Threads.h revision 6acd1d432f526ae9a055ddaece28bf93b474a776
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 MMAP // control thread for MMAP stream 34 }; 35 36 static const char *threadTypeToString(type_t type); 37 38 ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 39 audio_devices_t outDevice, audio_devices_t inDevice, type_t type, 40 bool systemReady); 41 virtual ~ThreadBase(); 42 43 virtual status_t readyToRun(); 44 45 void dumpBase(int fd, const Vector<String16>& args); 46 void dumpEffectChains(int fd, const Vector<String16>& args); 47 48 void clearPowerManager(); 49 50 // base for record and playback 51 enum { 52 CFG_EVENT_IO, 53 CFG_EVENT_PRIO, 54 CFG_EVENT_SET_PARAMETER, 55 CFG_EVENT_CREATE_AUDIO_PATCH, 56 CFG_EVENT_RELEASE_AUDIO_PATCH, 57 }; 58 59 class ConfigEventData: public RefBase { 60 public: 61 virtual ~ConfigEventData() {} 62 63 virtual void dump(char *buffer, size_t size) = 0; 64 protected: 65 ConfigEventData() {} 66 }; 67 68 // Config event sequence by client if status needed (e.g binder thread calling setParameters()): 69 // 1. create SetParameterConfigEvent. This sets mWaitStatus in config event 70 // 2. Lock mLock 71 // 3. Call sendConfigEvent_l(): Append to mConfigEvents and mWaitWorkCV.signal 72 // 4. sendConfigEvent_l() reads status from event->mStatus; 73 // 5. sendConfigEvent_l() returns status 74 // 6. Unlock 75 // 76 // Parameter sequence by server: threadLoop calling processConfigEvents_l(): 77 // 1. Lock mLock 78 // 2. If there is an entry in mConfigEvents proceed ... 79 // 3. Read first entry in mConfigEvents 80 // 4. Remove first entry from mConfigEvents 81 // 5. Process 82 // 6. Set event->mStatus 83 // 7. event->mCond.signal 84 // 8. Unlock 85 86 class ConfigEvent: public RefBase { 87 public: 88 virtual ~ConfigEvent() {} 89 90 void dump(char *buffer, size_t size) { mData->dump(buffer, size); } 91 92 const int mType; // event type e.g. CFG_EVENT_IO 93 Mutex mLock; // mutex associated with mCond 94 Condition mCond; // condition for status return 95 status_t mStatus; // status communicated to sender 96 bool mWaitStatus; // true if sender is waiting for status 97 bool mRequiresSystemReady; // true if must wait for system ready to enter event queue 98 sp<ConfigEventData> mData; // event specific parameter data 99 100 protected: 101 explicit ConfigEvent(int type, bool requiresSystemReady = false) : 102 mType(type), mStatus(NO_ERROR), mWaitStatus(false), 103 mRequiresSystemReady(requiresSystemReady), mData(NULL) {} 104 }; 105 106 class IoConfigEventData : public ConfigEventData { 107 public: 108 IoConfigEventData(audio_io_config_event event, pid_t pid) : 109 mEvent(event), mPid(pid) {} 110 111 virtual void dump(char *buffer, size_t size) { 112 snprintf(buffer, size, "IO event: event %d\n", mEvent); 113 } 114 115 const audio_io_config_event mEvent; 116 const pid_t mPid; 117 }; 118 119 class IoConfigEvent : public ConfigEvent { 120 public: 121 IoConfigEvent(audio_io_config_event event, pid_t pid) : 122 ConfigEvent(CFG_EVENT_IO) { 123 mData = new IoConfigEventData(event, pid); 124 } 125 virtual ~IoConfigEvent() {} 126 }; 127 128 class PrioConfigEventData : public ConfigEventData { 129 public: 130 PrioConfigEventData(pid_t pid, pid_t tid, int32_t prio) : 131 mPid(pid), mTid(tid), mPrio(prio) {} 132 133 virtual void dump(char *buffer, size_t size) { 134 snprintf(buffer, size, "Prio event: pid %d, tid %d, prio %d\n", mPid, mTid, mPrio); 135 } 136 137 const pid_t mPid; 138 const pid_t mTid; 139 const int32_t mPrio; 140 }; 141 142 class PrioConfigEvent : public ConfigEvent { 143 public: 144 PrioConfigEvent(pid_t pid, pid_t tid, int32_t prio) : 145 ConfigEvent(CFG_EVENT_PRIO, true) { 146 mData = new PrioConfigEventData(pid, tid, prio); 147 } 148 virtual ~PrioConfigEvent() {} 149 }; 150 151 class SetParameterConfigEventData : public ConfigEventData { 152 public: 153 explicit SetParameterConfigEventData(String8 keyValuePairs) : 154 mKeyValuePairs(keyValuePairs) {} 155 156 virtual void dump(char *buffer, size_t size) { 157 snprintf(buffer, size, "KeyValue: %s\n", mKeyValuePairs.string()); 158 } 159 160 const String8 mKeyValuePairs; 161 }; 162 163 class SetParameterConfigEvent : public ConfigEvent { 164 public: 165 explicit SetParameterConfigEvent(String8 keyValuePairs) : 166 ConfigEvent(CFG_EVENT_SET_PARAMETER) { 167 mData = new SetParameterConfigEventData(keyValuePairs); 168 mWaitStatus = true; 169 } 170 virtual ~SetParameterConfigEvent() {} 171 }; 172 173 class CreateAudioPatchConfigEventData : public ConfigEventData { 174 public: 175 CreateAudioPatchConfigEventData(const struct audio_patch patch, 176 audio_patch_handle_t handle) : 177 mPatch(patch), mHandle(handle) {} 178 179 virtual void dump(char *buffer, size_t size) { 180 snprintf(buffer, size, "Patch handle: %u\n", mHandle); 181 } 182 183 const struct audio_patch mPatch; 184 audio_patch_handle_t mHandle; 185 }; 186 187 class CreateAudioPatchConfigEvent : public ConfigEvent { 188 public: 189 CreateAudioPatchConfigEvent(const struct audio_patch patch, 190 audio_patch_handle_t handle) : 191 ConfigEvent(CFG_EVENT_CREATE_AUDIO_PATCH) { 192 mData = new CreateAudioPatchConfigEventData(patch, handle); 193 mWaitStatus = true; 194 } 195 virtual ~CreateAudioPatchConfigEvent() {} 196 }; 197 198 class ReleaseAudioPatchConfigEventData : public ConfigEventData { 199 public: 200 explicit ReleaseAudioPatchConfigEventData(const audio_patch_handle_t handle) : 201 mHandle(handle) {} 202 203 virtual void dump(char *buffer, size_t size) { 204 snprintf(buffer, size, "Patch handle: %u\n", mHandle); 205 } 206 207 audio_patch_handle_t mHandle; 208 }; 209 210 class ReleaseAudioPatchConfigEvent : public ConfigEvent { 211 public: 212 explicit ReleaseAudioPatchConfigEvent(const audio_patch_handle_t handle) : 213 ConfigEvent(CFG_EVENT_RELEASE_AUDIO_PATCH) { 214 mData = new ReleaseAudioPatchConfigEventData(handle); 215 mWaitStatus = true; 216 } 217 virtual ~ReleaseAudioPatchConfigEvent() {} 218 }; 219 220 class PMDeathRecipient : public IBinder::DeathRecipient { 221 public: 222 explicit PMDeathRecipient(const wp<ThreadBase>& thread) : mThread(thread) {} 223 virtual ~PMDeathRecipient() {} 224 225 // IBinder::DeathRecipient 226 virtual void binderDied(const wp<IBinder>& who); 227 228 private: 229 PMDeathRecipient(const PMDeathRecipient&); 230 PMDeathRecipient& operator = (const PMDeathRecipient&); 231 232 wp<ThreadBase> mThread; 233 }; 234 235 virtual status_t initCheck() const = 0; 236 237 // static externally-visible 238 type_t type() const { return mType; } 239 bool isDuplicating() const { return (mType == DUPLICATING); } 240 241 audio_io_handle_t id() const { return mId;} 242 243 // dynamic externally-visible 244 uint32_t sampleRate() const { return mSampleRate; } 245 audio_channel_mask_t channelMask() const { return mChannelMask; } 246 audio_format_t format() const { return mHALFormat; } 247 uint32_t channelCount() const { return mChannelCount; } 248 // Called by AudioFlinger::frameCount(audio_io_handle_t output) and effects, 249 // and returns the [normal mix] buffer's frame count. 250 virtual size_t frameCount() const = 0; 251 252 // Return's the HAL's frame count i.e. fast mixer buffer size. 253 size_t frameCountHAL() const { return mFrameCount; } 254 255 size_t frameSize() const { return mFrameSize; } 256 257 // Should be "virtual status_t requestExitAndWait()" and override same 258 // method in Thread, but Thread::requestExitAndWait() is not yet virtual. 259 void exit(); 260 virtual bool checkForNewParameter_l(const String8& keyValuePair, 261 status_t& status) = 0; 262 virtual status_t setParameters(const String8& keyValuePairs); 263 virtual String8 getParameters(const String8& keys) = 0; 264 virtual void ioConfigChanged(audio_io_config_event event, pid_t pid = 0) = 0; 265 // sendConfigEvent_l() must be called with ThreadBase::mLock held 266 // Can temporarily release the lock if waiting for a reply from 267 // processConfigEvents_l(). 268 status_t sendConfigEvent_l(sp<ConfigEvent>& event); 269 void sendIoConfigEvent(audio_io_config_event event, pid_t pid = 0); 270 void sendIoConfigEvent_l(audio_io_config_event event, pid_t pid = 0); 271 void sendPrioConfigEvent(pid_t pid, pid_t tid, int32_t prio); 272 void sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio); 273 status_t sendSetParameterConfigEvent_l(const String8& keyValuePair); 274 status_t sendCreateAudioPatchConfigEvent(const struct audio_patch *patch, 275 audio_patch_handle_t *handle); 276 status_t sendReleaseAudioPatchConfigEvent(audio_patch_handle_t handle); 277 void processConfigEvents_l(); 278 virtual void cacheParameters_l() = 0; 279 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 280 audio_patch_handle_t *handle) = 0; 281 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle) = 0; 282 virtual void getAudioPortConfig(struct audio_port_config *config) = 0; 283 284 285 // see note at declaration of mStandby, mOutDevice and mInDevice 286 bool standby() const { return mStandby; } 287 audio_devices_t outDevice() const { return mOutDevice; } 288 audio_devices_t inDevice() const { return mInDevice; } 289 290 virtual sp<StreamHalInterface> stream() const = 0; 291 292 sp<EffectHandle> createEffect_l( 293 const sp<AudioFlinger::Client>& client, 294 const sp<IEffectClient>& effectClient, 295 int32_t priority, 296 audio_session_t sessionId, 297 effect_descriptor_t *desc, 298 int *enabled, 299 status_t *status /*non-NULL*/, 300 bool pinned); 301 302 // return values for hasAudioSession (bit field) 303 enum effect_state { 304 EFFECT_SESSION = 0x1, // the audio session corresponds to at least one 305 // effect 306 TRACK_SESSION = 0x2, // the audio session corresponds to at least one 307 // track 308 FAST_SESSION = 0x4 // the audio session corresponds to at least one 309 // fast track 310 }; 311 312 // get effect chain corresponding to session Id. 313 sp<EffectChain> getEffectChain(audio_session_t sessionId); 314 // same as getEffectChain() but must be called with ThreadBase mutex locked 315 sp<EffectChain> getEffectChain_l(audio_session_t sessionId) const; 316 // add an effect chain to the chain list (mEffectChains) 317 virtual status_t addEffectChain_l(const sp<EffectChain>& chain) = 0; 318 // remove an effect chain from the chain list (mEffectChains) 319 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain) = 0; 320 // lock all effect chains Mutexes. Must be called before releasing the 321 // ThreadBase mutex before processing the mixer and effects. This guarantees the 322 // integrity of the chains during the process. 323 // Also sets the parameter 'effectChains' to current value of mEffectChains. 324 void lockEffectChains_l(Vector< sp<EffectChain> >& effectChains); 325 // unlock effect chains after process 326 void unlockEffectChains(const Vector< sp<EffectChain> >& effectChains); 327 // get a copy of mEffectChains vector 328 Vector< sp<EffectChain> > getEffectChains_l() const { return mEffectChains; }; 329 // set audio mode to all effect chains 330 void setMode(audio_mode_t mode); 331 // get effect module with corresponding ID on specified audio session 332 sp<AudioFlinger::EffectModule> getEffect(audio_session_t sessionId, int effectId); 333 sp<AudioFlinger::EffectModule> getEffect_l(audio_session_t sessionId, int effectId); 334 // add and effect module. Also creates the effect chain is none exists for 335 // the effects audio session 336 status_t addEffect_l(const sp< EffectModule>& effect); 337 // remove and effect module. Also removes the effect chain is this was the last 338 // effect 339 void removeEffect_l(const sp< EffectModule>& effect, bool release = false); 340 // disconnect an effect handle from module and destroy module if last handle 341 void disconnectEffectHandle(EffectHandle *handle, bool unpinIfLast); 342 // detach all tracks connected to an auxiliary effect 343 virtual void detachAuxEffect_l(int effectId __unused) {} 344 // returns a combination of: 345 // - EFFECT_SESSION if effects on this audio session exist in one chain 346 // - TRACK_SESSION if tracks on this audio session exist 347 // - FAST_SESSION if fast tracks on this audio session exist 348 virtual uint32_t hasAudioSession_l(audio_session_t sessionId) const = 0; 349 uint32_t hasAudioSession(audio_session_t sessionId) const { 350 Mutex::Autolock _l(mLock); 351 return hasAudioSession_l(sessionId); 352 } 353 354 // the value returned by default implementation is not important as the 355 // strategy is only meaningful for PlaybackThread which implements this method 356 virtual uint32_t getStrategyForSession_l(audio_session_t sessionId __unused) 357 { return 0; } 358 359 // suspend or restore effect according to the type of effect passed. a NULL 360 // type pointer means suspend all effects in the session 361 void setEffectSuspended(const effect_uuid_t *type, 362 bool suspend, 363 audio_session_t sessionId = AUDIO_SESSION_OUTPUT_MIX); 364 // check if some effects must be suspended/restored when an effect is enabled 365 // or disabled 366 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 367 bool enabled, 368 audio_session_t sessionId = 369 AUDIO_SESSION_OUTPUT_MIX); 370 void checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 371 bool enabled, 372 audio_session_t sessionId = 373 AUDIO_SESSION_OUTPUT_MIX); 374 375 virtual status_t setSyncEvent(const sp<SyncEvent>& event) = 0; 376 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const = 0; 377 378 // Return a reference to a per-thread heap which can be used to allocate IMemory 379 // objects that will be read-only to client processes, read/write to mediaserver, 380 // and shared by all client processes of the thread. 381 // The heap is per-thread rather than common across all threads, because 382 // clients can't be trusted not to modify the offset of the IMemory they receive. 383 // If a thread does not have such a heap, this method returns 0. 384 virtual sp<MemoryDealer> readOnlyHeap() const { return 0; } 385 386 virtual sp<IMemory> pipeMemory() const { return 0; } 387 388 void systemReady(); 389 390 // checkEffectCompatibility_l() must be called with ThreadBase::mLock held 391 virtual status_t checkEffectCompatibility_l(const effect_descriptor_t *desc, 392 audio_session_t sessionId) = 0; 393 394 void broadcast_l(); 395 396 mutable Mutex mLock; 397 398protected: 399 400 // entry describing an effect being suspended in mSuspendedSessions keyed vector 401 class SuspendedSessionDesc : public RefBase { 402 public: 403 SuspendedSessionDesc() : mRefCount(0) {} 404 405 int mRefCount; // number of active suspend requests 406 effect_uuid_t mType; // effect type UUID 407 }; 408 409 void acquireWakeLock(); 410 virtual void acquireWakeLock_l(); 411 void releaseWakeLock(); 412 void releaseWakeLock_l(); 413 void updateWakeLockUids_l(const SortedVector<uid_t> &uids); 414 void getPowerManager_l(); 415 void setEffectSuspended_l(const effect_uuid_t *type, 416 bool suspend, 417 audio_session_t sessionId); 418 // updated mSuspendedSessions when an effect suspended or restored 419 void updateSuspendedSessions_l(const effect_uuid_t *type, 420 bool suspend, 421 audio_session_t sessionId); 422 // check if some effects must be suspended when an effect chain is added 423 void checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain); 424 425 String16 getWakeLockTag(); 426 427 virtual void preExit() { } 428 virtual void setMasterMono_l(bool mono __unused) { } 429 virtual bool requireMonoBlend() { return false; } 430 431 friend class AudioFlinger; // for mEffectChains 432 433 const type_t mType; 434 435 // Used by parameters, config events, addTrack_l, exit 436 Condition mWaitWorkCV; 437 438 const sp<AudioFlinger> mAudioFlinger; 439 440 // updated by PlaybackThread::readOutputParameters_l() or 441 // RecordThread::readInputParameters_l() 442 uint32_t mSampleRate; 443 size_t mFrameCount; // output HAL, direct output, record 444 audio_channel_mask_t mChannelMask; 445 uint32_t mChannelCount; 446 size_t mFrameSize; 447 // not HAL frame size, this is for output sink (to pipe to fast mixer) 448 audio_format_t mFormat; // Source format for Recording and 449 // Sink format for Playback. 450 // Sink format may be different than 451 // HAL format if Fastmixer is used. 452 audio_format_t mHALFormat; 453 size_t mBufferSize; // HAL buffer size for read() or write() 454 455 Vector< sp<ConfigEvent> > mConfigEvents; 456 Vector< sp<ConfigEvent> > mPendingConfigEvents; // events awaiting system ready 457 458 // These fields are written and read by thread itself without lock or barrier, 459 // and read by other threads without lock or barrier via standby(), outDevice() 460 // and inDevice(). 461 // Because of the absence of a lock or barrier, any other thread that reads 462 // these fields must use the information in isolation, or be prepared to deal 463 // with possibility that it might be inconsistent with other information. 464 bool mStandby; // Whether thread is currently in standby. 465 audio_devices_t mOutDevice; // output device 466 audio_devices_t mInDevice; // input device 467 audio_devices_t mPrevOutDevice; // previous output device 468 audio_devices_t mPrevInDevice; // previous input device 469 struct audio_patch mPatch; 470 audio_source_t mAudioSource; 471 472 const audio_io_handle_t mId; 473 Vector< sp<EffectChain> > mEffectChains; 474 475 static const int kThreadNameLength = 16; // prctl(PR_SET_NAME) limit 476 char mThreadName[kThreadNameLength]; // guaranteed NUL-terminated 477 sp<IPowerManager> mPowerManager; 478 sp<IBinder> mWakeLockToken; 479 const sp<PMDeathRecipient> mDeathRecipient; 480 // list of suspended effects per session and per type. The first (outer) vector is 481 // keyed by session ID, the second (inner) by type UUID timeLow field 482 KeyedVector< audio_session_t, KeyedVector< int, sp<SuspendedSessionDesc> > > 483 mSuspendedSessions; 484 static const size_t kLogSize = 4 * 1024; 485 sp<NBLog::Writer> mNBLogWriter; 486 bool mSystemReady; 487 ExtendedTimestamp mTimestamp; 488 // A condition that must be evaluated by the thread loop has changed and 489 // we must not wait for async write callback in the thread loop before evaluating it 490 bool mSignalPending; 491 492 // ActiveTracks is a sorted vector of track type T representing the 493 // active tracks of threadLoop() to be considered by the locked prepare portion. 494 // ActiveTracks should be accessed with the ThreadBase lock held. 495 // 496 // During processing and I/O, the threadLoop does not hold the lock; 497 // hence it does not directly use ActiveTracks. Care should be taken 498 // to hold local strong references or defer removal of tracks 499 // if the threadLoop may still be accessing those tracks due to mix, etc. 500 // 501 // This class updates power information appropriately. 502 // 503 504 template <typename T> 505 class ActiveTracks { 506 public: 507 ActiveTracks() 508 : mActiveTracksGeneration(0) 509 , mLastActiveTracksGeneration(0) 510 { } 511 512 ~ActiveTracks() { 513 ALOGW_IF(!mActiveTracks.isEmpty(), 514 "ActiveTracks should be empty in destructor"); 515 } 516 // returns the last track added (even though it may have been 517 // subsequently removed from ActiveTracks). 518 // 519 // Used for DirectOutputThread to ensure a flush is called when transitioning 520 // to a new track (even though it may be on the same session). 521 // Used for OffloadThread to ensure that volume and mixer state is 522 // taken from the latest track added. 523 // 524 // The latest track is saved with a weak pointer to prevent keeping an 525 // otherwise useless track alive. Thus the function will return nullptr 526 // if the latest track has subsequently been removed and destroyed. 527 sp<T> getLatest() { 528 return mLatestActiveTrack.promote(); 529 } 530 531 // SortedVector methods 532 ssize_t add(const sp<T> &track); 533 ssize_t remove(const sp<T> &track); 534 size_t size() const { 535 return mActiveTracks.size(); 536 } 537 ssize_t indexOf(const sp<T>& item) { 538 return mActiveTracks.indexOf(item); 539 } 540 sp<T> operator[](size_t index) const { 541 return mActiveTracks[index]; 542 } 543 typename SortedVector<sp<T>>::iterator begin() { 544 return mActiveTracks.begin(); 545 } 546 typename SortedVector<sp<T>>::iterator end() { 547 return mActiveTracks.end(); 548 } 549 550 // Due to Binder recursion optimization, clear() and updatePowerState() 551 // cannot be called from a Binder thread because they may call back into 552 // the original calling process (system server) for BatteryNotifier 553 // (which requires a Java environment that may not be present). 554 // Hence, call clear() and updatePowerState() only from the 555 // ThreadBase thread. 556 void clear(); 557 // periodically called in the threadLoop() to update power state uids. 558 void updatePowerState(sp<ThreadBase> thread, bool force = false); 559 560 private: 561 SortedVector<uid_t> getWakeLockUids() { 562 SortedVector<uid_t> wakeLockUids; 563 for (const sp<T> &track : mActiveTracks) { 564 wakeLockUids.add(track->uid()); 565 } 566 return wakeLockUids; // moved by underlying SharedBuffer 567 } 568 569 std::map<uid_t, std::pair<ssize_t /* previous */, ssize_t /* current */>> 570 mBatteryCounter; 571 SortedVector<sp<T>> mActiveTracks; 572 int mActiveTracksGeneration; 573 int mLastActiveTracksGeneration; 574 wp<T> mLatestActiveTrack; // latest track added to ActiveTracks 575 }; 576}; 577 578class VolumeInterface { 579 public: 580 581 virtual ~VolumeInterface() {} 582 583 virtual void setMasterVolume(float value) = 0; 584 virtual void setMasterMute(bool muted) = 0; 585 virtual void setStreamVolume(audio_stream_type_t stream, float value) = 0; 586 virtual void setStreamMute(audio_stream_type_t stream, bool muted) = 0; 587 virtual float streamVolume(audio_stream_type_t stream) const = 0; 588 589}; 590 591// --- PlaybackThread --- 592class PlaybackThread : public ThreadBase, public StreamOutHalInterfaceCallback, 593 public VolumeInterface { 594public: 595 596#include "PlaybackTracks.h" 597 598 enum mixer_state { 599 MIXER_IDLE, // no active tracks 600 MIXER_TRACKS_ENABLED, // at least one active track, but no track has any data ready 601 MIXER_TRACKS_READY, // at least one active track, and at least one track has data 602 MIXER_DRAIN_TRACK, // drain currently playing track 603 MIXER_DRAIN_ALL, // fully drain the hardware 604 // standby mode does not have an enum value 605 // suspend by audio policy manager is orthogonal to mixer state 606 }; 607 608 // retry count before removing active track in case of underrun on offloaded thread: 609 // we need to make sure that AudioTrack client has enough time to send large buffers 610 //FIXME may be more appropriate if expressed in time units. Need to revise how underrun is 611 // handled for offloaded tracks 612 static const int8_t kMaxTrackRetriesOffload = 20; 613 static const int8_t kMaxTrackStartupRetriesOffload = 100; 614 static const int8_t kMaxTrackStopRetriesOffload = 2; 615 // 14 tracks max per client allows for 2 misbehaving application leaving 4 available tracks. 616 static const uint32_t kMaxTracksPerUid = 14; 617 618 PlaybackThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 619 audio_io_handle_t id, audio_devices_t device, type_t type, bool systemReady); 620 virtual ~PlaybackThread(); 621 622 void dump(int fd, const Vector<String16>& args); 623 624 // Thread virtuals 625 virtual bool threadLoop(); 626 627 // RefBase 628 virtual void onFirstRef(); 629 630 virtual status_t checkEffectCompatibility_l(const effect_descriptor_t *desc, 631 audio_session_t sessionId); 632 633protected: 634 // Code snippets that were lifted up out of threadLoop() 635 virtual void threadLoop_mix() = 0; 636 virtual void threadLoop_sleepTime() = 0; 637 virtual ssize_t threadLoop_write(); 638 virtual void threadLoop_drain(); 639 virtual void threadLoop_standby(); 640 virtual void threadLoop_exit(); 641 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 642 643 // prepareTracks_l reads and writes mActiveTracks, and returns 644 // the pending set of tracks to remove via Vector 'tracksToRemove'. The caller 645 // is responsible for clearing or destroying this Vector later on, when it 646 // is safe to do so. That will drop the final ref count and destroy the tracks. 647 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove) = 0; 648 void removeTracks_l(const Vector< sp<Track> >& tracksToRemove); 649 650 // StreamOutHalInterfaceCallback implementation 651 virtual void onWriteReady(); 652 virtual void onDrainReady(); 653 virtual void onError(); 654 655 void resetWriteBlocked(uint32_t sequence); 656 void resetDraining(uint32_t sequence); 657 658 virtual bool waitingAsyncCallback(); 659 virtual bool waitingAsyncCallback_l(); 660 virtual bool shouldStandby_l(); 661 virtual void onAddNewTrack_l(); 662 void onAsyncError(); // error reported by AsyncCallbackThread 663 664 // ThreadBase virtuals 665 virtual void preExit(); 666 667 virtual bool keepWakeLock() const { return true; } 668 virtual void acquireWakeLock_l() { 669 ThreadBase::acquireWakeLock_l(); 670 mActiveTracks.updatePowerState(this, true /* force */); 671 } 672 673public: 674 675 virtual status_t initCheck() const { return (mOutput == NULL) ? NO_INIT : NO_ERROR; } 676 677 // return estimated latency in milliseconds, as reported by HAL 678 uint32_t latency() const; 679 // same, but lock must already be held 680 uint32_t latency_l() const; 681 682 // VolumeInterface 683 virtual void setMasterVolume(float value); 684 virtual void setMasterMute(bool muted); 685 virtual void setStreamVolume(audio_stream_type_t stream, float value); 686 virtual void setStreamMute(audio_stream_type_t stream, bool muted); 687 virtual float streamVolume(audio_stream_type_t stream) const; 688 689 sp<Track> createTrack_l( 690 const sp<AudioFlinger::Client>& client, 691 audio_stream_type_t streamType, 692 uint32_t sampleRate, 693 audio_format_t format, 694 audio_channel_mask_t channelMask, 695 size_t *pFrameCount, 696 const sp<IMemory>& sharedBuffer, 697 audio_session_t sessionId, 698 audio_output_flags_t *flags, 699 pid_t tid, 700 uid_t uid, 701 status_t *status /*non-NULL*/, 702 audio_port_handle_t portId); 703 704 AudioStreamOut* getOutput() const; 705 AudioStreamOut* clearOutput(); 706 virtual sp<StreamHalInterface> stream() const; 707 708 // a very large number of suspend() will eventually wraparound, but unlikely 709 void suspend() { (void) android_atomic_inc(&mSuspended); } 710 void restore() 711 { 712 // if restore() is done without suspend(), get back into 713 // range so that the next suspend() will operate correctly 714 if (android_atomic_dec(&mSuspended) <= 0) { 715 android_atomic_release_store(0, &mSuspended); 716 } 717 } 718 bool isSuspended() const 719 { return android_atomic_acquire_load(&mSuspended) > 0; } 720 721 virtual String8 getParameters(const String8& keys); 722 virtual void ioConfigChanged(audio_io_config_event event, pid_t pid = 0); 723 status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames); 724 // FIXME rename mixBuffer() to sinkBuffer() and remove int16_t* dependency. 725 // Consider also removing and passing an explicit mMainBuffer initialization 726 // parameter to AF::PlaybackThread::Track::Track(). 727 int16_t *mixBuffer() const { 728 return reinterpret_cast<int16_t *>(mSinkBuffer); }; 729 730 virtual void detachAuxEffect_l(int effectId); 731 status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track>& track, 732 int EffectId); 733 status_t attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track>& track, 734 int EffectId); 735 736 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 737 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 738 virtual uint32_t hasAudioSession_l(audio_session_t sessionId) const; 739 virtual uint32_t getStrategyForSession_l(audio_session_t sessionId); 740 741 742 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 743 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 744 745 // called with AudioFlinger lock held 746 bool invalidateTracks_l(audio_stream_type_t streamType); 747 virtual void invalidateTracks(audio_stream_type_t streamType); 748 749 virtual size_t frameCount() const { return mNormalFrameCount; } 750 751 status_t getTimestamp_l(AudioTimestamp& timestamp); 752 753 void addPatchTrack(const sp<PatchTrack>& track); 754 void deletePatchTrack(const sp<PatchTrack>& track); 755 756 virtual void getAudioPortConfig(struct audio_port_config *config); 757 758protected: 759 // updated by readOutputParameters_l() 760 size_t mNormalFrameCount; // normal mixer and effects 761 762 bool mThreadThrottle; // throttle the thread processing 763 uint32_t mThreadThrottleTimeMs; // throttle time for MIXER threads 764 uint32_t mThreadThrottleEndMs; // notify once per throttling 765 uint32_t mHalfBufferMs; // half the buffer size in milliseconds 766 767 void* mSinkBuffer; // frame size aligned sink buffer 768 769 // TODO: 770 // Rearrange the buffer info into a struct/class with 771 // clear, copy, construction, destruction methods. 772 // 773 // mSinkBuffer also has associated with it: 774 // 775 // mSinkBufferSize: Sink Buffer Size 776 // mFormat: Sink Buffer Format 777 778 // Mixer Buffer (mMixerBuffer*) 779 // 780 // In the case of floating point or multichannel data, which is not in the 781 // sink format, it is required to accumulate in a higher precision or greater channel count 782 // buffer before downmixing or data conversion to the sink buffer. 783 784 // Set to "true" to enable the Mixer Buffer otherwise mixer output goes to sink buffer. 785 bool mMixerBufferEnabled; 786 787 // Storage, 32 byte aligned (may make this alignment a requirement later). 788 // Due to constraints on mNormalFrameCount, the buffer size is a multiple of 16 frames. 789 void* mMixerBuffer; 790 791 // Size of mMixerBuffer in bytes: mNormalFrameCount * #channels * sampsize. 792 size_t mMixerBufferSize; 793 794 // The audio format of mMixerBuffer. Set to AUDIO_FORMAT_PCM_(FLOAT|16_BIT) only. 795 audio_format_t mMixerBufferFormat; 796 797 // An internal flag set to true by MixerThread::prepareTracks_l() 798 // when mMixerBuffer contains valid data after mixing. 799 bool mMixerBufferValid; 800 801 // Effects Buffer (mEffectsBuffer*) 802 // 803 // In the case of effects data, which is not in the sink format, 804 // it is required to accumulate in a different buffer before data conversion 805 // to the sink buffer. 806 807 // Set to "true" to enable the Effects Buffer otherwise effects output goes to sink buffer. 808 bool mEffectBufferEnabled; 809 810 // Storage, 32 byte aligned (may make this alignment a requirement later). 811 // Due to constraints on mNormalFrameCount, the buffer size is a multiple of 16 frames. 812 void* mEffectBuffer; 813 814 // Size of mEffectsBuffer in bytes: mNormalFrameCount * #channels * sampsize. 815 size_t mEffectBufferSize; 816 817 // The audio format of mEffectsBuffer. Set to AUDIO_FORMAT_PCM_16_BIT only. 818 audio_format_t mEffectBufferFormat; 819 820 // An internal flag set to true by MixerThread::prepareTracks_l() 821 // when mEffectsBuffer contains valid data after mixing. 822 // 823 // When this is set, all mixer data is routed into the effects buffer 824 // for any processing (including output processing). 825 bool mEffectBufferValid; 826 827 // suspend count, > 0 means suspended. While suspended, the thread continues to pull from 828 // tracks and mix, but doesn't write to HAL. A2DP and SCO HAL implementations can't handle 829 // concurrent use of both of them, so Audio Policy Service suspends one of the threads to 830 // workaround that restriction. 831 // 'volatile' means accessed via atomic operations and no lock. 832 volatile int32_t mSuspended; 833 834 int64_t mBytesWritten; 835 int64_t mFramesWritten; // not reset on standby 836 int64_t mSuspendedFrames; // not reset on standby 837private: 838 // mMasterMute is in both PlaybackThread and in AudioFlinger. When a 839 // PlaybackThread needs to find out if master-muted, it checks it's local 840 // copy rather than the one in AudioFlinger. This optimization saves a lock. 841 bool mMasterMute; 842 void setMasterMute_l(bool muted) { mMasterMute = muted; } 843protected: 844 ActiveTracks<Track> mActiveTracks; 845 846 // Allocate a track name for a given channel mask. 847 // Returns name >= 0 if successful, -1 on failure. 848 virtual int getTrackName_l(audio_channel_mask_t channelMask, audio_format_t format, 849 audio_session_t sessionId, uid_t uid) = 0; 850 virtual void deleteTrackName_l(int name) = 0; 851 852 // Time to sleep between cycles when: 853 virtual uint32_t activeSleepTimeUs() const; // mixer state MIXER_TRACKS_ENABLED 854 virtual uint32_t idleSleepTimeUs() const = 0; // mixer state MIXER_IDLE 855 virtual uint32_t suspendSleepTimeUs() const = 0; // audio policy manager suspended us 856 // No sleep when mixer state == MIXER_TRACKS_READY; relies on audio HAL stream->write() 857 // No sleep in standby mode; waits on a condition 858 859 // Code snippets that are temporarily lifted up out of threadLoop() until the merge 860 void checkSilentMode_l(); 861 862 // Non-trivial for DUPLICATING only 863 virtual void saveOutputTracks() { } 864 virtual void clearOutputTracks() { } 865 866 // Cache various calculated values, at threadLoop() entry and after a parameter change 867 virtual void cacheParameters_l(); 868 869 virtual uint32_t correctLatency_l(uint32_t latency) const; 870 871 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 872 audio_patch_handle_t *handle); 873 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 874 875 bool usesHwAvSync() const { return (mType == DIRECT) && (mOutput != NULL) 876 && mHwSupportsPause 877 && (mOutput->flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC); } 878 879 uint32_t trackCountForUid_l(uid_t uid); 880 881private: 882 883 friend class AudioFlinger; // for numerous 884 885 PlaybackThread& operator = (const PlaybackThread&); 886 887 status_t addTrack_l(const sp<Track>& track); 888 bool destroyTrack_l(const sp<Track>& track); 889 void removeTrack_l(const sp<Track>& track); 890 891 void readOutputParameters_l(); 892 893 virtual void dumpInternals(int fd, const Vector<String16>& args); 894 void dumpTracks(int fd, const Vector<String16>& args); 895 896 SortedVector< sp<Track> > mTracks; 897 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 898 AudioStreamOut *mOutput; 899 900 float mMasterVolume; 901 nsecs_t mLastWriteTime; 902 int mNumWrites; 903 int mNumDelayedWrites; 904 bool mInWrite; 905 906 // FIXME rename these former local variables of threadLoop to standard "m" names 907 nsecs_t mStandbyTimeNs; 908 size_t mSinkBufferSize; 909 910 // cached copies of activeSleepTimeUs() and idleSleepTimeUs() made by cacheParameters_l() 911 uint32_t mActiveSleepTimeUs; 912 uint32_t mIdleSleepTimeUs; 913 914 uint32_t mSleepTimeUs; 915 916 // mixer status returned by prepareTracks_l() 917 mixer_state mMixerStatus; // current cycle 918 // previous cycle when in prepareTracks_l() 919 mixer_state mMixerStatusIgnoringFastTracks; 920 // FIXME or a separate ready state per track 921 922 // FIXME move these declarations into the specific sub-class that needs them 923 // MIXER only 924 uint32_t sleepTimeShift; 925 926 // same as AudioFlinger::mStandbyTimeInNsecs except for DIRECT which uses a shorter value 927 nsecs_t mStandbyDelayNs; 928 929 // MIXER only 930 nsecs_t maxPeriod; 931 932 // DUPLICATING only 933 uint32_t writeFrames; 934 935 size_t mBytesRemaining; 936 size_t mCurrentWriteLength; 937 bool mUseAsyncWrite; 938 // mWriteAckSequence contains current write sequence on bits 31-1. The write sequence is 939 // incremented each time a write(), a flush() or a standby() occurs. 940 // Bit 0 is set when a write blocks and indicates a callback is expected. 941 // Bit 0 is reset by the async callback thread calling resetWriteBlocked(). Out of sequence 942 // callbacks are ignored. 943 uint32_t mWriteAckSequence; 944 // mDrainSequence contains current drain sequence on bits 31-1. The drain sequence is 945 // incremented each time a drain is requested or a flush() or standby() occurs. 946 // Bit 0 is set when the drain() command is called at the HAL and indicates a callback is 947 // expected. 948 // Bit 0 is reset by the async callback thread calling resetDraining(). Out of sequence 949 // callbacks are ignored. 950 uint32_t mDrainSequence; 951 sp<AsyncCallbackThread> mCallbackThread; 952 953private: 954 // The HAL output sink is treated as non-blocking, but current implementation is blocking 955 sp<NBAIO_Sink> mOutputSink; 956 // If a fast mixer is present, the blocking pipe sink, otherwise clear 957 sp<NBAIO_Sink> mPipeSink; 958 // The current sink for the normal mixer to write it's (sub)mix, mOutputSink or mPipeSink 959 sp<NBAIO_Sink> mNormalSink; 960#ifdef TEE_SINK 961 // For dumpsys 962 sp<NBAIO_Sink> mTeeSink; 963 sp<NBAIO_Source> mTeeSource; 964#endif 965 uint32_t mScreenState; // cached copy of gScreenState 966 static const size_t kFastMixerLogSize = 4 * 1024; 967 sp<NBLog::Writer> mFastMixerNBLogWriter; 968 969 // Do not call from a sched_fifo thread as it uses a system time call 970 // and obtains a local mutex. 971 class LocalLog { 972 public: 973 void log(const char *fmt, ...) { 974 va_list val; 975 va_start(val, fmt); 976 977 // format to buffer 978 char buffer[512]; 979 int length = vsnprintf(buffer, sizeof(buffer), fmt, val); 980 if (length >= (signed)sizeof(buffer)) { 981 length = sizeof(buffer) - 1; 982 } 983 984 // strip out trailing newline 985 while (length > 0 && buffer[length - 1] == '\n') { 986 buffer[--length] = 0; 987 } 988 989 // store in circular array 990 AutoMutex _l(mLock); 991 mLog.emplace_back( 992 std::make_pair(systemTime(SYSTEM_TIME_REALTIME), std::string(buffer))); 993 if (mLog.size() > kLogSize) { 994 mLog.pop_front(); 995 } 996 997 va_end(val); 998 } 999 1000 void dump(int fd, const Vector<String16>& args, const char *prefix = "") { 1001 if (!AudioFlinger::dumpTryLock(mLock)) return; // a local lock, shouldn't happen 1002 if (mLog.size() > 0) { 1003 bool dumpAll = false; 1004 for (const auto &arg : args) { 1005 if (arg == String16("--locallog")) { 1006 dumpAll = true; 1007 } 1008 } 1009 1010 dprintf(fd, "Local Log:\n"); 1011 auto it = mLog.begin(); 1012 if (!dumpAll) { 1013 const size_t lines = 1014 (size_t)property_get_int32("audio.locallog.lines", kLogPrint); 1015 if (mLog.size() > lines) { 1016 it += (mLog.size() - lines); 1017 } 1018 } 1019 for (; it != mLog.end(); ++it) { 1020 const int64_t ns = it->first; 1021 const int ns_per_sec = 1000000000; 1022 const time_t sec = ns / ns_per_sec; 1023 struct tm tm; 1024 localtime_r(&sec, &tm); 1025 1026 dprintf(fd, "%s%02d-%02d %02d:%02d:%02d.%03d %s\n", 1027 prefix, 1028 tm.tm_mon + 1, // localtime_r uses months in 0 - 11 range 1029 tm.tm_mday, tm.tm_hour, tm.tm_min, tm.tm_sec, 1030 (int)(ns % ns_per_sec / 1000000), 1031 it->second.c_str()); 1032 } 1033 } 1034 mLock.unlock(); 1035 } 1036 1037 private: 1038 Mutex mLock; 1039 static const size_t kLogSize = 256; // full history 1040 static const size_t kLogPrint = 32; // default print history 1041 std::deque<std::pair<int64_t, std::string>> mLog; 1042 } mLocalLog; 1043 1044public: 1045 virtual bool hasFastMixer() const = 0; 1046 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex __unused) const 1047 { FastTrackUnderruns dummy; return dummy; } 1048 1049protected: 1050 // accessed by both binder threads and within threadLoop(), lock on mutex needed 1051 unsigned mFastTrackAvailMask; // bit i set if fast track [i] is available 1052 bool mHwSupportsPause; 1053 bool mHwPaused; 1054 bool mFlushPending; 1055}; 1056 1057class MixerThread : public PlaybackThread { 1058public: 1059 MixerThread(const sp<AudioFlinger>& audioFlinger, 1060 AudioStreamOut* output, 1061 audio_io_handle_t id, 1062 audio_devices_t device, 1063 bool systemReady, 1064 type_t type = MIXER); 1065 virtual ~MixerThread(); 1066 1067 // Thread virtuals 1068 1069 virtual bool checkForNewParameter_l(const String8& keyValuePair, 1070 status_t& status); 1071 virtual void dumpInternals(int fd, const Vector<String16>& args); 1072 1073protected: 1074 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1075 virtual int getTrackName_l(audio_channel_mask_t channelMask, audio_format_t format, 1076 audio_session_t sessionId, uid_t uid); 1077 virtual void deleteTrackName_l(int name); 1078 virtual uint32_t idleSleepTimeUs() const; 1079 virtual uint32_t suspendSleepTimeUs() const; 1080 virtual void cacheParameters_l(); 1081 1082 virtual void acquireWakeLock_l() { 1083 PlaybackThread::acquireWakeLock_l(); 1084 if (hasFastMixer()) { 1085 mFastMixer->setBoottimeOffset( 1086 mTimestamp.mTimebaseOffset[ExtendedTimestamp::TIMEBASE_BOOTTIME]); 1087 } 1088 } 1089 1090 // threadLoop snippets 1091 virtual ssize_t threadLoop_write(); 1092 virtual void threadLoop_standby(); 1093 virtual void threadLoop_mix(); 1094 virtual void threadLoop_sleepTime(); 1095 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 1096 virtual uint32_t correctLatency_l(uint32_t latency) const; 1097 1098 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 1099 audio_patch_handle_t *handle); 1100 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 1101 1102 AudioMixer* mAudioMixer; // normal mixer 1103private: 1104 // one-time initialization, no locks required 1105 sp<FastMixer> mFastMixer; // non-0 if there is also a fast mixer 1106 sp<AudioWatchdog> mAudioWatchdog; // non-0 if there is an audio watchdog thread 1107 1108 // contents are not guaranteed to be consistent, no locks required 1109 FastMixerDumpState mFastMixerDumpState; 1110#ifdef STATE_QUEUE_DUMP 1111 StateQueueObserverDump mStateQueueObserverDump; 1112 StateQueueMutatorDump mStateQueueMutatorDump; 1113#endif 1114 AudioWatchdogDump mAudioWatchdogDump; 1115 1116 // accessible only within the threadLoop(), no locks required 1117 // mFastMixer->sq() // for mutating and pushing state 1118 int32_t mFastMixerFutex; // for cold idle 1119 1120 std::atomic_bool mMasterMono; 1121public: 1122 virtual bool hasFastMixer() const { return mFastMixer != 0; } 1123 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const { 1124 ALOG_ASSERT(fastIndex < FastMixerState::sMaxFastTracks); 1125 return mFastMixerDumpState.mTracks[fastIndex].mUnderruns; 1126 } 1127 1128protected: 1129 virtual void setMasterMono_l(bool mono) { 1130 mMasterMono.store(mono); 1131 if (mFastMixer != nullptr) { /* hasFastMixer() */ 1132 mFastMixer->setMasterMono(mMasterMono); 1133 } 1134 } 1135 // the FastMixer performs mono blend if it exists. 1136 // Blending with limiter is not idempotent, 1137 // and blending without limiter is idempotent but inefficient to do twice. 1138 virtual bool requireMonoBlend() { return mMasterMono.load() && !hasFastMixer(); } 1139}; 1140 1141class DirectOutputThread : public PlaybackThread { 1142public: 1143 1144 DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1145 audio_io_handle_t id, audio_devices_t device, bool systemReady); 1146 virtual ~DirectOutputThread(); 1147 1148 // Thread virtuals 1149 1150 virtual bool checkForNewParameter_l(const String8& keyValuePair, 1151 status_t& status); 1152 virtual void flushHw_l(); 1153 1154protected: 1155 virtual int getTrackName_l(audio_channel_mask_t channelMask, audio_format_t format, 1156 audio_session_t sessionId, uid_t uid); 1157 virtual void deleteTrackName_l(int name); 1158 virtual uint32_t activeSleepTimeUs() const; 1159 virtual uint32_t idleSleepTimeUs() const; 1160 virtual uint32_t suspendSleepTimeUs() const; 1161 virtual void cacheParameters_l(); 1162 1163 // threadLoop snippets 1164 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1165 virtual void threadLoop_mix(); 1166 virtual void threadLoop_sleepTime(); 1167 virtual void threadLoop_exit(); 1168 virtual bool shouldStandby_l(); 1169 1170 virtual void onAddNewTrack_l(); 1171 1172 // volumes last sent to audio HAL with stream->set_volume() 1173 float mLeftVolFloat; 1174 float mRightVolFloat; 1175 1176 DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1177 audio_io_handle_t id, uint32_t device, ThreadBase::type_t type, 1178 bool systemReady); 1179 void processVolume_l(Track *track, bool lastTrack); 1180 1181 // prepareTracks_l() tells threadLoop_mix() the name of the single active track 1182 sp<Track> mActiveTrack; 1183 1184 wp<Track> mPreviousTrack; // used to detect track switch 1185 1186public: 1187 virtual bool hasFastMixer() const { return false; } 1188}; 1189 1190class OffloadThread : public DirectOutputThread { 1191public: 1192 1193 OffloadThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1194 audio_io_handle_t id, uint32_t device, bool systemReady); 1195 virtual ~OffloadThread() {}; 1196 virtual void flushHw_l(); 1197 1198protected: 1199 // threadLoop snippets 1200 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1201 virtual void threadLoop_exit(); 1202 1203 virtual bool waitingAsyncCallback(); 1204 virtual bool waitingAsyncCallback_l(); 1205 virtual void invalidateTracks(audio_stream_type_t streamType); 1206 1207 virtual bool keepWakeLock() const { return (mKeepWakeLock || (mDrainSequence & 1)); } 1208 1209private: 1210 size_t mPausedWriteLength; // length in bytes of write interrupted by pause 1211 size_t mPausedBytesRemaining; // bytes still waiting in mixbuffer after resume 1212 bool mKeepWakeLock; // keep wake lock while waiting for write callback 1213 uint64_t mOffloadUnderrunPosition; // Current frame position for offloaded playback 1214 // used and valid only during underrun. ~0 if 1215 // no underrun has occurred during playback and 1216 // is not reset on standby. 1217}; 1218 1219class AsyncCallbackThread : public Thread { 1220public: 1221 1222 explicit AsyncCallbackThread(const wp<PlaybackThread>& playbackThread); 1223 1224 virtual ~AsyncCallbackThread(); 1225 1226 // Thread virtuals 1227 virtual bool threadLoop(); 1228 1229 // RefBase 1230 virtual void onFirstRef(); 1231 1232 void exit(); 1233 void setWriteBlocked(uint32_t sequence); 1234 void resetWriteBlocked(); 1235 void setDraining(uint32_t sequence); 1236 void resetDraining(); 1237 void setAsyncError(); 1238 1239private: 1240 const wp<PlaybackThread> mPlaybackThread; 1241 // mWriteAckSequence corresponds to the last write sequence passed by the offload thread via 1242 // setWriteBlocked(). The sequence is shifted one bit to the left and the lsb is used 1243 // to indicate that the callback has been received via resetWriteBlocked() 1244 uint32_t mWriteAckSequence; 1245 // mDrainSequence corresponds to the last drain sequence passed by the offload thread via 1246 // setDraining(). The sequence is shifted one bit to the left and the lsb is used 1247 // to indicate that the callback has been received via resetDraining() 1248 uint32_t mDrainSequence; 1249 Condition mWaitWorkCV; 1250 Mutex mLock; 1251 bool mAsyncError; 1252}; 1253 1254class DuplicatingThread : public MixerThread { 1255public: 1256 DuplicatingThread(const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread, 1257 audio_io_handle_t id, bool systemReady); 1258 virtual ~DuplicatingThread(); 1259 1260 // Thread virtuals 1261 void addOutputTrack(MixerThread* thread); 1262 void removeOutputTrack(MixerThread* thread); 1263 uint32_t waitTimeMs() const { return mWaitTimeMs; } 1264protected: 1265 virtual uint32_t activeSleepTimeUs() const; 1266 1267private: 1268 bool outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks); 1269protected: 1270 // threadLoop snippets 1271 virtual void threadLoop_mix(); 1272 virtual void threadLoop_sleepTime(); 1273 virtual ssize_t threadLoop_write(); 1274 virtual void threadLoop_standby(); 1275 virtual void cacheParameters_l(); 1276 1277private: 1278 // called from threadLoop, addOutputTrack, removeOutputTrack 1279 virtual void updateWaitTime_l(); 1280protected: 1281 virtual void saveOutputTracks(); 1282 virtual void clearOutputTracks(); 1283private: 1284 1285 uint32_t mWaitTimeMs; 1286 SortedVector < sp<OutputTrack> > outputTracks; 1287 SortedVector < sp<OutputTrack> > mOutputTracks; 1288public: 1289 virtual bool hasFastMixer() const { return false; } 1290}; 1291 1292// record thread 1293class RecordThread : public ThreadBase 1294{ 1295public: 1296 1297 class RecordTrack; 1298 1299 /* The ResamplerBufferProvider is used to retrieve recorded input data from the 1300 * RecordThread. It maintains local state on the relative position of the read 1301 * position of the RecordTrack compared with the RecordThread. 1302 */ 1303 class ResamplerBufferProvider : public AudioBufferProvider 1304 { 1305 public: 1306 explicit ResamplerBufferProvider(RecordTrack* recordTrack) : 1307 mRecordTrack(recordTrack), 1308 mRsmpInUnrel(0), mRsmpInFront(0) { } 1309 virtual ~ResamplerBufferProvider() { } 1310 1311 // called to set the ResamplerBufferProvider to head of the RecordThread data buffer, 1312 // skipping any previous data read from the hal. 1313 virtual void reset(); 1314 1315 /* Synchronizes RecordTrack position with the RecordThread. 1316 * Calculates available frames and handle overruns if the RecordThread 1317 * has advanced faster than the ResamplerBufferProvider has retrieved data. 1318 * TODO: why not do this for every getNextBuffer? 1319 * 1320 * Parameters 1321 * framesAvailable: pointer to optional output size_t to store record track 1322 * frames available. 1323 * hasOverrun: pointer to optional boolean, returns true if track has overrun. 1324 */ 1325 1326 virtual void sync(size_t *framesAvailable = NULL, bool *hasOverrun = NULL); 1327 1328 // AudioBufferProvider interface 1329 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer); 1330 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 1331 private: 1332 RecordTrack * const mRecordTrack; 1333 size_t mRsmpInUnrel; // unreleased frames remaining from 1334 // most recent getNextBuffer 1335 // for debug only 1336 int32_t mRsmpInFront; // next available frame 1337 // rolling counter that is never cleared 1338 }; 1339 1340 /* The RecordBufferConverter is used for format, channel, and sample rate 1341 * conversion for a RecordTrack. 1342 * 1343 * TODO: Self contained, so move to a separate file later. 1344 * 1345 * RecordBufferConverter uses the convert() method rather than exposing a 1346 * buffer provider interface; this is to save a memory copy. 1347 */ 1348 class RecordBufferConverter 1349 { 1350 public: 1351 RecordBufferConverter( 1352 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 1353 uint32_t srcSampleRate, 1354 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 1355 uint32_t dstSampleRate); 1356 1357 ~RecordBufferConverter(); 1358 1359 /* Converts input data from an AudioBufferProvider by format, channelMask, 1360 * and sampleRate to a destination buffer. 1361 * 1362 * Parameters 1363 * dst: buffer to place the converted data. 1364 * provider: buffer provider to obtain source data. 1365 * frames: number of frames to convert 1366 * 1367 * Returns the number of frames converted. 1368 */ 1369 size_t convert(void *dst, AudioBufferProvider *provider, size_t frames); 1370 1371 // returns NO_ERROR if constructor was successful 1372 status_t initCheck() const { 1373 // mSrcChannelMask set on successful updateParameters 1374 return mSrcChannelMask != AUDIO_CHANNEL_INVALID ? NO_ERROR : NO_INIT; 1375 } 1376 1377 // allows dynamic reconfigure of all parameters 1378 status_t updateParameters( 1379 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 1380 uint32_t srcSampleRate, 1381 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 1382 uint32_t dstSampleRate); 1383 1384 // called to reset resampler buffers on record track discontinuity 1385 void reset() { 1386 if (mResampler != NULL) { 1387 mResampler->reset(); 1388 } 1389 } 1390 1391 private: 1392 // format conversion when not using resampler 1393 void convertNoResampler(void *dst, const void *src, size_t frames); 1394 1395 // format conversion when using resampler; modifies src in-place 1396 void convertResampler(void *dst, /*not-a-const*/ void *src, size_t frames); 1397 1398 // user provided information 1399 audio_channel_mask_t mSrcChannelMask; 1400 audio_format_t mSrcFormat; 1401 uint32_t mSrcSampleRate; 1402 audio_channel_mask_t mDstChannelMask; 1403 audio_format_t mDstFormat; 1404 uint32_t mDstSampleRate; 1405 1406 // derived information 1407 uint32_t mSrcChannelCount; 1408 uint32_t mDstChannelCount; 1409 size_t mDstFrameSize; 1410 1411 // format conversion buffer 1412 void *mBuf; 1413 size_t mBufFrames; 1414 size_t mBufFrameSize; 1415 1416 // resampler info 1417 AudioResampler *mResampler; 1418 1419 bool mIsLegacyDownmix; // legacy stereo to mono conversion needed 1420 bool mIsLegacyUpmix; // legacy mono to stereo conversion needed 1421 bool mRequiresFloat; // data processing requires float (e.g. resampler) 1422 PassthruBufferProvider *mInputConverterProvider; // converts input to float 1423 int8_t mIdxAry[sizeof(uint32_t) * 8]; // used for channel mask conversion 1424 }; 1425 1426#include "RecordTracks.h" 1427 1428 RecordThread(const sp<AudioFlinger>& audioFlinger, 1429 AudioStreamIn *input, 1430 audio_io_handle_t id, 1431 audio_devices_t outDevice, 1432 audio_devices_t inDevice, 1433 bool systemReady 1434#ifdef TEE_SINK 1435 , const sp<NBAIO_Sink>& teeSink 1436#endif 1437 ); 1438 virtual ~RecordThread(); 1439 1440 // no addTrack_l ? 1441 void destroyTrack_l(const sp<RecordTrack>& track); 1442 void removeTrack_l(const sp<RecordTrack>& track); 1443 1444 void dumpInternals(int fd, const Vector<String16>& args); 1445 void dumpTracks(int fd, const Vector<String16>& args); 1446 1447 // Thread virtuals 1448 virtual bool threadLoop(); 1449 1450 // RefBase 1451 virtual void onFirstRef(); 1452 1453 virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; } 1454 1455 virtual sp<MemoryDealer> readOnlyHeap() const { return mReadOnlyHeap; } 1456 1457 virtual sp<IMemory> pipeMemory() const { return mPipeMemory; } 1458 1459 sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l( 1460 const sp<AudioFlinger::Client>& client, 1461 uint32_t sampleRate, 1462 audio_format_t format, 1463 audio_channel_mask_t channelMask, 1464 size_t *pFrameCount, 1465 audio_session_t sessionId, 1466 size_t *notificationFrames, 1467 uid_t uid, 1468 audio_input_flags_t *flags, 1469 pid_t tid, 1470 status_t *status /*non-NULL*/, 1471 audio_port_handle_t portId); 1472 1473 status_t start(RecordTrack* recordTrack, 1474 AudioSystem::sync_event_t event, 1475 audio_session_t triggerSession); 1476 1477 // ask the thread to stop the specified track, and 1478 // return true if the caller should then do it's part of the stopping process 1479 bool stop(RecordTrack* recordTrack); 1480 1481 void dump(int fd, const Vector<String16>& args); 1482 AudioStreamIn* clearInput(); 1483 virtual sp<StreamHalInterface> stream() const; 1484 1485 1486 virtual bool checkForNewParameter_l(const String8& keyValuePair, 1487 status_t& status); 1488 virtual void cacheParameters_l() {} 1489 virtual String8 getParameters(const String8& keys); 1490 virtual void ioConfigChanged(audio_io_config_event event, pid_t pid = 0); 1491 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 1492 audio_patch_handle_t *handle); 1493 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 1494 1495 void addPatchRecord(const sp<PatchRecord>& record); 1496 void deletePatchRecord(const sp<PatchRecord>& record); 1497 1498 void readInputParameters_l(); 1499 virtual uint32_t getInputFramesLost(); 1500 1501 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1502 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1503 virtual uint32_t hasAudioSession_l(audio_session_t sessionId) const; 1504 1505 // Return the set of unique session IDs across all tracks. 1506 // The keys are the session IDs, and the associated values are meaningless. 1507 // FIXME replace by Set [and implement Bag/Multiset for other uses]. 1508 KeyedVector<audio_session_t, bool> sessionIds() const; 1509 1510 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 1511 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 1512 1513 static void syncStartEventCallback(const wp<SyncEvent>& event); 1514 1515 virtual size_t frameCount() const { return mFrameCount; } 1516 bool hasFastCapture() const { return mFastCapture != 0; } 1517 virtual void getAudioPortConfig(struct audio_port_config *config); 1518 1519 virtual status_t checkEffectCompatibility_l(const effect_descriptor_t *desc, 1520 audio_session_t sessionId); 1521 1522 virtual void acquireWakeLock_l() { 1523 ThreadBase::acquireWakeLock_l(); 1524 mActiveTracks.updatePowerState(this, true /* force */); 1525 } 1526 1527private: 1528 // Enter standby if not already in standby, and set mStandby flag 1529 void standbyIfNotAlreadyInStandby(); 1530 1531 // Call the HAL standby method unconditionally, and don't change mStandby flag 1532 void inputStandBy(); 1533 1534 AudioStreamIn *mInput; 1535 SortedVector < sp<RecordTrack> > mTracks; 1536 // mActiveTracks has dual roles: it indicates the current active track(s), and 1537 // is used together with mStartStopCond to indicate start()/stop() progress 1538 ActiveTracks<RecordTrack> mActiveTracks; 1539 1540 Condition mStartStopCond; 1541 1542 // resampler converts input at HAL Hz to output at AudioRecord client Hz 1543 void *mRsmpInBuffer; // size = mRsmpInFramesOA 1544 size_t mRsmpInFrames; // size of resampler input in frames 1545 size_t mRsmpInFramesP2;// size rounded up to a power-of-2 1546 size_t mRsmpInFramesOA;// mRsmpInFramesP2 + over-allocation 1547 1548 // rolling index that is never cleared 1549 int32_t mRsmpInRear; // last filled frame + 1 1550 1551 // For dumpsys 1552 const sp<NBAIO_Sink> mTeeSink; 1553 1554 const sp<MemoryDealer> mReadOnlyHeap; 1555 1556 // one-time initialization, no locks required 1557 sp<FastCapture> mFastCapture; // non-0 if there is also 1558 // a fast capture 1559 1560 // FIXME audio watchdog thread 1561 1562 // contents are not guaranteed to be consistent, no locks required 1563 FastCaptureDumpState mFastCaptureDumpState; 1564#ifdef STATE_QUEUE_DUMP 1565 // FIXME StateQueue observer and mutator dump fields 1566#endif 1567 // FIXME audio watchdog dump 1568 1569 // accessible only within the threadLoop(), no locks required 1570 // mFastCapture->sq() // for mutating and pushing state 1571 int32_t mFastCaptureFutex; // for cold idle 1572 1573 // The HAL input source is treated as non-blocking, 1574 // but current implementation is blocking 1575 sp<NBAIO_Source> mInputSource; 1576 // The source for the normal capture thread to read from: mInputSource or mPipeSource 1577 sp<NBAIO_Source> mNormalSource; 1578 // If a fast capture is present, the non-blocking pipe sink written to by fast capture, 1579 // otherwise clear 1580 sp<NBAIO_Sink> mPipeSink; 1581 // If a fast capture is present, the non-blocking pipe source read by normal thread, 1582 // otherwise clear 1583 sp<NBAIO_Source> mPipeSource; 1584 // Depth of pipe from fast capture to normal thread and fast clients, always power of 2 1585 size_t mPipeFramesP2; 1586 // If a fast capture is present, the Pipe as IMemory, otherwise clear 1587 sp<IMemory> mPipeMemory; 1588 1589 static const size_t kFastCaptureLogSize = 4 * 1024; 1590 sp<NBLog::Writer> mFastCaptureNBLogWriter; 1591 1592 bool mFastTrackAvail; // true if fast track available 1593}; 1594 1595class MmapThread : public ThreadBase 1596{ 1597 public: 1598 1599#include "MmapTracks.h" 1600 1601 MmapThread(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1602 AudioHwDevice *hwDev, sp<StreamHalInterface> stream, 1603 audio_devices_t outDevice, audio_devices_t inDevice, bool systemReady); 1604 virtual ~MmapThread(); 1605 1606 virtual void configure(const audio_attributes_t *attr, 1607 audio_stream_type_t streamType, 1608 audio_session_t sessionId, 1609 const sp<MmapStreamCallback>& callback, 1610 audio_port_handle_t portId); 1611 1612 void disconnect(); 1613 1614 // MmapStreamInterface 1615 status_t createMmapBuffer(int32_t minSizeFrames, 1616 struct audio_mmap_buffer_info *info); 1617 status_t getMmapPosition(struct audio_mmap_position *position); 1618 status_t start(const MmapStreamInterface::Client& client, audio_port_handle_t *handle); 1619 status_t stop(audio_port_handle_t handle); 1620 1621 // RefBase 1622 virtual void onFirstRef(); 1623 1624 // Thread virtuals 1625 virtual bool threadLoop(); 1626 1627 virtual void threadLoop_exit(); 1628 virtual void threadLoop_standby(); 1629 1630 virtual status_t initCheck() const { return (mHalStream == 0) ? NO_INIT : NO_ERROR; } 1631 virtual size_t frameCount() const { return mFrameCount; } 1632 virtual bool checkForNewParameter_l(const String8& keyValuePair, 1633 status_t& status); 1634 virtual String8 getParameters(const String8& keys); 1635 virtual void ioConfigChanged(audio_io_config_event event, pid_t pid = 0); 1636 void readHalParameters_l(); 1637 virtual void cacheParameters_l() {} 1638 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 1639 audio_patch_handle_t *handle); 1640 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 1641 virtual void getAudioPortConfig(struct audio_port_config *config); 1642 1643 virtual sp<StreamHalInterface> stream() const { return mHalStream; } 1644 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1645 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1646 virtual status_t checkEffectCompatibility_l(const effect_descriptor_t *desc, 1647 audio_session_t sessionId); 1648 1649 virtual uint32_t hasAudioSession_l(audio_session_t sessionId) const; 1650 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 1651 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 1652 1653 virtual void checkSilentMode_l() {} 1654 virtual void processVolume_l() {} 1655 void checkInvalidTracks_l(); 1656 1657 virtual audio_stream_type_t streamType() { return AUDIO_STREAM_DEFAULT; } 1658 1659 virtual void invalidateTracks(audio_stream_type_t streamType __unused) {} 1660 1661 void dump(int fd, const Vector<String16>& args); 1662 virtual void dumpInternals(int fd, const Vector<String16>& args); 1663 void dumpTracks(int fd, const Vector<String16>& args); 1664 1665 virtual bool isOutput() const = 0; 1666 1667 protected: 1668 1669 audio_attributes_t mAttr; 1670 audio_session_t mSessionId; 1671 audio_port_handle_t mPortId; 1672 1673 sp<MmapStreamCallback> mCallback; 1674 sp<StreamHalInterface> mHalStream; 1675 sp<DeviceHalInterface> mHalDevice; 1676 AudioHwDevice* const mAudioHwDev; 1677 ActiveTracks<MmapTrack> mActiveTracks; 1678}; 1679 1680class MmapPlaybackThread : public MmapThread, public VolumeInterface 1681{ 1682 1683public: 1684 MmapPlaybackThread(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1685 AudioHwDevice *hwDev, AudioStreamOut *output, 1686 audio_devices_t outDevice, audio_devices_t inDevice, bool systemReady); 1687 virtual ~MmapPlaybackThread() {} 1688 1689 virtual void configure(const audio_attributes_t *attr, 1690 audio_stream_type_t streamType, 1691 audio_session_t sessionId, 1692 const sp<MmapStreamCallback>& callback, 1693 audio_port_handle_t portId); 1694 1695 AudioStreamOut* clearOutput(); 1696 1697 // VolumeInterface 1698 virtual void setMasterVolume(float value); 1699 virtual void setMasterMute(bool muted); 1700 virtual void setStreamVolume(audio_stream_type_t stream, float value); 1701 virtual void setStreamMute(audio_stream_type_t stream, bool muted); 1702 virtual float streamVolume(audio_stream_type_t stream) const; 1703 1704 void setMasterMute_l(bool muted) { mMasterMute = muted; } 1705 1706 virtual void invalidateTracks(audio_stream_type_t streamType); 1707 1708 virtual audio_stream_type_t streamType() { return mStreamType; } 1709 virtual void checkSilentMode_l(); 1710 virtual void processVolume_l(); 1711 1712 virtual void dumpInternals(int fd, const Vector<String16>& args); 1713 1714 virtual bool isOutput() const { return true; } 1715 1716protected: 1717 1718 audio_stream_type_t mStreamType; 1719 float mMasterVolume; 1720 float mStreamVolume; 1721 bool mMasterMute; 1722 bool mStreamMute; 1723 float mHalVolFloat; 1724 AudioStreamOut* mOutput; 1725}; 1726 1727class MmapCaptureThread : public MmapThread 1728{ 1729 1730public: 1731 MmapCaptureThread(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1732 AudioHwDevice *hwDev, AudioStreamIn *input, 1733 audio_devices_t outDevice, audio_devices_t inDevice, bool systemReady); 1734 virtual ~MmapCaptureThread() {} 1735 1736 AudioStreamIn* clearInput(); 1737 1738 virtual bool isOutput() const { return false; } 1739 1740protected: 1741 1742 AudioStreamIn* mInput; 1743}; 1744