Threads.h revision d848eb48c121c119e8ba7583efc75415fe102570
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 audio_session_t 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(audio_session_t sessionId); 306 // same as getEffectChain() but must be called with ThreadBase mutex locked 307 sp<EffectChain> getEffectChain_l(audio_session_t 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(audio_session_t sessionId, int effectId); 325 sp<AudioFlinger::EffectModule> getEffect_l(audio_session_t 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(audio_session_t 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(audio_session_t sessionId __unused) 340 { return 0; } 341 342 // suspend or restore effect according to the type of effect passed. a NULL 343 // type pointer means suspend all effects in the session 344 void setEffectSuspended(const effect_uuid_t *type, 345 bool suspend, 346 audio_session_t sessionId = AUDIO_SESSION_OUTPUT_MIX); 347 // check if some effects must be suspended/restored when an effect is enabled 348 // or disabled 349 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 350 bool enabled, 351 audio_session_t sessionId = 352 AUDIO_SESSION_OUTPUT_MIX); 353 void checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 354 bool enabled, 355 audio_session_t sessionId = 356 AUDIO_SESSION_OUTPUT_MIX); 357 358 virtual status_t setSyncEvent(const sp<SyncEvent>& event) = 0; 359 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const = 0; 360 361 // Return a reference to a per-thread heap which can be used to allocate IMemory 362 // objects that will be read-only to client processes, read/write to mediaserver, 363 // and shared by all client processes of the thread. 364 // The heap is per-thread rather than common across all threads, because 365 // clients can't be trusted not to modify the offset of the IMemory they receive. 366 // If a thread does not have such a heap, this method returns 0. 367 virtual sp<MemoryDealer> readOnlyHeap() const { return 0; } 368 369 virtual sp<IMemory> pipeMemory() const { return 0; } 370 371 void systemReady(); 372 373 mutable Mutex mLock; 374 375protected: 376 377 // entry describing an effect being suspended in mSuspendedSessions keyed vector 378 class SuspendedSessionDesc : public RefBase { 379 public: 380 SuspendedSessionDesc() : mRefCount(0) {} 381 382 int mRefCount; // number of active suspend requests 383 effect_uuid_t mType; // effect type UUID 384 }; 385 386 void acquireWakeLock(int uid = -1); 387 virtual void acquireWakeLock_l(int uid = -1); 388 void releaseWakeLock(); 389 void releaseWakeLock_l(); 390 void updateWakeLockUids(const SortedVector<int> &uids); 391 void updateWakeLockUids_l(const SortedVector<int> &uids); 392 void getPowerManager_l(); 393 void setEffectSuspended_l(const effect_uuid_t *type, 394 bool suspend, 395 audio_session_t sessionId); 396 // updated mSuspendedSessions when an effect suspended or restored 397 void updateSuspendedSessions_l(const effect_uuid_t *type, 398 bool suspend, 399 audio_session_t sessionId); 400 // check if some effects must be suspended when an effect chain is added 401 void checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain); 402 403 String16 getWakeLockTag(); 404 405 virtual void preExit() { } 406 virtual void setMasterMono_l(bool mono __unused) { } 407 virtual bool requireMonoBlend() { return false; } 408 409 friend class AudioFlinger; // for mEffectChains 410 411 const type_t mType; 412 413 // Used by parameters, config events, addTrack_l, exit 414 Condition mWaitWorkCV; 415 416 const sp<AudioFlinger> mAudioFlinger; 417 418 // updated by PlaybackThread::readOutputParameters_l() or 419 // RecordThread::readInputParameters_l() 420 uint32_t mSampleRate; 421 size_t mFrameCount; // output HAL, direct output, record 422 audio_channel_mask_t mChannelMask; 423 uint32_t mChannelCount; 424 size_t mFrameSize; 425 // not HAL frame size, this is for output sink (to pipe to fast mixer) 426 audio_format_t mFormat; // Source format for Recording and 427 // Sink format for Playback. 428 // Sink format may be different than 429 // HAL format if Fastmixer is used. 430 audio_format_t mHALFormat; 431 size_t mBufferSize; // HAL buffer size for read() or write() 432 433 Vector< sp<ConfigEvent> > mConfigEvents; 434 Vector< sp<ConfigEvent> > mPendingConfigEvents; // events awaiting system ready 435 436 // These fields are written and read by thread itself without lock or barrier, 437 // and read by other threads without lock or barrier via standby(), outDevice() 438 // and inDevice(). 439 // Because of the absence of a lock or barrier, any other thread that reads 440 // these fields must use the information in isolation, or be prepared to deal 441 // with possibility that it might be inconsistent with other information. 442 bool mStandby; // Whether thread is currently in standby. 443 audio_devices_t mOutDevice; // output device 444 audio_devices_t mInDevice; // input device 445 audio_devices_t mPrevOutDevice; // previous output device 446 audio_devices_t mPrevInDevice; // previous input device 447 struct audio_patch mPatch; 448 audio_source_t mAudioSource; 449 450 const audio_io_handle_t mId; 451 Vector< sp<EffectChain> > mEffectChains; 452 453 static const int kThreadNameLength = 16; // prctl(PR_SET_NAME) limit 454 char mThreadName[kThreadNameLength]; // guaranteed NUL-terminated 455 sp<IPowerManager> mPowerManager; 456 sp<IBinder> mWakeLockToken; 457 const sp<PMDeathRecipient> mDeathRecipient; 458 // list of suspended effects per session and per type. The first (outer) vector is 459 // keyed by session ID, the second (inner) by type UUID timeLow field 460 KeyedVector< audio_session_t, KeyedVector< int, sp<SuspendedSessionDesc> > > 461 mSuspendedSessions; 462 static const size_t kLogSize = 4 * 1024; 463 sp<NBLog::Writer> mNBLogWriter; 464 bool mSystemReady; 465 bool mNotifiedBatteryStart; 466 ExtendedTimestamp mTimestamp; 467}; 468 469// --- PlaybackThread --- 470class PlaybackThread : public ThreadBase { 471public: 472 473#include "PlaybackTracks.h" 474 475 enum mixer_state { 476 MIXER_IDLE, // no active tracks 477 MIXER_TRACKS_ENABLED, // at least one active track, but no track has any data ready 478 MIXER_TRACKS_READY, // at least one active track, and at least one track has data 479 MIXER_DRAIN_TRACK, // drain currently playing track 480 MIXER_DRAIN_ALL, // fully drain the hardware 481 // standby mode does not have an enum value 482 // suspend by audio policy manager is orthogonal to mixer state 483 }; 484 485 PlaybackThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 486 audio_io_handle_t id, audio_devices_t device, type_t type, bool systemReady, 487 uint32_t bitRate = 0); 488 virtual ~PlaybackThread(); 489 490 void dump(int fd, const Vector<String16>& args); 491 492 // Thread virtuals 493 virtual bool threadLoop(); 494 495 // RefBase 496 virtual void onFirstRef(); 497 498protected: 499 // Code snippets that were lifted up out of threadLoop() 500 virtual void threadLoop_mix() = 0; 501 virtual void threadLoop_sleepTime() = 0; 502 virtual ssize_t threadLoop_write(); 503 virtual void threadLoop_drain(); 504 virtual void threadLoop_standby(); 505 virtual void threadLoop_exit(); 506 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 507 508 // prepareTracks_l reads and writes mActiveTracks, and returns 509 // the pending set of tracks to remove via Vector 'tracksToRemove'. The caller 510 // is responsible for clearing or destroying this Vector later on, when it 511 // is safe to do so. That will drop the final ref count and destroy the tracks. 512 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove) = 0; 513 void removeTracks_l(const Vector< sp<Track> >& tracksToRemove); 514 515 void writeCallback(); 516 void resetWriteBlocked(uint32_t sequence); 517 void drainCallback(); 518 void resetDraining(uint32_t sequence); 519 520 static int asyncCallback(stream_callback_event_t event, void *param, void *cookie); 521 522 virtual bool waitingAsyncCallback(); 523 virtual bool waitingAsyncCallback_l(); 524 virtual bool shouldStandby_l(); 525 virtual void onAddNewTrack_l(); 526 527 // ThreadBase virtuals 528 virtual void preExit(); 529 530public: 531 532 virtual status_t initCheck() const { return (mOutput == NULL) ? NO_INIT : NO_ERROR; } 533 534 // return estimated latency in milliseconds, as reported by HAL 535 uint32_t latency() const; 536 // same, but lock must already be held 537 uint32_t latency_l() const; 538 539 void setMasterVolume(float value); 540 void setMasterMute(bool muted); 541 542 void setStreamVolume(audio_stream_type_t stream, float value); 543 void setStreamMute(audio_stream_type_t stream, bool muted); 544 545 float streamVolume(audio_stream_type_t stream) const; 546 547 sp<Track> createTrack_l( 548 const sp<AudioFlinger::Client>& client, 549 audio_stream_type_t streamType, 550 uint32_t sampleRate, 551 audio_format_t format, 552 audio_channel_mask_t channelMask, 553 size_t *pFrameCount, 554 const sp<IMemory>& sharedBuffer, 555 audio_session_t sessionId, 556 IAudioFlinger::track_flags_t *flags, 557 pid_t tid, 558 int uid, 559 status_t *status /*non-NULL*/); 560 561 AudioStreamOut* getOutput() const; 562 AudioStreamOut* clearOutput(); 563 virtual audio_stream_t* stream() const; 564 565 // a very large number of suspend() will eventually wraparound, but unlikely 566 void suspend() { (void) android_atomic_inc(&mSuspended); } 567 void restore() 568 { 569 // if restore() is done without suspend(), get back into 570 // range so that the next suspend() will operate correctly 571 if (android_atomic_dec(&mSuspended) <= 0) { 572 android_atomic_release_store(0, &mSuspended); 573 } 574 } 575 bool isSuspended() const 576 { return android_atomic_acquire_load(&mSuspended) > 0; } 577 578 virtual String8 getParameters(const String8& keys); 579 virtual void ioConfigChanged(audio_io_config_event event, pid_t pid = 0); 580 status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames); 581 // FIXME rename mixBuffer() to sinkBuffer() and remove int16_t* dependency. 582 // Consider also removing and passing an explicit mMainBuffer initialization 583 // parameter to AF::PlaybackThread::Track::Track(). 584 int16_t *mixBuffer() const { 585 return reinterpret_cast<int16_t *>(mSinkBuffer); }; 586 587 virtual void detachAuxEffect_l(int effectId); 588 status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track> track, 589 int EffectId); 590 status_t attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track> track, 591 int EffectId); 592 593 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 594 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 595 virtual uint32_t hasAudioSession(audio_session_t sessionId) const; 596 virtual uint32_t getStrategyForSession_l(audio_session_t sessionId); 597 598 599 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 600 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 601 602 // called with AudioFlinger lock held 603 void invalidateTracks(audio_stream_type_t streamType); 604 605 virtual size_t frameCount() const { return mNormalFrameCount; } 606 607 // Return's the HAL's frame count i.e. fast mixer buffer size. 608 size_t frameCountHAL() const { return mFrameCount; } 609 610 status_t getTimestamp_l(AudioTimestamp& timestamp); 611 612 void addPatchTrack(const sp<PatchTrack>& track); 613 void deletePatchTrack(const sp<PatchTrack>& track); 614 615 virtual void getAudioPortConfig(struct audio_port_config *config); 616 617protected: 618 // updated by readOutputParameters_l() 619 size_t mNormalFrameCount; // normal mixer and effects 620 621 bool mThreadThrottle; // throttle the thread processing 622 uint32_t mThreadThrottleTimeMs; // throttle time for MIXER threads 623 uint32_t mThreadThrottleEndMs; // notify once per throttling 624 uint32_t mHalfBufferMs; // half the buffer size in milliseconds 625 626 void* mSinkBuffer; // frame size aligned sink buffer 627 628 // TODO: 629 // Rearrange the buffer info into a struct/class with 630 // clear, copy, construction, destruction methods. 631 // 632 // mSinkBuffer also has associated with it: 633 // 634 // mSinkBufferSize: Sink Buffer Size 635 // mFormat: Sink Buffer Format 636 637 // Mixer Buffer (mMixerBuffer*) 638 // 639 // In the case of floating point or multichannel data, which is not in the 640 // sink format, it is required to accumulate in a higher precision or greater channel count 641 // buffer before downmixing or data conversion to the sink buffer. 642 643 // Set to "true" to enable the Mixer Buffer otherwise mixer output goes to sink buffer. 644 bool mMixerBufferEnabled; 645 646 // Storage, 32 byte aligned (may make this alignment a requirement later). 647 // Due to constraints on mNormalFrameCount, the buffer size is a multiple of 16 frames. 648 void* mMixerBuffer; 649 650 // Size of mMixerBuffer in bytes: mNormalFrameCount * #channels * sampsize. 651 size_t mMixerBufferSize; 652 653 // The audio format of mMixerBuffer. Set to AUDIO_FORMAT_PCM_(FLOAT|16_BIT) only. 654 audio_format_t mMixerBufferFormat; 655 656 // An internal flag set to true by MixerThread::prepareTracks_l() 657 // when mMixerBuffer contains valid data after mixing. 658 bool mMixerBufferValid; 659 660 // Effects Buffer (mEffectsBuffer*) 661 // 662 // In the case of effects data, which is not in the sink format, 663 // it is required to accumulate in a different buffer before data conversion 664 // to the sink buffer. 665 666 // Set to "true" to enable the Effects Buffer otherwise effects output goes to sink buffer. 667 bool mEffectBufferEnabled; 668 669 // Storage, 32 byte aligned (may make this alignment a requirement later). 670 // Due to constraints on mNormalFrameCount, the buffer size is a multiple of 16 frames. 671 void* mEffectBuffer; 672 673 // Size of mEffectsBuffer in bytes: mNormalFrameCount * #channels * sampsize. 674 size_t mEffectBufferSize; 675 676 // The audio format of mEffectsBuffer. Set to AUDIO_FORMAT_PCM_16_BIT only. 677 audio_format_t mEffectBufferFormat; 678 679 // An internal flag set to true by MixerThread::prepareTracks_l() 680 // when mEffectsBuffer contains valid data after mixing. 681 // 682 // When this is set, all mixer data is routed into the effects buffer 683 // for any processing (including output processing). 684 bool mEffectBufferValid; 685 686 // suspend count, > 0 means suspended. While suspended, the thread continues to pull from 687 // tracks and mix, but doesn't write to HAL. A2DP and SCO HAL implementations can't handle 688 // concurrent use of both of them, so Audio Policy Service suspends one of the threads to 689 // workaround that restriction. 690 // 'volatile' means accessed via atomic operations and no lock. 691 volatile int32_t mSuspended; 692 693 int64_t mBytesWritten; 694 int64_t mFramesWritten; // not reset on standby 695private: 696 // mMasterMute is in both PlaybackThread and in AudioFlinger. When a 697 // PlaybackThread needs to find out if master-muted, it checks it's local 698 // copy rather than the one in AudioFlinger. This optimization saves a lock. 699 bool mMasterMute; 700 void setMasterMute_l(bool muted) { mMasterMute = muted; } 701protected: 702 SortedVector< wp<Track> > mActiveTracks; // FIXME check if this could be sp<> 703 SortedVector<int> mWakeLockUids; 704 int mActiveTracksGeneration; 705 wp<Track> mLatestActiveTrack; // latest track added to mActiveTracks 706 707 // Allocate a track name for a given channel mask. 708 // Returns name >= 0 if successful, -1 on failure. 709 virtual int getTrackName_l(audio_channel_mask_t channelMask, 710 audio_format_t format, audio_session_t sessionId) = 0; 711 virtual void deleteTrackName_l(int name) = 0; 712 713 // Time to sleep between cycles when: 714 virtual uint32_t activeSleepTimeUs() const; // mixer state MIXER_TRACKS_ENABLED 715 virtual uint32_t idleSleepTimeUs() const = 0; // mixer state MIXER_IDLE 716 virtual uint32_t suspendSleepTimeUs() const = 0; // audio policy manager suspended us 717 // No sleep when mixer state == MIXER_TRACKS_READY; relies on audio HAL stream->write() 718 // No sleep in standby mode; waits on a condition 719 720 // Code snippets that are temporarily lifted up out of threadLoop() until the merge 721 void checkSilentMode_l(); 722 723 // Non-trivial for DUPLICATING only 724 virtual void saveOutputTracks() { } 725 virtual void clearOutputTracks() { } 726 727 // Cache various calculated values, at threadLoop() entry and after a parameter change 728 virtual void cacheParameters_l(); 729 730 virtual uint32_t correctLatency_l(uint32_t latency) const; 731 732 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 733 audio_patch_handle_t *handle); 734 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 735 736 bool usesHwAvSync() const { return (mType == DIRECT) && (mOutput != NULL) 737 && mHwSupportsPause 738 && (mOutput->flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC); } 739 740private: 741 742 friend class AudioFlinger; // for numerous 743 744 PlaybackThread& operator = (const PlaybackThread&); 745 746 status_t addTrack_l(const sp<Track>& track); 747 bool destroyTrack_l(const sp<Track>& track); 748 void removeTrack_l(const sp<Track>& track); 749 void broadcast_l(); 750 751 void readOutputParameters_l(); 752 753 virtual void dumpInternals(int fd, const Vector<String16>& args); 754 void dumpTracks(int fd, const Vector<String16>& args); 755 756 SortedVector< sp<Track> > mTracks; 757 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 758 AudioStreamOut *mOutput; 759 760 float mMasterVolume; 761 nsecs_t mLastWriteTime; 762 int mNumWrites; 763 int mNumDelayedWrites; 764 bool mInWrite; 765 766 // FIXME rename these former local variables of threadLoop to standard "m" names 767 nsecs_t mStandbyTimeNs; 768 size_t mSinkBufferSize; 769 770 // cached copies of activeSleepTimeUs() and idleSleepTimeUs() made by cacheParameters_l() 771 uint32_t mActiveSleepTimeUs; 772 uint32_t mIdleSleepTimeUs; 773 774 uint32_t mSleepTimeUs; 775 776 // mixer status returned by prepareTracks_l() 777 mixer_state mMixerStatus; // current cycle 778 // previous cycle when in prepareTracks_l() 779 mixer_state mMixerStatusIgnoringFastTracks; 780 // FIXME or a separate ready state per track 781 782 // FIXME move these declarations into the specific sub-class that needs them 783 // MIXER only 784 uint32_t sleepTimeShift; 785 786 // same as AudioFlinger::mStandbyTimeInNsecs except for DIRECT which uses a shorter value 787 nsecs_t mStandbyDelayNs; 788 789 // MIXER only 790 nsecs_t maxPeriod; 791 792 // DUPLICATING only 793 uint32_t writeFrames; 794 795 size_t mBytesRemaining; 796 size_t mCurrentWriteLength; 797 bool mUseAsyncWrite; 798 // mWriteAckSequence contains current write sequence on bits 31-1. The write sequence is 799 // incremented each time a write(), a flush() or a standby() occurs. 800 // Bit 0 is set when a write blocks and indicates a callback is expected. 801 // Bit 0 is reset by the async callback thread calling resetWriteBlocked(). Out of sequence 802 // callbacks are ignored. 803 uint32_t mWriteAckSequence; 804 // mDrainSequence contains current drain sequence on bits 31-1. The drain sequence is 805 // incremented each time a drain is requested or a flush() or standby() occurs. 806 // Bit 0 is set when the drain() command is called at the HAL and indicates a callback is 807 // expected. 808 // Bit 0 is reset by the async callback thread calling resetDraining(). Out of sequence 809 // callbacks are ignored. 810 uint32_t mDrainSequence; 811 // A condition that must be evaluated by prepareTrack_l() has changed and we must not wait 812 // for async write callback in the thread loop before evaluating it 813 bool mSignalPending; 814 sp<AsyncCallbackThread> mCallbackThread; 815 816private: 817 // The HAL output sink is treated as non-blocking, but current implementation is blocking 818 sp<NBAIO_Sink> mOutputSink; 819 // If a fast mixer is present, the blocking pipe sink, otherwise clear 820 sp<NBAIO_Sink> mPipeSink; 821 // The current sink for the normal mixer to write it's (sub)mix, mOutputSink or mPipeSink 822 sp<NBAIO_Sink> mNormalSink; 823#ifdef TEE_SINK 824 // For dumpsys 825 sp<NBAIO_Sink> mTeeSink; 826 sp<NBAIO_Source> mTeeSource; 827#endif 828 uint32_t mScreenState; // cached copy of gScreenState 829 static const size_t kFastMixerLogSize = 4 * 1024; 830 sp<NBLog::Writer> mFastMixerNBLogWriter; 831public: 832 virtual bool hasFastMixer() const = 0; 833 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex __unused) const 834 { FastTrackUnderruns dummy; return dummy; } 835 836protected: 837 // accessed by both binder threads and within threadLoop(), lock on mutex needed 838 unsigned mFastTrackAvailMask; // bit i set if fast track [i] is available 839 bool mHwSupportsPause; 840 bool mHwPaused; 841 bool mFlushPending; 842 uint32_t mBufferDurationUs; // estimated duration of an audio HAL buffer 843 // based on initial bit rate (offload only) 844}; 845 846class MixerThread : public PlaybackThread { 847public: 848 MixerThread(const sp<AudioFlinger>& audioFlinger, 849 AudioStreamOut* output, 850 audio_io_handle_t id, 851 audio_devices_t device, 852 bool systemReady, 853 type_t type = MIXER); 854 virtual ~MixerThread(); 855 856 // Thread virtuals 857 858 virtual bool checkForNewParameter_l(const String8& keyValuePair, 859 status_t& status); 860 virtual void dumpInternals(int fd, const Vector<String16>& args); 861 862protected: 863 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 864 virtual int getTrackName_l(audio_channel_mask_t channelMask, 865 audio_format_t format, audio_session_t sessionId); 866 virtual void deleteTrackName_l(int name); 867 virtual uint32_t idleSleepTimeUs() const; 868 virtual uint32_t suspendSleepTimeUs() const; 869 virtual void cacheParameters_l(); 870 871 virtual void acquireWakeLock_l(int uid = -1) { 872 PlaybackThread::acquireWakeLock_l(uid); 873 if (hasFastMixer()) { 874 mFastMixer->setBoottimeOffset( 875 mTimestamp.mTimebaseOffset[ExtendedTimestamp::TIMEBASE_BOOTTIME]); 876 } 877 } 878 879 // threadLoop snippets 880 virtual ssize_t threadLoop_write(); 881 virtual void threadLoop_standby(); 882 virtual void threadLoop_mix(); 883 virtual void threadLoop_sleepTime(); 884 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 885 virtual uint32_t correctLatency_l(uint32_t latency) const; 886 887 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 888 audio_patch_handle_t *handle); 889 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 890 891 AudioMixer* mAudioMixer; // normal mixer 892private: 893 // one-time initialization, no locks required 894 sp<FastMixer> mFastMixer; // non-0 if there is also a fast mixer 895 sp<AudioWatchdog> mAudioWatchdog; // non-0 if there is an audio watchdog thread 896 897 // contents are not guaranteed to be consistent, no locks required 898 FastMixerDumpState mFastMixerDumpState; 899#ifdef STATE_QUEUE_DUMP 900 StateQueueObserverDump mStateQueueObserverDump; 901 StateQueueMutatorDump mStateQueueMutatorDump; 902#endif 903 AudioWatchdogDump mAudioWatchdogDump; 904 905 // accessible only within the threadLoop(), no locks required 906 // mFastMixer->sq() // for mutating and pushing state 907 int32_t mFastMixerFutex; // for cold idle 908 909 std::atomic_bool mMasterMono; 910public: 911 virtual bool hasFastMixer() const { return mFastMixer != 0; } 912 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const { 913 ALOG_ASSERT(fastIndex < FastMixerState::kMaxFastTracks); 914 return mFastMixerDumpState.mTracks[fastIndex].mUnderruns; 915 } 916 917protected: 918 virtual void setMasterMono_l(bool mono) { 919 mMasterMono.store(mono); 920 if (mFastMixer != nullptr) { /* hasFastMixer() */ 921 mFastMixer->setMasterMono(mMasterMono); 922 } 923 } 924 // the FastMixer performs mono blend if it exists. 925 // Blending with limiter is not idempotent, 926 // and blending without limiter is idempotent but inefficient to do twice. 927 virtual bool requireMonoBlend() { return mMasterMono.load() && !hasFastMixer(); } 928}; 929 930class DirectOutputThread : public PlaybackThread { 931public: 932 933 DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 934 audio_io_handle_t id, audio_devices_t device, bool systemReady, 935 uint32_t bitRate = 0); 936 virtual ~DirectOutputThread(); 937 938 // Thread virtuals 939 940 virtual bool checkForNewParameter_l(const String8& keyValuePair, 941 status_t& status); 942 virtual void flushHw_l(); 943 944protected: 945 virtual int getTrackName_l(audio_channel_mask_t channelMask, 946 audio_format_t format, audio_session_t sessionId); 947 virtual void deleteTrackName_l(int name); 948 virtual uint32_t activeSleepTimeUs() const; 949 virtual uint32_t idleSleepTimeUs() const; 950 virtual uint32_t suspendSleepTimeUs() const; 951 virtual void cacheParameters_l(); 952 953 // threadLoop snippets 954 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 955 virtual void threadLoop_mix(); 956 virtual void threadLoop_sleepTime(); 957 virtual void threadLoop_exit(); 958 virtual bool shouldStandby_l(); 959 960 virtual void onAddNewTrack_l(); 961 962 // volumes last sent to audio HAL with stream->set_volume() 963 float mLeftVolFloat; 964 float mRightVolFloat; 965 966 DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 967 audio_io_handle_t id, uint32_t device, ThreadBase::type_t type, 968 bool systemReady, uint32_t bitRate = 0); 969 void processVolume_l(Track *track, bool lastTrack); 970 971 // prepareTracks_l() tells threadLoop_mix() the name of the single active track 972 sp<Track> mActiveTrack; 973 974 wp<Track> mPreviousTrack; // used to detect track switch 975 976public: 977 virtual bool hasFastMixer() const { return false; } 978}; 979 980class OffloadThread : public DirectOutputThread { 981public: 982 983 OffloadThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 984 audio_io_handle_t id, uint32_t device, 985 bool systemReady, uint32_t bitRate); 986 virtual ~OffloadThread() {}; 987 virtual void flushHw_l(); 988 989protected: 990 // threadLoop snippets 991 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 992 virtual void threadLoop_exit(); 993 994 virtual uint32_t activeSleepTimeUs() const; 995 996 virtual bool waitingAsyncCallback(); 997 virtual bool waitingAsyncCallback_l(); 998 999private: 1000 size_t mPausedWriteLength; // length in bytes of write interrupted by pause 1001 size_t mPausedBytesRemaining; // bytes still waiting in mixbuffer after resume 1002}; 1003 1004class AsyncCallbackThread : public Thread { 1005public: 1006 1007 AsyncCallbackThread(const wp<PlaybackThread>& playbackThread); 1008 1009 virtual ~AsyncCallbackThread(); 1010 1011 // Thread virtuals 1012 virtual bool threadLoop(); 1013 1014 // RefBase 1015 virtual void onFirstRef(); 1016 1017 void exit(); 1018 void setWriteBlocked(uint32_t sequence); 1019 void resetWriteBlocked(); 1020 void setDraining(uint32_t sequence); 1021 void resetDraining(); 1022 1023private: 1024 const wp<PlaybackThread> mPlaybackThread; 1025 // mWriteAckSequence corresponds to the last write sequence passed by the offload thread via 1026 // setWriteBlocked(). The sequence is shifted one bit to the left and the lsb is used 1027 // to indicate that the callback has been received via resetWriteBlocked() 1028 uint32_t mWriteAckSequence; 1029 // mDrainSequence corresponds to the last drain sequence passed by the offload thread via 1030 // setDraining(). The sequence is shifted one bit to the left and the lsb is used 1031 // to indicate that the callback has been received via resetDraining() 1032 uint32_t mDrainSequence; 1033 Condition mWaitWorkCV; 1034 Mutex mLock; 1035}; 1036 1037class DuplicatingThread : public MixerThread { 1038public: 1039 DuplicatingThread(const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread, 1040 audio_io_handle_t id, bool systemReady); 1041 virtual ~DuplicatingThread(); 1042 1043 // Thread virtuals 1044 void addOutputTrack(MixerThread* thread); 1045 void removeOutputTrack(MixerThread* thread); 1046 uint32_t waitTimeMs() const { return mWaitTimeMs; } 1047protected: 1048 virtual uint32_t activeSleepTimeUs() const; 1049 1050private: 1051 bool outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks); 1052protected: 1053 // threadLoop snippets 1054 virtual void threadLoop_mix(); 1055 virtual void threadLoop_sleepTime(); 1056 virtual ssize_t threadLoop_write(); 1057 virtual void threadLoop_standby(); 1058 virtual void cacheParameters_l(); 1059 1060private: 1061 // called from threadLoop, addOutputTrack, removeOutputTrack 1062 virtual void updateWaitTime_l(); 1063protected: 1064 virtual void saveOutputTracks(); 1065 virtual void clearOutputTracks(); 1066private: 1067 1068 uint32_t mWaitTimeMs; 1069 SortedVector < sp<OutputTrack> > outputTracks; 1070 SortedVector < sp<OutputTrack> > mOutputTracks; 1071public: 1072 virtual bool hasFastMixer() const { return false; } 1073}; 1074 1075 1076// record thread 1077class RecordThread : public ThreadBase 1078{ 1079public: 1080 1081 class RecordTrack; 1082 1083 /* The ResamplerBufferProvider is used to retrieve recorded input data from the 1084 * RecordThread. It maintains local state on the relative position of the read 1085 * position of the RecordTrack compared with the RecordThread. 1086 */ 1087 class ResamplerBufferProvider : public AudioBufferProvider 1088 { 1089 public: 1090 ResamplerBufferProvider(RecordTrack* recordTrack) : 1091 mRecordTrack(recordTrack), 1092 mRsmpInUnrel(0), mRsmpInFront(0) { } 1093 virtual ~ResamplerBufferProvider() { } 1094 1095 // called to set the ResamplerBufferProvider to head of the RecordThread data buffer, 1096 // skipping any previous data read from the hal. 1097 virtual void reset(); 1098 1099 /* Synchronizes RecordTrack position with the RecordThread. 1100 * Calculates available frames and handle overruns if the RecordThread 1101 * has advanced faster than the ResamplerBufferProvider has retrieved data. 1102 * TODO: why not do this for every getNextBuffer? 1103 * 1104 * Parameters 1105 * framesAvailable: pointer to optional output size_t to store record track 1106 * frames available. 1107 * hasOverrun: pointer to optional boolean, returns true if track has overrun. 1108 */ 1109 1110 virtual void sync(size_t *framesAvailable = NULL, bool *hasOverrun = NULL); 1111 1112 // AudioBufferProvider interface 1113 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer); 1114 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 1115 private: 1116 RecordTrack * const mRecordTrack; 1117 size_t mRsmpInUnrel; // unreleased frames remaining from 1118 // most recent getNextBuffer 1119 // for debug only 1120 int32_t mRsmpInFront; // next available frame 1121 // rolling counter that is never cleared 1122 }; 1123 1124 /* The RecordBufferConverter is used for format, channel, and sample rate 1125 * conversion for a RecordTrack. 1126 * 1127 * TODO: Self contained, so move to a separate file later. 1128 * 1129 * RecordBufferConverter uses the convert() method rather than exposing a 1130 * buffer provider interface; this is to save a memory copy. 1131 */ 1132 class RecordBufferConverter 1133 { 1134 public: 1135 RecordBufferConverter( 1136 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 1137 uint32_t srcSampleRate, 1138 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 1139 uint32_t dstSampleRate); 1140 1141 ~RecordBufferConverter(); 1142 1143 /* Converts input data from an AudioBufferProvider by format, channelMask, 1144 * and sampleRate to a destination buffer. 1145 * 1146 * Parameters 1147 * dst: buffer to place the converted data. 1148 * provider: buffer provider to obtain source data. 1149 * frames: number of frames to convert 1150 * 1151 * Returns the number of frames converted. 1152 */ 1153 size_t convert(void *dst, AudioBufferProvider *provider, size_t frames); 1154 1155 // returns NO_ERROR if constructor was successful 1156 status_t initCheck() const { 1157 // mSrcChannelMask set on successful updateParameters 1158 return mSrcChannelMask != AUDIO_CHANNEL_INVALID ? NO_ERROR : NO_INIT; 1159 } 1160 1161 // allows dynamic reconfigure of all parameters 1162 status_t updateParameters( 1163 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 1164 uint32_t srcSampleRate, 1165 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 1166 uint32_t dstSampleRate); 1167 1168 // called to reset resampler buffers on record track discontinuity 1169 void reset() { 1170 if (mResampler != NULL) { 1171 mResampler->reset(); 1172 } 1173 } 1174 1175 private: 1176 // format conversion when not using resampler 1177 void convertNoResampler(void *dst, const void *src, size_t frames); 1178 1179 // format conversion when using resampler; modifies src in-place 1180 void convertResampler(void *dst, /*not-a-const*/ void *src, size_t frames); 1181 1182 // user provided information 1183 audio_channel_mask_t mSrcChannelMask; 1184 audio_format_t mSrcFormat; 1185 uint32_t mSrcSampleRate; 1186 audio_channel_mask_t mDstChannelMask; 1187 audio_format_t mDstFormat; 1188 uint32_t mDstSampleRate; 1189 1190 // derived information 1191 uint32_t mSrcChannelCount; 1192 uint32_t mDstChannelCount; 1193 size_t mDstFrameSize; 1194 1195 // format conversion buffer 1196 void *mBuf; 1197 size_t mBufFrames; 1198 size_t mBufFrameSize; 1199 1200 // resampler info 1201 AudioResampler *mResampler; 1202 1203 bool mIsLegacyDownmix; // legacy stereo to mono conversion needed 1204 bool mIsLegacyUpmix; // legacy mono to stereo conversion needed 1205 bool mRequiresFloat; // data processing requires float (e.g. resampler) 1206 PassthruBufferProvider *mInputConverterProvider; // converts input to float 1207 int8_t mIdxAry[sizeof(uint32_t) * 8]; // used for channel mask conversion 1208 }; 1209 1210#include "RecordTracks.h" 1211 1212 RecordThread(const sp<AudioFlinger>& audioFlinger, 1213 AudioStreamIn *input, 1214 audio_io_handle_t id, 1215 audio_devices_t outDevice, 1216 audio_devices_t inDevice, 1217 bool systemReady 1218#ifdef TEE_SINK 1219 , const sp<NBAIO_Sink>& teeSink 1220#endif 1221 ); 1222 virtual ~RecordThread(); 1223 1224 // no addTrack_l ? 1225 void destroyTrack_l(const sp<RecordTrack>& track); 1226 void removeTrack_l(const sp<RecordTrack>& track); 1227 1228 void dumpInternals(int fd, const Vector<String16>& args); 1229 void dumpTracks(int fd, const Vector<String16>& args); 1230 1231 // Thread virtuals 1232 virtual bool threadLoop(); 1233 1234 // RefBase 1235 virtual void onFirstRef(); 1236 1237 virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; } 1238 1239 virtual sp<MemoryDealer> readOnlyHeap() const { return mReadOnlyHeap; } 1240 1241 virtual sp<IMemory> pipeMemory() const { return mPipeMemory; } 1242 1243 sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l( 1244 const sp<AudioFlinger::Client>& client, 1245 uint32_t sampleRate, 1246 audio_format_t format, 1247 audio_channel_mask_t channelMask, 1248 size_t *pFrameCount, 1249 audio_session_t sessionId, 1250 size_t *notificationFrames, 1251 int uid, 1252 IAudioFlinger::track_flags_t *flags, 1253 pid_t tid, 1254 status_t *status /*non-NULL*/); 1255 1256 status_t start(RecordTrack* recordTrack, 1257 AudioSystem::sync_event_t event, 1258 audio_session_t triggerSession); 1259 1260 // ask the thread to stop the specified track, and 1261 // return true if the caller should then do it's part of the stopping process 1262 bool stop(RecordTrack* recordTrack); 1263 1264 void dump(int fd, const Vector<String16>& args); 1265 AudioStreamIn* clearInput(); 1266 virtual audio_stream_t* stream() const; 1267 1268 1269 virtual bool checkForNewParameter_l(const String8& keyValuePair, 1270 status_t& status); 1271 virtual void cacheParameters_l() {} 1272 virtual String8 getParameters(const String8& keys); 1273 virtual void ioConfigChanged(audio_io_config_event event, pid_t pid = 0); 1274 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 1275 audio_patch_handle_t *handle); 1276 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 1277 1278 void addPatchRecord(const sp<PatchRecord>& record); 1279 void deletePatchRecord(const sp<PatchRecord>& record); 1280 1281 void readInputParameters_l(); 1282 virtual uint32_t getInputFramesLost(); 1283 1284 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1285 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1286 virtual uint32_t hasAudioSession(audio_session_t sessionId) const; 1287 1288 // Return the set of unique session IDs across all tracks. 1289 // The keys are the session IDs, and the associated values are meaningless. 1290 // FIXME replace by Set [and implement Bag/Multiset for other uses]. 1291 KeyedVector<audio_session_t, bool> sessionIds() const; 1292 1293 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 1294 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 1295 1296 static void syncStartEventCallback(const wp<SyncEvent>& event); 1297 1298 virtual size_t frameCount() const { return mFrameCount; } 1299 bool hasFastCapture() const { return mFastCapture != 0; } 1300 virtual void getAudioPortConfig(struct audio_port_config *config); 1301 1302private: 1303 // Enter standby if not already in standby, and set mStandby flag 1304 void standbyIfNotAlreadyInStandby(); 1305 1306 // Call the HAL standby method unconditionally, and don't change mStandby flag 1307 void inputStandBy(); 1308 1309 AudioStreamIn *mInput; 1310 SortedVector < sp<RecordTrack> > mTracks; 1311 // mActiveTracks has dual roles: it indicates the current active track(s), and 1312 // is used together with mStartStopCond to indicate start()/stop() progress 1313 SortedVector< sp<RecordTrack> > mActiveTracks; 1314 // generation counter for mActiveTracks 1315 int mActiveTracksGen; 1316 Condition mStartStopCond; 1317 1318 // resampler converts input at HAL Hz to output at AudioRecord client Hz 1319 void *mRsmpInBuffer; // 1320 size_t mRsmpInFrames; // size of resampler input in frames 1321 size_t mRsmpInFramesP2;// size rounded up to a power-of-2 1322 1323 // rolling index that is never cleared 1324 int32_t mRsmpInRear; // last filled frame + 1 1325 1326 // For dumpsys 1327 const sp<NBAIO_Sink> mTeeSink; 1328 1329 const sp<MemoryDealer> mReadOnlyHeap; 1330 1331 // one-time initialization, no locks required 1332 sp<FastCapture> mFastCapture; // non-0 if there is also 1333 // a fast capture 1334 1335 // FIXME audio watchdog thread 1336 1337 // contents are not guaranteed to be consistent, no locks required 1338 FastCaptureDumpState mFastCaptureDumpState; 1339#ifdef STATE_QUEUE_DUMP 1340 // FIXME StateQueue observer and mutator dump fields 1341#endif 1342 // FIXME audio watchdog dump 1343 1344 // accessible only within the threadLoop(), no locks required 1345 // mFastCapture->sq() // for mutating and pushing state 1346 int32_t mFastCaptureFutex; // for cold idle 1347 1348 // The HAL input source is treated as non-blocking, 1349 // but current implementation is blocking 1350 sp<NBAIO_Source> mInputSource; 1351 // The source for the normal capture thread to read from: mInputSource or mPipeSource 1352 sp<NBAIO_Source> mNormalSource; 1353 // If a fast capture is present, the non-blocking pipe sink written to by fast capture, 1354 // otherwise clear 1355 sp<NBAIO_Sink> mPipeSink; 1356 // If a fast capture is present, the non-blocking pipe source read by normal thread, 1357 // otherwise clear 1358 sp<NBAIO_Source> mPipeSource; 1359 // Depth of pipe from fast capture to normal thread and fast clients, always power of 2 1360 size_t mPipeFramesP2; 1361 // If a fast capture is present, the Pipe as IMemory, otherwise clear 1362 sp<IMemory> mPipeMemory; 1363 1364 static const size_t kFastCaptureLogSize = 4 * 1024; 1365 sp<NBLog::Writer> mFastCaptureNBLogWriter; 1366 1367 bool mFastTrackAvail; // true if fast track available 1368}; 1369