Threads.h revision d79072e9dff59f767cce2cda1caab80ce5a0815b
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 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 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 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 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 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 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 size_t frameSize() const { return mFrameSize; } 251 252 // Should be "virtual status_t requestExitAndWait()" and override same 253 // method in Thread, but Thread::requestExitAndWait() is not yet virtual. 254 void exit(); 255 virtual bool checkForNewParameter_l(const String8& keyValuePair, 256 status_t& status) = 0; 257 virtual status_t setParameters(const String8& keyValuePairs); 258 virtual String8 getParameters(const String8& keys) = 0; 259 virtual void ioConfigChanged(audio_io_config_event event, pid_t pid = 0) = 0; 260 // sendConfigEvent_l() must be called with ThreadBase::mLock held 261 // Can temporarily release the lock if waiting for a reply from 262 // processConfigEvents_l(). 263 status_t sendConfigEvent_l(sp<ConfigEvent>& event); 264 void sendIoConfigEvent(audio_io_config_event event, pid_t pid = 0); 265 void sendIoConfigEvent_l(audio_io_config_event event, pid_t pid = 0); 266 void sendPrioConfigEvent(pid_t pid, pid_t tid, int32_t prio); 267 void sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio); 268 status_t sendSetParameterConfigEvent_l(const String8& keyValuePair); 269 status_t sendCreateAudioPatchConfigEvent(const struct audio_patch *patch, 270 audio_patch_handle_t *handle); 271 status_t sendReleaseAudioPatchConfigEvent(audio_patch_handle_t handle); 272 void processConfigEvents_l(); 273 virtual void cacheParameters_l() = 0; 274 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 275 audio_patch_handle_t *handle) = 0; 276 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle) = 0; 277 virtual void getAudioPortConfig(struct audio_port_config *config) = 0; 278 279 280 // see note at declaration of mStandby, mOutDevice and mInDevice 281 bool standby() const { return mStandby; } 282 audio_devices_t outDevice() const { return mOutDevice; } 283 audio_devices_t inDevice() const { return mInDevice; } 284 285 virtual audio_stream_t* stream() const = 0; 286 287 sp<EffectHandle> createEffect_l( 288 const sp<AudioFlinger::Client>& client, 289 const sp<IEffectClient>& effectClient, 290 int32_t priority, 291 int sessionId, 292 effect_descriptor_t *desc, 293 int *enabled, 294 status_t *status /*non-NULL*/); 295 296 // return values for hasAudioSession (bit field) 297 enum effect_state { 298 EFFECT_SESSION = 0x1, // the audio session corresponds to at least one 299 // effect 300 TRACK_SESSION = 0x2 // the audio session corresponds to at least one 301 // track 302 }; 303 304 // get effect chain corresponding to session Id. 305 sp<EffectChain> getEffectChain(int sessionId); 306 // same as getEffectChain() but must be called with ThreadBase mutex locked 307 sp<EffectChain> getEffectChain_l(int sessionId) const; 308 // add an effect chain to the chain list (mEffectChains) 309 virtual status_t addEffectChain_l(const sp<EffectChain>& chain) = 0; 310 // remove an effect chain from the chain list (mEffectChains) 311 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain) = 0; 312 // lock all effect chains Mutexes. Must be called before releasing the 313 // ThreadBase mutex before processing the mixer and effects. This guarantees the 314 // integrity of the chains during the process. 315 // Also sets the parameter 'effectChains' to current value of mEffectChains. 316 void lockEffectChains_l(Vector< sp<EffectChain> >& effectChains); 317 // unlock effect chains after process 318 void unlockEffectChains(const Vector< sp<EffectChain> >& effectChains); 319 // get a copy of mEffectChains vector 320 Vector< sp<EffectChain> > getEffectChains_l() const { return mEffectChains; }; 321 // set audio mode to all effect chains 322 void setMode(audio_mode_t mode); 323 // get effect module with corresponding ID on specified audio session 324 sp<AudioFlinger::EffectModule> getEffect(int sessionId, int effectId); 325 sp<AudioFlinger::EffectModule> getEffect_l(int sessionId, int effectId); 326 // add and effect module. Also creates the effect chain is none exists for 327 // the effects audio session 328 status_t addEffect_l(const sp< EffectModule>& effect); 329 // remove and effect module. Also removes the effect chain is this was the last 330 // effect 331 void removeEffect_l(const sp< EffectModule>& effect); 332 // detach all tracks connected to an auxiliary effect 333 virtual void detachAuxEffect_l(int effectId __unused) {} 334 // returns either EFFECT_SESSION if effects on this audio session exist in one 335 // chain, or TRACK_SESSION if tracks on this audio session exist, or both 336 virtual uint32_t hasAudioSession(int sessionId) const = 0; 337 // the value returned by default implementation is not important as the 338 // strategy is only meaningful for PlaybackThread which implements this method 339 virtual uint32_t getStrategyForSession_l(int sessionId __unused) { return 0; } 340 341 // suspend or restore effect according to the type of effect passed. a NULL 342 // type pointer means suspend all effects in the session 343 void setEffectSuspended(const effect_uuid_t *type, 344 bool suspend, 345 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 346 // check if some effects must be suspended/restored when an effect is enabled 347 // or disabled 348 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 349 bool enabled, 350 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 351 void checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 352 bool enabled, 353 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 354 355 virtual status_t setSyncEvent(const sp<SyncEvent>& event) = 0; 356 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const = 0; 357 358 // Return a reference to a per-thread heap which can be used to allocate IMemory 359 // objects that will be read-only to client processes, read/write to mediaserver, 360 // and shared by all client processes of the thread. 361 // The heap is per-thread rather than common across all threads, because 362 // clients can't be trusted not to modify the offset of the IMemory they receive. 363 // If a thread does not have such a heap, this method returns 0. 364 virtual sp<MemoryDealer> readOnlyHeap() const { return 0; } 365 366 virtual sp<IMemory> pipeMemory() const { return 0; } 367 368 void systemReady(); 369 370 mutable Mutex mLock; 371 372protected: 373 374 // entry describing an effect being suspended in mSuspendedSessions keyed vector 375 class SuspendedSessionDesc : public RefBase { 376 public: 377 SuspendedSessionDesc() : mRefCount(0) {} 378 379 int mRefCount; // number of active suspend requests 380 effect_uuid_t mType; // effect type UUID 381 }; 382 383 void acquireWakeLock(int uid = -1); 384 void acquireWakeLock_l(int uid = -1); 385 void releaseWakeLock(); 386 void releaseWakeLock_l(); 387 void updateWakeLockUids(const SortedVector<int> &uids); 388 void updateWakeLockUids_l(const SortedVector<int> &uids); 389 void getPowerManager_l(); 390 void setEffectSuspended_l(const effect_uuid_t *type, 391 bool suspend, 392 int sessionId); 393 // updated mSuspendedSessions when an effect suspended or restored 394 void updateSuspendedSessions_l(const effect_uuid_t *type, 395 bool suspend, 396 int sessionId); 397 // check if some effects must be suspended when an effect chain is added 398 void checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain); 399 400 String16 getWakeLockTag(); 401 402 virtual void preExit() { } 403 virtual void setMasterMono_l(bool mono __unused) { } 404 virtual bool requireMonoBlend() { return false; } 405 406 friend class AudioFlinger; // for mEffectChains 407 408 const type_t mType; 409 410 // Used by parameters, config events, addTrack_l, exit 411 Condition mWaitWorkCV; 412 413 const sp<AudioFlinger> mAudioFlinger; 414 415 // updated by PlaybackThread::readOutputParameters_l() or 416 // RecordThread::readInputParameters_l() 417 uint32_t mSampleRate; 418 size_t mFrameCount; // output HAL, direct output, record 419 audio_channel_mask_t mChannelMask; 420 uint32_t mChannelCount; 421 size_t mFrameSize; 422 // not HAL frame size, this is for output sink (to pipe to fast mixer) 423 audio_format_t mFormat; // Source format for Recording and 424 // Sink format for Playback. 425 // Sink format may be different than 426 // HAL format if Fastmixer is used. 427 audio_format_t mHALFormat; 428 size_t mBufferSize; // HAL buffer size for read() or write() 429 430 Vector< sp<ConfigEvent> > mConfigEvents; 431 Vector< sp<ConfigEvent> > mPendingConfigEvents; // events awaiting system ready 432 433 // These fields are written and read by thread itself without lock or barrier, 434 // and read by other threads without lock or barrier via standby(), outDevice() 435 // and inDevice(). 436 // Because of the absence of a lock or barrier, any other thread that reads 437 // these fields must use the information in isolation, or be prepared to deal 438 // with possibility that it might be inconsistent with other information. 439 bool mStandby; // Whether thread is currently in standby. 440 audio_devices_t mOutDevice; // output device 441 audio_devices_t mInDevice; // input device 442 audio_devices_t mPrevOutDevice; // previous output device 443 audio_devices_t mPrevInDevice; // previous input device 444 struct audio_patch mPatch; 445 audio_source_t mAudioSource; 446 447 const audio_io_handle_t mId; 448 Vector< sp<EffectChain> > mEffectChains; 449 450 static const int kThreadNameLength = 16; // prctl(PR_SET_NAME) limit 451 char mThreadName[kThreadNameLength]; // guaranteed NUL-terminated 452 sp<IPowerManager> mPowerManager; 453 sp<IBinder> mWakeLockToken; 454 const sp<PMDeathRecipient> mDeathRecipient; 455 // list of suspended effects per session and per type. The first vector is 456 // keyed by session ID, the second by type UUID timeLow field 457 KeyedVector< int, KeyedVector< int, sp<SuspendedSessionDesc> > > 458 mSuspendedSessions; 459 static const size_t kLogSize = 4 * 1024; 460 sp<NBLog::Writer> mNBLogWriter; 461 bool mSystemReady; 462 bool mNotifiedBatteryStart; 463}; 464 465// --- PlaybackThread --- 466class PlaybackThread : public ThreadBase { 467public: 468 469#include "PlaybackTracks.h" 470 471 enum mixer_state { 472 MIXER_IDLE, // no active tracks 473 MIXER_TRACKS_ENABLED, // at least one active track, but no track has any data ready 474 MIXER_TRACKS_READY, // at least one active track, and at least one track has data 475 MIXER_DRAIN_TRACK, // drain currently playing track 476 MIXER_DRAIN_ALL, // fully drain the hardware 477 // standby mode does not have an enum value 478 // suspend by audio policy manager is orthogonal to mixer state 479 }; 480 481 // retry count before removing active track in case of underrun on offloaded thread: 482 // we need to make sure that AudioTrack client has enough time to send large buffers 483//FIXME may be more appropriate if expressed in time units. Need to revise how underrun is handled 484 // for offloaded tracks 485 static const int8_t kMaxTrackRetriesOffload = 20; 486 487 PlaybackThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 488 audio_io_handle_t id, audio_devices_t device, type_t type, bool systemReady); 489 virtual ~PlaybackThread(); 490 491 void dump(int fd, const Vector<String16>& args); 492 493 // Thread virtuals 494 virtual bool threadLoop(); 495 496 // RefBase 497 virtual void onFirstRef(); 498 499protected: 500 // Code snippets that were lifted up out of threadLoop() 501 virtual void threadLoop_mix() = 0; 502 virtual void threadLoop_sleepTime() = 0; 503 virtual ssize_t threadLoop_write(); 504 virtual void threadLoop_drain(); 505 virtual void threadLoop_standby(); 506 virtual void threadLoop_exit(); 507 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 508 509 // prepareTracks_l reads and writes mActiveTracks, and returns 510 // the pending set of tracks to remove via Vector 'tracksToRemove'. The caller 511 // is responsible for clearing or destroying this Vector later on, when it 512 // is safe to do so. That will drop the final ref count and destroy the tracks. 513 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove) = 0; 514 void removeTracks_l(const Vector< sp<Track> >& tracksToRemove); 515 516 void writeCallback(); 517 void resetWriteBlocked(uint32_t sequence); 518 void drainCallback(); 519 void resetDraining(uint32_t sequence); 520 521 static int asyncCallback(stream_callback_event_t event, void *param, void *cookie); 522 523 virtual bool waitingAsyncCallback(); 524 virtual bool waitingAsyncCallback_l(); 525 virtual bool shouldStandby_l(); 526 virtual void onAddNewTrack_l(); 527 528 // ThreadBase virtuals 529 virtual void preExit(); 530 531public: 532 533 virtual status_t initCheck() const { return (mOutput == NULL) ? NO_INIT : NO_ERROR; } 534 535 // return estimated latency in milliseconds, as reported by HAL 536 uint32_t latency() const; 537 // same, but lock must already be held 538 uint32_t latency_l() const; 539 540 void setMasterVolume(float value); 541 void setMasterMute(bool muted); 542 543 void setStreamVolume(audio_stream_type_t stream, float value); 544 void setStreamMute(audio_stream_type_t stream, bool muted); 545 546 float streamVolume(audio_stream_type_t stream) const; 547 548 sp<Track> createTrack_l( 549 const sp<AudioFlinger::Client>& client, 550 audio_stream_type_t streamType, 551 uint32_t sampleRate, 552 audio_format_t format, 553 audio_channel_mask_t channelMask, 554 size_t *pFrameCount, 555 const sp<IMemory>& sharedBuffer, 556 int sessionId, 557 IAudioFlinger::track_flags_t *flags, 558 pid_t tid, 559 int uid, 560 status_t *status /*non-NULL*/); 561 562 AudioStreamOut* getOutput() const; 563 AudioStreamOut* clearOutput(); 564 virtual audio_stream_t* stream() const; 565 566 // a very large number of suspend() will eventually wraparound, but unlikely 567 void suspend() { (void) android_atomic_inc(&mSuspended); } 568 void restore() 569 { 570 // if restore() is done without suspend(), get back into 571 // range so that the next suspend() will operate correctly 572 if (android_atomic_dec(&mSuspended) <= 0) { 573 android_atomic_release_store(0, &mSuspended); 574 } 575 } 576 bool isSuspended() const 577 { return android_atomic_acquire_load(&mSuspended) > 0; } 578 579 virtual String8 getParameters(const String8& keys); 580 virtual void ioConfigChanged(audio_io_config_event event, pid_t pid = 0); 581 status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames); 582 // FIXME rename mixBuffer() to sinkBuffer() and remove int16_t* dependency. 583 // Consider also removing and passing an explicit mMainBuffer initialization 584 // parameter to AF::PlaybackThread::Track::Track(). 585 int16_t *mixBuffer() const { 586 return reinterpret_cast<int16_t *>(mSinkBuffer); }; 587 588 virtual void detachAuxEffect_l(int effectId); 589 status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track> track, 590 int EffectId); 591 status_t attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track> track, 592 int EffectId); 593 594 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 595 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 596 virtual uint32_t hasAudioSession(int sessionId) const; 597 virtual uint32_t getStrategyForSession_l(int sessionId); 598 599 600 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 601 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 602 603 // called with AudioFlinger lock held 604 void invalidateTracks(audio_stream_type_t streamType); 605 606 virtual size_t frameCount() const { return mNormalFrameCount; } 607 608 // Return's the HAL's frame count i.e. fast mixer buffer size. 609 size_t frameCountHAL() const { return mFrameCount; } 610 611 status_t getTimestamp_l(AudioTimestamp& timestamp); 612 613 void addPatchTrack(const sp<PatchTrack>& track); 614 void deletePatchTrack(const sp<PatchTrack>& track); 615 616 virtual void getAudioPortConfig(struct audio_port_config *config); 617 618protected: 619 // updated by readOutputParameters_l() 620 size_t mNormalFrameCount; // normal mixer and effects 621 622 bool mThreadThrottle; // throttle the thread processing 623 uint32_t mThreadThrottleTimeMs; // throttle time for MIXER threads 624 uint32_t mThreadThrottleEndMs; // notify once per throttling 625 uint32_t mHalfBufferMs; // half the buffer size in milliseconds 626 627 void* mSinkBuffer; // frame size aligned sink buffer 628 629 // TODO: 630 // Rearrange the buffer info into a struct/class with 631 // clear, copy, construction, destruction methods. 632 // 633 // mSinkBuffer also has associated with it: 634 // 635 // mSinkBufferSize: Sink Buffer Size 636 // mFormat: Sink Buffer Format 637 638 // Mixer Buffer (mMixerBuffer*) 639 // 640 // In the case of floating point or multichannel data, which is not in the 641 // sink format, it is required to accumulate in a higher precision or greater channel count 642 // buffer before downmixing or data conversion to the sink buffer. 643 644 // Set to "true" to enable the Mixer Buffer otherwise mixer output goes to sink buffer. 645 bool mMixerBufferEnabled; 646 647 // Storage, 32 byte aligned (may make this alignment a requirement later). 648 // Due to constraints on mNormalFrameCount, the buffer size is a multiple of 16 frames. 649 void* mMixerBuffer; 650 651 // Size of mMixerBuffer in bytes: mNormalFrameCount * #channels * sampsize. 652 size_t mMixerBufferSize; 653 654 // The audio format of mMixerBuffer. Set to AUDIO_FORMAT_PCM_(FLOAT|16_BIT) only. 655 audio_format_t mMixerBufferFormat; 656 657 // An internal flag set to true by MixerThread::prepareTracks_l() 658 // when mMixerBuffer contains valid data after mixing. 659 bool mMixerBufferValid; 660 661 // Effects Buffer (mEffectsBuffer*) 662 // 663 // In the case of effects data, which is not in the sink format, 664 // it is required to accumulate in a different buffer before data conversion 665 // to the sink buffer. 666 667 // Set to "true" to enable the Effects Buffer otherwise effects output goes to sink buffer. 668 bool mEffectBufferEnabled; 669 670 // Storage, 32 byte aligned (may make this alignment a requirement later). 671 // Due to constraints on mNormalFrameCount, the buffer size is a multiple of 16 frames. 672 void* mEffectBuffer; 673 674 // Size of mEffectsBuffer in bytes: mNormalFrameCount * #channels * sampsize. 675 size_t mEffectBufferSize; 676 677 // The audio format of mEffectsBuffer. Set to AUDIO_FORMAT_PCM_16_BIT only. 678 audio_format_t mEffectBufferFormat; 679 680 // An internal flag set to true by MixerThread::prepareTracks_l() 681 // when mEffectsBuffer contains valid data after mixing. 682 // 683 // When this is set, all mixer data is routed into the effects buffer 684 // for any processing (including output processing). 685 bool mEffectBufferValid; 686 687 // suspend count, > 0 means suspended. While suspended, the thread continues to pull from 688 // tracks and mix, but doesn't write to HAL. A2DP and SCO HAL implementations can't handle 689 // concurrent use of both of them, so Audio Policy Service suspends one of the threads to 690 // workaround that restriction. 691 // 'volatile' means accessed via atomic operations and no lock. 692 volatile int32_t mSuspended; 693 694 // FIXME overflows every 6+ hours at 44.1 kHz stereo 16-bit samples 695 // mFramesWritten would be better, or 64-bit even better 696 size_t mBytesWritten; 697private: 698 // mMasterMute is in both PlaybackThread and in AudioFlinger. When a 699 // PlaybackThread needs to find out if master-muted, it checks it's local 700 // copy rather than the one in AudioFlinger. This optimization saves a lock. 701 bool mMasterMute; 702 void setMasterMute_l(bool muted) { mMasterMute = muted; } 703protected: 704 SortedVector< wp<Track> > mActiveTracks; // FIXME check if this could be sp<> 705 SortedVector<int> mWakeLockUids; 706 int mActiveTracksGeneration; 707 wp<Track> mLatestActiveTrack; // latest track added to mActiveTracks 708 709 // Allocate a track name for a given channel mask. 710 // Returns name >= 0 if successful, -1 on failure. 711 virtual int getTrackName_l(audio_channel_mask_t channelMask, 712 audio_format_t format, int sessionId) = 0; 713 virtual void deleteTrackName_l(int name) = 0; 714 715 // Time to sleep between cycles when: 716 virtual uint32_t activeSleepTimeUs() const; // mixer state MIXER_TRACKS_ENABLED 717 virtual uint32_t idleSleepTimeUs() const = 0; // mixer state MIXER_IDLE 718 virtual uint32_t suspendSleepTimeUs() const = 0; // audio policy manager suspended us 719 // No sleep when mixer state == MIXER_TRACKS_READY; relies on audio HAL stream->write() 720 // No sleep in standby mode; waits on a condition 721 722 // Code snippets that are temporarily lifted up out of threadLoop() until the merge 723 void checkSilentMode_l(); 724 725 // Non-trivial for DUPLICATING only 726 virtual void saveOutputTracks() { } 727 virtual void clearOutputTracks() { } 728 729 // Cache various calculated values, at threadLoop() entry and after a parameter change 730 virtual void cacheParameters_l(); 731 732 virtual uint32_t correctLatency_l(uint32_t latency) const; 733 734 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 735 audio_patch_handle_t *handle); 736 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 737 738 bool usesHwAvSync() const { return (mType == DIRECT) && (mOutput != NULL) 739 && mHwSupportsPause 740 && (mOutput->flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC); } 741 742private: 743 744 friend class AudioFlinger; // for numerous 745 746 PlaybackThread& operator = (const PlaybackThread&); 747 748 status_t addTrack_l(const sp<Track>& track); 749 bool destroyTrack_l(const sp<Track>& track); 750 void removeTrack_l(const sp<Track>& track); 751 void broadcast_l(); 752 753 void readOutputParameters_l(); 754 755 virtual void dumpInternals(int fd, const Vector<String16>& args); 756 void dumpTracks(int fd, const Vector<String16>& args); 757 758 SortedVector< sp<Track> > mTracks; 759 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 760 AudioStreamOut *mOutput; 761 762 float mMasterVolume; 763 nsecs_t mLastWriteTime; 764 int mNumWrites; 765 int mNumDelayedWrites; 766 bool mInWrite; 767 768 // FIXME rename these former local variables of threadLoop to standard "m" names 769 nsecs_t mStandbyTimeNs; 770 size_t mSinkBufferSize; 771 772 // cached copies of activeSleepTimeUs() and idleSleepTimeUs() made by cacheParameters_l() 773 uint32_t mActiveSleepTimeUs; 774 uint32_t mIdleSleepTimeUs; 775 776 uint32_t mSleepTimeUs; 777 778 // mixer status returned by prepareTracks_l() 779 mixer_state mMixerStatus; // current cycle 780 // previous cycle when in prepareTracks_l() 781 mixer_state mMixerStatusIgnoringFastTracks; 782 // FIXME or a separate ready state per track 783 784 // FIXME move these declarations into the specific sub-class that needs them 785 // MIXER only 786 uint32_t sleepTimeShift; 787 788 // same as AudioFlinger::mStandbyTimeInNsecs except for DIRECT which uses a shorter value 789 nsecs_t mStandbyDelayNs; 790 791 // MIXER only 792 nsecs_t maxPeriod; 793 794 // DUPLICATING only 795 uint32_t writeFrames; 796 797 size_t mBytesRemaining; 798 size_t mCurrentWriteLength; 799 bool mUseAsyncWrite; 800 // mWriteAckSequence contains current write sequence on bits 31-1. The write sequence is 801 // incremented each time a write(), a flush() or a standby() occurs. 802 // Bit 0 is set when a write blocks and indicates a callback is expected. 803 // Bit 0 is reset by the async callback thread calling resetWriteBlocked(). Out of sequence 804 // callbacks are ignored. 805 uint32_t mWriteAckSequence; 806 // mDrainSequence contains current drain sequence on bits 31-1. The drain sequence is 807 // incremented each time a drain is requested or a flush() or standby() occurs. 808 // Bit 0 is set when the drain() command is called at the HAL and indicates a callback is 809 // expected. 810 // Bit 0 is reset by the async callback thread calling resetDraining(). Out of sequence 811 // callbacks are ignored. 812 uint32_t mDrainSequence; 813 // A condition that must be evaluated by prepareTrack_l() has changed and we must not wait 814 // for async write callback in the thread loop before evaluating it 815 bool mSignalPending; 816 sp<AsyncCallbackThread> mCallbackThread; 817 818private: 819 // The HAL output sink is treated as non-blocking, but current implementation is blocking 820 sp<NBAIO_Sink> mOutputSink; 821 // If a fast mixer is present, the blocking pipe sink, otherwise clear 822 sp<NBAIO_Sink> mPipeSink; 823 // The current sink for the normal mixer to write it's (sub)mix, mOutputSink or mPipeSink 824 sp<NBAIO_Sink> mNormalSink; 825#ifdef TEE_SINK 826 // For dumpsys 827 sp<NBAIO_Sink> mTeeSink; 828 sp<NBAIO_Source> mTeeSource; 829#endif 830 uint32_t mScreenState; // cached copy of gScreenState 831 static const size_t kFastMixerLogSize = 4 * 1024; 832 sp<NBLog::Writer> mFastMixerNBLogWriter; 833public: 834 virtual bool hasFastMixer() const = 0; 835 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex __unused) const 836 { FastTrackUnderruns dummy; return dummy; } 837 838protected: 839 // accessed by both binder threads and within threadLoop(), lock on mutex needed 840 unsigned mFastTrackAvailMask; // bit i set if fast track [i] is available 841 bool mHwSupportsPause; 842 bool mHwPaused; 843 bool mFlushPending; 844private: 845 // timestamp latch: 846 // D input is written by threadLoop_write while mutex is unlocked, and read while locked 847 // Q output is written while locked, and read while locked 848 struct { 849 AudioTimestamp mTimestamp; 850 uint32_t mUnpresentedFrames; 851 KeyedVector<Track *, uint32_t> mFramesReleased; 852 } mLatchD, mLatchQ; 853 bool mLatchDValid; // true means mLatchD is valid 854 // (except for mFramesReleased which is filled in later), 855 // and clock it into latch at next opportunity 856 bool mLatchQValid; // true means mLatchQ is valid 857}; 858 859class MixerThread : public PlaybackThread { 860public: 861 MixerThread(const sp<AudioFlinger>& audioFlinger, 862 AudioStreamOut* output, 863 audio_io_handle_t id, 864 audio_devices_t device, 865 bool systemReady, 866 type_t type = MIXER); 867 virtual ~MixerThread(); 868 869 // Thread virtuals 870 871 virtual bool checkForNewParameter_l(const String8& keyValuePair, 872 status_t& status); 873 virtual void dumpInternals(int fd, const Vector<String16>& args); 874 875protected: 876 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 877 virtual int getTrackName_l(audio_channel_mask_t channelMask, 878 audio_format_t format, int sessionId); 879 virtual void deleteTrackName_l(int name); 880 virtual uint32_t idleSleepTimeUs() const; 881 virtual uint32_t suspendSleepTimeUs() const; 882 virtual void cacheParameters_l(); 883 884 // threadLoop snippets 885 virtual ssize_t threadLoop_write(); 886 virtual void threadLoop_standby(); 887 virtual void threadLoop_mix(); 888 virtual void threadLoop_sleepTime(); 889 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 890 virtual uint32_t correctLatency_l(uint32_t latency) const; 891 892 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 893 audio_patch_handle_t *handle); 894 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 895 896 AudioMixer* mAudioMixer; // normal mixer 897private: 898 // one-time initialization, no locks required 899 sp<FastMixer> mFastMixer; // non-0 if there is also a fast mixer 900 sp<AudioWatchdog> mAudioWatchdog; // non-0 if there is an audio watchdog thread 901 902 // contents are not guaranteed to be consistent, no locks required 903 FastMixerDumpState mFastMixerDumpState; 904#ifdef STATE_QUEUE_DUMP 905 StateQueueObserverDump mStateQueueObserverDump; 906 StateQueueMutatorDump mStateQueueMutatorDump; 907#endif 908 AudioWatchdogDump mAudioWatchdogDump; 909 910 // accessible only within the threadLoop(), no locks required 911 // mFastMixer->sq() // for mutating and pushing state 912 int32_t mFastMixerFutex; // for cold idle 913 914 std::atomic_bool mMasterMono; 915public: 916 virtual bool hasFastMixer() const { return mFastMixer != 0; } 917 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const { 918 ALOG_ASSERT(fastIndex < FastMixerState::kMaxFastTracks); 919 return mFastMixerDumpState.mTracks[fastIndex].mUnderruns; 920 } 921 922protected: 923 virtual void setMasterMono_l(bool mono) { 924 mMasterMono.store(mono); 925 if (mFastMixer != nullptr) { /* hasFastMixer() */ 926 mFastMixer->setMasterMono(mMasterMono); 927 } 928 } 929 // the FastMixer performs mono blend if it exists. 930 virtual bool requireMonoBlend() { return mMasterMono.load() && !hasFastMixer(); } 931}; 932 933class DirectOutputThread : public PlaybackThread { 934public: 935 936 DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 937 audio_io_handle_t id, audio_devices_t device, bool systemReady); 938 virtual ~DirectOutputThread(); 939 940 // Thread virtuals 941 942 virtual bool checkForNewParameter_l(const String8& keyValuePair, 943 status_t& status); 944 virtual void flushHw_l(); 945 946protected: 947 virtual int getTrackName_l(audio_channel_mask_t channelMask, 948 audio_format_t format, int sessionId); 949 virtual void deleteTrackName_l(int name); 950 virtual uint32_t activeSleepTimeUs() const; 951 virtual uint32_t idleSleepTimeUs() const; 952 virtual uint32_t suspendSleepTimeUs() const; 953 virtual void cacheParameters_l(); 954 955 // threadLoop snippets 956 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 957 virtual void threadLoop_mix(); 958 virtual void threadLoop_sleepTime(); 959 virtual void threadLoop_exit(); 960 virtual bool shouldStandby_l(); 961 962 virtual void onAddNewTrack_l(); 963 964 // volumes last sent to audio HAL with stream->set_volume() 965 float mLeftVolFloat; 966 float mRightVolFloat; 967 968 DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 969 audio_io_handle_t id, uint32_t device, ThreadBase::type_t type, 970 bool systemReady); 971 void processVolume_l(Track *track, bool lastTrack); 972 973 // prepareTracks_l() tells threadLoop_mix() the name of the single active track 974 sp<Track> mActiveTrack; 975 976 wp<Track> mPreviousTrack; // used to detect track switch 977 978public: 979 virtual bool hasFastMixer() const { return false; } 980}; 981 982class OffloadThread : public DirectOutputThread { 983public: 984 985 OffloadThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 986 audio_io_handle_t id, uint32_t device, bool systemReady); 987 virtual ~OffloadThread() {}; 988 virtual void flushHw_l(); 989 990protected: 991 // threadLoop snippets 992 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 993 virtual void threadLoop_exit(); 994 995 virtual bool waitingAsyncCallback(); 996 virtual bool waitingAsyncCallback_l(); 997 998private: 999 size_t mPausedWriteLength; // length in bytes of write interrupted by pause 1000 size_t mPausedBytesRemaining; // bytes still waiting in mixbuffer after resume 1001}; 1002 1003class AsyncCallbackThread : public Thread { 1004public: 1005 1006 AsyncCallbackThread(const wp<PlaybackThread>& playbackThread); 1007 1008 virtual ~AsyncCallbackThread(); 1009 1010 // Thread virtuals 1011 virtual bool threadLoop(); 1012 1013 // RefBase 1014 virtual void onFirstRef(); 1015 1016 void exit(); 1017 void setWriteBlocked(uint32_t sequence); 1018 void resetWriteBlocked(); 1019 void setDraining(uint32_t sequence); 1020 void resetDraining(); 1021 1022private: 1023 const wp<PlaybackThread> mPlaybackThread; 1024 // mWriteAckSequence corresponds to the last write sequence passed by the offload thread via 1025 // setWriteBlocked(). The sequence is shifted one bit to the left and the lsb is used 1026 // to indicate that the callback has been received via resetWriteBlocked() 1027 uint32_t mWriteAckSequence; 1028 // mDrainSequence corresponds to the last drain sequence passed by the offload thread via 1029 // setDraining(). The sequence is shifted one bit to the left and the lsb is used 1030 // to indicate that the callback has been received via resetDraining() 1031 uint32_t mDrainSequence; 1032 Condition mWaitWorkCV; 1033 Mutex mLock; 1034}; 1035 1036class DuplicatingThread : public MixerThread { 1037public: 1038 DuplicatingThread(const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread, 1039 audio_io_handle_t id, bool systemReady); 1040 virtual ~DuplicatingThread(); 1041 1042 // Thread virtuals 1043 void addOutputTrack(MixerThread* thread); 1044 void removeOutputTrack(MixerThread* thread); 1045 uint32_t waitTimeMs() const { return mWaitTimeMs; } 1046protected: 1047 virtual uint32_t activeSleepTimeUs() const; 1048 1049private: 1050 bool outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks); 1051protected: 1052 // threadLoop snippets 1053 virtual void threadLoop_mix(); 1054 virtual void threadLoop_sleepTime(); 1055 virtual ssize_t threadLoop_write(); 1056 virtual void threadLoop_standby(); 1057 virtual void cacheParameters_l(); 1058 1059private: 1060 // called from threadLoop, addOutputTrack, removeOutputTrack 1061 virtual void updateWaitTime_l(); 1062protected: 1063 virtual void saveOutputTracks(); 1064 virtual void clearOutputTracks(); 1065private: 1066 1067 uint32_t mWaitTimeMs; 1068 SortedVector < sp<OutputTrack> > outputTracks; 1069 SortedVector < sp<OutputTrack> > mOutputTracks; 1070public: 1071 virtual bool hasFastMixer() const { return false; } 1072}; 1073 1074 1075// record thread 1076class RecordThread : public ThreadBase 1077{ 1078public: 1079 1080 class RecordTrack; 1081 1082 /* The ResamplerBufferProvider is used to retrieve recorded input data from the 1083 * RecordThread. It maintains local state on the relative position of the read 1084 * position of the RecordTrack compared with the RecordThread. 1085 */ 1086 class ResamplerBufferProvider : public AudioBufferProvider 1087 { 1088 public: 1089 ResamplerBufferProvider(RecordTrack* recordTrack) : 1090 mRecordTrack(recordTrack), 1091 mRsmpInUnrel(0), mRsmpInFront(0) { } 1092 virtual ~ResamplerBufferProvider() { } 1093 1094 // called to set the ResamplerBufferProvider to head of the RecordThread data buffer, 1095 // skipping any previous data read from the hal. 1096 virtual void reset(); 1097 1098 /* Synchronizes RecordTrack position with the RecordThread. 1099 * Calculates available frames and handle overruns if the RecordThread 1100 * has advanced faster than the ResamplerBufferProvider has retrieved data. 1101 * TODO: why not do this for every getNextBuffer? 1102 * 1103 * Parameters 1104 * framesAvailable: pointer to optional output size_t to store record track 1105 * frames available. 1106 * hasOverrun: pointer to optional boolean, returns true if track has overrun. 1107 */ 1108 1109 virtual void sync(size_t *framesAvailable = NULL, bool *hasOverrun = NULL); 1110 1111 // AudioBufferProvider interface 1112 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer); 1113 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 1114 private: 1115 RecordTrack * const mRecordTrack; 1116 size_t mRsmpInUnrel; // unreleased frames remaining from 1117 // most recent getNextBuffer 1118 // for debug only 1119 int32_t mRsmpInFront; // next available frame 1120 // rolling counter that is never cleared 1121 }; 1122 1123 /* The RecordBufferConverter is used for format, channel, and sample rate 1124 * conversion for a RecordTrack. 1125 * 1126 * TODO: Self contained, so move to a separate file later. 1127 * 1128 * RecordBufferConverter uses the convert() method rather than exposing a 1129 * buffer provider interface; this is to save a memory copy. 1130 */ 1131 class RecordBufferConverter 1132 { 1133 public: 1134 RecordBufferConverter( 1135 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 1136 uint32_t srcSampleRate, 1137 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 1138 uint32_t dstSampleRate); 1139 1140 ~RecordBufferConverter(); 1141 1142 /* Converts input data from an AudioBufferProvider by format, channelMask, 1143 * and sampleRate to a destination buffer. 1144 * 1145 * Parameters 1146 * dst: buffer to place the converted data. 1147 * provider: buffer provider to obtain source data. 1148 * frames: number of frames to convert 1149 * 1150 * Returns the number of frames converted. 1151 */ 1152 size_t convert(void *dst, AudioBufferProvider *provider, size_t frames); 1153 1154 // returns NO_ERROR if constructor was successful 1155 status_t initCheck() const { 1156 // mSrcChannelMask set on successful updateParameters 1157 return mSrcChannelMask != AUDIO_CHANNEL_INVALID ? NO_ERROR : NO_INIT; 1158 } 1159 1160 // allows dynamic reconfigure of all parameters 1161 status_t updateParameters( 1162 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 1163 uint32_t srcSampleRate, 1164 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 1165 uint32_t dstSampleRate); 1166 1167 // called to reset resampler buffers on record track discontinuity 1168 void reset() { 1169 if (mResampler != NULL) { 1170 mResampler->reset(); 1171 } 1172 } 1173 1174 private: 1175 // format conversion when not using resampler 1176 void convertNoResampler(void *dst, const void *src, size_t frames); 1177 1178 // format conversion when using resampler; modifies src in-place 1179 void convertResampler(void *dst, /*not-a-const*/ void *src, size_t frames); 1180 1181 // user provided information 1182 audio_channel_mask_t mSrcChannelMask; 1183 audio_format_t mSrcFormat; 1184 uint32_t mSrcSampleRate; 1185 audio_channel_mask_t mDstChannelMask; 1186 audio_format_t mDstFormat; 1187 uint32_t mDstSampleRate; 1188 1189 // derived information 1190 uint32_t mSrcChannelCount; 1191 uint32_t mDstChannelCount; 1192 size_t mDstFrameSize; 1193 1194 // format conversion buffer 1195 void *mBuf; 1196 size_t mBufFrames; 1197 size_t mBufFrameSize; 1198 1199 // resampler info 1200 AudioResampler *mResampler; 1201 1202 bool mIsLegacyDownmix; // legacy stereo to mono conversion needed 1203 bool mIsLegacyUpmix; // legacy mono to stereo conversion needed 1204 bool mRequiresFloat; // data processing requires float (e.g. resampler) 1205 PassthruBufferProvider *mInputConverterProvider; // converts input to float 1206 int8_t mIdxAry[sizeof(uint32_t) * 8]; // used for channel mask conversion 1207 }; 1208 1209#include "RecordTracks.h" 1210 1211 RecordThread(const sp<AudioFlinger>& audioFlinger, 1212 AudioStreamIn *input, 1213 audio_io_handle_t id, 1214 audio_devices_t outDevice, 1215 audio_devices_t inDevice, 1216 bool systemReady 1217#ifdef TEE_SINK 1218 , const sp<NBAIO_Sink>& teeSink 1219#endif 1220 ); 1221 virtual ~RecordThread(); 1222 1223 // no addTrack_l ? 1224 void destroyTrack_l(const sp<RecordTrack>& track); 1225 void removeTrack_l(const sp<RecordTrack>& track); 1226 1227 void dumpInternals(int fd, const Vector<String16>& args); 1228 void dumpTracks(int fd, const Vector<String16>& args); 1229 1230 // Thread virtuals 1231 virtual bool threadLoop(); 1232 1233 // RefBase 1234 virtual void onFirstRef(); 1235 1236 virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; } 1237 1238 virtual sp<MemoryDealer> readOnlyHeap() const { return mReadOnlyHeap; } 1239 1240 virtual sp<IMemory> pipeMemory() const { return mPipeMemory; } 1241 1242 sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l( 1243 const sp<AudioFlinger::Client>& client, 1244 uint32_t sampleRate, 1245 audio_format_t format, 1246 audio_channel_mask_t channelMask, 1247 size_t *pFrameCount, 1248 int sessionId, 1249 size_t *notificationFrames, 1250 int uid, 1251 IAudioFlinger::track_flags_t *flags, 1252 pid_t tid, 1253 status_t *status /*non-NULL*/); 1254 1255 status_t start(RecordTrack* recordTrack, 1256 AudioSystem::sync_event_t event, 1257 int triggerSession); 1258 1259 // ask the thread to stop the specified track, and 1260 // return true if the caller should then do it's part of the stopping process 1261 bool stop(RecordTrack* recordTrack); 1262 1263 void dump(int fd, const Vector<String16>& args); 1264 AudioStreamIn* clearInput(); 1265 virtual audio_stream_t* stream() const; 1266 1267 1268 virtual bool checkForNewParameter_l(const String8& keyValuePair, 1269 status_t& status); 1270 virtual void cacheParameters_l() {} 1271 virtual String8 getParameters(const String8& keys); 1272 virtual void ioConfigChanged(audio_io_config_event event, pid_t pid = 0); 1273 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 1274 audio_patch_handle_t *handle); 1275 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 1276 1277 void addPatchRecord(const sp<PatchRecord>& record); 1278 void deletePatchRecord(const sp<PatchRecord>& record); 1279 1280 void readInputParameters_l(); 1281 virtual uint32_t getInputFramesLost(); 1282 1283 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1284 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1285 virtual uint32_t hasAudioSession(int sessionId) const; 1286 1287 // Return the set of unique session IDs across all tracks. 1288 // The keys are the session IDs, and the associated values are meaningless. 1289 // FIXME replace by Set [and implement Bag/Multiset for other uses]. 1290 KeyedVector<int, bool> sessionIds() const; 1291 1292 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 1293 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 1294 1295 static void syncStartEventCallback(const wp<SyncEvent>& event); 1296 1297 virtual size_t frameCount() const { return mFrameCount; } 1298 bool hasFastCapture() const { return mFastCapture != 0; } 1299 virtual void getAudioPortConfig(struct audio_port_config *config); 1300 1301private: 1302 // Enter standby if not already in standby, and set mStandby flag 1303 void standbyIfNotAlreadyInStandby(); 1304 1305 // Call the HAL standby method unconditionally, and don't change mStandby flag 1306 void inputStandBy(); 1307 1308 AudioStreamIn *mInput; 1309 SortedVector < sp<RecordTrack> > mTracks; 1310 // mActiveTracks has dual roles: it indicates the current active track(s), and 1311 // is used together with mStartStopCond to indicate start()/stop() progress 1312 SortedVector< sp<RecordTrack> > mActiveTracks; 1313 // generation counter for mActiveTracks 1314 int mActiveTracksGen; 1315 Condition mStartStopCond; 1316 1317 // resampler converts input at HAL Hz to output at AudioRecord client Hz 1318 void *mRsmpInBuffer; // 1319 size_t mRsmpInFrames; // size of resampler input in frames 1320 size_t mRsmpInFramesP2;// size rounded up to a power-of-2 1321 1322 // rolling index that is never cleared 1323 int32_t mRsmpInRear; // last filled frame + 1 1324 1325 // For dumpsys 1326 const sp<NBAIO_Sink> mTeeSink; 1327 1328 const sp<MemoryDealer> mReadOnlyHeap; 1329 1330 // one-time initialization, no locks required 1331 sp<FastCapture> mFastCapture; // non-0 if there is also 1332 // a fast capture 1333 1334 // FIXME audio watchdog thread 1335 1336 // contents are not guaranteed to be consistent, no locks required 1337 FastCaptureDumpState mFastCaptureDumpState; 1338#ifdef STATE_QUEUE_DUMP 1339 // FIXME StateQueue observer and mutator dump fields 1340#endif 1341 // FIXME audio watchdog dump 1342 1343 // accessible only within the threadLoop(), no locks required 1344 // mFastCapture->sq() // for mutating and pushing state 1345 int32_t mFastCaptureFutex; // for cold idle 1346 1347 // The HAL input source is treated as non-blocking, 1348 // but current implementation is blocking 1349 sp<NBAIO_Source> mInputSource; 1350 // The source for the normal capture thread to read from: mInputSource or mPipeSource 1351 sp<NBAIO_Source> mNormalSource; 1352 // If a fast capture is present, the non-blocking pipe sink written to by fast capture, 1353 // otherwise clear 1354 sp<NBAIO_Sink> mPipeSink; 1355 // If a fast capture is present, the non-blocking pipe source read by normal thread, 1356 // otherwise clear 1357 sp<NBAIO_Source> mPipeSource; 1358 // Depth of pipe from fast capture to normal thread and fast clients, always power of 2 1359 size_t mPipeFramesP2; 1360 // If a fast capture is present, the Pipe as IMemory, otherwise clear 1361 sp<IMemory> mPipeMemory; 1362 1363 static const size_t kFastCaptureLogSize = 4 * 1024; 1364 sp<NBLog::Writer> mFastCaptureNBLogWriter; 1365 1366 bool mFastTrackAvail; // true if fast track available 1367}; 1368