Threads.h revision dae27707fc7d8370eb200d25d1a7c6dd7ad5e201
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 bool pinned); 300 301 // return values for hasAudioSession (bit field) 302 enum effect_state { 303 EFFECT_SESSION = 0x1, // the audio session corresponds to at least one 304 // effect 305 TRACK_SESSION = 0x2, // the audio session corresponds to at least one 306 // track 307 FAST_SESSION = 0x4 // the audio session corresponds to at least one 308 // fast track 309 }; 310 311 // get effect chain corresponding to session Id. 312 sp<EffectChain> getEffectChain(audio_session_t sessionId); 313 // same as getEffectChain() but must be called with ThreadBase mutex locked 314 sp<EffectChain> getEffectChain_l(audio_session_t sessionId) const; 315 // add an effect chain to the chain list (mEffectChains) 316 virtual status_t addEffectChain_l(const sp<EffectChain>& chain) = 0; 317 // remove an effect chain from the chain list (mEffectChains) 318 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain) = 0; 319 // lock all effect chains Mutexes. Must be called before releasing the 320 // ThreadBase mutex before processing the mixer and effects. This guarantees the 321 // integrity of the chains during the process. 322 // Also sets the parameter 'effectChains' to current value of mEffectChains. 323 void lockEffectChains_l(Vector< sp<EffectChain> >& effectChains); 324 // unlock effect chains after process 325 void unlockEffectChains(const Vector< sp<EffectChain> >& effectChains); 326 // get a copy of mEffectChains vector 327 Vector< sp<EffectChain> > getEffectChains_l() const { return mEffectChains; }; 328 // set audio mode to all effect chains 329 void setMode(audio_mode_t mode); 330 // get effect module with corresponding ID on specified audio session 331 sp<AudioFlinger::EffectModule> getEffect(audio_session_t sessionId, int effectId); 332 sp<AudioFlinger::EffectModule> getEffect_l(audio_session_t sessionId, int effectId); 333 // add and effect module. Also creates the effect chain is none exists for 334 // the effects audio session 335 status_t addEffect_l(const sp< EffectModule>& effect); 336 // remove and effect module. Also removes the effect chain is this was the last 337 // effect 338 void removeEffect_l(const sp< EffectModule>& effect, bool release = false); 339 // disconnect an effect handle from module and destroy module if last handle 340 void disconnectEffectHandle(EffectHandle *handle, bool unpinIfLast); 341 // detach all tracks connected to an auxiliary effect 342 virtual void detachAuxEffect_l(int effectId __unused) {} 343 // returns a combination of: 344 // - EFFECT_SESSION if effects on this audio session exist in one chain 345 // - TRACK_SESSION if tracks on this audio session exist 346 // - FAST_SESSION if fast tracks on this audio session exist 347 virtual uint32_t hasAudioSession_l(audio_session_t sessionId) const = 0; 348 uint32_t hasAudioSession(audio_session_t sessionId) const { 349 Mutex::Autolock _l(mLock); 350 return hasAudioSession_l(sessionId); 351 } 352 353 // the value returned by default implementation is not important as the 354 // strategy is only meaningful for PlaybackThread which implements this method 355 virtual uint32_t getStrategyForSession_l(audio_session_t sessionId __unused) 356 { return 0; } 357 358 // suspend or restore effect according to the type of effect passed. a NULL 359 // type pointer means suspend all effects in the session 360 void setEffectSuspended(const effect_uuid_t *type, 361 bool suspend, 362 audio_session_t sessionId = AUDIO_SESSION_OUTPUT_MIX); 363 // check if some effects must be suspended/restored when an effect is enabled 364 // or disabled 365 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 366 bool enabled, 367 audio_session_t sessionId = 368 AUDIO_SESSION_OUTPUT_MIX); 369 void checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 370 bool enabled, 371 audio_session_t sessionId = 372 AUDIO_SESSION_OUTPUT_MIX); 373 374 virtual status_t setSyncEvent(const sp<SyncEvent>& event) = 0; 375 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const = 0; 376 377 // Return a reference to a per-thread heap which can be used to allocate IMemory 378 // objects that will be read-only to client processes, read/write to mediaserver, 379 // and shared by all client processes of the thread. 380 // The heap is per-thread rather than common across all threads, because 381 // clients can't be trusted not to modify the offset of the IMemory they receive. 382 // If a thread does not have such a heap, this method returns 0. 383 virtual sp<MemoryDealer> readOnlyHeap() const { return 0; } 384 385 virtual sp<IMemory> pipeMemory() const { return 0; } 386 387 void systemReady(); 388 389 // checkEffectCompatibility_l() must be called with ThreadBase::mLock held 390 virtual status_t checkEffectCompatibility_l(const effect_descriptor_t *desc, 391 audio_session_t sessionId) = 0; 392 393 mutable Mutex mLock; 394 395protected: 396 397 // entry describing an effect being suspended in mSuspendedSessions keyed vector 398 class SuspendedSessionDesc : public RefBase { 399 public: 400 SuspendedSessionDesc() : mRefCount(0) {} 401 402 int mRefCount; // number of active suspend requests 403 effect_uuid_t mType; // effect type UUID 404 }; 405 406 void acquireWakeLock(); 407 virtual void acquireWakeLock_l(); 408 void releaseWakeLock(); 409 void releaseWakeLock_l(); 410 void updateWakeLockUids_l(const SortedVector<int> &uids); 411 void getPowerManager_l(); 412 void setEffectSuspended_l(const effect_uuid_t *type, 413 bool suspend, 414 audio_session_t sessionId); 415 // updated mSuspendedSessions when an effect suspended or restored 416 void updateSuspendedSessions_l(const effect_uuid_t *type, 417 bool suspend, 418 audio_session_t sessionId); 419 // check if some effects must be suspended when an effect chain is added 420 void checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain); 421 422 String16 getWakeLockTag(); 423 424 virtual void preExit() { } 425 virtual void setMasterMono_l(bool mono __unused) { } 426 virtual bool requireMonoBlend() { return false; } 427 428 friend class AudioFlinger; // for mEffectChains 429 430 const type_t mType; 431 432 // Used by parameters, config events, addTrack_l, exit 433 Condition mWaitWorkCV; 434 435 const sp<AudioFlinger> mAudioFlinger; 436 437 // updated by PlaybackThread::readOutputParameters_l() or 438 // RecordThread::readInputParameters_l() 439 uint32_t mSampleRate; 440 size_t mFrameCount; // output HAL, direct output, record 441 audio_channel_mask_t mChannelMask; 442 uint32_t mChannelCount; 443 size_t mFrameSize; 444 // not HAL frame size, this is for output sink (to pipe to fast mixer) 445 audio_format_t mFormat; // Source format for Recording and 446 // Sink format for Playback. 447 // Sink format may be different than 448 // HAL format if Fastmixer is used. 449 audio_format_t mHALFormat; 450 size_t mBufferSize; // HAL buffer size for read() or write() 451 452 Vector< sp<ConfigEvent> > mConfigEvents; 453 Vector< sp<ConfigEvent> > mPendingConfigEvents; // events awaiting system ready 454 455 // These fields are written and read by thread itself without lock or barrier, 456 // and read by other threads without lock or barrier via standby(), outDevice() 457 // and inDevice(). 458 // Because of the absence of a lock or barrier, any other thread that reads 459 // these fields must use the information in isolation, or be prepared to deal 460 // with possibility that it might be inconsistent with other information. 461 bool mStandby; // Whether thread is currently in standby. 462 audio_devices_t mOutDevice; // output device 463 audio_devices_t mInDevice; // input device 464 audio_devices_t mPrevOutDevice; // previous output device 465 audio_devices_t mPrevInDevice; // previous input device 466 struct audio_patch mPatch; 467 audio_source_t mAudioSource; 468 469 const audio_io_handle_t mId; 470 Vector< sp<EffectChain> > mEffectChains; 471 472 static const int kThreadNameLength = 16; // prctl(PR_SET_NAME) limit 473 char mThreadName[kThreadNameLength]; // guaranteed NUL-terminated 474 sp<IPowerManager> mPowerManager; 475 sp<IBinder> mWakeLockToken; 476 const sp<PMDeathRecipient> mDeathRecipient; 477 // list of suspended effects per session and per type. The first (outer) vector is 478 // keyed by session ID, the second (inner) by type UUID timeLow field 479 KeyedVector< audio_session_t, KeyedVector< int, sp<SuspendedSessionDesc> > > 480 mSuspendedSessions; 481 static const size_t kLogSize = 4 * 1024; 482 sp<NBLog::Writer> mNBLogWriter; 483 bool mSystemReady; 484 ExtendedTimestamp mTimestamp; 485 486 // ActiveTracks is a sorted vector of track type T representing the 487 // active tracks of threadLoop() to be considered by the locked prepare portion. 488 // ActiveTracks should be accessed with the ThreadBase lock held. 489 // 490 // During processing and I/O, the threadLoop does not hold the lock; 491 // hence it does not directly use ActiveTracks. Care should be taken 492 // to hold local strong references or defer removal of tracks 493 // if the threadLoop may still be accessing those tracks due to mix, etc. 494 // 495 // This class updates power information appropriately. 496 // 497 498 template <typename T> 499 class ActiveTracks { 500 public: 501 ActiveTracks() 502 : mActiveTracksGeneration(0) 503 , mLastActiveTracksGeneration(0) 504 { } 505 506 ~ActiveTracks() { 507 ALOGW_IF(!mActiveTracks.isEmpty(), 508 "ActiveTracks should be empty in destructor"); 509 } 510 // returns the last track added (even though it may have been 511 // subsequently removed from ActiveTracks). 512 // 513 // Used for DirectOutputThread to ensure a flush is called when transitioning 514 // to a new track (even though it may be on the same session). 515 // Used for OffloadThread to ensure that volume and mixer state is 516 // taken from the latest track added. 517 // 518 // The latest track is saved with a weak pointer to prevent keeping an 519 // otherwise useless track alive. Thus the function will return nullptr 520 // if the latest track has subsequently been removed and destroyed. 521 sp<T> getLatest() { 522 return mLatestActiveTrack.promote(); 523 } 524 525 // SortedVector methods 526 ssize_t add(const sp<T> &track); 527 ssize_t remove(const sp<T> &track); 528 size_t size() const { 529 return mActiveTracks.size(); 530 } 531 ssize_t indexOf(const sp<T>& item) { 532 return mActiveTracks.indexOf(item); 533 } 534 sp<T> operator[](size_t index) const { 535 return mActiveTracks[index]; 536 } 537 typename SortedVector<sp<T>>::iterator begin() { 538 return mActiveTracks.begin(); 539 } 540 typename SortedVector<sp<T>>::iterator end() { 541 return mActiveTracks.end(); 542 } 543 544 // Due to Binder recursion optimization, clear() and updatePowerState() 545 // cannot be called from a Binder thread because they may call back into 546 // the original calling process (system server) for BatteryNotifier 547 // (which requires a Java environment that may not be present). 548 // Hence, call clear() and updatePowerState() only from the 549 // ThreadBase thread. 550 void clear(); 551 // periodically called in the threadLoop() to update power state uids. 552 void updatePowerState(sp<ThreadBase> thread, bool force = false); 553 554 private: 555 SortedVector<int> getWakeLockUids() { 556 SortedVector<int> wakeLockUids; 557 for (const sp<T> &track : mActiveTracks) { 558 wakeLockUids.add(track->uid()); 559 } 560 return wakeLockUids; // moved by underlying SharedBuffer 561 } 562 563 std::map<uid_t, std::pair<ssize_t /* previous */, ssize_t /* current */>> 564 mBatteryCounter; 565 SortedVector<sp<T>> mActiveTracks; 566 int mActiveTracksGeneration; 567 int mLastActiveTracksGeneration; 568 wp<T> mLatestActiveTrack; // latest track added to ActiveTracks 569 }; 570}; 571 572// --- PlaybackThread --- 573class PlaybackThread : public ThreadBase, public StreamOutHalInterfaceCallback { 574public: 575 576#include "PlaybackTracks.h" 577 578 enum mixer_state { 579 MIXER_IDLE, // no active tracks 580 MIXER_TRACKS_ENABLED, // at least one active track, but no track has any data ready 581 MIXER_TRACKS_READY, // at least one active track, and at least one track has data 582 MIXER_DRAIN_TRACK, // drain currently playing track 583 MIXER_DRAIN_ALL, // fully drain the hardware 584 // standby mode does not have an enum value 585 // suspend by audio policy manager is orthogonal to mixer state 586 }; 587 588 // retry count before removing active track in case of underrun on offloaded thread: 589 // we need to make sure that AudioTrack client has enough time to send large buffers 590 //FIXME may be more appropriate if expressed in time units. Need to revise how underrun is 591 // handled for offloaded tracks 592 static const int8_t kMaxTrackRetriesOffload = 20; 593 static const int8_t kMaxTrackStartupRetriesOffload = 100; 594 static const int8_t kMaxTrackStopRetriesOffload = 2; 595 // 14 tracks max per client allows for 2 misbehaving application leaving 4 available tracks. 596 static const uint32_t kMaxTracksPerUid = 14; 597 598 PlaybackThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 599 audio_io_handle_t id, audio_devices_t device, type_t type, bool systemReady); 600 virtual ~PlaybackThread(); 601 602 void dump(int fd, const Vector<String16>& args); 603 604 // Thread virtuals 605 virtual bool threadLoop(); 606 607 // RefBase 608 virtual void onFirstRef(); 609 610 virtual status_t checkEffectCompatibility_l(const effect_descriptor_t *desc, 611 audio_session_t sessionId); 612 613protected: 614 // Code snippets that were lifted up out of threadLoop() 615 virtual void threadLoop_mix() = 0; 616 virtual void threadLoop_sleepTime() = 0; 617 virtual ssize_t threadLoop_write(); 618 virtual void threadLoop_drain(); 619 virtual void threadLoop_standby(); 620 virtual void threadLoop_exit(); 621 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 622 623 // prepareTracks_l reads and writes mActiveTracks, and returns 624 // the pending set of tracks to remove via Vector 'tracksToRemove'. The caller 625 // is responsible for clearing or destroying this Vector later on, when it 626 // is safe to do so. That will drop the final ref count and destroy the tracks. 627 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove) = 0; 628 void removeTracks_l(const Vector< sp<Track> >& tracksToRemove); 629 630 // StreamOutHalInterfaceCallback implementation 631 virtual void onWriteReady(); 632 virtual void onDrainReady(); 633 virtual void onError(); 634 635 void resetWriteBlocked(uint32_t sequence); 636 void resetDraining(uint32_t sequence); 637 638 virtual bool waitingAsyncCallback(); 639 virtual bool waitingAsyncCallback_l(); 640 virtual bool shouldStandby_l(); 641 virtual void onAddNewTrack_l(); 642 void onAsyncError(); // error reported by AsyncCallbackThread 643 644 // ThreadBase virtuals 645 virtual void preExit(); 646 647 virtual bool keepWakeLock() const { return true; } 648 virtual void acquireWakeLock_l() { 649 ThreadBase::acquireWakeLock_l(); 650 mActiveTracks.updatePowerState(this, true /* force */); 651 } 652 653public: 654 655 virtual status_t initCheck() const { return (mOutput == NULL) ? NO_INIT : NO_ERROR; } 656 657 // return estimated latency in milliseconds, as reported by HAL 658 uint32_t latency() const; 659 // same, but lock must already be held 660 uint32_t latency_l() const; 661 662 void setMasterVolume(float value); 663 void setMasterMute(bool muted); 664 665 void setStreamVolume(audio_stream_type_t stream, float value); 666 void setStreamMute(audio_stream_type_t stream, bool muted); 667 668 float streamVolume(audio_stream_type_t stream) const; 669 670 sp<Track> createTrack_l( 671 const sp<AudioFlinger::Client>& client, 672 audio_stream_type_t streamType, 673 uint32_t sampleRate, 674 audio_format_t format, 675 audio_channel_mask_t channelMask, 676 size_t *pFrameCount, 677 const sp<IMemory>& sharedBuffer, 678 audio_session_t sessionId, 679 audio_output_flags_t *flags, 680 pid_t tid, 681 uid_t uid, 682 status_t *status /*non-NULL*/); 683 684 AudioStreamOut* getOutput() const; 685 AudioStreamOut* clearOutput(); 686 virtual sp<StreamHalInterface> stream() const; 687 688 // a very large number of suspend() will eventually wraparound, but unlikely 689 void suspend() { (void) android_atomic_inc(&mSuspended); } 690 void restore() 691 { 692 // if restore() is done without suspend(), get back into 693 // range so that the next suspend() will operate correctly 694 if (android_atomic_dec(&mSuspended) <= 0) { 695 android_atomic_release_store(0, &mSuspended); 696 } 697 } 698 bool isSuspended() const 699 { return android_atomic_acquire_load(&mSuspended) > 0; } 700 701 virtual String8 getParameters(const String8& keys); 702 virtual void ioConfigChanged(audio_io_config_event event, pid_t pid = 0); 703 status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames); 704 // FIXME rename mixBuffer() to sinkBuffer() and remove int16_t* dependency. 705 // Consider also removing and passing an explicit mMainBuffer initialization 706 // parameter to AF::PlaybackThread::Track::Track(). 707 int16_t *mixBuffer() const { 708 return reinterpret_cast<int16_t *>(mSinkBuffer); }; 709 710 virtual void detachAuxEffect_l(int effectId); 711 status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track>& track, 712 int EffectId); 713 status_t attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track>& track, 714 int EffectId); 715 716 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 717 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 718 virtual uint32_t hasAudioSession_l(audio_session_t sessionId) const; 719 virtual uint32_t getStrategyForSession_l(audio_session_t sessionId); 720 721 722 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 723 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 724 725 // called with AudioFlinger lock held 726 bool invalidateTracks_l(audio_stream_type_t streamType); 727 virtual void invalidateTracks(audio_stream_type_t streamType); 728 729 virtual size_t frameCount() const { return mNormalFrameCount; } 730 731 status_t getTimestamp_l(AudioTimestamp& timestamp); 732 733 void addPatchTrack(const sp<PatchTrack>& track); 734 void deletePatchTrack(const sp<PatchTrack>& track); 735 736 virtual void getAudioPortConfig(struct audio_port_config *config); 737 738protected: 739 // updated by readOutputParameters_l() 740 size_t mNormalFrameCount; // normal mixer and effects 741 742 bool mThreadThrottle; // throttle the thread processing 743 uint32_t mThreadThrottleTimeMs; // throttle time for MIXER threads 744 uint32_t mThreadThrottleEndMs; // notify once per throttling 745 uint32_t mHalfBufferMs; // half the buffer size in milliseconds 746 747 void* mSinkBuffer; // frame size aligned sink buffer 748 749 // TODO: 750 // Rearrange the buffer info into a struct/class with 751 // clear, copy, construction, destruction methods. 752 // 753 // mSinkBuffer also has associated with it: 754 // 755 // mSinkBufferSize: Sink Buffer Size 756 // mFormat: Sink Buffer Format 757 758 // Mixer Buffer (mMixerBuffer*) 759 // 760 // In the case of floating point or multichannel data, which is not in the 761 // sink format, it is required to accumulate in a higher precision or greater channel count 762 // buffer before downmixing or data conversion to the sink buffer. 763 764 // Set to "true" to enable the Mixer Buffer otherwise mixer output goes to sink buffer. 765 bool mMixerBufferEnabled; 766 767 // Storage, 32 byte aligned (may make this alignment a requirement later). 768 // Due to constraints on mNormalFrameCount, the buffer size is a multiple of 16 frames. 769 void* mMixerBuffer; 770 771 // Size of mMixerBuffer in bytes: mNormalFrameCount * #channels * sampsize. 772 size_t mMixerBufferSize; 773 774 // The audio format of mMixerBuffer. Set to AUDIO_FORMAT_PCM_(FLOAT|16_BIT) only. 775 audio_format_t mMixerBufferFormat; 776 777 // An internal flag set to true by MixerThread::prepareTracks_l() 778 // when mMixerBuffer contains valid data after mixing. 779 bool mMixerBufferValid; 780 781 // Effects Buffer (mEffectsBuffer*) 782 // 783 // In the case of effects data, which is not in the sink format, 784 // it is required to accumulate in a different buffer before data conversion 785 // to the sink buffer. 786 787 // Set to "true" to enable the Effects Buffer otherwise effects output goes to sink buffer. 788 bool mEffectBufferEnabled; 789 790 // Storage, 32 byte aligned (may make this alignment a requirement later). 791 // Due to constraints on mNormalFrameCount, the buffer size is a multiple of 16 frames. 792 void* mEffectBuffer; 793 794 // Size of mEffectsBuffer in bytes: mNormalFrameCount * #channels * sampsize. 795 size_t mEffectBufferSize; 796 797 // The audio format of mEffectsBuffer. Set to AUDIO_FORMAT_PCM_16_BIT only. 798 audio_format_t mEffectBufferFormat; 799 800 // An internal flag set to true by MixerThread::prepareTracks_l() 801 // when mEffectsBuffer contains valid data after mixing. 802 // 803 // When this is set, all mixer data is routed into the effects buffer 804 // for any processing (including output processing). 805 bool mEffectBufferValid; 806 807 // suspend count, > 0 means suspended. While suspended, the thread continues to pull from 808 // tracks and mix, but doesn't write to HAL. A2DP and SCO HAL implementations can't handle 809 // concurrent use of both of them, so Audio Policy Service suspends one of the threads to 810 // workaround that restriction. 811 // 'volatile' means accessed via atomic operations and no lock. 812 volatile int32_t mSuspended; 813 814 int64_t mBytesWritten; 815 int64_t mFramesWritten; // not reset on standby 816 int64_t mSuspendedFrames; // not reset on standby 817private: 818 // mMasterMute is in both PlaybackThread and in AudioFlinger. When a 819 // PlaybackThread needs to find out if master-muted, it checks it's local 820 // copy rather than the one in AudioFlinger. This optimization saves a lock. 821 bool mMasterMute; 822 void setMasterMute_l(bool muted) { mMasterMute = muted; } 823protected: 824 ActiveTracks<Track> mActiveTracks; 825 826 // Allocate a track name for a given channel mask. 827 // Returns name >= 0 if successful, -1 on failure. 828 virtual int getTrackName_l(audio_channel_mask_t channelMask, audio_format_t format, 829 audio_session_t sessionId, uid_t uid) = 0; 830 virtual void deleteTrackName_l(int name) = 0; 831 832 // Time to sleep between cycles when: 833 virtual uint32_t activeSleepTimeUs() const; // mixer state MIXER_TRACKS_ENABLED 834 virtual uint32_t idleSleepTimeUs() const = 0; // mixer state MIXER_IDLE 835 virtual uint32_t suspendSleepTimeUs() const = 0; // audio policy manager suspended us 836 // No sleep when mixer state == MIXER_TRACKS_READY; relies on audio HAL stream->write() 837 // No sleep in standby mode; waits on a condition 838 839 // Code snippets that are temporarily lifted up out of threadLoop() until the merge 840 void checkSilentMode_l(); 841 842 // Non-trivial for DUPLICATING only 843 virtual void saveOutputTracks() { } 844 virtual void clearOutputTracks() { } 845 846 // Cache various calculated values, at threadLoop() entry and after a parameter change 847 virtual void cacheParameters_l(); 848 849 virtual uint32_t correctLatency_l(uint32_t latency) const; 850 851 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 852 audio_patch_handle_t *handle); 853 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 854 855 bool usesHwAvSync() const { return (mType == DIRECT) && (mOutput != NULL) 856 && mHwSupportsPause 857 && (mOutput->flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC); } 858 859 uint32_t trackCountForUid_l(uid_t uid); 860 861private: 862 863 friend class AudioFlinger; // for numerous 864 865 PlaybackThread& operator = (const PlaybackThread&); 866 867 status_t addTrack_l(const sp<Track>& track); 868 bool destroyTrack_l(const sp<Track>& track); 869 void removeTrack_l(const sp<Track>& track); 870 void broadcast_l(); 871 872 void readOutputParameters_l(); 873 874 virtual void dumpInternals(int fd, const Vector<String16>& args); 875 void dumpTracks(int fd, const Vector<String16>& args); 876 877 SortedVector< sp<Track> > mTracks; 878 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 879 AudioStreamOut *mOutput; 880 881 float mMasterVolume; 882 nsecs_t mLastWriteTime; 883 int mNumWrites; 884 int mNumDelayedWrites; 885 bool mInWrite; 886 887 // FIXME rename these former local variables of threadLoop to standard "m" names 888 nsecs_t mStandbyTimeNs; 889 size_t mSinkBufferSize; 890 891 // cached copies of activeSleepTimeUs() and idleSleepTimeUs() made by cacheParameters_l() 892 uint32_t mActiveSleepTimeUs; 893 uint32_t mIdleSleepTimeUs; 894 895 uint32_t mSleepTimeUs; 896 897 // mixer status returned by prepareTracks_l() 898 mixer_state mMixerStatus; // current cycle 899 // previous cycle when in prepareTracks_l() 900 mixer_state mMixerStatusIgnoringFastTracks; 901 // FIXME or a separate ready state per track 902 903 // FIXME move these declarations into the specific sub-class that needs them 904 // MIXER only 905 uint32_t sleepTimeShift; 906 907 // same as AudioFlinger::mStandbyTimeInNsecs except for DIRECT which uses a shorter value 908 nsecs_t mStandbyDelayNs; 909 910 // MIXER only 911 nsecs_t maxPeriod; 912 913 // DUPLICATING only 914 uint32_t writeFrames; 915 916 size_t mBytesRemaining; 917 size_t mCurrentWriteLength; 918 bool mUseAsyncWrite; 919 // mWriteAckSequence contains current write sequence on bits 31-1. The write sequence is 920 // incremented each time a write(), a flush() or a standby() occurs. 921 // Bit 0 is set when a write blocks and indicates a callback is expected. 922 // Bit 0 is reset by the async callback thread calling resetWriteBlocked(). Out of sequence 923 // callbacks are ignored. 924 uint32_t mWriteAckSequence; 925 // mDrainSequence contains current drain sequence on bits 31-1. The drain sequence is 926 // incremented each time a drain is requested or a flush() or standby() occurs. 927 // Bit 0 is set when the drain() command is called at the HAL and indicates a callback is 928 // expected. 929 // Bit 0 is reset by the async callback thread calling resetDraining(). Out of sequence 930 // callbacks are ignored. 931 uint32_t mDrainSequence; 932 // A condition that must be evaluated by prepareTrack_l() has changed and we must not wait 933 // for async write callback in the thread loop before evaluating it 934 bool mSignalPending; 935 sp<AsyncCallbackThread> mCallbackThread; 936 937private: 938 // The HAL output sink is treated as non-blocking, but current implementation is blocking 939 sp<NBAIO_Sink> mOutputSink; 940 // If a fast mixer is present, the blocking pipe sink, otherwise clear 941 sp<NBAIO_Sink> mPipeSink; 942 // The current sink for the normal mixer to write it's (sub)mix, mOutputSink or mPipeSink 943 sp<NBAIO_Sink> mNormalSink; 944#ifdef TEE_SINK 945 // For dumpsys 946 sp<NBAIO_Sink> mTeeSink; 947 sp<NBAIO_Source> mTeeSource; 948#endif 949 uint32_t mScreenState; // cached copy of gScreenState 950 static const size_t kFastMixerLogSize = 4 * 1024; 951 sp<NBLog::Writer> mFastMixerNBLogWriter; 952 953 // Do not call from a sched_fifo thread as it uses a system time call 954 // and obtains a local mutex. 955 class LocalLog { 956 public: 957 void log(const char *fmt, ...) { 958 va_list val; 959 va_start(val, fmt); 960 961 // format to buffer 962 char buffer[512]; 963 int length = vsnprintf(buffer, sizeof(buffer), fmt, val); 964 if (length >= (signed)sizeof(buffer)) { 965 length = sizeof(buffer) - 1; 966 } 967 968 // strip out trailing newline 969 while (length > 0 && buffer[length - 1] == '\n') { 970 buffer[--length] = 0; 971 } 972 973 // store in circular array 974 AutoMutex _l(mLock); 975 mLog.emplace_back( 976 std::make_pair(systemTime(SYSTEM_TIME_REALTIME), std::string(buffer))); 977 if (mLog.size() > kLogSize) { 978 mLog.pop_front(); 979 } 980 981 va_end(val); 982 } 983 984 void dump(int fd, const Vector<String16>& args, const char *prefix = "") { 985 if (!AudioFlinger::dumpTryLock(mLock)) return; // a local lock, shouldn't happen 986 if (mLog.size() > 0) { 987 bool dumpAll = false; 988 for (const auto &arg : args) { 989 if (arg == String16("--locallog")) { 990 dumpAll = true; 991 } 992 } 993 994 dprintf(fd, "Local Log:\n"); 995 auto it = mLog.begin(); 996 if (!dumpAll && mLog.size() > kLogPrint) { 997 it += (mLog.size() - kLogPrint); 998 } 999 for (; it != mLog.end(); ++it) { 1000 const int64_t ns = it->first; 1001 const int ns_per_sec = 1000000000; 1002 const time_t sec = ns / ns_per_sec; 1003 struct tm tm; 1004 localtime_r(&sec, &tm); 1005 1006 dprintf(fd, "%s%02d-%02d %02d:%02d:%02d.%03d %s\n", 1007 prefix, 1008 tm.tm_mon + 1, // localtime_r uses months in 0 - 11 range 1009 tm.tm_mday, tm.tm_hour, tm.tm_min, tm.tm_sec, 1010 (int)(ns % ns_per_sec / 1000000), 1011 it->second.c_str()); 1012 } 1013 } 1014 mLock.unlock(); 1015 } 1016 1017 private: 1018 Mutex mLock; 1019 static const size_t kLogSize = 256; // full history 1020 static const size_t kLogPrint = 32; // default print history 1021 std::deque<std::pair<int64_t, std::string>> mLog; 1022 } mLocalLog; 1023 1024public: 1025 virtual bool hasFastMixer() const = 0; 1026 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex __unused) const 1027 { FastTrackUnderruns dummy; return dummy; } 1028 1029protected: 1030 // accessed by both binder threads and within threadLoop(), lock on mutex needed 1031 unsigned mFastTrackAvailMask; // bit i set if fast track [i] is available 1032 bool mHwSupportsPause; 1033 bool mHwPaused; 1034 bool mFlushPending; 1035}; 1036 1037class MixerThread : public PlaybackThread { 1038public: 1039 MixerThread(const sp<AudioFlinger>& audioFlinger, 1040 AudioStreamOut* output, 1041 audio_io_handle_t id, 1042 audio_devices_t device, 1043 bool systemReady, 1044 type_t type = MIXER); 1045 virtual ~MixerThread(); 1046 1047 // Thread virtuals 1048 1049 virtual bool checkForNewParameter_l(const String8& keyValuePair, 1050 status_t& status); 1051 virtual void dumpInternals(int fd, const Vector<String16>& args); 1052 1053protected: 1054 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1055 virtual int getTrackName_l(audio_channel_mask_t channelMask, audio_format_t format, 1056 audio_session_t sessionId, uid_t uid); 1057 virtual void deleteTrackName_l(int name); 1058 virtual uint32_t idleSleepTimeUs() const; 1059 virtual uint32_t suspendSleepTimeUs() const; 1060 virtual void cacheParameters_l(); 1061 1062 virtual void acquireWakeLock_l() { 1063 PlaybackThread::acquireWakeLock_l(); 1064 if (hasFastMixer()) { 1065 mFastMixer->setBoottimeOffset( 1066 mTimestamp.mTimebaseOffset[ExtendedTimestamp::TIMEBASE_BOOTTIME]); 1067 } 1068 } 1069 1070 // threadLoop snippets 1071 virtual ssize_t threadLoop_write(); 1072 virtual void threadLoop_standby(); 1073 virtual void threadLoop_mix(); 1074 virtual void threadLoop_sleepTime(); 1075 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 1076 virtual uint32_t correctLatency_l(uint32_t latency) const; 1077 1078 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 1079 audio_patch_handle_t *handle); 1080 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 1081 1082 AudioMixer* mAudioMixer; // normal mixer 1083private: 1084 // one-time initialization, no locks required 1085 sp<FastMixer> mFastMixer; // non-0 if there is also a fast mixer 1086 sp<AudioWatchdog> mAudioWatchdog; // non-0 if there is an audio watchdog thread 1087 1088 // contents are not guaranteed to be consistent, no locks required 1089 FastMixerDumpState mFastMixerDumpState; 1090#ifdef STATE_QUEUE_DUMP 1091 StateQueueObserverDump mStateQueueObserverDump; 1092 StateQueueMutatorDump mStateQueueMutatorDump; 1093#endif 1094 AudioWatchdogDump mAudioWatchdogDump; 1095 1096 // accessible only within the threadLoop(), no locks required 1097 // mFastMixer->sq() // for mutating and pushing state 1098 int32_t mFastMixerFutex; // for cold idle 1099 1100 std::atomic_bool mMasterMono; 1101public: 1102 virtual bool hasFastMixer() const { return mFastMixer != 0; } 1103 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const { 1104 ALOG_ASSERT(fastIndex < FastMixerState::sMaxFastTracks); 1105 return mFastMixerDumpState.mTracks[fastIndex].mUnderruns; 1106 } 1107 1108protected: 1109 virtual void setMasterMono_l(bool mono) { 1110 mMasterMono.store(mono); 1111 if (mFastMixer != nullptr) { /* hasFastMixer() */ 1112 mFastMixer->setMasterMono(mMasterMono); 1113 } 1114 } 1115 // the FastMixer performs mono blend if it exists. 1116 // Blending with limiter is not idempotent, 1117 // and blending without limiter is idempotent but inefficient to do twice. 1118 virtual bool requireMonoBlend() { return mMasterMono.load() && !hasFastMixer(); } 1119}; 1120 1121class DirectOutputThread : public PlaybackThread { 1122public: 1123 1124 DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1125 audio_io_handle_t id, audio_devices_t device, bool systemReady); 1126 virtual ~DirectOutputThread(); 1127 1128 // Thread virtuals 1129 1130 virtual bool checkForNewParameter_l(const String8& keyValuePair, 1131 status_t& status); 1132 virtual void flushHw_l(); 1133 1134protected: 1135 virtual int getTrackName_l(audio_channel_mask_t channelMask, audio_format_t format, 1136 audio_session_t sessionId, uid_t uid); 1137 virtual void deleteTrackName_l(int name); 1138 virtual uint32_t activeSleepTimeUs() const; 1139 virtual uint32_t idleSleepTimeUs() const; 1140 virtual uint32_t suspendSleepTimeUs() const; 1141 virtual void cacheParameters_l(); 1142 1143 // threadLoop snippets 1144 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1145 virtual void threadLoop_mix(); 1146 virtual void threadLoop_sleepTime(); 1147 virtual void threadLoop_exit(); 1148 virtual bool shouldStandby_l(); 1149 1150 virtual void onAddNewTrack_l(); 1151 1152 // volumes last sent to audio HAL with stream->set_volume() 1153 float mLeftVolFloat; 1154 float mRightVolFloat; 1155 1156 DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1157 audio_io_handle_t id, uint32_t device, ThreadBase::type_t type, 1158 bool systemReady); 1159 void processVolume_l(Track *track, bool lastTrack); 1160 1161 // prepareTracks_l() tells threadLoop_mix() the name of the single active track 1162 sp<Track> mActiveTrack; 1163 1164 wp<Track> mPreviousTrack; // used to detect track switch 1165 1166public: 1167 virtual bool hasFastMixer() const { return false; } 1168}; 1169 1170class OffloadThread : public DirectOutputThread { 1171public: 1172 1173 OffloadThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1174 audio_io_handle_t id, uint32_t device, bool systemReady); 1175 virtual ~OffloadThread() {}; 1176 virtual void flushHw_l(); 1177 1178protected: 1179 // threadLoop snippets 1180 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1181 virtual void threadLoop_exit(); 1182 1183 virtual bool waitingAsyncCallback(); 1184 virtual bool waitingAsyncCallback_l(); 1185 virtual void invalidateTracks(audio_stream_type_t streamType); 1186 1187 virtual bool keepWakeLock() const { return (mKeepWakeLock || (mDrainSequence & 1)); } 1188 1189private: 1190 size_t mPausedWriteLength; // length in bytes of write interrupted by pause 1191 size_t mPausedBytesRemaining; // bytes still waiting in mixbuffer after resume 1192 bool mKeepWakeLock; // keep wake lock while waiting for write callback 1193 uint64_t mOffloadUnderrunPosition; // Current frame position for offloaded playback 1194 // used and valid only during underrun. ~0 if 1195 // no underrun has occurred during playback and 1196 // is not reset on standby. 1197}; 1198 1199class AsyncCallbackThread : public Thread { 1200public: 1201 1202 explicit AsyncCallbackThread(const wp<PlaybackThread>& playbackThread); 1203 1204 virtual ~AsyncCallbackThread(); 1205 1206 // Thread virtuals 1207 virtual bool threadLoop(); 1208 1209 // RefBase 1210 virtual void onFirstRef(); 1211 1212 void exit(); 1213 void setWriteBlocked(uint32_t sequence); 1214 void resetWriteBlocked(); 1215 void setDraining(uint32_t sequence); 1216 void resetDraining(); 1217 void setAsyncError(); 1218 1219private: 1220 const wp<PlaybackThread> mPlaybackThread; 1221 // mWriteAckSequence corresponds to the last write sequence passed by the offload thread via 1222 // setWriteBlocked(). The sequence is shifted one bit to the left and the lsb is used 1223 // to indicate that the callback has been received via resetWriteBlocked() 1224 uint32_t mWriteAckSequence; 1225 // mDrainSequence corresponds to the last drain sequence passed by the offload thread via 1226 // setDraining(). The sequence is shifted one bit to the left and the lsb is used 1227 // to indicate that the callback has been received via resetDraining() 1228 uint32_t mDrainSequence; 1229 Condition mWaitWorkCV; 1230 Mutex mLock; 1231 bool mAsyncError; 1232}; 1233 1234class DuplicatingThread : public MixerThread { 1235public: 1236 DuplicatingThread(const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread, 1237 audio_io_handle_t id, bool systemReady); 1238 virtual ~DuplicatingThread(); 1239 1240 // Thread virtuals 1241 void addOutputTrack(MixerThread* thread); 1242 void removeOutputTrack(MixerThread* thread); 1243 uint32_t waitTimeMs() const { return mWaitTimeMs; } 1244protected: 1245 virtual uint32_t activeSleepTimeUs() const; 1246 1247private: 1248 bool outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks); 1249protected: 1250 // threadLoop snippets 1251 virtual void threadLoop_mix(); 1252 virtual void threadLoop_sleepTime(); 1253 virtual ssize_t threadLoop_write(); 1254 virtual void threadLoop_standby(); 1255 virtual void cacheParameters_l(); 1256 1257private: 1258 // called from threadLoop, addOutputTrack, removeOutputTrack 1259 virtual void updateWaitTime_l(); 1260protected: 1261 virtual void saveOutputTracks(); 1262 virtual void clearOutputTracks(); 1263private: 1264 1265 uint32_t mWaitTimeMs; 1266 SortedVector < sp<OutputTrack> > outputTracks; 1267 SortedVector < sp<OutputTrack> > mOutputTracks; 1268public: 1269 virtual bool hasFastMixer() const { return false; } 1270}; 1271 1272 1273// record thread 1274class RecordThread : public ThreadBase 1275{ 1276public: 1277 1278 class RecordTrack; 1279 1280 /* The ResamplerBufferProvider is used to retrieve recorded input data from the 1281 * RecordThread. It maintains local state on the relative position of the read 1282 * position of the RecordTrack compared with the RecordThread. 1283 */ 1284 class ResamplerBufferProvider : public AudioBufferProvider 1285 { 1286 public: 1287 explicit ResamplerBufferProvider(RecordTrack* recordTrack) : 1288 mRecordTrack(recordTrack), 1289 mRsmpInUnrel(0), mRsmpInFront(0) { } 1290 virtual ~ResamplerBufferProvider() { } 1291 1292 // called to set the ResamplerBufferProvider to head of the RecordThread data buffer, 1293 // skipping any previous data read from the hal. 1294 virtual void reset(); 1295 1296 /* Synchronizes RecordTrack position with the RecordThread. 1297 * Calculates available frames and handle overruns if the RecordThread 1298 * has advanced faster than the ResamplerBufferProvider has retrieved data. 1299 * TODO: why not do this for every getNextBuffer? 1300 * 1301 * Parameters 1302 * framesAvailable: pointer to optional output size_t to store record track 1303 * frames available. 1304 * hasOverrun: pointer to optional boolean, returns true if track has overrun. 1305 */ 1306 1307 virtual void sync(size_t *framesAvailable = NULL, bool *hasOverrun = NULL); 1308 1309 // AudioBufferProvider interface 1310 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer); 1311 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 1312 private: 1313 RecordTrack * const mRecordTrack; 1314 size_t mRsmpInUnrel; // unreleased frames remaining from 1315 // most recent getNextBuffer 1316 // for debug only 1317 int32_t mRsmpInFront; // next available frame 1318 // rolling counter that is never cleared 1319 }; 1320 1321 /* The RecordBufferConverter is used for format, channel, and sample rate 1322 * conversion for a RecordTrack. 1323 * 1324 * TODO: Self contained, so move to a separate file later. 1325 * 1326 * RecordBufferConverter uses the convert() method rather than exposing a 1327 * buffer provider interface; this is to save a memory copy. 1328 */ 1329 class RecordBufferConverter 1330 { 1331 public: 1332 RecordBufferConverter( 1333 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 1334 uint32_t srcSampleRate, 1335 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 1336 uint32_t dstSampleRate); 1337 1338 ~RecordBufferConverter(); 1339 1340 /* Converts input data from an AudioBufferProvider by format, channelMask, 1341 * and sampleRate to a destination buffer. 1342 * 1343 * Parameters 1344 * dst: buffer to place the converted data. 1345 * provider: buffer provider to obtain source data. 1346 * frames: number of frames to convert 1347 * 1348 * Returns the number of frames converted. 1349 */ 1350 size_t convert(void *dst, AudioBufferProvider *provider, size_t frames); 1351 1352 // returns NO_ERROR if constructor was successful 1353 status_t initCheck() const { 1354 // mSrcChannelMask set on successful updateParameters 1355 return mSrcChannelMask != AUDIO_CHANNEL_INVALID ? NO_ERROR : NO_INIT; 1356 } 1357 1358 // allows dynamic reconfigure of all parameters 1359 status_t updateParameters( 1360 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 1361 uint32_t srcSampleRate, 1362 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 1363 uint32_t dstSampleRate); 1364 1365 // called to reset resampler buffers on record track discontinuity 1366 void reset() { 1367 if (mResampler != NULL) { 1368 mResampler->reset(); 1369 } 1370 } 1371 1372 private: 1373 // format conversion when not using resampler 1374 void convertNoResampler(void *dst, const void *src, size_t frames); 1375 1376 // format conversion when using resampler; modifies src in-place 1377 void convertResampler(void *dst, /*not-a-const*/ void *src, size_t frames); 1378 1379 // user provided information 1380 audio_channel_mask_t mSrcChannelMask; 1381 audio_format_t mSrcFormat; 1382 uint32_t mSrcSampleRate; 1383 audio_channel_mask_t mDstChannelMask; 1384 audio_format_t mDstFormat; 1385 uint32_t mDstSampleRate; 1386 1387 // derived information 1388 uint32_t mSrcChannelCount; 1389 uint32_t mDstChannelCount; 1390 size_t mDstFrameSize; 1391 1392 // format conversion buffer 1393 void *mBuf; 1394 size_t mBufFrames; 1395 size_t mBufFrameSize; 1396 1397 // resampler info 1398 AudioResampler *mResampler; 1399 1400 bool mIsLegacyDownmix; // legacy stereo to mono conversion needed 1401 bool mIsLegacyUpmix; // legacy mono to stereo conversion needed 1402 bool mRequiresFloat; // data processing requires float (e.g. resampler) 1403 PassthruBufferProvider *mInputConverterProvider; // converts input to float 1404 int8_t mIdxAry[sizeof(uint32_t) * 8]; // used for channel mask conversion 1405 }; 1406 1407#include "RecordTracks.h" 1408 1409 RecordThread(const sp<AudioFlinger>& audioFlinger, 1410 AudioStreamIn *input, 1411 audio_io_handle_t id, 1412 audio_devices_t outDevice, 1413 audio_devices_t inDevice, 1414 bool systemReady 1415#ifdef TEE_SINK 1416 , const sp<NBAIO_Sink>& teeSink 1417#endif 1418 ); 1419 virtual ~RecordThread(); 1420 1421 // no addTrack_l ? 1422 void destroyTrack_l(const sp<RecordTrack>& track); 1423 void removeTrack_l(const sp<RecordTrack>& track); 1424 1425 void dumpInternals(int fd, const Vector<String16>& args); 1426 void dumpTracks(int fd, const Vector<String16>& args); 1427 1428 // Thread virtuals 1429 virtual bool threadLoop(); 1430 1431 // RefBase 1432 virtual void onFirstRef(); 1433 1434 virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; } 1435 1436 virtual sp<MemoryDealer> readOnlyHeap() const { return mReadOnlyHeap; } 1437 1438 virtual sp<IMemory> pipeMemory() const { return mPipeMemory; } 1439 1440 sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l( 1441 const sp<AudioFlinger::Client>& client, 1442 uint32_t sampleRate, 1443 audio_format_t format, 1444 audio_channel_mask_t channelMask, 1445 size_t *pFrameCount, 1446 audio_session_t sessionId, 1447 size_t *notificationFrames, 1448 uid_t uid, 1449 audio_input_flags_t *flags, 1450 pid_t tid, 1451 status_t *status /*non-NULL*/); 1452 1453 status_t start(RecordTrack* recordTrack, 1454 AudioSystem::sync_event_t event, 1455 audio_session_t triggerSession); 1456 1457 // ask the thread to stop the specified track, and 1458 // return true if the caller should then do it's part of the stopping process 1459 bool stop(RecordTrack* recordTrack); 1460 1461 void dump(int fd, const Vector<String16>& args); 1462 AudioStreamIn* clearInput(); 1463 virtual sp<StreamHalInterface> stream() const; 1464 1465 1466 virtual bool checkForNewParameter_l(const String8& keyValuePair, 1467 status_t& status); 1468 virtual void cacheParameters_l() {} 1469 virtual String8 getParameters(const String8& keys); 1470 virtual void ioConfigChanged(audio_io_config_event event, pid_t pid = 0); 1471 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 1472 audio_patch_handle_t *handle); 1473 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 1474 1475 void addPatchRecord(const sp<PatchRecord>& record); 1476 void deletePatchRecord(const sp<PatchRecord>& record); 1477 1478 void readInputParameters_l(); 1479 virtual uint32_t getInputFramesLost(); 1480 1481 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1482 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1483 virtual uint32_t hasAudioSession_l(audio_session_t sessionId) const; 1484 1485 // Return the set of unique session IDs across all tracks. 1486 // The keys are the session IDs, and the associated values are meaningless. 1487 // FIXME replace by Set [and implement Bag/Multiset for other uses]. 1488 KeyedVector<audio_session_t, bool> sessionIds() const; 1489 1490 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 1491 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 1492 1493 static void syncStartEventCallback(const wp<SyncEvent>& event); 1494 1495 virtual size_t frameCount() const { return mFrameCount; } 1496 bool hasFastCapture() const { return mFastCapture != 0; } 1497 virtual void getAudioPortConfig(struct audio_port_config *config); 1498 1499 virtual status_t checkEffectCompatibility_l(const effect_descriptor_t *desc, 1500 audio_session_t sessionId); 1501 1502 virtual void acquireWakeLock_l() { 1503 ThreadBase::acquireWakeLock_l(); 1504 mActiveTracks.updatePowerState(this, true /* force */); 1505 } 1506 1507private: 1508 // Enter standby if not already in standby, and set mStandby flag 1509 void standbyIfNotAlreadyInStandby(); 1510 1511 // Call the HAL standby method unconditionally, and don't change mStandby flag 1512 void inputStandBy(); 1513 1514 AudioStreamIn *mInput; 1515 SortedVector < sp<RecordTrack> > mTracks; 1516 // mActiveTracks has dual roles: it indicates the current active track(s), and 1517 // is used together with mStartStopCond to indicate start()/stop() progress 1518 ActiveTracks<RecordTrack> mActiveTracks; 1519 1520 Condition mStartStopCond; 1521 1522 // resampler converts input at HAL Hz to output at AudioRecord client Hz 1523 void *mRsmpInBuffer; // size = mRsmpInFramesOA 1524 size_t mRsmpInFrames; // size of resampler input in frames 1525 size_t mRsmpInFramesP2;// size rounded up to a power-of-2 1526 size_t mRsmpInFramesOA;// mRsmpInFramesP2 + over-allocation 1527 1528 // rolling index that is never cleared 1529 int32_t mRsmpInRear; // last filled frame + 1 1530 1531 // For dumpsys 1532 const sp<NBAIO_Sink> mTeeSink; 1533 1534 const sp<MemoryDealer> mReadOnlyHeap; 1535 1536 // one-time initialization, no locks required 1537 sp<FastCapture> mFastCapture; // non-0 if there is also 1538 // a fast capture 1539 1540 // FIXME audio watchdog thread 1541 1542 // contents are not guaranteed to be consistent, no locks required 1543 FastCaptureDumpState mFastCaptureDumpState; 1544#ifdef STATE_QUEUE_DUMP 1545 // FIXME StateQueue observer and mutator dump fields 1546#endif 1547 // FIXME audio watchdog dump 1548 1549 // accessible only within the threadLoop(), no locks required 1550 // mFastCapture->sq() // for mutating and pushing state 1551 int32_t mFastCaptureFutex; // for cold idle 1552 1553 // The HAL input source is treated as non-blocking, 1554 // but current implementation is blocking 1555 sp<NBAIO_Source> mInputSource; 1556 // The source for the normal capture thread to read from: mInputSource or mPipeSource 1557 sp<NBAIO_Source> mNormalSource; 1558 // If a fast capture is present, the non-blocking pipe sink written to by fast capture, 1559 // otherwise clear 1560 sp<NBAIO_Sink> mPipeSink; 1561 // If a fast capture is present, the non-blocking pipe source read by normal thread, 1562 // otherwise clear 1563 sp<NBAIO_Source> mPipeSource; 1564 // Depth of pipe from fast capture to normal thread and fast clients, always power of 2 1565 size_t mPipeFramesP2; 1566 // If a fast capture is present, the Pipe as IMemory, otherwise clear 1567 sp<IMemory> mPipeMemory; 1568 1569 static const size_t kFastCaptureLogSize = 4 * 1024; 1570 sp<NBLog::Writer> mFastCaptureNBLogWriter; 1571 1572 bool mFastTrackAvail; // true if fast track available 1573}; 1574