Threads.h revision d01b0f18491c355d808a57cb272404480e69618f
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<uid_t> &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<uid_t> getWakeLockUids() { 556 SortedVector<uid_t> 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 audio_port_handle_t portId); 684 685 AudioStreamOut* getOutput() const; 686 AudioStreamOut* clearOutput(); 687 virtual sp<StreamHalInterface> stream() const; 688 689 // a very large number of suspend() will eventually wraparound, but unlikely 690 void suspend() { (void) android_atomic_inc(&mSuspended); } 691 void restore() 692 { 693 // if restore() is done without suspend(), get back into 694 // range so that the next suspend() will operate correctly 695 if (android_atomic_dec(&mSuspended) <= 0) { 696 android_atomic_release_store(0, &mSuspended); 697 } 698 } 699 bool isSuspended() const 700 { return android_atomic_acquire_load(&mSuspended) > 0; } 701 702 virtual String8 getParameters(const String8& keys); 703 virtual void ioConfigChanged(audio_io_config_event event, pid_t pid = 0); 704 status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames); 705 // FIXME rename mixBuffer() to sinkBuffer() and remove int16_t* dependency. 706 // Consider also removing and passing an explicit mMainBuffer initialization 707 // parameter to AF::PlaybackThread::Track::Track(). 708 int16_t *mixBuffer() const { 709 return reinterpret_cast<int16_t *>(mSinkBuffer); }; 710 711 virtual void detachAuxEffect_l(int effectId); 712 status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track>& track, 713 int EffectId); 714 status_t attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track>& track, 715 int EffectId); 716 717 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 718 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 719 virtual uint32_t hasAudioSession_l(audio_session_t sessionId) const; 720 virtual uint32_t getStrategyForSession_l(audio_session_t sessionId); 721 722 723 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 724 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 725 726 // called with AudioFlinger lock held 727 bool invalidateTracks_l(audio_stream_type_t streamType); 728 virtual void invalidateTracks(audio_stream_type_t streamType); 729 730 virtual size_t frameCount() const { return mNormalFrameCount; } 731 732 status_t getTimestamp_l(AudioTimestamp& timestamp); 733 734 void addPatchTrack(const sp<PatchTrack>& track); 735 void deletePatchTrack(const sp<PatchTrack>& track); 736 737 virtual void getAudioPortConfig(struct audio_port_config *config); 738 739protected: 740 // updated by readOutputParameters_l() 741 size_t mNormalFrameCount; // normal mixer and effects 742 743 bool mThreadThrottle; // throttle the thread processing 744 uint32_t mThreadThrottleTimeMs; // throttle time for MIXER threads 745 uint32_t mThreadThrottleEndMs; // notify once per throttling 746 uint32_t mHalfBufferMs; // half the buffer size in milliseconds 747 748 void* mSinkBuffer; // frame size aligned sink buffer 749 750 // TODO: 751 // Rearrange the buffer info into a struct/class with 752 // clear, copy, construction, destruction methods. 753 // 754 // mSinkBuffer also has associated with it: 755 // 756 // mSinkBufferSize: Sink Buffer Size 757 // mFormat: Sink Buffer Format 758 759 // Mixer Buffer (mMixerBuffer*) 760 // 761 // In the case of floating point or multichannel data, which is not in the 762 // sink format, it is required to accumulate in a higher precision or greater channel count 763 // buffer before downmixing or data conversion to the sink buffer. 764 765 // Set to "true" to enable the Mixer Buffer otherwise mixer output goes to sink buffer. 766 bool mMixerBufferEnabled; 767 768 // Storage, 32 byte aligned (may make this alignment a requirement later). 769 // Due to constraints on mNormalFrameCount, the buffer size is a multiple of 16 frames. 770 void* mMixerBuffer; 771 772 // Size of mMixerBuffer in bytes: mNormalFrameCount * #channels * sampsize. 773 size_t mMixerBufferSize; 774 775 // The audio format of mMixerBuffer. Set to AUDIO_FORMAT_PCM_(FLOAT|16_BIT) only. 776 audio_format_t mMixerBufferFormat; 777 778 // An internal flag set to true by MixerThread::prepareTracks_l() 779 // when mMixerBuffer contains valid data after mixing. 780 bool mMixerBufferValid; 781 782 // Effects Buffer (mEffectsBuffer*) 783 // 784 // In the case of effects data, which is not in the sink format, 785 // it is required to accumulate in a different buffer before data conversion 786 // to the sink buffer. 787 788 // Set to "true" to enable the Effects Buffer otherwise effects output goes to sink buffer. 789 bool mEffectBufferEnabled; 790 791 // Storage, 32 byte aligned (may make this alignment a requirement later). 792 // Due to constraints on mNormalFrameCount, the buffer size is a multiple of 16 frames. 793 void* mEffectBuffer; 794 795 // Size of mEffectsBuffer in bytes: mNormalFrameCount * #channels * sampsize. 796 size_t mEffectBufferSize; 797 798 // The audio format of mEffectsBuffer. Set to AUDIO_FORMAT_PCM_16_BIT only. 799 audio_format_t mEffectBufferFormat; 800 801 // An internal flag set to true by MixerThread::prepareTracks_l() 802 // when mEffectsBuffer contains valid data after mixing. 803 // 804 // When this is set, all mixer data is routed into the effects buffer 805 // for any processing (including output processing). 806 bool mEffectBufferValid; 807 808 // suspend count, > 0 means suspended. While suspended, the thread continues to pull from 809 // tracks and mix, but doesn't write to HAL. A2DP and SCO HAL implementations can't handle 810 // concurrent use of both of them, so Audio Policy Service suspends one of the threads to 811 // workaround that restriction. 812 // 'volatile' means accessed via atomic operations and no lock. 813 volatile int32_t mSuspended; 814 815 int64_t mBytesWritten; 816 int64_t mFramesWritten; // not reset on standby 817 int64_t mSuspendedFrames; // not reset on standby 818private: 819 // mMasterMute is in both PlaybackThread and in AudioFlinger. When a 820 // PlaybackThread needs to find out if master-muted, it checks it's local 821 // copy rather than the one in AudioFlinger. This optimization saves a lock. 822 bool mMasterMute; 823 void setMasterMute_l(bool muted) { mMasterMute = muted; } 824protected: 825 ActiveTracks<Track> mActiveTracks; 826 827 // Allocate a track name for a given channel mask. 828 // Returns name >= 0 if successful, -1 on failure. 829 virtual int getTrackName_l(audio_channel_mask_t channelMask, audio_format_t format, 830 audio_session_t sessionId, uid_t uid) = 0; 831 virtual void deleteTrackName_l(int name) = 0; 832 833 // Time to sleep between cycles when: 834 virtual uint32_t activeSleepTimeUs() const; // mixer state MIXER_TRACKS_ENABLED 835 virtual uint32_t idleSleepTimeUs() const = 0; // mixer state MIXER_IDLE 836 virtual uint32_t suspendSleepTimeUs() const = 0; // audio policy manager suspended us 837 // No sleep when mixer state == MIXER_TRACKS_READY; relies on audio HAL stream->write() 838 // No sleep in standby mode; waits on a condition 839 840 // Code snippets that are temporarily lifted up out of threadLoop() until the merge 841 void checkSilentMode_l(); 842 843 // Non-trivial for DUPLICATING only 844 virtual void saveOutputTracks() { } 845 virtual void clearOutputTracks() { } 846 847 // Cache various calculated values, at threadLoop() entry and after a parameter change 848 virtual void cacheParameters_l(); 849 850 virtual uint32_t correctLatency_l(uint32_t latency) const; 851 852 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 853 audio_patch_handle_t *handle); 854 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 855 856 bool usesHwAvSync() const { return (mType == DIRECT) && (mOutput != NULL) 857 && mHwSupportsPause 858 && (mOutput->flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC); } 859 860 uint32_t trackCountForUid_l(uid_t uid); 861 862private: 863 864 friend class AudioFlinger; // for numerous 865 866 PlaybackThread& operator = (const PlaybackThread&); 867 868 status_t addTrack_l(const sp<Track>& track); 869 bool destroyTrack_l(const sp<Track>& track); 870 void removeTrack_l(const sp<Track>& track); 871 void broadcast_l(); 872 873 void readOutputParameters_l(); 874 875 virtual void dumpInternals(int fd, const Vector<String16>& args); 876 void dumpTracks(int fd, const Vector<String16>& args); 877 878 SortedVector< sp<Track> > mTracks; 879 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 880 AudioStreamOut *mOutput; 881 882 float mMasterVolume; 883 nsecs_t mLastWriteTime; 884 int mNumWrites; 885 int mNumDelayedWrites; 886 bool mInWrite; 887 888 // FIXME rename these former local variables of threadLoop to standard "m" names 889 nsecs_t mStandbyTimeNs; 890 size_t mSinkBufferSize; 891 892 // cached copies of activeSleepTimeUs() and idleSleepTimeUs() made by cacheParameters_l() 893 uint32_t mActiveSleepTimeUs; 894 uint32_t mIdleSleepTimeUs; 895 896 uint32_t mSleepTimeUs; 897 898 // mixer status returned by prepareTracks_l() 899 mixer_state mMixerStatus; // current cycle 900 // previous cycle when in prepareTracks_l() 901 mixer_state mMixerStatusIgnoringFastTracks; 902 // FIXME or a separate ready state per track 903 904 // FIXME move these declarations into the specific sub-class that needs them 905 // MIXER only 906 uint32_t sleepTimeShift; 907 908 // same as AudioFlinger::mStandbyTimeInNsecs except for DIRECT which uses a shorter value 909 nsecs_t mStandbyDelayNs; 910 911 // MIXER only 912 nsecs_t maxPeriod; 913 914 // DUPLICATING only 915 uint32_t writeFrames; 916 917 size_t mBytesRemaining; 918 size_t mCurrentWriteLength; 919 bool mUseAsyncWrite; 920 // mWriteAckSequence contains current write sequence on bits 31-1. The write sequence is 921 // incremented each time a write(), a flush() or a standby() occurs. 922 // Bit 0 is set when a write blocks and indicates a callback is expected. 923 // Bit 0 is reset by the async callback thread calling resetWriteBlocked(). Out of sequence 924 // callbacks are ignored. 925 uint32_t mWriteAckSequence; 926 // mDrainSequence contains current drain sequence on bits 31-1. The drain sequence is 927 // incremented each time a drain is requested or a flush() or standby() occurs. 928 // Bit 0 is set when the drain() command is called at the HAL and indicates a callback is 929 // expected. 930 // Bit 0 is reset by the async callback thread calling resetDraining(). Out of sequence 931 // callbacks are ignored. 932 uint32_t mDrainSequence; 933 // A condition that must be evaluated by prepareTrack_l() has changed and we must not wait 934 // for async write callback in the thread loop before evaluating it 935 bool mSignalPending; 936 sp<AsyncCallbackThread> mCallbackThread; 937 938private: 939 // The HAL output sink is treated as non-blocking, but current implementation is blocking 940 sp<NBAIO_Sink> mOutputSink; 941 // If a fast mixer is present, the blocking pipe sink, otherwise clear 942 sp<NBAIO_Sink> mPipeSink; 943 // The current sink for the normal mixer to write it's (sub)mix, mOutputSink or mPipeSink 944 sp<NBAIO_Sink> mNormalSink; 945#ifdef TEE_SINK 946 // For dumpsys 947 sp<NBAIO_Sink> mTeeSink; 948 sp<NBAIO_Source> mTeeSource; 949#endif 950 uint32_t mScreenState; // cached copy of gScreenState 951 static const size_t kFastMixerLogSize = 4 * 1024; 952 sp<NBLog::Writer> mFastMixerNBLogWriter; 953 954 // Do not call from a sched_fifo thread as it uses a system time call 955 // and obtains a local mutex. 956 class LocalLog { 957 public: 958 void log(const char *fmt, ...) { 959 va_list val; 960 va_start(val, fmt); 961 962 // format to buffer 963 char buffer[512]; 964 int length = vsnprintf(buffer, sizeof(buffer), fmt, val); 965 if (length >= (signed)sizeof(buffer)) { 966 length = sizeof(buffer) - 1; 967 } 968 969 // strip out trailing newline 970 while (length > 0 && buffer[length - 1] == '\n') { 971 buffer[--length] = 0; 972 } 973 974 // store in circular array 975 AutoMutex _l(mLock); 976 mLog.emplace_back( 977 std::make_pair(systemTime(SYSTEM_TIME_REALTIME), std::string(buffer))); 978 if (mLog.size() > kLogSize) { 979 mLog.pop_front(); 980 } 981 982 va_end(val); 983 } 984 985 void dump(int fd, const Vector<String16>& args, const char *prefix = "") { 986 if (!AudioFlinger::dumpTryLock(mLock)) return; // a local lock, shouldn't happen 987 if (mLog.size() > 0) { 988 bool dumpAll = false; 989 for (const auto &arg : args) { 990 if (arg == String16("--locallog")) { 991 dumpAll = true; 992 } 993 } 994 995 dprintf(fd, "Local Log:\n"); 996 auto it = mLog.begin(); 997 if (!dumpAll) { 998 const size_t lines = 999 (size_t)property_get_int32("audio.locallog.lines", kLogPrint); 1000 if (mLog.size() > lines) { 1001 it += (mLog.size() - lines); 1002 } 1003 } 1004 for (; it != mLog.end(); ++it) { 1005 const int64_t ns = it->first; 1006 const int ns_per_sec = 1000000000; 1007 const time_t sec = ns / ns_per_sec; 1008 struct tm tm; 1009 localtime_r(&sec, &tm); 1010 1011 dprintf(fd, "%s%02d-%02d %02d:%02d:%02d.%03d %s\n", 1012 prefix, 1013 tm.tm_mon + 1, // localtime_r uses months in 0 - 11 range 1014 tm.tm_mday, tm.tm_hour, tm.tm_min, tm.tm_sec, 1015 (int)(ns % ns_per_sec / 1000000), 1016 it->second.c_str()); 1017 } 1018 } 1019 mLock.unlock(); 1020 } 1021 1022 private: 1023 Mutex mLock; 1024 static const size_t kLogSize = 256; // full history 1025 static const size_t kLogPrint = 32; // default print history 1026 std::deque<std::pair<int64_t, std::string>> mLog; 1027 } mLocalLog; 1028 1029public: 1030 virtual bool hasFastMixer() const = 0; 1031 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex __unused) const 1032 { FastTrackUnderruns dummy; return dummy; } 1033 1034protected: 1035 // accessed by both binder threads and within threadLoop(), lock on mutex needed 1036 unsigned mFastTrackAvailMask; // bit i set if fast track [i] is available 1037 bool mHwSupportsPause; 1038 bool mHwPaused; 1039 bool mFlushPending; 1040}; 1041 1042class MixerThread : public PlaybackThread { 1043public: 1044 MixerThread(const sp<AudioFlinger>& audioFlinger, 1045 AudioStreamOut* output, 1046 audio_io_handle_t id, 1047 audio_devices_t device, 1048 bool systemReady, 1049 type_t type = MIXER); 1050 virtual ~MixerThread(); 1051 1052 // Thread virtuals 1053 1054 virtual bool checkForNewParameter_l(const String8& keyValuePair, 1055 status_t& status); 1056 virtual void dumpInternals(int fd, const Vector<String16>& args); 1057 1058protected: 1059 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1060 virtual int getTrackName_l(audio_channel_mask_t channelMask, audio_format_t format, 1061 audio_session_t sessionId, uid_t uid); 1062 virtual void deleteTrackName_l(int name); 1063 virtual uint32_t idleSleepTimeUs() const; 1064 virtual uint32_t suspendSleepTimeUs() const; 1065 virtual void cacheParameters_l(); 1066 1067 virtual void acquireWakeLock_l() { 1068 PlaybackThread::acquireWakeLock_l(); 1069 if (hasFastMixer()) { 1070 mFastMixer->setBoottimeOffset( 1071 mTimestamp.mTimebaseOffset[ExtendedTimestamp::TIMEBASE_BOOTTIME]); 1072 } 1073 } 1074 1075 // threadLoop snippets 1076 virtual ssize_t threadLoop_write(); 1077 virtual void threadLoop_standby(); 1078 virtual void threadLoop_mix(); 1079 virtual void threadLoop_sleepTime(); 1080 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 1081 virtual uint32_t correctLatency_l(uint32_t latency) const; 1082 1083 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 1084 audio_patch_handle_t *handle); 1085 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 1086 1087 AudioMixer* mAudioMixer; // normal mixer 1088private: 1089 // one-time initialization, no locks required 1090 sp<FastMixer> mFastMixer; // non-0 if there is also a fast mixer 1091 sp<AudioWatchdog> mAudioWatchdog; // non-0 if there is an audio watchdog thread 1092 1093 // contents are not guaranteed to be consistent, no locks required 1094 FastMixerDumpState mFastMixerDumpState; 1095#ifdef STATE_QUEUE_DUMP 1096 StateQueueObserverDump mStateQueueObserverDump; 1097 StateQueueMutatorDump mStateQueueMutatorDump; 1098#endif 1099 AudioWatchdogDump mAudioWatchdogDump; 1100 1101 // accessible only within the threadLoop(), no locks required 1102 // mFastMixer->sq() // for mutating and pushing state 1103 int32_t mFastMixerFutex; // for cold idle 1104 1105 std::atomic_bool mMasterMono; 1106public: 1107 virtual bool hasFastMixer() const { return mFastMixer != 0; } 1108 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const { 1109 ALOG_ASSERT(fastIndex < FastMixerState::sMaxFastTracks); 1110 return mFastMixerDumpState.mTracks[fastIndex].mUnderruns; 1111 } 1112 1113protected: 1114 virtual void setMasterMono_l(bool mono) { 1115 mMasterMono.store(mono); 1116 if (mFastMixer != nullptr) { /* hasFastMixer() */ 1117 mFastMixer->setMasterMono(mMasterMono); 1118 } 1119 } 1120 // the FastMixer performs mono blend if it exists. 1121 // Blending with limiter is not idempotent, 1122 // and blending without limiter is idempotent but inefficient to do twice. 1123 virtual bool requireMonoBlend() { return mMasterMono.load() && !hasFastMixer(); } 1124}; 1125 1126class DirectOutputThread : public PlaybackThread { 1127public: 1128 1129 DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1130 audio_io_handle_t id, audio_devices_t device, bool systemReady); 1131 virtual ~DirectOutputThread(); 1132 1133 // Thread virtuals 1134 1135 virtual bool checkForNewParameter_l(const String8& keyValuePair, 1136 status_t& status); 1137 virtual void flushHw_l(); 1138 1139protected: 1140 virtual int getTrackName_l(audio_channel_mask_t channelMask, audio_format_t format, 1141 audio_session_t sessionId, uid_t uid); 1142 virtual void deleteTrackName_l(int name); 1143 virtual uint32_t activeSleepTimeUs() const; 1144 virtual uint32_t idleSleepTimeUs() const; 1145 virtual uint32_t suspendSleepTimeUs() const; 1146 virtual void cacheParameters_l(); 1147 1148 // threadLoop snippets 1149 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1150 virtual void threadLoop_mix(); 1151 virtual void threadLoop_sleepTime(); 1152 virtual void threadLoop_exit(); 1153 virtual bool shouldStandby_l(); 1154 1155 virtual void onAddNewTrack_l(); 1156 1157 // volumes last sent to audio HAL with stream->set_volume() 1158 float mLeftVolFloat; 1159 float mRightVolFloat; 1160 1161 DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1162 audio_io_handle_t id, uint32_t device, ThreadBase::type_t type, 1163 bool systemReady); 1164 void processVolume_l(Track *track, bool lastTrack); 1165 1166 // prepareTracks_l() tells threadLoop_mix() the name of the single active track 1167 sp<Track> mActiveTrack; 1168 1169 wp<Track> mPreviousTrack; // used to detect track switch 1170 1171public: 1172 virtual bool hasFastMixer() const { return false; } 1173}; 1174 1175class OffloadThread : public DirectOutputThread { 1176public: 1177 1178 OffloadThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1179 audio_io_handle_t id, uint32_t device, bool systemReady); 1180 virtual ~OffloadThread() {}; 1181 virtual void flushHw_l(); 1182 1183protected: 1184 // threadLoop snippets 1185 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1186 virtual void threadLoop_exit(); 1187 1188 virtual bool waitingAsyncCallback(); 1189 virtual bool waitingAsyncCallback_l(); 1190 virtual void invalidateTracks(audio_stream_type_t streamType); 1191 1192 virtual bool keepWakeLock() const { return (mKeepWakeLock || (mDrainSequence & 1)); } 1193 1194private: 1195 size_t mPausedWriteLength; // length in bytes of write interrupted by pause 1196 size_t mPausedBytesRemaining; // bytes still waiting in mixbuffer after resume 1197 bool mKeepWakeLock; // keep wake lock while waiting for write callback 1198 uint64_t mOffloadUnderrunPosition; // Current frame position for offloaded playback 1199 // used and valid only during underrun. ~0 if 1200 // no underrun has occurred during playback and 1201 // is not reset on standby. 1202}; 1203 1204class AsyncCallbackThread : public Thread { 1205public: 1206 1207 explicit AsyncCallbackThread(const wp<PlaybackThread>& playbackThread); 1208 1209 virtual ~AsyncCallbackThread(); 1210 1211 // Thread virtuals 1212 virtual bool threadLoop(); 1213 1214 // RefBase 1215 virtual void onFirstRef(); 1216 1217 void exit(); 1218 void setWriteBlocked(uint32_t sequence); 1219 void resetWriteBlocked(); 1220 void setDraining(uint32_t sequence); 1221 void resetDraining(); 1222 void setAsyncError(); 1223 1224private: 1225 const wp<PlaybackThread> mPlaybackThread; 1226 // mWriteAckSequence corresponds to the last write sequence passed by the offload thread via 1227 // setWriteBlocked(). The sequence is shifted one bit to the left and the lsb is used 1228 // to indicate that the callback has been received via resetWriteBlocked() 1229 uint32_t mWriteAckSequence; 1230 // mDrainSequence corresponds to the last drain sequence passed by the offload thread via 1231 // setDraining(). The sequence is shifted one bit to the left and the lsb is used 1232 // to indicate that the callback has been received via resetDraining() 1233 uint32_t mDrainSequence; 1234 Condition mWaitWorkCV; 1235 Mutex mLock; 1236 bool mAsyncError; 1237}; 1238 1239class DuplicatingThread : public MixerThread { 1240public: 1241 DuplicatingThread(const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread, 1242 audio_io_handle_t id, bool systemReady); 1243 virtual ~DuplicatingThread(); 1244 1245 // Thread virtuals 1246 void addOutputTrack(MixerThread* thread); 1247 void removeOutputTrack(MixerThread* thread); 1248 uint32_t waitTimeMs() const { return mWaitTimeMs; } 1249protected: 1250 virtual uint32_t activeSleepTimeUs() const; 1251 1252private: 1253 bool outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks); 1254protected: 1255 // threadLoop snippets 1256 virtual void threadLoop_mix(); 1257 virtual void threadLoop_sleepTime(); 1258 virtual ssize_t threadLoop_write(); 1259 virtual void threadLoop_standby(); 1260 virtual void cacheParameters_l(); 1261 1262private: 1263 // called from threadLoop, addOutputTrack, removeOutputTrack 1264 virtual void updateWaitTime_l(); 1265protected: 1266 virtual void saveOutputTracks(); 1267 virtual void clearOutputTracks(); 1268private: 1269 1270 uint32_t mWaitTimeMs; 1271 SortedVector < sp<OutputTrack> > outputTracks; 1272 SortedVector < sp<OutputTrack> > mOutputTracks; 1273public: 1274 virtual bool hasFastMixer() const { return false; } 1275}; 1276 1277 1278// record thread 1279class RecordThread : public ThreadBase 1280{ 1281public: 1282 1283 class RecordTrack; 1284 1285 /* The ResamplerBufferProvider is used to retrieve recorded input data from the 1286 * RecordThread. It maintains local state on the relative position of the read 1287 * position of the RecordTrack compared with the RecordThread. 1288 */ 1289 class ResamplerBufferProvider : public AudioBufferProvider 1290 { 1291 public: 1292 explicit ResamplerBufferProvider(RecordTrack* recordTrack) : 1293 mRecordTrack(recordTrack), 1294 mRsmpInUnrel(0), mRsmpInFront(0) { } 1295 virtual ~ResamplerBufferProvider() { } 1296 1297 // called to set the ResamplerBufferProvider to head of the RecordThread data buffer, 1298 // skipping any previous data read from the hal. 1299 virtual void reset(); 1300 1301 /* Synchronizes RecordTrack position with the RecordThread. 1302 * Calculates available frames and handle overruns if the RecordThread 1303 * has advanced faster than the ResamplerBufferProvider has retrieved data. 1304 * TODO: why not do this for every getNextBuffer? 1305 * 1306 * Parameters 1307 * framesAvailable: pointer to optional output size_t to store record track 1308 * frames available. 1309 * hasOverrun: pointer to optional boolean, returns true if track has overrun. 1310 */ 1311 1312 virtual void sync(size_t *framesAvailable = NULL, bool *hasOverrun = NULL); 1313 1314 // AudioBufferProvider interface 1315 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer); 1316 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 1317 private: 1318 RecordTrack * const mRecordTrack; 1319 size_t mRsmpInUnrel; // unreleased frames remaining from 1320 // most recent getNextBuffer 1321 // for debug only 1322 int32_t mRsmpInFront; // next available frame 1323 // rolling counter that is never cleared 1324 }; 1325 1326 /* The RecordBufferConverter is used for format, channel, and sample rate 1327 * conversion for a RecordTrack. 1328 * 1329 * TODO: Self contained, so move to a separate file later. 1330 * 1331 * RecordBufferConverter uses the convert() method rather than exposing a 1332 * buffer provider interface; this is to save a memory copy. 1333 */ 1334 class RecordBufferConverter 1335 { 1336 public: 1337 RecordBufferConverter( 1338 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 1339 uint32_t srcSampleRate, 1340 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 1341 uint32_t dstSampleRate); 1342 1343 ~RecordBufferConverter(); 1344 1345 /* Converts input data from an AudioBufferProvider by format, channelMask, 1346 * and sampleRate to a destination buffer. 1347 * 1348 * Parameters 1349 * dst: buffer to place the converted data. 1350 * provider: buffer provider to obtain source data. 1351 * frames: number of frames to convert 1352 * 1353 * Returns the number of frames converted. 1354 */ 1355 size_t convert(void *dst, AudioBufferProvider *provider, size_t frames); 1356 1357 // returns NO_ERROR if constructor was successful 1358 status_t initCheck() const { 1359 // mSrcChannelMask set on successful updateParameters 1360 return mSrcChannelMask != AUDIO_CHANNEL_INVALID ? NO_ERROR : NO_INIT; 1361 } 1362 1363 // allows dynamic reconfigure of all parameters 1364 status_t updateParameters( 1365 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 1366 uint32_t srcSampleRate, 1367 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 1368 uint32_t dstSampleRate); 1369 1370 // called to reset resampler buffers on record track discontinuity 1371 void reset() { 1372 if (mResampler != NULL) { 1373 mResampler->reset(); 1374 } 1375 } 1376 1377 private: 1378 // format conversion when not using resampler 1379 void convertNoResampler(void *dst, const void *src, size_t frames); 1380 1381 // format conversion when using resampler; modifies src in-place 1382 void convertResampler(void *dst, /*not-a-const*/ void *src, size_t frames); 1383 1384 // user provided information 1385 audio_channel_mask_t mSrcChannelMask; 1386 audio_format_t mSrcFormat; 1387 uint32_t mSrcSampleRate; 1388 audio_channel_mask_t mDstChannelMask; 1389 audio_format_t mDstFormat; 1390 uint32_t mDstSampleRate; 1391 1392 // derived information 1393 uint32_t mSrcChannelCount; 1394 uint32_t mDstChannelCount; 1395 size_t mDstFrameSize; 1396 1397 // format conversion buffer 1398 void *mBuf; 1399 size_t mBufFrames; 1400 size_t mBufFrameSize; 1401 1402 // resampler info 1403 AudioResampler *mResampler; 1404 1405 bool mIsLegacyDownmix; // legacy stereo to mono conversion needed 1406 bool mIsLegacyUpmix; // legacy mono to stereo conversion needed 1407 bool mRequiresFloat; // data processing requires float (e.g. resampler) 1408 PassthruBufferProvider *mInputConverterProvider; // converts input to float 1409 int8_t mIdxAry[sizeof(uint32_t) * 8]; // used for channel mask conversion 1410 }; 1411 1412#include "RecordTracks.h" 1413 1414 RecordThread(const sp<AudioFlinger>& audioFlinger, 1415 AudioStreamIn *input, 1416 audio_io_handle_t id, 1417 audio_devices_t outDevice, 1418 audio_devices_t inDevice, 1419 bool systemReady 1420#ifdef TEE_SINK 1421 , const sp<NBAIO_Sink>& teeSink 1422#endif 1423 ); 1424 virtual ~RecordThread(); 1425 1426 // no addTrack_l ? 1427 void destroyTrack_l(const sp<RecordTrack>& track); 1428 void removeTrack_l(const sp<RecordTrack>& track); 1429 1430 void dumpInternals(int fd, const Vector<String16>& args); 1431 void dumpTracks(int fd, const Vector<String16>& args); 1432 1433 // Thread virtuals 1434 virtual bool threadLoop(); 1435 1436 // RefBase 1437 virtual void onFirstRef(); 1438 1439 virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; } 1440 1441 virtual sp<MemoryDealer> readOnlyHeap() const { return mReadOnlyHeap; } 1442 1443 virtual sp<IMemory> pipeMemory() const { return mPipeMemory; } 1444 1445 sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l( 1446 const sp<AudioFlinger::Client>& client, 1447 uint32_t sampleRate, 1448 audio_format_t format, 1449 audio_channel_mask_t channelMask, 1450 size_t *pFrameCount, 1451 audio_session_t sessionId, 1452 size_t *notificationFrames, 1453 uid_t uid, 1454 audio_input_flags_t *flags, 1455 pid_t tid, 1456 status_t *status /*non-NULL*/, 1457 audio_port_handle_t portId); 1458 1459 status_t start(RecordTrack* recordTrack, 1460 AudioSystem::sync_event_t event, 1461 audio_session_t triggerSession); 1462 1463 // ask the thread to stop the specified track, and 1464 // return true if the caller should then do it's part of the stopping process 1465 bool stop(RecordTrack* recordTrack); 1466 1467 void dump(int fd, const Vector<String16>& args); 1468 AudioStreamIn* clearInput(); 1469 virtual sp<StreamHalInterface> stream() const; 1470 1471 1472 virtual bool checkForNewParameter_l(const String8& keyValuePair, 1473 status_t& status); 1474 virtual void cacheParameters_l() {} 1475 virtual String8 getParameters(const String8& keys); 1476 virtual void ioConfigChanged(audio_io_config_event event, pid_t pid = 0); 1477 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 1478 audio_patch_handle_t *handle); 1479 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 1480 1481 void addPatchRecord(const sp<PatchRecord>& record); 1482 void deletePatchRecord(const sp<PatchRecord>& record); 1483 1484 void readInputParameters_l(); 1485 virtual uint32_t getInputFramesLost(); 1486 1487 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1488 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1489 virtual uint32_t hasAudioSession_l(audio_session_t sessionId) const; 1490 1491 // Return the set of unique session IDs across all tracks. 1492 // The keys are the session IDs, and the associated values are meaningless. 1493 // FIXME replace by Set [and implement Bag/Multiset for other uses]. 1494 KeyedVector<audio_session_t, bool> sessionIds() const; 1495 1496 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 1497 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 1498 1499 static void syncStartEventCallback(const wp<SyncEvent>& event); 1500 1501 virtual size_t frameCount() const { return mFrameCount; } 1502 bool hasFastCapture() const { return mFastCapture != 0; } 1503 virtual void getAudioPortConfig(struct audio_port_config *config); 1504 1505 virtual status_t checkEffectCompatibility_l(const effect_descriptor_t *desc, 1506 audio_session_t sessionId); 1507 1508 virtual void acquireWakeLock_l() { 1509 ThreadBase::acquireWakeLock_l(); 1510 mActiveTracks.updatePowerState(this, true /* force */); 1511 } 1512 1513private: 1514 // Enter standby if not already in standby, and set mStandby flag 1515 void standbyIfNotAlreadyInStandby(); 1516 1517 // Call the HAL standby method unconditionally, and don't change mStandby flag 1518 void inputStandBy(); 1519 1520 AudioStreamIn *mInput; 1521 SortedVector < sp<RecordTrack> > mTracks; 1522 // mActiveTracks has dual roles: it indicates the current active track(s), and 1523 // is used together with mStartStopCond to indicate start()/stop() progress 1524 ActiveTracks<RecordTrack> mActiveTracks; 1525 1526 Condition mStartStopCond; 1527 1528 // resampler converts input at HAL Hz to output at AudioRecord client Hz 1529 void *mRsmpInBuffer; // size = mRsmpInFramesOA 1530 size_t mRsmpInFrames; // size of resampler input in frames 1531 size_t mRsmpInFramesP2;// size rounded up to a power-of-2 1532 size_t mRsmpInFramesOA;// mRsmpInFramesP2 + over-allocation 1533 1534 // rolling index that is never cleared 1535 int32_t mRsmpInRear; // last filled frame + 1 1536 1537 // For dumpsys 1538 const sp<NBAIO_Sink> mTeeSink; 1539 1540 const sp<MemoryDealer> mReadOnlyHeap; 1541 1542 // one-time initialization, no locks required 1543 sp<FastCapture> mFastCapture; // non-0 if there is also 1544 // a fast capture 1545 1546 // FIXME audio watchdog thread 1547 1548 // contents are not guaranteed to be consistent, no locks required 1549 FastCaptureDumpState mFastCaptureDumpState; 1550#ifdef STATE_QUEUE_DUMP 1551 // FIXME StateQueue observer and mutator dump fields 1552#endif 1553 // FIXME audio watchdog dump 1554 1555 // accessible only within the threadLoop(), no locks required 1556 // mFastCapture->sq() // for mutating and pushing state 1557 int32_t mFastCaptureFutex; // for cold idle 1558 1559 // The HAL input source is treated as non-blocking, 1560 // but current implementation is blocking 1561 sp<NBAIO_Source> mInputSource; 1562 // The source for the normal capture thread to read from: mInputSource or mPipeSource 1563 sp<NBAIO_Source> mNormalSource; 1564 // If a fast capture is present, the non-blocking pipe sink written to by fast capture, 1565 // otherwise clear 1566 sp<NBAIO_Sink> mPipeSink; 1567 // If a fast capture is present, the non-blocking pipe source read by normal thread, 1568 // otherwise clear 1569 sp<NBAIO_Source> mPipeSource; 1570 // Depth of pipe from fast capture to normal thread and fast clients, always power of 2 1571 size_t mPipeFramesP2; 1572 // If a fast capture is present, the Pipe as IMemory, otherwise clear 1573 sp<IMemory> mPipeMemory; 1574 1575 static const size_t kFastCaptureLogSize = 4 * 1024; 1576 sp<NBLog::Writer> mFastCaptureNBLogWriter; 1577 1578 bool mFastTrackAvail; // true if fast track available 1579}; 1580