback(); 632 virtual bool waitingAsyncCallback_l(); 633 virtual bool shouldStandby_l(); 634 virtual void onAddNewTrack_l(); 635 void onAsyncError(); // error reported by AsyncCallbackThread 636 637 // ThreadBase virtuals 638 virtual void preExit(); 639 640 virtual bool keepWakeLock() const { return true; } 641 virtual void acquireWakeLock_l() { 642 ThreadBase::acquireWakeLock_l(); 643 mActiveTracks.updateWakeLockUids(this, true /* force */); 644 } 645 646public: 647 648 virtual status_t initCheck() const { return (mOutput == NULL) ? NO_INIT : NO_ERROR; } 649 650 // return estimated latency in milliseconds, as reported by HAL 651 uint32_t latency() const; 652 // same, but lock must already be held 653 uint32_t latency_l() const; 654 655 void setMasterVolume(float value); 656 void setMasterMute(bool muted); 657 658 void setStreamVolume(audio_stream_type_t stream, float value); 659 void setStreamMute(audio_stream_type_t stream, bool muted); 660 661 float streamVolume(audio_stream_type_t stream) const; 662 663 sp<Track> createTrack_l( 664 const sp<AudioFlinger::Client>& client, 665 audio_stream_type_t streamType, 666 uint32_t sampleRate, 667 audio_format_t format, 668 audio_channel_mask_t channelMask, 669 size_t *pFrameCount, 670 const sp<IMemory>& sharedBuffer, 671 audio_session_t sessionId, 672 audio_output_flags_t *flags, 673 pid_t tid, 674 int uid, 675 status_t *status /*non-NULL*/); 676 677 AudioStreamOut* getOutput() const; 678 AudioStreamOut* clearOutput(); 679 virtual sp<StreamHalInterface> stream() const; 680 681 // a very large number of suspend() will eventually wraparound, but unlikely 682 void suspend() { (void) android_atomic_inc(&mSuspended); } 683 void restore() 684 { 685 // if restore() is done without suspend(), get back into 686 // range so that the next suspend() will operate correctly 687 if (android_atomic_dec(&mSuspended) <= 0) { 688 android_atomic_release_store(0, &mSuspended); 689 } 690 } 691 bool isSuspended() const 692 { return android_atomic_acquire_load(&mSuspended) > 0; } 693 694 virtual String8 getParameters(const String8& keys); 695 virtual void ioConfigChanged(audio_io_config_event event, pid_t pid = 0); 696 status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames); 697 // FIXME rename mixBuffer() to sinkBuffer() and remove int16_t* dependency. 698 // Consider also removing and passing an explicit mMainBuffer initialization 699 // parameter to AF::PlaybackThread::Track::Track(). 700 int16_t *mixBuffer() const { 701 return reinterpret_cast<int16_t *>(mSinkBuffer); }; 702 703 virtual void detachAuxEffect_l(int effectId); 704 status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track>& track, 705 int EffectId); 706 status_t attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track>& track, 707 int EffectId); 708 709 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 710 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 711 virtual uint32_t hasAudioSession_l(audio_session_t sessionId) const; 712 virtual uint32_t getStrategyForSession_l(audio_session_t sessionId); 713 714 715 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 716 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 717 718 // called with AudioFlinger lock held 719 bool invalidateTracks_l(audio_stream_type_t streamType); 720 virtual void invalidateTracks(audio_stream_type_t streamType); 721 722 virtual size_t frameCount() const { return mNormalFrameCount; } 723 724 status_t getTimestamp_l(AudioTimestamp& timestamp); 725 726 void addPatchTrack(const sp<PatchTrack>& track); 727 void deletePatchTrack(const sp<PatchTrack>& track); 728 729 virtual void getAudioPortConfig(struct audio_port_config *config); 730 731protected: 732 // updated by readOutputParameters_l() 733 size_t mNormalFrameCount; // normal mixer and effects 734 735 bool mThreadThrottle; // throttle the thread processing 736 uint32_t mThreadThrottleTimeMs; // throttle time for MIXER threads 737 uint32_t mThreadThrottleEndMs; // notify once per throttling 738 uint32_t mHalfBufferMs; // half the buffer size in milliseconds 739 740 void* mSinkBuffer; // frame size aligned sink buffer 741 742 // TODO: 743 // Rearrange the buffer info into a struct/class with 744 // clear, copy, construction, destruction methods. 745 // 746 // mSinkBuffer also has associated with it: 747 // 748 // mSinkBufferSize: Sink Buffer Size 749 // mFormat: Sink Buffer Format 750 751 // Mixer Buffer (mMixerBuffer*) 752 // 753 // In the case of floating point or multichannel data, which is not in the 754 // sink format, it is required to accumulate in a higher precision or greater channel count 755 // buffer before downmixing or data conversion to the sink buffer. 756 757 // Set to "true" to enable the Mixer Buffer otherwise mixer output goes to sink buffer. 758 bool mMixerBufferEnabled; 759 760 // Storage, 32 byte aligned (may make this alignment a requirement later). 761 // Due to constraints on mNormalFrameCount, the buffer size is a multiple of 16 frames. 762 void* mMixerBuffer; 763 764 // Size of mMixerBuffer in bytes: mNormalFrameCount * #channels * sampsize. 765 size_t mMixerBufferSize; 766 767 // The audio format of mMixerBuffer. Set to AUDIO_FORMAT_PCM_(FLOAT|16_BIT) only. 768 audio_format_t mMixerBufferFormat; 769 770 // An internal flag set to true by MixerThread::prepareTracks_l() 771 // when mMixerBuffer contains valid data after mixing. 772 bool mMixerBufferValid; 773 774 // Effects Buffer (mEffectsBuffer*) 775 // 776 // In the case of effects data, which is not in the sink format, 777 // it is required to accumulate in a different buffer before data conversion 778 // to the sink buffer. 779 780 // Set to "true" to enable the Effects Buffer otherwise effects output goes to sink buffer. 781 bool mEffectBufferEnabled; 782 783 // Storage, 32 byte aligned (may make this alignment a requirement later). 784 // Due to constraints on mNormalFrameCount, the buffer size is a multiple of 16 frames. 785 void* mEffectBuffer; 786 787 // Size of mEffectsBuffer in bytes: mNormalFrameCount * #channels * sampsize. 788 size_t mEffectBufferSize; 789 790 // The audio format of mEffectsBuffer. Set to AUDIO_FORMAT_PCM_16_BIT only. 791 audio_format_t mEffectBufferFormat; 792 793 // An internal flag set to true by MixerThread::prepareTracks_l() 794 // when mEffectsBuffer contains valid data after mixing. 795 // 796 // When this is set, all mixer data is routed into the effects buffer 797 // for any processing (including output processing). 798 bool mEffectBufferValid; 799 800 // suspend count, > 0 means suspended. While suspended, the thread continues to pull from 801 // tracks and mix, but doesn't write to HAL. A2DP and SCO HAL implementations can't handle 802 // concurrent use of both of them, so Audio Policy Service suspends one of the threads to 803 // workaround that restriction. 804 // 'volatile' means accessed via atomic operations and no lock. 805 volatile int32_t mSuspended; 806 807 int64_t mBytesWritten; 808 int64_t mFramesWritten; // not reset on standby 809 int64_t mSuspendedFrames; // not reset on standby 810private: 811 // mMasterMute is in both PlaybackThread and in AudioFlinger. When a 812 // PlaybackThread needs to find out if master-muted, it checks it's local 813 // copy rather than the one in AudioFlinger. This optimization saves a lock. 814 bool mMasterMute; 815 void setMasterMute_l(bool muted) { mMasterMute = muted; } 816protected: 817 ActiveTracks<Track> mActiveTracks; 818 819 // Allocate a track name for a given channel mask. 820 // Returns name >= 0 if successful, -1 on failure. 821 virtual int getTrackName_l(audio_channel_mask_t channelMask, audio_format_t format, 822 audio_session_t sessionId, uid_t uid) = 0; 823 virtual void deleteTrackName_l(int name) = 0; 824 825 // Time to sleep between cycles when: 826 virtual uint32_t activeSleepTimeUs() const; // mixer state MIXER_TRACKS_ENABLED 827 virtual uint32_t idleSleepTimeUs() const = 0; // mixer state MIXER_IDLE 828 virtual uint32_t suspendSleepTimeUs() const = 0; // audio policy manager suspended us 829 // No sleep when mixer state == MIXER_TRACKS_READY; relies on audio HAL stream->write() 830 // No sleep in standby mode; waits on a condition 831 832 // Code snippets that are temporarily lifted up out of threadLoop() until the merge 833 void checkSilentMode_l(); 834 835 // Non-trivial for DUPLICATING only 836 virtual void saveOutputTracks() { } 837 virtual void clearOutputTracks() { } 838 839 // Cache various calculated values, at threadLoop() entry and after a parameter change 840 virtual void cacheParameters_l(); 841 842 virtual uint32_t correctLatency_l(uint32_t latency) const; 843 844 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 845 audio_patch_handle_t *handle); 846 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 847 848 bool usesHwAvSync() const { return (mType == DIRECT) && (mOutput != NULL) 849 && mHwSupportsPause 850 && (mOutput->flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC); } 851 852 uint32_t trackCountForUid_l(uid_t uid); 853 854private: 855 856 friend class AudioFlinger; // for numerous 857 858 PlaybackThread& operator = (const PlaybackThread&); 859 860 status_t addTrack_l(const sp<Track>& track); 861 bool destroyTrack_l(const sp<Track>& track); 862 void removeTrack_l(const sp<Track>& track); 863 void broadcast_l(); 864 865 void readOutputParameters_l(); 866 867 virtual void dumpInternals(int fd, const Vector<String16>& args); 868 void dumpTracks(int fd, const Vector<String16>& args); 869 870 SortedVector< sp<Track> > mTracks; 871 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 872 AudioStreamOut *mOutput; 873 874 float mMasterVolume; 875 nsecs_t mLastWriteTime; 876 int mNumWrites; 877 int mNumDelayedWrites; 878 bool mInWrite; 879 880 // FIXME rename these former local variables of threadLoop to standard "m" names 881 nsecs_t mStandbyTimeNs; 882 size_t mSinkBufferSize; 883 884 // cached copies of activeSleepTimeUs() and idleSleepTimeUs() made by cacheParameters_l() 885 uint32_t mActiveSleepTimeUs; 886 uint32_t mIdleSleepTimeUs; 887 888 uint32_t mSleepTimeUs; 889 890 // mixer status returned by prepareTracks_l() 891 mixer_state mMixerStatus; // current cycle 892 // previous cycle when in prepareTracks_l() 893 mixer_state mMixerStatusIgnoringFastTracks; 894 // FIXME or a separate ready state per track 895 896 // FIXME move these declarations into the specific sub-class that needs them 897 // MIXER only 898 uint32_t sleepTimeShift; 899 900 // same as AudioFlinger::mStandbyTimeInNsecs except for DIRECT which uses a shorter value 901 nsecs_t mStandbyDelayNs; 902 903 // MIXER only 904 nsecs_t maxPeriod; 905 906 // DUPLICATING only 907 uint32_t writeFrames; 908 909 size_t mBytesRemaining; 910 size_t mCurrentWriteLength; 911 bool mUseAsyncWrite; 912 // mWriteAckSequence contains current write sequence on bits 31-1. The write sequence is 913 // incremented each time a write(), a flush() or a standby() occurs. 914 // Bit 0 is set when a write blocks and indicates a callback is expected. 915 // Bit 0 is reset by the async callback thread calling resetWriteBlocked(). Out of sequence 916 // callbacks are ignored. 917 uint32_t mWriteAckSequence; 918 // mDrainSequence contains current drain sequence on bits 31-1. The drain sequence is 919 // incremented each time a drain is requested or a flush() or standby() occurs. 920 // Bit 0 is set when the drain() command is called at the HAL and indicates a callback is 921 // expected. 922 // Bit 0 is reset by the async callback thread calling resetDraining(). Out of sequence 923 // callbacks are ignored. 924 uint32_t mDrainSequence; 925 // A condition that must be evaluated by prepareTrack_l() has changed and we must not wait 926 // for async write callback in the thread loop before evaluating it 927 bool mSignalPending; 928 sp<AsyncCallbackThread> mCallbackThread; 929 930private: 931 // The HAL output sink is treated as non-blocking, but current implementation is blocking 932 sp<NBAIO_Sink> mOutputSink; 933 // If a fast mixer is present, the blocking pipe sink, otherwise clear 934 sp<NBAIO_Sink> mPipeSink; 935 // The current sink for the normal mixer to write it's (sub)mix, mOutputSink or mPipeSink 936 sp<NBAIO_Sink> mNormalSink; 937#ifdef TEE_SINK 938 // For dumpsys 939 sp<NBAIO_Sink> mTeeSink; 940 sp<NBAIO_Source> mTeeSource; 941#endif 942 uint32_t mScreenState; // cached copy of gScreenState 943 static const size_t kFastMixerLogSize = 4 * 1024; 944 sp<NBLog::Writer> mFastMixerNBLogWriter; 945public: 946 virtual bool hasFastMixer() const = 0; 947 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex __unused) const 948 { FastTrackUnderruns dummy; return dummy; } 949 950protected: 951 // accessed by both binder threads and within threadLoop(), lock on mutex needed 952 unsigned mFastTrackAvailMask; // bit i set if fast track [i] is available 953 bool mHwSupportsPause; 954 bool mHwPaused; 955 bool mFlushPending; 956}; 957 958class MixerThread : public PlaybackThread { 959public: 960 MixerThread(const sp<AudioFlinger>& audioFlinger, 961 AudioStreamOut* output, 962 audio_io_handle_t id, 963 audio_devices_t device, 964 bool systemReady, 965 type_t type = MIXER); 966 virtual ~MixerThread(); 967 968 // Thread virtuals 969 970 virtual bool checkForNewParameter_l(const String8& keyValuePair, 971 status_t& status); 972 virtual void dumpInternals(int fd, const Vector<String16>& args); 973 974protected: 975 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 976 virtual int getTrackName_l(audio_channel_mask_t channelMask, audio_format_t format, 977 audio_session_t sessionId, uid_t uid); 978 virtual void deleteTrackName_l(int name); 979 virtual uint32_t idleSleepTimeUs() const; 980 virtual uint32_t suspendSleepTimeUs() const; 981 virtual void cacheParameters_l(); 982 983 virtual void acquireWakeLock_l() { 984 PlaybackThread::acquireWakeLock_l(); 985 if (hasFastMixer()) { 986 mFastMixer->setBoottimeOffset( 987 mTimestamp.mTimebaseOffset[ExtendedTimestamp::TIMEBASE_BOOTTIME]); 988 } 989 } 990 991 // threadLoop snippets 992 virtual ssize_t threadLoop_write(); 993 virtual void threadLoop_standby(); 994 virtual void threadLoop_mix(); 995 virtual void threadLoop_sleepTime(); 996 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 997 virtual uint32_t correctLatency_l(uint32_t latency) const; 998 999 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 1000 audio_patch_handle_t *handle); 1001 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 1002 1003 AudioMixer* mAudioMixer; // normal mixer 1004private: 1005 // one-time initialization, no locks required 1006 sp<FastMixer> mFastMixer; // non-0 if there is also a fast mixer 1007 sp<AudioWatchdog> mAudioWatchdog; // non-0 if there is an audio watchdog thread 1008 1009 // contents are not guaranteed to be consistent, no locks required 1010 FastMixerDumpState mFastMixerDumpState; 1011#ifdef STATE_QUEUE_DUMP 1012 StateQueueObserverDump mStateQueueObserverDump; 1013 StateQueueMutatorDump mStateQueueMutatorDump; 1014#endif 1015 AudioWatchdogDump mAudioWatchdogDump; 1016 1017 // accessible only within the threadLoop(), no locks required 1018 // mFastMixer->sq() // for mutating and pushing state 1019 int32_t mFastMixerFutex; // for cold idle 1020 1021 std::atomic_bool mMasterMono; 1022public: 1023 virtual bool hasFastMixer() const { return mFastMixer != 0; } 1024 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const { 1025 ALOG_ASSERT(fastIndex < FastMixerState::sMaxFastTracks); 1026 return mFastMixerDumpState.mTracks[fastIndex].mUnderruns; 1027 } 1028 1029protected: 1030 virtual void setMasterMono_l(bool mono) { 1031 mMasterMono.store(mono); 1032 if (mFastMixer != nullptr) { /* hasFastMixer() */ 1033 mFastMixer->setMasterMono(mMasterMono); 1034 } 1035 } 1036 // the FastMixer performs mono blend if it exists. 1037 // Blending with limiter is not idempotent, 1038 // and blending without limiter is idempotent but inefficient to do twice. 1039 virtual bool requireMonoBlend() { return mMasterMono.load() && !hasFastMixer(); } 1040}; 1041 1042class DirectOutputThread : public PlaybackThread { 1043public: 1044 1045 DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1046 audio_io_handle_t id, audio_devices_t device, bool systemReady); 1047 virtual ~DirectOutputThread(); 1048 1049 // Thread virtuals 1050 1051 virtual bool checkForNewParameter_l(const String8& keyValuePair, 1052 status_t& status); 1053 virtual void flushHw_l(); 1054 1055protected: 1056 virtual int getTrackName_l(audio_channel_mask_t channelMask, audio_format_t format, 1057 audio_session_t sessionId, uid_t uid); 1058 virtual void deleteTrackName_l(int name); 1059 virtual uint32_t activeSleepTimeUs() const; 1060 virtual uint32_t idleSleepTimeUs() const; 1061 virtual uint32_t suspendSleepTimeUs() const; 1062 virtual void cacheParameters_l(); 1063 1064 // threadLoop snippets 1065 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1066 virtual void threadLoop_mix(); 1067 virtual void threadLoop_sleepTime(); 1068 virtual void threadLoop_exit(); 1069 virtual bool shouldStandby_l(); 1070 1071 virtual void onAddNewTrack_l(); 1072 1073 // volumes last sent to audio HAL with stream->set_volume() 1074 float mLeftVolFloat; 1075 float mRightVolFloat; 1076 1077 DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1078 audio_io_handle_t id, uint32_t device, ThreadBase::type_t type, 1079 bool systemReady); 1080 void processVolume_l(Track *track, bool lastTrack); 1081 1082 // prepareTracks_l() tells threadLoop_mix() the name of the single active track 1083 sp<Track> mActiveTrack; 1084 1085 wp<Track> mPreviousTrack; // used to detect track switch 1086 1087public: 1088 virtual bool hasFastMixer() const { return false; } 1089}; 1090 1091class OffloadThread : public DirectOutputThread { 1092public: 1093 1094 OffloadThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1095 audio_io_handle_t id, uint32_t device, bool systemReady); 1096 virtual ~OffloadThread() {}; 1097 virtual void flushHw_l(); 1098 1099protected: 1100 // threadLoop snippets 1101 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1102 virtual void threadLoop_exit(); 1103 1104 virtual bool waitingAsyncCallback(); 1105 virtual bool waitingAsyncCallback_l(); 1106 virtual void invalidateTracks(audio_stream_type_t streamType); 1107 1108 virtual bool keepWakeLock() const { return (mKeepWakeLock || (mDrainSequence & 1)); } 1109 1110private: 1111 size_t mPausedWriteLength; // length in bytes of write interrupted by pause 1112 size_t mPausedBytesRemaining; // bytes still waiting in mixbuffer after resume 1113 bool mKeepWakeLock; // keep wake lock while waiting for write callback 1114 uint64_t mOffloadUnderrunPosition; // Current frame position for offloaded playback 1115 // used and valid only during underrun. ~0 if 1116 // no underrun has occurred during playback and 1117 // is not reset on standby. 1118}; 1119 1120class AsyncCallbackThread : public Thread { 1121public: 1122 1123 explicit AsyncCallbackThread(const wp<PlaybackThread>& playbackThread); 1124 1125 virtual ~AsyncCallbackThread(); 1126 1127 // Thread virtuals 1128 virtual bool threadLoop(); 1129 1130 // RefBase 1131 virtual void onFirstRef(); 1132 1133 void exit(); 1134 void setWriteBlocked(uint32_t sequence); 1135 void resetWriteBlocked(); 1136 void setDraining(uint32_t sequence); 1137 void resetDraining(); 1138 void setAsyncError(); 1139 1140private: 1141 const wp<PlaybackThread> mPlaybackThread; 1142 // mWriteAckSequence corresponds to the last write sequence passed by the offload thread via 1143 // setWriteBlocked(). The sequence is shifted one bit to the left and the lsb is used 1144 // to indicate that the callback has been received via resetWriteBlocked() 1145 uint32_t mWriteAckSequence; 1146 // mDrainSequence corresponds to the last drain sequence passed by the offload thread via 1147 // setDraining(). The sequence is shifted one bit to the left and the lsb is used 1148 // to indicate that the callback has been received via resetDraining() 1149 uint32_t mDrainSequence; 1150 Condition mWaitWorkCV; 1151 Mutex mLock; 1152 bool mAsyncError; 1153}; 1154 1155class DuplicatingThread : public MixerThread { 1156public: 1157 DuplicatingThread(const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread, 1158 audio_io_handle_t id, bool systemReady); 1159 virtual ~DuplicatingThread(); 1160 1161 // Thread virtuals 1162 void addOutputTrack(MixerThread* thread); 1163 void removeOutputTrack(MixerThread* thread); 1164 uint32_t waitTimeMs() const { return mWaitTimeMs; } 1165protected: 1166 virtual uint32_t activeSleepTimeUs() const; 1167 1168private: 1169 bool outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks); 1170protected: 1171 // threadLoop snippets 1172 virtual void threadLoop_mix(); 1173 virtual void threadLoop_sleepTime(); 1174 virtual ssize_t threadLoop_write(); 1175 virtual void threadLoop_standby(); 1176 virtual void cacheParameters_l(); 1177 1178private: 1179 // called from threadLoop, addOutputTrack, removeOutputTrack 1180 virtual void updateWaitTime_l(); 1181protected: 1182 virtual void saveOutputTracks(); 1183 virtual void clearOutputTracks(); 1184private: 1185 1186 uint32_t mWaitTimeMs; 1187 SortedVector < sp<OutputTrack> > outputTracks; 1188 SortedVector < sp<OutputTrack> > mOutputTracks; 1189public: 1190 virtual bool hasFastMixer() const { return false; } 1191}; 1192 1193 1194// record thread 1195class RecordThread : public ThreadBase 1196{ 1197public: 1198 1199 class RecordTrack; 1200 1201 /* The ResamplerBufferProvider is used to retrieve recorded input data from the 1202 * RecordThread. It maintains local state on the relative position of the read 1203 * position of the RecordTrack compared with the RecordThread. 1204 */ 1205 class ResamplerBufferProvider : public AudioBufferProvider 1206 { 1207 public: 1208 explicit ResamplerBufferProvider(RecordTrack* recordTrack) : 1209 mRecordTrack(recordTrack), 1210 mRsmpInUnrel(0), mRsmpInFront(0) { } 1211 virtual ~ResamplerBufferProvider() { } 1212 1213 // called to set the ResamplerBufferProvider to head of the RecordThread data buffer, 1214 // skipping any previous data read from the hal. 1215 virtual void reset(); 1216 1217 /* Synchronizes RecordTrack position with the RecordThread. 1218 * Calculates available frames and handle overruns if the RecordThread 1219 * has advanced faster than the ResamplerBufferProvider has retrieved data. 1220 * TODO: why not do this for every getNextBuffer? 1221 * 1222 * Parameters 1223 * framesAvailable: pointer to optional output size_t to store record track 1224 * frames available. 1225 * hasOverrun: pointer to optional boolean, returns true if track has overrun. 1226 */ 1227 1228 virtual void sync(size_t *framesAvailable = NULL, bool *hasOverrun = NULL); 1229 1230 // AudioBufferProvider interface 1231 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer); 1232 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 1233 private: 1234 RecordTrack * const mRecordTrack; 1235 size_t mRsmpInUnrel; // unreleased frames remaining from 1236 // most recent getNextBuffer 1237 // for debug only 1238 int32_t mRsmpInFront; // next available frame 1239 // rolling counter that is never cleared 1240 }; 1241 1242 /* The RecordBufferConverter is used for format, channel, and sample rate 1243 * conversion for a RecordTrack. 1244 * 1245 * TODO: Self contained, so move to a separate file later. 1246 * 1247 * RecordBufferConverter uses the convert() method rather than exposing a 1248 * buffer provider interface; this is to save a memory copy. 1249 */ 1250 class RecordBufferConverter 1251 { 1252 public: 1253 RecordBufferConverter( 1254 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 1255 uint32_t srcSampleRate, 1256 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 1257 uint32_t dstSampleRate); 1258 1259 ~RecordBufferConverter(); 1260 1261 /* Converts input data from an AudioBufferProvider by format, channelMask, 1262 * and sampleRate to a destination buffer. 1263 * 1264 * Parameters 1265 * dst: buffer to place the converted data. 1266 * provider: buffer provider to obtain source data. 1267 * frames: number of frames to convert 1268 * 1269 * Returns the number of frames converted. 1270 */ 1271 size_t convert(void *dst, AudioBufferProvider *provider, size_t frames); 1272 1273 // returns NO_ERROR if constructor was successful 1274 status_t initCheck() const { 1275 // mSrcChannelMask set on successful updateParameters 1276 return mSrcChannelMask != AUDIO_CHANNEL_INVALID ? NO_ERROR : NO_INIT; 1277 } 1278 1279 // allows dynamic reconfigure of all parameters 1280 status_t updateParameters( 1281 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 1282 uint32_t srcSampleRate, 1283 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 1284 uint32_t dstSampleRate); 1285 1286 // called to reset resampler buffers on record track discontinuity 1287 void reset() { 1288 if (mResampler != NULL) { 1289 mResampler->reset(); 1290 } 1291 } 1292 1293 private: 1294 // format conversion when not using resampler 1295 void convertNoResampler(void *dst, const void *src, size_t frames); 1296 1297 // format conversion when using resampler; modifies src in-place 1298 void convertResampler(void *dst, /*not-a-const*/ void *src, size_t frames); 1299 1300 // user provided information 1301 audio_channel_mask_t mSrcChannelMask; 1302 audio_format_t mSrcFormat; 1303 uint32_t mSrcSampleRate; 1304 audio_channel_mask_t mDstChannelMask; 1305 audio_format_t mDstFormat; 1306 uint32_t mDstSampleRate; 1307 1308 // derived information 1309 uint32_t mSrcChannelCount; 1310 uint32_t mDstChannelCount; 1311 size_t mDstFrameSize; 1312 1313 // format conversion buffer 1314 void *mBuf; 1315 size_t mBufFrames; 1316 size_t mBufFrameSize; 1317 1318 // resampler info 1319 AudioResampler *mResampler; 1320 1321 bool mIsLegacyDownmix; // legacy stereo to mono conversion needed 1322 bool mIsLegacyUpmix; // legacy mono to stereo conversion needed 1323 bool mRequiresFloat; // data processing requires float (e.g. resampler) 1324 PassthruBufferProvider *mInputConverterProvider; // converts input to float 1325 int8_t mIdxAry[sizeof(uint32_t) * 8]; // used for channel mask conversion 1326 }; 1327 1328#include "RecordTracks.h" 1329 1330 RecordThread(const sp<AudioFlinger>& audioFlinger, 1331 AudioStreamIn *input, 1332 audio_io_handle_t id, 1333 audio_devices_t outDevice, 1334 audio_devices_t inDevice, 1335 bool systemReady 1336#ifdef TEE_SINK 1337 , const sp<NBAIO_Sink>& teeSink 1338#endif 1339 ); 1340 virtual ~RecordThread(); 1341 1342 // no addTrack_l ? 1343 void destroyTrack_l(const sp<RecordTrack>& track); 1344 void removeTrack_l(const sp<RecordTrack>& track); 1345 1346 void dumpInternals(int fd, const Vector<String16>& args); 1347 void dumpTracks(int fd, const Vector<String16>& args); 1348 1349 // Thread virtuals 1350 virtual bool threadLoop(); 1351 1352 // RefBase 1353 virtual void onFirstRef(); 1354 1355 virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; } 1356 1357 virtual sp<MemoryDealer> readOnlyHeap() const { return mReadOnlyHeap; } 1358 1359 virtual sp<IMemory> pipeMemory() const { return mPipeMemory; } 1360 1361 sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l( 1362 const sp<AudioFlinger::Client>& client, 1363 uint32_t sampleRate, 1364 audio_format_t format, 1365 audio_channel_mask_t channelMask, 1366 size_t *pFrameCount, 1367 audio_session_t sessionId, 1368 size_t *notificationFrames, 1369 int uid, 1370 audio_input_flags_t *flags, 1371 pid_t tid, 1372 status_t *status /*non-NULL*/); 1373 1374 status_t start(RecordTrack* recordTrack, 1375 AudioSystem::sync_event_t event, 1376 audio_session_t triggerSession); 1377 1378 // ask the thread to stop the specified track, and 1379 // return true if the caller should then do it's part of the stopping process 1380 bool stop(RecordTrack* recordTrack); 1381 1382 void dump(int fd, const Vector<String16>& args); 1383 AudioStreamIn* clearInput(); 1384 virtual sp<StreamHalInterface> stream() const; 1385 1386 1387 virtual bool checkForNewParameter_l(const String8& keyValuePair, 1388 status_t& status); 1389 virtual void cacheParameters_l() {} 1390 virtual String8 getParameters(const String8& keys); 1391 virtual void ioConfigChanged(audio_io_config_event event, pid_t pid = 0); 1392 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 1393 audio_patch_handle_t *handle); 1394 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 1395 1396 void addPatchRecord(const sp<PatchRecord>& record); 1397 void deletePatchRecord(const sp<PatchRecord>& record); 1398 1399 void readInputParameters_l(); 1400 virtual uint32_t getInputFramesLost(); 1401 1402 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1403 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1404 virtual uint32_t hasAudioSession_l(audio_session_t sessionId) const; 1405 1406 // Return the set of unique session IDs across all tracks. 1407 // The keys are the session IDs, and the associated values are meaningless. 1408 // FIXME replace by Set [and implement Bag/Multiset for other uses]. 1409 KeyedVector<audio_session_t, bool> sessionIds() const; 1410 1411 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 1412 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 1413 1414 static void syncStartEventCallback(const wp<SyncEvent>& event); 1415 1416 virtual size_t frameCount() const { return mFrameCount; } 1417 bool hasFastCapture() const { return mFastCapture != 0; } 1418 virtual void getAudioPortConfig(struct audio_port_config *config); 1419 1420 virtual status_t checkEffectCompatibility_l(const effect_descriptor_t *desc, 1421 audio_session_t sessionId); 1422 1423 virtual void acquireWakeLock_l() { 1424 ThreadBase::acquireWakeLock_l(); 1425 mActiveTracks.updateWakeLockUids(this, true /* force */); 1426 } 1427 1428private: 1429 // Enter standby if not already in standby, and set mStandby flag 1430 void standbyIfNotAlreadyInStandby(); 1431 1432 // Call the HAL standby method unconditionally, and don't change mStandby flag 1433 void inputStandBy(); 1434 1435 AudioStreamIn *mInput; 1436 SortedVector < sp<RecordTrack> > mTracks; 1437 // mActiveTracks has dual roles: it indicates the current active track(s), and 1438 // is used together with mStartStopCond to indicate start()/stop() progress 1439 ActiveTracks<RecordTrack> mActiveTracks; 1440 1441 Condition mStartStopCond; 1442 1443 // resampler converts input at HAL Hz to output at AudioRecord client Hz 1444 void *mRsmpInBuffer; // size = mRsmpInFramesOA 1445 size_t mRsmpInFrames; // size of resampler input in frames 1446 size_t mRsmpInFramesP2;// size rounded up to a power-of-2 1447 size_t mRsmpInFramesOA;// mRsmpInFramesP2 + over-allocation 1448 1449 // rolling index that is never cleared 1450 int32_t mRsmpInRear; // last filled frame + 1 1451 1452 // For dumpsys 1453 const sp<NBAIO_Sink> mTeeSink; 1454 1455 const sp<MemoryDealer> mReadOnlyHeap; 1456 1457 // one-time initialization, no locks required 1458 sp<FastCapture> mFastCapture; // non-0 if there is also 1459 // a fast capture 1460 1461 // FIXME audio watchdog thread 1462 1463 // contents are not guaranteed to be consistent, no locks required 1464 FastCaptureDumpState mFastCaptureDumpState; 1465#ifdef STATE_QUEUE_DUMP 1466 // FIXME StateQueue observer and mutator dump fields 1467#endif 1468 // FIXME audio watchdog dump 1469 1470 // accessible only within the threadLoop(), no locks required 1471 // mFastCapture->sq() // for mutating and pushing state 1472 int32_t mFastCaptureFutex; // for cold idle 1473 1474 // The HAL input source is treated as non-blocking, 1475 // but current implementation is blocking 1476 sp<NBAIO_Source> mInputSource; 1477 // The source for the normal capture thread to read from: mInputSource or mPipeSource 1478 sp<NBAIO_Source> mNormalSource; 1479 // If a fast capture is present, the non-blocking pipe sink written to by fast capture, 1480 // otherwise clear 1481 sp<NBAIO_Sink> mPipeSink; 1482 // If a fast capture is present, the non-blocking pipe source read by normal thread, 1483 // otherwise clear 1484 sp<NBAIO_Source> mPipeSource; 1485 // Depth of pipe from fast capture to normal thread and fast clients, always power of 2 1486 size_t mPipeFramesP2; 1487 // If a fast capture is present, the Pipe as IMemory, otherwise clear 1488 sp<IMemory> mPipeMemory; 1489 1490 static const size_t kFastCaptureLogSize = 4 * 1024; 1491 sp<NBLog::Writer> mFastCaptureNBLogWriter; 1492 1493 bool mFastTrackAvail; // true if fast track available 1494}; 1495