1 /* 2 ** 3 ** Copyright 2012, The Android Open Source Project 4 ** 5 ** Licensed under the Apache License, Version 2.0 (the "License"); 6 ** you may not use this file except in compliance with the License. 7 ** You may obtain a copy of the License at 8 ** 9 ** http://www.apache.org/licenses/LICENSE-2.0 10 ** 11 ** Unless required by applicable law or agreed to in writing, software 12 ** distributed under the License is distributed on an "AS IS" BASIS, 13 ** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14 ** See the License for the specific language governing permissions and 15 ** limitations under the License. 16 */ 17 18 #ifndef INCLUDING_FROM_AUDIOFLINGER_H 19 #error This header file should only be included from AudioFlinger.h 20 #endif 21 22 class ThreadBase : public Thread { 23 public: 24 25 #include "TrackBase.h" 26 27 enum type_t { 28 MIXER, // Thread class is MixerThread 29 DIRECT, // Thread class is DirectOutputThread 30 DUPLICATING, // Thread class is DuplicatingThread 31 RECORD, // Thread class is RecordThread 32 OFFLOAD, // Thread class is OffloadThread 33 MMAP // control thread for MMAP stream 34 // If you add any values here, also update ThreadBase::threadTypeToString() 35 }; 36 37 static const char *threadTypeToString(type_t type); 38 39 ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 40 audio_devices_t outDevice, audio_devices_t inDevice, type_t type, 41 bool systemReady); 42 virtual ~ThreadBase(); 43 44 virtual status_t readyToRun(); 45 46 void dumpBase(int fd, const Vector<String16>& args); 47 void dumpEffectChains(int fd, const Vector<String16>& args); 48 49 void clearPowerManager(); 50 51 // base for record and playback 52 enum { 53 CFG_EVENT_IO, 54 CFG_EVENT_PRIO, 55 CFG_EVENT_SET_PARAMETER, 56 CFG_EVENT_CREATE_AUDIO_PATCH, 57 CFG_EVENT_RELEASE_AUDIO_PATCH, 58 }; 59 60 class ConfigEventData: public RefBase { 61 public: 62 virtual ~ConfigEventData() {} 63 64 virtual void dump(char *buffer, size_t size) = 0; 65 protected: 66 ConfigEventData() {} 67 }; 68 69 // Config event sequence by client if status needed (e.g binder thread calling setParameters()): 70 // 1. create SetParameterConfigEvent. This sets mWaitStatus in config event 71 // 2. Lock mLock 72 // 3. Call sendConfigEvent_l(): Append to mConfigEvents and mWaitWorkCV.signal 73 // 4. sendConfigEvent_l() reads status from event->mStatus; 74 // 5. sendConfigEvent_l() returns status 75 // 6. Unlock 76 // 77 // Parameter sequence by server: threadLoop calling processConfigEvents_l(): 78 // 1. Lock mLock 79 // 2. If there is an entry in mConfigEvents proceed ... 80 // 3. Read first entry in mConfigEvents 81 // 4. Remove first entry from mConfigEvents 82 // 5. Process 83 // 6. Set event->mStatus 84 // 7. event->mCond.signal 85 // 8. Unlock 86 87 class ConfigEvent: public RefBase { 88 public: 89 virtual ~ConfigEvent() {} 90 91 void dump(char *buffer, size_t size) { mData->dump(buffer, size); } 92 93 const int mType; // event type e.g. CFG_EVENT_IO 94 Mutex mLock; // mutex associated with mCond 95 Condition mCond; // condition for status return 96 status_t mStatus; // status communicated to sender 97 bool mWaitStatus; // true if sender is waiting for status 98 bool mRequiresSystemReady; // true if must wait for system ready to enter event queue 99 sp<ConfigEventData> mData; // event specific parameter data 100 101 protected: 102 explicit ConfigEvent(int type, bool requiresSystemReady = false) : 103 mType(type), mStatus(NO_ERROR), mWaitStatus(false), 104 mRequiresSystemReady(requiresSystemReady), mData(NULL) {} 105 }; 106 107 class IoConfigEventData : public ConfigEventData { 108 public: 109 IoConfigEventData(audio_io_config_event event, pid_t pid) : 110 mEvent(event), mPid(pid) {} 111 112 virtual void dump(char *buffer, size_t size) { 113 snprintf(buffer, size, "IO event: event %d\n", mEvent); 114 } 115 116 const audio_io_config_event mEvent; 117 const pid_t mPid; 118 }; 119 120 class IoConfigEvent : public ConfigEvent { 121 public: 122 IoConfigEvent(audio_io_config_event event, pid_t pid) : 123 ConfigEvent(CFG_EVENT_IO) { 124 mData = new IoConfigEventData(event, pid); 125 } 126 virtual ~IoConfigEvent() {} 127 }; 128 129 class PrioConfigEventData : public ConfigEventData { 130 public: 131 PrioConfigEventData(pid_t pid, pid_t tid, int32_t prio, bool forApp) : 132 mPid(pid), mTid(tid), mPrio(prio), mForApp(forApp) {} 133 134 virtual void dump(char *buffer, size_t size) { 135 snprintf(buffer, size, "Prio event: pid %d, tid %d, prio %d, for app? %d\n", 136 mPid, mTid, mPrio, mForApp); 137 } 138 139 const pid_t mPid; 140 const pid_t mTid; 141 const int32_t mPrio; 142 const bool mForApp; 143 }; 144 145 class PrioConfigEvent : public ConfigEvent { 146 public: 147 PrioConfigEvent(pid_t pid, pid_t tid, int32_t prio, bool forApp) : 148 ConfigEvent(CFG_EVENT_PRIO, true) { 149 mData = new PrioConfigEventData(pid, tid, prio, forApp); 150 } 151 virtual ~PrioConfigEvent() {} 152 }; 153 154 class SetParameterConfigEventData : public ConfigEventData { 155 public: 156 explicit SetParameterConfigEventData(String8 keyValuePairs) : 157 mKeyValuePairs(keyValuePairs) {} 158 159 virtual void dump(char *buffer, size_t size) { 160 snprintf(buffer, size, "KeyValue: %s\n", mKeyValuePairs.string()); 161 } 162 163 const String8 mKeyValuePairs; 164 }; 165 166 class SetParameterConfigEvent : public ConfigEvent { 167 public: 168 explicit SetParameterConfigEvent(String8 keyValuePairs) : 169 ConfigEvent(CFG_EVENT_SET_PARAMETER) { 170 mData = new SetParameterConfigEventData(keyValuePairs); 171 mWaitStatus = true; 172 } 173 virtual ~SetParameterConfigEvent() {} 174 }; 175 176 class CreateAudioPatchConfigEventData : public ConfigEventData { 177 public: 178 CreateAudioPatchConfigEventData(const struct audio_patch patch, 179 audio_patch_handle_t handle) : 180 mPatch(patch), mHandle(handle) {} 181 182 virtual void dump(char *buffer, size_t size) { 183 snprintf(buffer, size, "Patch handle: %u\n", mHandle); 184 } 185 186 const struct audio_patch mPatch; 187 audio_patch_handle_t mHandle; 188 }; 189 190 class CreateAudioPatchConfigEvent : public ConfigEvent { 191 public: 192 CreateAudioPatchConfigEvent(const struct audio_patch patch, 193 audio_patch_handle_t handle) : 194 ConfigEvent(CFG_EVENT_CREATE_AUDIO_PATCH) { 195 mData = new CreateAudioPatchConfigEventData(patch, handle); 196 mWaitStatus = true; 197 } 198 virtual ~CreateAudioPatchConfigEvent() {} 199 }; 200 201 class ReleaseAudioPatchConfigEventData : public ConfigEventData { 202 public: 203 explicit ReleaseAudioPatchConfigEventData(const audio_patch_handle_t handle) : 204 mHandle(handle) {} 205 206 virtual void dump(char *buffer, size_t size) { 207 snprintf(buffer, size, "Patch handle: %u\n", mHandle); 208 } 209 210 audio_patch_handle_t mHandle; 211 }; 212 213 class ReleaseAudioPatchConfigEvent : public ConfigEvent { 214 public: 215 explicit ReleaseAudioPatchConfigEvent(const audio_patch_handle_t handle) : 216 ConfigEvent(CFG_EVENT_RELEASE_AUDIO_PATCH) { 217 mData = new ReleaseAudioPatchConfigEventData(handle); 218 mWaitStatus = true; 219 } 220 virtual ~ReleaseAudioPatchConfigEvent() {} 221 }; 222 223 class PMDeathRecipient : public IBinder::DeathRecipient { 224 public: 225 explicit PMDeathRecipient(const wp<ThreadBase>& thread) : mThread(thread) {} 226 virtual ~PMDeathRecipient() {} 227 228 // IBinder::DeathRecipient 229 virtual void binderDied(const wp<IBinder>& who); 230 231 private: 232 DISALLOW_COPY_AND_ASSIGN(PMDeathRecipient); 233 234 wp<ThreadBase> mThread; 235 }; 236 237 virtual status_t initCheck() const = 0; 238 239 // static externally-visible 240 type_t type() const { return mType; } 241 bool isDuplicating() const { return (mType == DUPLICATING); } 242 243 audio_io_handle_t id() const { return mId;} 244 245 // dynamic externally-visible 246 uint32_t sampleRate() const { return mSampleRate; } 247 audio_channel_mask_t channelMask() const { return mChannelMask; } 248 audio_format_t format() const { return mHALFormat; } 249 uint32_t channelCount() const { return mChannelCount; } 250 // Called by AudioFlinger::frameCount(audio_io_handle_t output) and effects, 251 // and returns the [normal mix] buffer's frame count. 252 virtual size_t frameCount() const = 0; 253 254 // Return's the HAL's frame count i.e. fast mixer buffer size. 255 size_t frameCountHAL() const { return mFrameCount; } 256 257 size_t frameSize() const { return mFrameSize; } 258 259 // Should be "virtual status_t requestExitAndWait()" and override same 260 // method in Thread, but Thread::requestExitAndWait() is not yet virtual. 261 void exit(); 262 virtual bool checkForNewParameter_l(const String8& keyValuePair, 263 status_t& status) = 0; 264 virtual status_t setParameters(const String8& keyValuePairs); 265 virtual String8 getParameters(const String8& keys) = 0; 266 virtual void ioConfigChanged(audio_io_config_event event, pid_t pid = 0) = 0; 267 // sendConfigEvent_l() must be called with ThreadBase::mLock held 268 // Can temporarily release the lock if waiting for a reply from 269 // processConfigEvents_l(). 270 status_t sendConfigEvent_l(sp<ConfigEvent>& event); 271 void sendIoConfigEvent(audio_io_config_event event, pid_t pid = 0); 272 void sendIoConfigEvent_l(audio_io_config_event event, pid_t pid = 0); 273 void sendPrioConfigEvent(pid_t pid, pid_t tid, int32_t prio, bool forApp); 274 void sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio, bool forApp); 275 status_t sendSetParameterConfigEvent_l(const String8& keyValuePair); 276 status_t sendCreateAudioPatchConfigEvent(const struct audio_patch *patch, 277 audio_patch_handle_t *handle); 278 status_t sendReleaseAudioPatchConfigEvent(audio_patch_handle_t handle); 279 void processConfigEvents_l(); 280 virtual void cacheParameters_l() = 0; 281 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 282 audio_patch_handle_t *handle) = 0; 283 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle) = 0; 284 virtual void getAudioPortConfig(struct audio_port_config *config) = 0; 285 286 287 // see note at declaration of mStandby, mOutDevice and mInDevice 288 bool standby() const { return mStandby; } 289 audio_devices_t outDevice() const { return mOutDevice; } 290 audio_devices_t inDevice() const { return mInDevice; } 291 audio_devices_t getDevice() const { return isOutput() ? mOutDevice : mInDevice; } 292 293 virtual bool isOutput() const = 0; 294 295 virtual sp<StreamHalInterface> stream() const = 0; 296 297 sp<EffectHandle> createEffect_l( 298 const sp<AudioFlinger::Client>& client, 299 const sp<IEffectClient>& effectClient, 300 int32_t priority, 301 audio_session_t sessionId, 302 effect_descriptor_t *desc, 303 int *enabled, 304 status_t *status /*non-NULL*/, 305 bool pinned); 306 307 // return values for hasAudioSession (bit field) 308 enum effect_state { 309 EFFECT_SESSION = 0x1, // the audio session corresponds to at least one 310 // effect 311 TRACK_SESSION = 0x2, // the audio session corresponds to at least one 312 // track 313 FAST_SESSION = 0x4 // the audio session corresponds to at least one 314 // fast track 315 }; 316 317 // get effect chain corresponding to session Id. 318 sp<EffectChain> getEffectChain(audio_session_t sessionId); 319 // same as getEffectChain() but must be called with ThreadBase mutex locked 320 sp<EffectChain> getEffectChain_l(audio_session_t sessionId) const; 321 // add an effect chain to the chain list (mEffectChains) 322 virtual status_t addEffectChain_l(const sp<EffectChain>& chain) = 0; 323 // remove an effect chain from the chain list (mEffectChains) 324 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain) = 0; 325 // lock all effect chains Mutexes. Must be called before releasing the 326 // ThreadBase mutex before processing the mixer and effects. This guarantees the 327 // integrity of the chains during the process. 328 // Also sets the parameter 'effectChains' to current value of mEffectChains. 329 void lockEffectChains_l(Vector< sp<EffectChain> >& effectChains); 330 // unlock effect chains after process 331 void unlockEffectChains(const Vector< sp<EffectChain> >& effectChains); 332 // get a copy of mEffectChains vector 333 Vector< sp<EffectChain> > getEffectChains_l() const { return mEffectChains; }; 334 // set audio mode to all effect chains 335 void setMode(audio_mode_t mode); 336 // get effect module with corresponding ID on specified audio session 337 sp<AudioFlinger::EffectModule> getEffect(audio_session_t sessionId, int effectId); 338 sp<AudioFlinger::EffectModule> getEffect_l(audio_session_t sessionId, int effectId); 339 // add and effect module. Also creates the effect chain is none exists for 340 // the effects audio session 341 status_t addEffect_l(const sp< EffectModule>& effect); 342 // remove and effect module. Also removes the effect chain is this was the last 343 // effect 344 void removeEffect_l(const sp< EffectModule>& effect, bool release = false); 345 // disconnect an effect handle from module and destroy module if last handle 346 void disconnectEffectHandle(EffectHandle *handle, bool unpinIfLast); 347 // detach all tracks connected to an auxiliary effect 348 virtual void detachAuxEffect_l(int effectId __unused) {} 349 // returns a combination of: 350 // - EFFECT_SESSION if effects on this audio session exist in one chain 351 // - TRACK_SESSION if tracks on this audio session exist 352 // - FAST_SESSION if fast tracks on this audio session exist 353 virtual uint32_t hasAudioSession_l(audio_session_t sessionId) const = 0; 354 uint32_t hasAudioSession(audio_session_t sessionId) const { 355 Mutex::Autolock _l(mLock); 356 return hasAudioSession_l(sessionId); 357 } 358 359 // the value returned by default implementation is not important as the 360 // strategy is only meaningful for PlaybackThread which implements this method 361 virtual uint32_t getStrategyForSession_l(audio_session_t sessionId __unused) 362 { return 0; } 363 364 // check if some effects must be suspended/restored when an effect is enabled 365 // or disabled 366 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 367 bool enabled, 368 audio_session_t sessionId = 369 AUDIO_SESSION_OUTPUT_MIX); 370 void checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 371 bool enabled, 372 audio_session_t sessionId = 373 AUDIO_SESSION_OUTPUT_MIX); 374 375 virtual status_t setSyncEvent(const sp<SyncEvent>& event) = 0; 376 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const = 0; 377 378 // Return a reference to a per-thread heap which can be used to allocate IMemory 379 // objects that will be read-only to client processes, read/write to mediaserver, 380 // and shared by all client processes of the thread. 381 // The heap is per-thread rather than common across all threads, because 382 // clients can't be trusted not to modify the offset of the IMemory they receive. 383 // If a thread does not have such a heap, this method returns 0. 384 virtual sp<MemoryDealer> readOnlyHeap() const { return 0; } 385 386 virtual sp<IMemory> pipeMemory() const { return 0; } 387 388 void systemReady(); 389 390 // checkEffectCompatibility_l() must be called with ThreadBase::mLock held 391 virtual status_t checkEffectCompatibility_l(const effect_descriptor_t *desc, 392 audio_session_t sessionId) = 0; 393 394 void broadcast_l(); 395 396 mutable Mutex mLock; 397 398 protected: 399 400 // entry describing an effect being suspended in mSuspendedSessions keyed vector 401 class SuspendedSessionDesc : public RefBase { 402 public: 403 SuspendedSessionDesc() : mRefCount(0) {} 404 405 int mRefCount; // number of active suspend requests 406 effect_uuid_t mType; // effect type UUID 407 }; 408 409 void acquireWakeLock(); 410 virtual void acquireWakeLock_l(); 411 void releaseWakeLock(); 412 void releaseWakeLock_l(); 413 void updateWakeLockUids_l(const SortedVector<uid_t> &uids); 414 void getPowerManager_l(); 415 // suspend or restore effects of the specified type (or all if type is NULL) 416 // on a given session. The number of suspend requests is counted and restore 417 // occurs when all suspend requests are cancelled. 418 void setEffectSuspended_l(const effect_uuid_t *type, 419 bool suspend, 420 audio_session_t sessionId); 421 // updated mSuspendedSessions when an effect is suspended or restored 422 void updateSuspendedSessions_l(const effect_uuid_t *type, 423 bool suspend, 424 audio_session_t sessionId); 425 // check if some effects must be suspended when an effect chain is added 426 void checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain); 427 428 // sends the metadata of the active tracks to the HAL 429 virtual void updateMetadata_l() = 0; 430 431 String16 getWakeLockTag(); 432 433 virtual void preExit() { } 434 virtual void setMasterMono_l(bool mono __unused) { } 435 virtual bool requireMonoBlend() { return false; } 436 437 friend class AudioFlinger; // for mEffectChains 438 439 const type_t mType; 440 441 // Used by parameters, config events, addTrack_l, exit 442 Condition mWaitWorkCV; 443 444 const sp<AudioFlinger> mAudioFlinger; 445 446 // updated by PlaybackThread::readOutputParameters_l() or 447 // RecordThread::readInputParameters_l() 448 uint32_t mSampleRate; 449 size_t mFrameCount; // output HAL, direct output, record 450 audio_channel_mask_t mChannelMask; 451 uint32_t mChannelCount; 452 size_t mFrameSize; 453 // not HAL frame size, this is for output sink (to pipe to fast mixer) 454 audio_format_t mFormat; // Source format for Recording and 455 // Sink format for Playback. 456 // Sink format may be different than 457 // HAL format if Fastmixer is used. 458 audio_format_t mHALFormat; 459 size_t mBufferSize; // HAL buffer size for read() or write() 460 461 Vector< sp<ConfigEvent> > mConfigEvents; 462 Vector< sp<ConfigEvent> > mPendingConfigEvents; // events awaiting system ready 463 464 // These fields are written and read by thread itself without lock or barrier, 465 // and read by other threads without lock or barrier via standby(), outDevice() 466 // and inDevice(). 467 // Because of the absence of a lock or barrier, any other thread that reads 468 // these fields must use the information in isolation, or be prepared to deal 469 // with possibility that it might be inconsistent with other information. 470 bool mStandby; // Whether thread is currently in standby. 471 audio_devices_t mOutDevice; // output device 472 audio_devices_t mInDevice; // input device 473 audio_devices_t mPrevOutDevice; // previous output device 474 audio_devices_t mPrevInDevice; // previous input device 475 struct audio_patch mPatch; 476 audio_source_t mAudioSource; 477 478 const audio_io_handle_t mId; 479 Vector< sp<EffectChain> > mEffectChains; 480 481 static const int kThreadNameLength = 16; // prctl(PR_SET_NAME) limit 482 char mThreadName[kThreadNameLength]; // guaranteed NUL-terminated 483 sp<IPowerManager> mPowerManager; 484 sp<IBinder> mWakeLockToken; 485 const sp<PMDeathRecipient> mDeathRecipient; 486 // list of suspended effects per session and per type. The first (outer) vector is 487 // keyed by session ID, the second (inner) by type UUID timeLow field 488 // Updated by updateSuspendedSessions_l() only. 489 KeyedVector< audio_session_t, KeyedVector< int, sp<SuspendedSessionDesc> > > 490 mSuspendedSessions; 491 // TODO: add comment and adjust size as needed 492 static const size_t kLogSize = 4 * 1024; 493 sp<NBLog::Writer> mNBLogWriter; 494 bool mSystemReady; 495 ExtendedTimestamp mTimestamp; 496 // A condition that must be evaluated by the thread loop has changed and 497 // we must not wait for async write callback in the thread loop before evaluating it 498 bool mSignalPending; 499 500 // ActiveTracks is a sorted vector of track type T representing the 501 // active tracks of threadLoop() to be considered by the locked prepare portion. 502 // ActiveTracks should be accessed with the ThreadBase lock held. 503 // 504 // During processing and I/O, the threadLoop does not hold the lock; 505 // hence it does not directly use ActiveTracks. Care should be taken 506 // to hold local strong references or defer removal of tracks 507 // if the threadLoop may still be accessing those tracks due to mix, etc. 508 // 509 // This class updates power information appropriately. 510 // 511 512 template <typename T> 513 class ActiveTracks { 514 public: 515 explicit ActiveTracks(SimpleLog *localLog = nullptr) 516 : mActiveTracksGeneration(0) 517 , mLastActiveTracksGeneration(0) 518 , mLocalLog(localLog) 519 { } 520 521 ~ActiveTracks() { 522 ALOGW_IF(!mActiveTracks.isEmpty(), 523 "ActiveTracks should be empty in destructor"); 524 } 525 // returns the last track added (even though it may have been 526 // subsequently removed from ActiveTracks). 527 // 528 // Used for DirectOutputThread to ensure a flush is called when transitioning 529 // to a new track (even though it may be on the same session). 530 // Used for OffloadThread to ensure that volume and mixer state is 531 // taken from the latest track added. 532 // 533 // The latest track is saved with a weak pointer to prevent keeping an 534 // otherwise useless track alive. Thus the function will return nullptr 535 // if the latest track has subsequently been removed and destroyed. 536 sp<T> getLatest() { 537 return mLatestActiveTrack.promote(); 538 } 539 540 // SortedVector methods 541 ssize_t add(const sp<T> &track); 542 ssize_t remove(const sp<T> &track); 543 size_t size() const { 544 return mActiveTracks.size(); 545 } 546 ssize_t indexOf(const sp<T>& item) { 547 return mActiveTracks.indexOf(item); 548 } 549 sp<T> operator[](size_t index) const { 550 return mActiveTracks[index]; 551 } 552 typename SortedVector<sp<T>>::iterator begin() { 553 return mActiveTracks.begin(); 554 } 555 typename SortedVector<sp<T>>::iterator end() { 556 return mActiveTracks.end(); 557 } 558 559 // Due to Binder recursion optimization, clear() and updatePowerState() 560 // cannot be called from a Binder thread because they may call back into 561 // the original calling process (system server) for BatteryNotifier 562 // (which requires a Java environment that may not be present). 563 // Hence, call clear() and updatePowerState() only from the 564 // ThreadBase thread. 565 void clear(); 566 // periodically called in the threadLoop() to update power state uids. 567 void updatePowerState(sp<ThreadBase> thread, bool force = false); 568 569 /** @return true if one or move active tracks was added or removed since the 570 * last time this function was called or the vector was created. */ 571 bool readAndClearHasChanged(); 572 573 private: 574 void logTrack(const char *funcName, const sp<T> &track) const; 575 576 SortedVector<uid_t> getWakeLockUids() { 577 SortedVector<uid_t> wakeLockUids; 578 for (const sp<T> &track : mActiveTracks) { 579 wakeLockUids.add(track->uid()); 580 } 581 return wakeLockUids; // moved by underlying SharedBuffer 582 } 583 584 std::map<uid_t, std::pair<ssize_t /* previous */, ssize_t /* current */>> 585 mBatteryCounter; 586 SortedVector<sp<T>> mActiveTracks; 587 int mActiveTracksGeneration; 588 int mLastActiveTracksGeneration; 589 wp<T> mLatestActiveTrack; // latest track added to ActiveTracks 590 SimpleLog * const mLocalLog; 591 // If the vector has changed since last call to readAndClearHasChanged 592 bool mHasChanged = false; 593 }; 594 595 SimpleLog mLocalLog; 596 }; 597 598 class VolumeInterface { 599 public: 600 601 virtual ~VolumeInterface() {} 602 603 virtual void setMasterVolume(float value) = 0; 604 virtual void setMasterMute(bool muted) = 0; 605 virtual void setStreamVolume(audio_stream_type_t stream, float value) = 0; 606 virtual void setStreamMute(audio_stream_type_t stream, bool muted) = 0; 607 virtual float streamVolume(audio_stream_type_t stream) const = 0; 608 609 }; 610 611 // --- PlaybackThread --- 612 class PlaybackThread : public ThreadBase, public StreamOutHalInterfaceCallback, 613 public VolumeInterface { 614 public: 615 616 #include "PlaybackTracks.h" 617 618 enum mixer_state { 619 MIXER_IDLE, // no active tracks 620 MIXER_TRACKS_ENABLED, // at least one active track, but no track has any data ready 621 MIXER_TRACKS_READY, // at least one active track, and at least one track has data 622 MIXER_DRAIN_TRACK, // drain currently playing track 623 MIXER_DRAIN_ALL, // fully drain the hardware 624 // standby mode does not have an enum value 625 // suspend by audio policy manager is orthogonal to mixer state 626 }; 627 628 // retry count before removing active track in case of underrun on offloaded thread: 629 // we need to make sure that AudioTrack client has enough time to send large buffers 630 //FIXME may be more appropriate if expressed in time units. Need to revise how underrun is 631 // handled for offloaded tracks 632 static const int8_t kMaxTrackRetriesOffload = 20; 633 static const int8_t kMaxTrackStartupRetriesOffload = 100; 634 static const int8_t kMaxTrackStopRetriesOffload = 2; 635 static constexpr uint32_t kMaxTracksPerUid = 40; 636 static constexpr size_t kMaxTracks = 256; 637 638 // Maximum delay (in nanoseconds) for upcoming buffers in suspend mode, otherwise 639 // if delay is greater, the estimated time for timeLoopNextNs is reset. 640 // This allows for catch-up to be done for small delays, while resetting the estimate 641 // for initial conditions or large delays. 642 static const nsecs_t kMaxNextBufferDelayNs = 100000000; 643 644 PlaybackThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 645 audio_io_handle_t id, audio_devices_t device, type_t type, bool systemReady); 646 virtual ~PlaybackThread(); 647 648 void dump(int fd, const Vector<String16>& args); 649 650 // Thread virtuals 651 virtual bool threadLoop(); 652 653 // RefBase 654 virtual void onFirstRef(); 655 656 virtual status_t checkEffectCompatibility_l(const effect_descriptor_t *desc, 657 audio_session_t sessionId); 658 659 protected: 660 // Code snippets that were lifted up out of threadLoop() 661 virtual void threadLoop_mix() = 0; 662 virtual void threadLoop_sleepTime() = 0; 663 virtual ssize_t threadLoop_write(); 664 virtual void threadLoop_drain(); 665 virtual void threadLoop_standby(); 666 virtual void threadLoop_exit(); 667 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 668 669 // prepareTracks_l reads and writes mActiveTracks, and returns 670 // the pending set of tracks to remove via Vector 'tracksToRemove'. The caller 671 // is responsible for clearing or destroying this Vector later on, when it 672 // is safe to do so. That will drop the final ref count and destroy the tracks. 673 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove) = 0; 674 void removeTracks_l(const Vector< sp<Track> >& tracksToRemove); 675 676 // StreamOutHalInterfaceCallback implementation 677 virtual void onWriteReady(); 678 virtual void onDrainReady(); 679 virtual void onError(); 680 681 void resetWriteBlocked(uint32_t sequence); 682 void resetDraining(uint32_t sequence); 683 684 virtual bool waitingAsyncCallback(); 685 virtual bool waitingAsyncCallback_l(); 686 virtual bool shouldStandby_l(); 687 virtual void onAddNewTrack_l(); 688 void onAsyncError(); // error reported by AsyncCallbackThread 689 690 // ThreadBase virtuals 691 virtual void preExit(); 692 693 virtual bool keepWakeLock() const { return true; } 694 virtual void acquireWakeLock_l() { 695 ThreadBase::acquireWakeLock_l(); 696 mActiveTracks.updatePowerState(this, true /* force */); 697 } 698 699 public: 700 701 virtual status_t initCheck() const { return (mOutput == NULL) ? NO_INIT : NO_ERROR; } 702 703 // return estimated latency in milliseconds, as reported by HAL 704 uint32_t latency() const; 705 // same, but lock must already be held 706 uint32_t latency_l() const; 707 708 // VolumeInterface 709 virtual void setMasterVolume(float value); 710 virtual void setMasterMute(bool muted); 711 virtual void setStreamVolume(audio_stream_type_t stream, float value); 712 virtual void setStreamMute(audio_stream_type_t stream, bool muted); 713 virtual float streamVolume(audio_stream_type_t stream) const; 714 715 sp<Track> createTrack_l( 716 const sp<AudioFlinger::Client>& client, 717 audio_stream_type_t streamType, 718 const audio_attributes_t& attr, 719 uint32_t *sampleRate, 720 audio_format_t format, 721 audio_channel_mask_t channelMask, 722 size_t *pFrameCount, 723 size_t *pNotificationFrameCount, 724 uint32_t notificationsPerBuffer, 725 float speed, 726 const sp<IMemory>& sharedBuffer, 727 audio_session_t sessionId, 728 audio_output_flags_t *flags, 729 pid_t tid, 730 uid_t uid, 731 status_t *status /*non-NULL*/, 732 audio_port_handle_t portId); 733 734 AudioStreamOut* getOutput() const; 735 AudioStreamOut* clearOutput(); 736 virtual sp<StreamHalInterface> stream() const; 737 738 // a very large number of suspend() will eventually wraparound, but unlikely 739 void suspend() { (void) android_atomic_inc(&mSuspended); } 740 void restore() 741 { 742 // if restore() is done without suspend(), get back into 743 // range so that the next suspend() will operate correctly 744 if (android_atomic_dec(&mSuspended) <= 0) { 745 android_atomic_release_store(0, &mSuspended); 746 } 747 } 748 bool isSuspended() const 749 { return android_atomic_acquire_load(&mSuspended) > 0; } 750 751 virtual String8 getParameters(const String8& keys); 752 virtual void ioConfigChanged(audio_io_config_event event, pid_t pid = 0); 753 status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames); 754 // Consider also removing and passing an explicit mMainBuffer initialization 755 // parameter to AF::PlaybackThread::Track::Track(). 756 effect_buffer_t *sinkBuffer() const { 757 return reinterpret_cast<effect_buffer_t *>(mSinkBuffer); }; 758 759 virtual void detachAuxEffect_l(int effectId); 760 status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track>& track, 761 int EffectId); 762 status_t attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track>& track, 763 int EffectId); 764 765 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 766 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 767 virtual uint32_t hasAudioSession_l(audio_session_t sessionId) const; 768 virtual uint32_t getStrategyForSession_l(audio_session_t sessionId); 769 770 771 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 772 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 773 774 // called with AudioFlinger lock held 775 bool invalidateTracks_l(audio_stream_type_t streamType); 776 virtual void invalidateTracks(audio_stream_type_t streamType); 777 778 virtual size_t frameCount() const { return mNormalFrameCount; } 779 780 status_t getTimestamp_l(AudioTimestamp& timestamp); 781 782 void addPatchTrack(const sp<PatchTrack>& track); 783 void deletePatchTrack(const sp<PatchTrack>& track); 784 785 virtual void getAudioPortConfig(struct audio_port_config *config); 786 787 // Return the asynchronous signal wait time. 788 virtual int64_t computeWaitTimeNs_l() const { return INT64_MAX; } 789 790 virtual bool isOutput() const override { return true; } 791 792 // returns true if the track is allowed to be added to the thread. 793 virtual bool isTrackAllowed_l( 794 audio_channel_mask_t channelMask __unused, 795 audio_format_t format __unused, 796 audio_session_t sessionId __unused, 797 uid_t uid) const { 798 return trackCountForUid_l(uid) < PlaybackThread::kMaxTracksPerUid 799 && mTracks.size() < PlaybackThread::kMaxTracks; 800 } 801 802 protected: 803 // updated by readOutputParameters_l() 804 size_t mNormalFrameCount; // normal mixer and effects 805 806 bool mThreadThrottle; // throttle the thread processing 807 uint32_t mThreadThrottleTimeMs; // throttle time for MIXER threads 808 uint32_t mThreadThrottleEndMs; // notify once per throttling 809 uint32_t mHalfBufferMs; // half the buffer size in milliseconds 810 811 void* mSinkBuffer; // frame size aligned sink buffer 812 813 // TODO: 814 // Rearrange the buffer info into a struct/class with 815 // clear, copy, construction, destruction methods. 816 // 817 // mSinkBuffer also has associated with it: 818 // 819 // mSinkBufferSize: Sink Buffer Size 820 // mFormat: Sink Buffer Format 821 822 // Mixer Buffer (mMixerBuffer*) 823 // 824 // In the case of floating point or multichannel data, which is not in the 825 // sink format, it is required to accumulate in a higher precision or greater channel count 826 // buffer before downmixing or data conversion to the sink buffer. 827 828 // Set to "true" to enable the Mixer Buffer otherwise mixer output goes to sink buffer. 829 bool mMixerBufferEnabled; 830 831 // Storage, 32 byte aligned (may make this alignment a requirement later). 832 // Due to constraints on mNormalFrameCount, the buffer size is a multiple of 16 frames. 833 void* mMixerBuffer; 834 835 // Size of mMixerBuffer in bytes: mNormalFrameCount * #channels * sampsize. 836 size_t mMixerBufferSize; 837 838 // The audio format of mMixerBuffer. Set to AUDIO_FORMAT_PCM_(FLOAT|16_BIT) only. 839 audio_format_t mMixerBufferFormat; 840 841 // An internal flag set to true by MixerThread::prepareTracks_l() 842 // when mMixerBuffer contains valid data after mixing. 843 bool mMixerBufferValid; 844 845 // Effects Buffer (mEffectsBuffer*) 846 // 847 // In the case of effects data, which is not in the sink format, 848 // it is required to accumulate in a different buffer before data conversion 849 // to the sink buffer. 850 851 // Set to "true" to enable the Effects Buffer otherwise effects output goes to sink buffer. 852 bool mEffectBufferEnabled; 853 854 // Storage, 32 byte aligned (may make this alignment a requirement later). 855 // Due to constraints on mNormalFrameCount, the buffer size is a multiple of 16 frames. 856 void* mEffectBuffer; 857 858 // Size of mEffectsBuffer in bytes: mNormalFrameCount * #channels * sampsize. 859 size_t mEffectBufferSize; 860 861 // The audio format of mEffectsBuffer. Set to AUDIO_FORMAT_PCM_16_BIT only. 862 audio_format_t mEffectBufferFormat; 863 864 // An internal flag set to true by MixerThread::prepareTracks_l() 865 // when mEffectsBuffer contains valid data after mixing. 866 // 867 // When this is set, all mixer data is routed into the effects buffer 868 // for any processing (including output processing). 869 bool mEffectBufferValid; 870 871 // suspend count, > 0 means suspended. While suspended, the thread continues to pull from 872 // tracks and mix, but doesn't write to HAL. A2DP and SCO HAL implementations can't handle 873 // concurrent use of both of them, so Audio Policy Service suspends one of the threads to 874 // workaround that restriction. 875 // 'volatile' means accessed via atomic operations and no lock. 876 volatile int32_t mSuspended; 877 878 int64_t mBytesWritten; 879 int64_t mFramesWritten; // not reset on standby 880 int64_t mSuspendedFrames; // not reset on standby 881 private: 882 // mMasterMute is in both PlaybackThread and in AudioFlinger. When a 883 // PlaybackThread needs to find out if master-muted, it checks it's local 884 // copy rather than the one in AudioFlinger. This optimization saves a lock. 885 bool mMasterMute; 886 void setMasterMute_l(bool muted) { mMasterMute = muted; } 887 protected: 888 ActiveTracks<Track> mActiveTracks; 889 890 // Time to sleep between cycles when: 891 virtual uint32_t activeSleepTimeUs() const; // mixer state MIXER_TRACKS_ENABLED 892 virtual uint32_t idleSleepTimeUs() const = 0; // mixer state MIXER_IDLE 893 virtual uint32_t suspendSleepTimeUs() const = 0; // audio policy manager suspended us 894 // No sleep when mixer state == MIXER_TRACKS_READY; relies on audio HAL stream->write() 895 // No sleep in standby mode; waits on a condition 896 897 // Code snippets that are temporarily lifted up out of threadLoop() until the merge 898 void checkSilentMode_l(); 899 900 // Non-trivial for DUPLICATING only 901 virtual void saveOutputTracks() { } 902 virtual void clearOutputTracks() { } 903 904 // Cache various calculated values, at threadLoop() entry and after a parameter change 905 virtual void cacheParameters_l(); 906 907 virtual uint32_t correctLatency_l(uint32_t latency) const; 908 909 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 910 audio_patch_handle_t *handle); 911 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 912 913 bool usesHwAvSync() const { return (mType == DIRECT) && (mOutput != NULL) 914 && mHwSupportsPause 915 && (mOutput->flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC); } 916 917 uint32_t trackCountForUid_l(uid_t uid) const; 918 919 private: 920 921 friend class AudioFlinger; // for numerous 922 923 DISALLOW_COPY_AND_ASSIGN(PlaybackThread); 924 925 status_t addTrack_l(const sp<Track>& track); 926 bool destroyTrack_l(const sp<Track>& track); 927 void removeTrack_l(const sp<Track>& track); 928 929 void readOutputParameters_l(); 930 void updateMetadata_l() final; 931 virtual void sendMetadataToBackend_l(const StreamOutHalInterface::SourceMetadata& metadata); 932 933 virtual void dumpInternals(int fd, const Vector<String16>& args); 934 void dumpTracks(int fd, const Vector<String16>& args); 935 936 // The Tracks class manages names for all tracks 937 // added and removed from the Thread. 938 template <typename T> 939 class Tracks { 940 public: 941 Tracks(bool saveDeletedTrackNames) : 942 mSaveDeletedTrackNames(saveDeletedTrackNames) { } 943 944 // SortedVector methods 945 ssize_t add(const sp<T> &track); 946 ssize_t remove(const sp<T> &track); 947 size_t size() const { 948 return mTracks.size(); 949 } 950 bool isEmpty() const { 951 return mTracks.isEmpty(); 952 } 953 ssize_t indexOf(const sp<T> &item) { 954 return mTracks.indexOf(item); 955 } 956 sp<T> operator[](size_t index) const { 957 return mTracks[index]; 958 } 959 typename SortedVector<sp<T>>::iterator begin() { 960 return mTracks.begin(); 961 } 962 typename SortedVector<sp<T>>::iterator end() { 963 return mTracks.end(); 964 } 965 966 size_t processDeletedTrackNames(std::function<void(int)> f) { 967 const size_t size = mDeletedTrackNames.size(); 968 if (size > 0) { 969 for (const int name : mDeletedTrackNames) { 970 f(name); 971 } 972 } 973 return size; 974 } 975 976 void clearDeletedTrackNames() { mDeletedTrackNames.clear(); } 977 978 private: 979 // Track names pending deletion for MIXER type threads 980 const bool mSaveDeletedTrackNames; // true to enable tracking 981 std::set<int> mDeletedTrackNames; 982 983 // Fast lookup of previously deleted track names for reuse. 984 // This is an arbitrary decision (actually any non-negative 985 // integer that isn't in mTracks[*]->names() could be used) - we attempt 986 // to use the smallest possible available name. 987 std::set<int> mUnusedTrackNames; 988 989 SortedVector<sp<T>> mTracks; // wrapped SortedVector. 990 }; 991 992 Tracks<Track> mTracks; 993 994 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 995 AudioStreamOut *mOutput; 996 997 float mMasterVolume; 998 nsecs_t mLastWriteTime; 999 int mNumWrites; 1000 int mNumDelayedWrites; 1001 bool mInWrite; 1002 1003 // FIXME rename these former local variables of threadLoop to standard "m" names 1004 nsecs_t mStandbyTimeNs; 1005 size_t mSinkBufferSize; 1006 1007 // cached copies of activeSleepTimeUs() and idleSleepTimeUs() made by cacheParameters_l() 1008 uint32_t mActiveSleepTimeUs; 1009 uint32_t mIdleSleepTimeUs; 1010 1011 uint32_t mSleepTimeUs; 1012 1013 // mixer status returned by prepareTracks_l() 1014 mixer_state mMixerStatus; // current cycle 1015 // previous cycle when in prepareTracks_l() 1016 mixer_state mMixerStatusIgnoringFastTracks; 1017 // FIXME or a separate ready state per track 1018 1019 // FIXME move these declarations into the specific sub-class that needs them 1020 // MIXER only 1021 uint32_t sleepTimeShift; 1022 1023 // same as AudioFlinger::mStandbyTimeInNsecs except for DIRECT which uses a shorter value 1024 nsecs_t mStandbyDelayNs; 1025 1026 // MIXER only 1027 nsecs_t maxPeriod; 1028 1029 // DUPLICATING only 1030 uint32_t writeFrames; 1031 1032 size_t mBytesRemaining; 1033 size_t mCurrentWriteLength; 1034 bool mUseAsyncWrite; 1035 // mWriteAckSequence contains current write sequence on bits 31-1. The write sequence is 1036 // incremented each time a write(), a flush() or a standby() occurs. 1037 // Bit 0 is set when a write blocks and indicates a callback is expected. 1038 // Bit 0 is reset by the async callback thread calling resetWriteBlocked(). Out of sequence 1039 // callbacks are ignored. 1040 uint32_t mWriteAckSequence; 1041 // mDrainSequence contains current drain sequence on bits 31-1. The drain sequence is 1042 // incremented each time a drain is requested or a flush() or standby() occurs. 1043 // Bit 0 is set when the drain() command is called at the HAL and indicates a callback is 1044 // expected. 1045 // Bit 0 is reset by the async callback thread calling resetDraining(). Out of sequence 1046 // callbacks are ignored. 1047 uint32_t mDrainSequence; 1048 sp<AsyncCallbackThread> mCallbackThread; 1049 1050 private: 1051 // The HAL output sink is treated as non-blocking, but current implementation is blocking 1052 sp<NBAIO_Sink> mOutputSink; 1053 // If a fast mixer is present, the blocking pipe sink, otherwise clear 1054 sp<NBAIO_Sink> mPipeSink; 1055 // The current sink for the normal mixer to write it's (sub)mix, mOutputSink or mPipeSink 1056 sp<NBAIO_Sink> mNormalSink; 1057 #ifdef TEE_SINK 1058 // For dumpsys 1059 sp<NBAIO_Sink> mTeeSink; 1060 sp<NBAIO_Source> mTeeSource; 1061 #endif 1062 uint32_t mScreenState; // cached copy of gScreenState 1063 // TODO: add comment and adjust size as needed 1064 static const size_t kFastMixerLogSize = 8 * 1024; 1065 sp<NBLog::Writer> mFastMixerNBLogWriter; 1066 1067 1068 public: 1069 virtual bool hasFastMixer() const = 0; 1070 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex __unused) const 1071 { FastTrackUnderruns dummy; return dummy; } 1072 1073 protected: 1074 // accessed by both binder threads and within threadLoop(), lock on mutex needed 1075 unsigned mFastTrackAvailMask; // bit i set if fast track [i] is available 1076 bool mHwSupportsPause; 1077 bool mHwPaused; 1078 bool mFlushPending; 1079 // volumes last sent to audio HAL with stream->setVolume() 1080 float mLeftVolFloat; 1081 float mRightVolFloat; 1082 }; 1083 1084 class MixerThread : public PlaybackThread { 1085 public: 1086 MixerThread(const sp<AudioFlinger>& audioFlinger, 1087 AudioStreamOut* output, 1088 audio_io_handle_t id, 1089 audio_devices_t device, 1090 bool systemReady, 1091 type_t type = MIXER); 1092 virtual ~MixerThread(); 1093 1094 // Thread virtuals 1095 1096 virtual bool checkForNewParameter_l(const String8& keyValuePair, 1097 status_t& status); 1098 virtual void dumpInternals(int fd, const Vector<String16>& args); 1099 1100 virtual bool isTrackAllowed_l( 1101 audio_channel_mask_t channelMask, audio_format_t format, 1102 audio_session_t sessionId, uid_t uid) const override; 1103 protected: 1104 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1105 virtual uint32_t idleSleepTimeUs() const; 1106 virtual uint32_t suspendSleepTimeUs() const; 1107 virtual void cacheParameters_l(); 1108 1109 virtual void acquireWakeLock_l() { 1110 PlaybackThread::acquireWakeLock_l(); 1111 if (hasFastMixer()) { 1112 mFastMixer->setBoottimeOffset( 1113 mTimestamp.mTimebaseOffset[ExtendedTimestamp::TIMEBASE_BOOTTIME]); 1114 } 1115 } 1116 1117 // threadLoop snippets 1118 virtual ssize_t threadLoop_write(); 1119 virtual void threadLoop_standby(); 1120 virtual void threadLoop_mix(); 1121 virtual void threadLoop_sleepTime(); 1122 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 1123 virtual uint32_t correctLatency_l(uint32_t latency) const; 1124 1125 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 1126 audio_patch_handle_t *handle); 1127 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 1128 1129 AudioMixer* mAudioMixer; // normal mixer 1130 private: 1131 // one-time initialization, no locks required 1132 sp<FastMixer> mFastMixer; // non-0 if there is also a fast mixer 1133 sp<AudioWatchdog> mAudioWatchdog; // non-0 if there is an audio watchdog thread 1134 1135 // contents are not guaranteed to be consistent, no locks required 1136 FastMixerDumpState mFastMixerDumpState; 1137 #ifdef STATE_QUEUE_DUMP 1138 StateQueueObserverDump mStateQueueObserverDump; 1139 StateQueueMutatorDump mStateQueueMutatorDump; 1140 #endif 1141 AudioWatchdogDump mAudioWatchdogDump; 1142 1143 // accessible only within the threadLoop(), no locks required 1144 // mFastMixer->sq() // for mutating and pushing state 1145 int32_t mFastMixerFutex; // for cold idle 1146 1147 std::atomic_bool mMasterMono; 1148 public: 1149 virtual bool hasFastMixer() const { return mFastMixer != 0; } 1150 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const { 1151 ALOG_ASSERT(fastIndex < FastMixerState::sMaxFastTracks); 1152 return mFastMixerDumpState.mTracks[fastIndex].mUnderruns; 1153 } 1154 1155 protected: 1156 virtual void setMasterMono_l(bool mono) { 1157 mMasterMono.store(mono); 1158 if (mFastMixer != nullptr) { /* hasFastMixer() */ 1159 mFastMixer->setMasterMono(mMasterMono); 1160 } 1161 } 1162 // the FastMixer performs mono blend if it exists. 1163 // Blending with limiter is not idempotent, 1164 // and blending without limiter is idempotent but inefficient to do twice. 1165 virtual bool requireMonoBlend() { return mMasterMono.load() && !hasFastMixer(); } 1166 }; 1167 1168 class DirectOutputThread : public PlaybackThread { 1169 public: 1170 1171 DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1172 audio_io_handle_t id, audio_devices_t device, bool systemReady); 1173 virtual ~DirectOutputThread(); 1174 1175 // Thread virtuals 1176 1177 virtual bool checkForNewParameter_l(const String8& keyValuePair, 1178 status_t& status); 1179 virtual void flushHw_l(); 1180 1181 protected: 1182 virtual uint32_t activeSleepTimeUs() const; 1183 virtual uint32_t idleSleepTimeUs() const; 1184 virtual uint32_t suspendSleepTimeUs() const; 1185 virtual void cacheParameters_l(); 1186 1187 // threadLoop snippets 1188 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1189 virtual void threadLoop_mix(); 1190 virtual void threadLoop_sleepTime(); 1191 virtual void threadLoop_exit(); 1192 virtual bool shouldStandby_l(); 1193 1194 virtual void onAddNewTrack_l(); 1195 1196 bool mVolumeShaperActive; 1197 1198 DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1199 audio_io_handle_t id, uint32_t device, ThreadBase::type_t type, 1200 bool systemReady); 1201 void processVolume_l(Track *track, bool lastTrack); 1202 1203 // prepareTracks_l() tells threadLoop_mix() the name of the single active track 1204 sp<Track> mActiveTrack; 1205 1206 wp<Track> mPreviousTrack; // used to detect track switch 1207 1208 public: 1209 virtual bool hasFastMixer() const { return false; } 1210 1211 virtual int64_t computeWaitTimeNs_l() const override; 1212 }; 1213 1214 class OffloadThread : public DirectOutputThread { 1215 public: 1216 1217 OffloadThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1218 audio_io_handle_t id, uint32_t device, bool systemReady); 1219 virtual ~OffloadThread() {}; 1220 virtual void flushHw_l(); 1221 1222 protected: 1223 // threadLoop snippets 1224 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1225 virtual void threadLoop_exit(); 1226 1227 virtual bool waitingAsyncCallback(); 1228 virtual bool waitingAsyncCallback_l(); 1229 virtual void invalidateTracks(audio_stream_type_t streamType); 1230 1231 virtual bool keepWakeLock() const { return (mKeepWakeLock || (mDrainSequence & 1)); } 1232 1233 private: 1234 size_t mPausedWriteLength; // length in bytes of write interrupted by pause 1235 size_t mPausedBytesRemaining; // bytes still waiting in mixbuffer after resume 1236 bool mKeepWakeLock; // keep wake lock while waiting for write callback 1237 uint64_t mOffloadUnderrunPosition; // Current frame position for offloaded playback 1238 // used and valid only during underrun. ~0 if 1239 // no underrun has occurred during playback and 1240 // is not reset on standby. 1241 }; 1242 1243 class AsyncCallbackThread : public Thread { 1244 public: 1245 1246 explicit AsyncCallbackThread(const wp<PlaybackThread>& playbackThread); 1247 1248 virtual ~AsyncCallbackThread(); 1249 1250 // Thread virtuals 1251 virtual bool threadLoop(); 1252 1253 // RefBase 1254 virtual void onFirstRef(); 1255 1256 void exit(); 1257 void setWriteBlocked(uint32_t sequence); 1258 void resetWriteBlocked(); 1259 void setDraining(uint32_t sequence); 1260 void resetDraining(); 1261 void setAsyncError(); 1262 1263 private: 1264 const wp<PlaybackThread> mPlaybackThread; 1265 // mWriteAckSequence corresponds to the last write sequence passed by the offload thread via 1266 // setWriteBlocked(). The sequence is shifted one bit to the left and the lsb is used 1267 // to indicate that the callback has been received via resetWriteBlocked() 1268 uint32_t mWriteAckSequence; 1269 // mDrainSequence corresponds to the last drain sequence passed by the offload thread via 1270 // setDraining(). The sequence is shifted one bit to the left and the lsb is used 1271 // to indicate that the callback has been received via resetDraining() 1272 uint32_t mDrainSequence; 1273 Condition mWaitWorkCV; 1274 Mutex mLock; 1275 bool mAsyncError; 1276 }; 1277 1278 class DuplicatingThread : public MixerThread { 1279 public: 1280 DuplicatingThread(const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread, 1281 audio_io_handle_t id, bool systemReady); 1282 virtual ~DuplicatingThread(); 1283 1284 // Thread virtuals 1285 virtual void dumpInternals(int fd, const Vector<String16>& args) override; 1286 1287 void addOutputTrack(MixerThread* thread); 1288 void removeOutputTrack(MixerThread* thread); 1289 uint32_t waitTimeMs() const { return mWaitTimeMs; } 1290 1291 void sendMetadataToBackend_l( 1292 const StreamOutHalInterface::SourceMetadata& metadata) override; 1293 protected: 1294 virtual uint32_t activeSleepTimeUs() const; 1295 1296 private: 1297 bool outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks); 1298 protected: 1299 // threadLoop snippets 1300 virtual void threadLoop_mix(); 1301 virtual void threadLoop_sleepTime(); 1302 virtual ssize_t threadLoop_write(); 1303 virtual void threadLoop_standby(); 1304 virtual void cacheParameters_l(); 1305 1306 private: 1307 // called from threadLoop, addOutputTrack, removeOutputTrack 1308 virtual void updateWaitTime_l(); 1309 protected: 1310 virtual void saveOutputTracks(); 1311 virtual void clearOutputTracks(); 1312 private: 1313 1314 uint32_t mWaitTimeMs; 1315 SortedVector < sp<OutputTrack> > outputTracks; 1316 SortedVector < sp<OutputTrack> > mOutputTracks; 1317 public: 1318 virtual bool hasFastMixer() const { return false; } 1319 }; 1320 1321 // record thread 1322 class RecordThread : public ThreadBase 1323 { 1324 public: 1325 1326 class RecordTrack; 1327 1328 /* The ResamplerBufferProvider is used to retrieve recorded input data from the 1329 * RecordThread. It maintains local state on the relative position of the read 1330 * position of the RecordTrack compared with the RecordThread. 1331 */ 1332 class ResamplerBufferProvider : public AudioBufferProvider 1333 { 1334 public: 1335 explicit ResamplerBufferProvider(RecordTrack* recordTrack) : 1336 mRecordTrack(recordTrack), 1337 mRsmpInUnrel(0), mRsmpInFront(0) { } 1338 virtual ~ResamplerBufferProvider() { } 1339 1340 // called to set the ResamplerBufferProvider to head of the RecordThread data buffer, 1341 // skipping any previous data read from the hal. 1342 virtual void reset(); 1343 1344 /* Synchronizes RecordTrack position with the RecordThread. 1345 * Calculates available frames and handle overruns if the RecordThread 1346 * has advanced faster than the ResamplerBufferProvider has retrieved data. 1347 * TODO: why not do this for every getNextBuffer? 1348 * 1349 * Parameters 1350 * framesAvailable: pointer to optional output size_t to store record track 1351 * frames available. 1352 * hasOverrun: pointer to optional boolean, returns true if track has overrun. 1353 */ 1354 1355 virtual void sync(size_t *framesAvailable = NULL, bool *hasOverrun = NULL); 1356 1357 // AudioBufferProvider interface 1358 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer); 1359 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 1360 private: 1361 RecordTrack * const mRecordTrack; 1362 size_t mRsmpInUnrel; // unreleased frames remaining from 1363 // most recent getNextBuffer 1364 // for debug only 1365 int32_t mRsmpInFront; // next available frame 1366 // rolling counter that is never cleared 1367 }; 1368 1369 #include "RecordTracks.h" 1370 1371 RecordThread(const sp<AudioFlinger>& audioFlinger, 1372 AudioStreamIn *input, 1373 audio_io_handle_t id, 1374 audio_devices_t outDevice, 1375 audio_devices_t inDevice, 1376 bool systemReady 1377 #ifdef TEE_SINK 1378 , const sp<NBAIO_Sink>& teeSink 1379 #endif 1380 ); 1381 virtual ~RecordThread(); 1382 1383 // no addTrack_l ? 1384 void destroyTrack_l(const sp<RecordTrack>& track); 1385 void removeTrack_l(const sp<RecordTrack>& track); 1386 1387 void dumpInternals(int fd, const Vector<String16>& args); 1388 void dumpTracks(int fd, const Vector<String16>& args); 1389 1390 // Thread virtuals 1391 virtual bool threadLoop(); 1392 virtual void preExit(); 1393 1394 // RefBase 1395 virtual void onFirstRef(); 1396 1397 virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; } 1398 1399 virtual sp<MemoryDealer> readOnlyHeap() const { return mReadOnlyHeap; } 1400 1401 virtual sp<IMemory> pipeMemory() const { return mPipeMemory; } 1402 1403 sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l( 1404 const sp<AudioFlinger::Client>& client, 1405 const audio_attributes_t& attr, 1406 uint32_t *pSampleRate, 1407 audio_format_t format, 1408 audio_channel_mask_t channelMask, 1409 size_t *pFrameCount, 1410 audio_session_t sessionId, 1411 size_t *pNotificationFrameCount, 1412 uid_t uid, 1413 audio_input_flags_t *flags, 1414 pid_t tid, 1415 status_t *status /*non-NULL*/, 1416 audio_port_handle_t portId); 1417 1418 status_t start(RecordTrack* recordTrack, 1419 AudioSystem::sync_event_t event, 1420 audio_session_t triggerSession); 1421 1422 // ask the thread to stop the specified track, and 1423 // return true if the caller should then do it's part of the stopping process 1424 bool stop(RecordTrack* recordTrack); 1425 1426 void dump(int fd, const Vector<String16>& args); 1427 AudioStreamIn* clearInput(); 1428 virtual sp<StreamHalInterface> stream() const; 1429 1430 1431 virtual bool checkForNewParameter_l(const String8& keyValuePair, 1432 status_t& status); 1433 virtual void cacheParameters_l() {} 1434 virtual String8 getParameters(const String8& keys); 1435 virtual void ioConfigChanged(audio_io_config_event event, pid_t pid = 0); 1436 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 1437 audio_patch_handle_t *handle); 1438 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 1439 1440 void addPatchRecord(const sp<PatchRecord>& record); 1441 void deletePatchRecord(const sp<PatchRecord>& record); 1442 1443 void readInputParameters_l(); 1444 virtual uint32_t getInputFramesLost(); 1445 1446 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1447 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1448 virtual uint32_t hasAudioSession_l(audio_session_t sessionId) const; 1449 1450 // Return the set of unique session IDs across all tracks. 1451 // The keys are the session IDs, and the associated values are meaningless. 1452 // FIXME replace by Set [and implement Bag/Multiset for other uses]. 1453 KeyedVector<audio_session_t, bool> sessionIds() const; 1454 1455 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 1456 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 1457 1458 static void syncStartEventCallback(const wp<SyncEvent>& event); 1459 1460 virtual size_t frameCount() const { return mFrameCount; } 1461 bool hasFastCapture() const { return mFastCapture != 0; } 1462 virtual void getAudioPortConfig(struct audio_port_config *config); 1463 1464 virtual status_t checkEffectCompatibility_l(const effect_descriptor_t *desc, 1465 audio_session_t sessionId); 1466 1467 virtual void acquireWakeLock_l() { 1468 ThreadBase::acquireWakeLock_l(); 1469 mActiveTracks.updatePowerState(this, true /* force */); 1470 } 1471 virtual bool isOutput() const override { return false; } 1472 1473 void checkBtNrec(); 1474 1475 // Sets the UID records silence 1476 void setRecordSilenced(uid_t uid, bool silenced); 1477 1478 status_t getActiveMicrophones(std::vector<media::MicrophoneInfo>* activeMicrophones); 1479 1480 void updateMetadata_l() override; 1481 1482 private: 1483 // Enter standby if not already in standby, and set mStandby flag 1484 void standbyIfNotAlreadyInStandby(); 1485 1486 // Call the HAL standby method unconditionally, and don't change mStandby flag 1487 void inputStandBy(); 1488 1489 void checkBtNrec_l(); 1490 1491 AudioStreamIn *mInput; 1492 SortedVector < sp<RecordTrack> > mTracks; 1493 // mActiveTracks has dual roles: it indicates the current active track(s), and 1494 // is used together with mStartStopCond to indicate start()/stop() progress 1495 ActiveTracks<RecordTrack> mActiveTracks; 1496 1497 Condition mStartStopCond; 1498 1499 // resampler converts input at HAL Hz to output at AudioRecord client Hz 1500 void *mRsmpInBuffer; // size = mRsmpInFramesOA 1501 size_t mRsmpInFrames; // size of resampler input in frames 1502 size_t mRsmpInFramesP2;// size rounded up to a power-of-2 1503 size_t mRsmpInFramesOA;// mRsmpInFramesP2 + over-allocation 1504 1505 // rolling index that is never cleared 1506 int32_t mRsmpInRear; // last filled frame + 1 1507 1508 // For dumpsys 1509 const sp<NBAIO_Sink> mTeeSink; 1510 1511 const sp<MemoryDealer> mReadOnlyHeap; 1512 1513 // one-time initialization, no locks required 1514 sp<FastCapture> mFastCapture; // non-0 if there is also 1515 // a fast capture 1516 1517 // FIXME audio watchdog thread 1518 1519 // contents are not guaranteed to be consistent, no locks required 1520 FastCaptureDumpState mFastCaptureDumpState; 1521 #ifdef STATE_QUEUE_DUMP 1522 // FIXME StateQueue observer and mutator dump fields 1523 #endif 1524 // FIXME audio watchdog dump 1525 1526 // accessible only within the threadLoop(), no locks required 1527 // mFastCapture->sq() // for mutating and pushing state 1528 int32_t mFastCaptureFutex; // for cold idle 1529 1530 // The HAL input source is treated as non-blocking, 1531 // but current implementation is blocking 1532 sp<NBAIO_Source> mInputSource; 1533 // The source for the normal capture thread to read from: mInputSource or mPipeSource 1534 sp<NBAIO_Source> mNormalSource; 1535 // If a fast capture is present, the non-blocking pipe sink written to by fast capture, 1536 // otherwise clear 1537 sp<NBAIO_Sink> mPipeSink; 1538 // If a fast capture is present, the non-blocking pipe source read by normal thread, 1539 // otherwise clear 1540 sp<NBAIO_Source> mPipeSource; 1541 // Depth of pipe from fast capture to normal thread and fast clients, always power of 2 1542 size_t mPipeFramesP2; 1543 // If a fast capture is present, the Pipe as IMemory, otherwise clear 1544 sp<IMemory> mPipeMemory; 1545 1546 // TODO: add comment and adjust size as needed 1547 static const size_t kFastCaptureLogSize = 4 * 1024; 1548 sp<NBLog::Writer> mFastCaptureNBLogWriter; 1549 1550 bool mFastTrackAvail; // true if fast track available 1551 // common state to all record threads 1552 std::atomic_bool mBtNrecSuspended; 1553 }; 1554 1555 class MmapThread : public ThreadBase 1556 { 1557 public: 1558 1559 #include "MmapTracks.h" 1560 1561 MmapThread(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1562 AudioHwDevice *hwDev, sp<StreamHalInterface> stream, 1563 audio_devices_t outDevice, audio_devices_t inDevice, bool systemReady); 1564 virtual ~MmapThread(); 1565 1566 virtual void configure(const audio_attributes_t *attr, 1567 audio_stream_type_t streamType, 1568 audio_session_t sessionId, 1569 const sp<MmapStreamCallback>& callback, 1570 audio_port_handle_t deviceId, 1571 audio_port_handle_t portId); 1572 1573 void disconnect(); 1574 1575 // MmapStreamInterface 1576 status_t createMmapBuffer(int32_t minSizeFrames, 1577 struct audio_mmap_buffer_info *info); 1578 status_t getMmapPosition(struct audio_mmap_position *position); 1579 status_t start(const AudioClient& client, audio_port_handle_t *handle); 1580 status_t stop(audio_port_handle_t handle); 1581 status_t standby(); 1582 1583 // RefBase 1584 virtual void onFirstRef(); 1585 1586 // Thread virtuals 1587 virtual bool threadLoop(); 1588 1589 virtual void threadLoop_exit(); 1590 virtual void threadLoop_standby(); 1591 virtual bool shouldStandby_l() { return false; } 1592 virtual status_t exitStandby(); 1593 1594 virtual status_t initCheck() const { return (mHalStream == 0) ? NO_INIT : NO_ERROR; } 1595 virtual size_t frameCount() const { return mFrameCount; } 1596 virtual bool checkForNewParameter_l(const String8& keyValuePair, 1597 status_t& status); 1598 virtual String8 getParameters(const String8& keys); 1599 virtual void ioConfigChanged(audio_io_config_event event, pid_t pid = 0); 1600 void readHalParameters_l(); 1601 virtual void cacheParameters_l() {} 1602 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 1603 audio_patch_handle_t *handle); 1604 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 1605 virtual void getAudioPortConfig(struct audio_port_config *config); 1606 1607 virtual sp<StreamHalInterface> stream() const { return mHalStream; } 1608 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1609 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1610 virtual status_t checkEffectCompatibility_l(const effect_descriptor_t *desc, 1611 audio_session_t sessionId); 1612 1613 virtual uint32_t hasAudioSession_l(audio_session_t sessionId) const; 1614 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 1615 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 1616 1617 virtual void checkSilentMode_l() {} 1618 virtual void processVolume_l() {} 1619 void checkInvalidTracks_l(); 1620 1621 virtual audio_stream_type_t streamType() { return AUDIO_STREAM_DEFAULT; } 1622 1623 virtual void invalidateTracks(audio_stream_type_t streamType __unused) {} 1624 1625 // Sets the UID records silence 1626 virtual void setRecordSilenced(uid_t uid __unused, bool silenced __unused) {} 1627 1628 void dump(int fd, const Vector<String16>& args); 1629 virtual void dumpInternals(int fd, const Vector<String16>& args); 1630 void dumpTracks(int fd, const Vector<String16>& args); 1631 1632 protected: 1633 1634 audio_attributes_t mAttr; 1635 audio_session_t mSessionId; 1636 audio_port_handle_t mDeviceId; 1637 audio_port_handle_t mPortId; 1638 1639 wp<MmapStreamCallback> mCallback; 1640 sp<StreamHalInterface> mHalStream; 1641 sp<DeviceHalInterface> mHalDevice; 1642 AudioHwDevice* const mAudioHwDev; 1643 ActiveTracks<MmapTrack> mActiveTracks; 1644 float mHalVolFloat; 1645 1646 int32_t mNoCallbackWarningCount; 1647 static constexpr int32_t kMaxNoCallbackWarnings = 5; 1648 }; 1649 1650 class MmapPlaybackThread : public MmapThread, public VolumeInterface 1651 { 1652 1653 public: 1654 MmapPlaybackThread(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1655 AudioHwDevice *hwDev, AudioStreamOut *output, 1656 audio_devices_t outDevice, audio_devices_t inDevice, bool systemReady); 1657 virtual ~MmapPlaybackThread() {} 1658 1659 virtual void configure(const audio_attributes_t *attr, 1660 audio_stream_type_t streamType, 1661 audio_session_t sessionId, 1662 const sp<MmapStreamCallback>& callback, 1663 audio_port_handle_t deviceId, 1664 audio_port_handle_t portId); 1665 1666 AudioStreamOut* clearOutput(); 1667 1668 // VolumeInterface 1669 virtual void setMasterVolume(float value); 1670 virtual void setMasterMute(bool muted); 1671 virtual void setStreamVolume(audio_stream_type_t stream, float value); 1672 virtual void setStreamMute(audio_stream_type_t stream, bool muted); 1673 virtual float streamVolume(audio_stream_type_t stream) const; 1674 1675 void setMasterMute_l(bool muted) { mMasterMute = muted; } 1676 1677 virtual void invalidateTracks(audio_stream_type_t streamType); 1678 1679 virtual audio_stream_type_t streamType() { return mStreamType; } 1680 virtual void checkSilentMode_l(); 1681 void processVolume_l() override; 1682 1683 virtual void dumpInternals(int fd, const Vector<String16>& args); 1684 1685 virtual bool isOutput() const override { return true; } 1686 1687 void updateMetadata_l() override; 1688 1689 protected: 1690 1691 audio_stream_type_t mStreamType; 1692 float mMasterVolume; 1693 float mStreamVolume; 1694 bool mMasterMute; 1695 bool mStreamMute; 1696 AudioStreamOut* mOutput; 1697 }; 1698 1699 class MmapCaptureThread : public MmapThread 1700 { 1701 1702 public: 1703 MmapCaptureThread(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1704 AudioHwDevice *hwDev, AudioStreamIn *input, 1705 audio_devices_t outDevice, audio_devices_t inDevice, bool systemReady); 1706 virtual ~MmapCaptureThread() {} 1707 1708 AudioStreamIn* clearInput(); 1709 1710 status_t exitStandby() override; 1711 virtual bool isOutput() const override { return false; } 1712 1713 void updateMetadata_l() override; 1714 void processVolume_l() override; 1715 void setRecordSilenced(uid_t uid, bool silenced) override; 1716 1717 protected: 1718 1719 AudioStreamIn* mInput; 1720 }; 1721
