Threads.h revision 4a8308b11b92e608cdaf29f73f7919e75706f9a2
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; 250 251 // Return's the HAL's frame count i.e. fast mixer buffer size. 252 size_t frameCountHAL() const { return mFrameCount; } 253 254 size_t frameSize() const { return mFrameSize; } 255 256 // Should be "virtual status_t requestExitAndWait()" and override same 257 // method in Thread, but Thread::requestExitAndWait() is not yet virtual. 258 void exit(); 259 virtual bool checkForNewParameter_l(const String8& keyValuePair, 260 status_t& status) = 0; 261 virtual status_t setParameters(const String8& keyValuePairs); 262 virtual String8 getParameters(const String8& keys) = 0; 263 virtual void ioConfigChanged(audio_io_config_event event, pid_t pid = 0) = 0; 264 // sendConfigEvent_l() must be called with ThreadBase::mLock held 265 // Can temporarily release the lock if waiting for a reply from 266 // processConfigEvents_l(). 267 status_t sendConfigEvent_l(sp<ConfigEvent>& event); 268 void sendIoConfigEvent(audio_io_config_event event, pid_t pid = 0); 269 void sendIoConfigEvent_l(audio_io_config_event event, pid_t pid = 0); 270 void sendPrioConfigEvent(pid_t pid, pid_t tid, int32_t prio); 271 void sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio); 272 status_t sendSetParameterConfigEvent_l(const String8& keyValuePair); 273 status_t sendCreateAudioPatchConfigEvent(const struct audio_patch *patch, 274 audio_patch_handle_t *handle); 275 status_t sendReleaseAudioPatchConfigEvent(audio_patch_handle_t handle); 276 void processConfigEvents_l(); 277 virtual void cacheParameters_l() = 0; 278 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 279 audio_patch_handle_t *handle) = 0; 280 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle) = 0; 281 virtual void getAudioPortConfig(struct audio_port_config *config) = 0; 282 283 284 // see note at declaration of mStandby, mOutDevice and mInDevice 285 bool standby() const { return mStandby; } 286 audio_devices_t outDevice() const { return mOutDevice; } 287 audio_devices_t inDevice() const { return mInDevice; } 288 289 virtual audio_stream_t* stream() const = 0; 290 291 sp<EffectHandle> createEffect_l( 292 const sp<AudioFlinger::Client>& client, 293 const sp<IEffectClient>& effectClient, 294 int32_t priority, 295 audio_session_t sessionId, 296 effect_descriptor_t *desc, 297 int *enabled, 298 status_t *status /*non-NULL*/); 299 300 // return values for hasAudioSession (bit field) 301 enum effect_state { 302 EFFECT_SESSION = 0x1, // the audio session corresponds to at least one 303 // effect 304 TRACK_SESSION = 0x2 // the audio session corresponds to at least one 305 // track 306 }; 307 308 // get effect chain corresponding to session Id. 309 sp<EffectChain> getEffectChain(audio_session_t sessionId); 310 // same as getEffectChain() but must be called with ThreadBase mutex locked 311 sp<EffectChain> getEffectChain_l(audio_session_t sessionId) const; 312 // add an effect chain to the chain list (mEffectChains) 313 virtual status_t addEffectChain_l(const sp<EffectChain>& chain) = 0; 314 // remove an effect chain from the chain list (mEffectChains) 315 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain) = 0; 316 // lock all effect chains Mutexes. Must be called before releasing the 317 // ThreadBase mutex before processing the mixer and effects. This guarantees the 318 // integrity of the chains during the process. 319 // Also sets the parameter 'effectChains' to current value of mEffectChains. 320 void lockEffectChains_l(Vector< sp<EffectChain> >& effectChains); 321 // unlock effect chains after process 322 void unlockEffectChains(const Vector< sp<EffectChain> >& effectChains); 323 // get a copy of mEffectChains vector 324 Vector< sp<EffectChain> > getEffectChains_l() const { return mEffectChains; }; 325 // set audio mode to all effect chains 326 void setMode(audio_mode_t mode); 327 // get effect module with corresponding ID on specified audio session 328 sp<AudioFlinger::EffectModule> getEffect(audio_session_t sessionId, int effectId); 329 sp<AudioFlinger::EffectModule> getEffect_l(audio_session_t sessionId, int effectId); 330 // add and effect module. Also creates the effect chain is none exists for 331 // the effects audio session 332 status_t addEffect_l(const sp< EffectModule>& effect); 333 // remove and effect module. Also removes the effect chain is this was the last 334 // effect 335 void removeEffect_l(const sp< EffectModule>& effect); 336 // detach all tracks connected to an auxiliary effect 337 virtual void detachAuxEffect_l(int effectId __unused) {} 338 // returns either EFFECT_SESSION if effects on this audio session exist in one 339 // chain, or TRACK_SESSION if tracks on this audio session exist, or both 340 virtual uint32_t hasAudioSession(audio_session_t sessionId) const = 0; 341 // the value returned by default implementation is not important as the 342 // strategy is only meaningful for PlaybackThread which implements this method 343 virtual uint32_t getStrategyForSession_l(audio_session_t sessionId __unused) 344 { return 0; } 345 346 // suspend or restore effect according to the type of effect passed. a NULL 347 // type pointer means suspend all effects in the session 348 void setEffectSuspended(const effect_uuid_t *type, 349 bool suspend, 350 audio_session_t sessionId = AUDIO_SESSION_OUTPUT_MIX); 351 // check if some effects must be suspended/restored when an effect is enabled 352 // or disabled 353 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 354 bool enabled, 355 audio_session_t sessionId = 356 AUDIO_SESSION_OUTPUT_MIX); 357 void checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 358 bool enabled, 359 audio_session_t sessionId = 360 AUDIO_SESSION_OUTPUT_MIX); 361 362 virtual status_t setSyncEvent(const sp<SyncEvent>& event) = 0; 363 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const = 0; 364 365 // Return a reference to a per-thread heap which can be used to allocate IMemory 366 // objects that will be read-only to client processes, read/write to mediaserver, 367 // and shared by all client processes of the thread. 368 // The heap is per-thread rather than common across all threads, because 369 // clients can't be trusted not to modify the offset of the IMemory they receive. 370 // If a thread does not have such a heap, this method returns 0. 371 virtual sp<MemoryDealer> readOnlyHeap() const { return 0; } 372 373 virtual sp<IMemory> pipeMemory() const { return 0; } 374 375 void systemReady(); 376 377 mutable Mutex mLock; 378 379protected: 380 381 // entry describing an effect being suspended in mSuspendedSessions keyed vector 382 class SuspendedSessionDesc : public RefBase { 383 public: 384 SuspendedSessionDesc() : mRefCount(0) {} 385 386 int mRefCount; // number of active suspend requests 387 effect_uuid_t mType; // effect type UUID 388 }; 389 390 void acquireWakeLock(int uid = -1); 391 virtual void acquireWakeLock_l(int uid = -1); 392 void releaseWakeLock(); 393 void releaseWakeLock_l(); 394 void updateWakeLockUids(const SortedVector<int> &uids); 395 void updateWakeLockUids_l(const SortedVector<int> &uids); 396 void getPowerManager_l(); 397 void setEffectSuspended_l(const effect_uuid_t *type, 398 bool suspend, 399 audio_session_t sessionId); 400 // updated mSuspendedSessions when an effect suspended or restored 401 void updateSuspendedSessions_l(const effect_uuid_t *type, 402 bool suspend, 403 audio_session_t sessionId); 404 // check if some effects must be suspended when an effect chain is added 405 void checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain); 406 407 String16 getWakeLockTag(); 408 409 virtual void preExit() { } 410 virtual void setMasterMono_l(bool mono __unused) { } 411 virtual bool requireMonoBlend() { return false; } 412 413 friend class AudioFlinger; // for mEffectChains 414 415 const type_t mType; 416 417 // Used by parameters, config events, addTrack_l, exit 418 Condition mWaitWorkCV; 419 420 const sp<AudioFlinger> mAudioFlinger; 421 422 // updated by PlaybackThread::readOutputParameters_l() or 423 // RecordThread::readInputParameters_l() 424 uint32_t mSampleRate; 425 size_t mFrameCount; // output HAL, direct output, record 426 audio_channel_mask_t mChannelMask; 427 uint32_t mChannelCount; 428 size_t mFrameSize; 429 // not HAL frame size, this is for output sink (to pipe to fast mixer) 430 audio_format_t mFormat; // Source format for Recording and 431 // Sink format for Playback. 432 // Sink format may be different than 433 // HAL format if Fastmixer is used. 434 audio_format_t mHALFormat; 435 size_t mBufferSize; // HAL buffer size for read() or write() 436 437 Vector< sp<ConfigEvent> > mConfigEvents; 438 Vector< sp<ConfigEvent> > mPendingConfigEvents; // events awaiting system ready 439 440 // These fields are written and read by thread itself without lock or barrier, 441 // and read by other threads without lock or barrier via standby(), outDevice() 442 // and inDevice(). 443 // Because of the absence of a lock or barrier, any other thread that reads 444 // these fields must use the information in isolation, or be prepared to deal 445 // with possibility that it might be inconsistent with other information. 446 bool mStandby; // Whether thread is currently in standby. 447 audio_devices_t mOutDevice; // output device 448 audio_devices_t mInDevice; // input device 449 audio_devices_t mPrevOutDevice; // previous output device 450 audio_devices_t mPrevInDevice; // previous input device 451 struct audio_patch mPatch; 452 audio_source_t mAudioSource; 453 454 const audio_io_handle_t mId; 455 Vector< sp<EffectChain> > mEffectChains; 456 457 static const int kThreadNameLength = 16; // prctl(PR_SET_NAME) limit 458 char mThreadName[kThreadNameLength]; // guaranteed NUL-terminated 459 sp<IPowerManager> mPowerManager; 460 sp<IBinder> mWakeLockToken; 461 const sp<PMDeathRecipient> mDeathRecipient; 462 // list of suspended effects per session and per type. The first (outer) vector is 463 // keyed by session ID, the second (inner) by type UUID timeLow field 464 KeyedVector< audio_session_t, KeyedVector< int, sp<SuspendedSessionDesc> > > 465 mSuspendedSessions; 466 static const size_t kLogSize = 4 * 1024; 467 sp<NBLog::Writer> mNBLogWriter; 468 bool mSystemReady; 469 bool mNotifiedBatteryStart; 470 ExtendedTimestamp mTimestamp; 471}; 472 473// --- PlaybackThread --- 474class PlaybackThread : public ThreadBase { 475public: 476 477#include "PlaybackTracks.h" 478 479 enum mixer_state { 480 MIXER_IDLE, // no active tracks 481 MIXER_TRACKS_ENABLED, // at least one active track, but no track has any data ready 482 MIXER_TRACKS_READY, // at least one active track, and at least one track has data 483 MIXER_DRAIN_TRACK, // drain currently playing track 484 MIXER_DRAIN_ALL, // fully drain the hardware 485 // standby mode does not have an enum value 486 // suspend by audio policy manager is orthogonal to mixer state 487 }; 488 489 PlaybackThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 490 audio_io_handle_t id, audio_devices_t device, type_t type, bool systemReady, 491 uint32_t bitRate = 0); 492 virtual ~PlaybackThread(); 493 494 void dump(int fd, const Vector<String16>& args); 495 496 // Thread virtuals 497 virtual bool threadLoop(); 498 499 // RefBase 500 virtual void onFirstRef(); 501 502protected: 503 // Code snippets that were lifted up out of threadLoop() 504 virtual void threadLoop_mix() = 0; 505 virtual void threadLoop_sleepTime() = 0; 506 virtual ssize_t threadLoop_write(); 507 virtual void threadLoop_drain(); 508 virtual void threadLoop_standby(); 509 virtual void threadLoop_exit(); 510 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 511 512 // prepareTracks_l reads and writes mActiveTracks, and returns 513 // the pending set of tracks to remove via Vector 'tracksToRemove'. The caller 514 // is responsible for clearing or destroying this Vector later on, when it 515 // is safe to do so. That will drop the final ref count and destroy the tracks. 516 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove) = 0; 517 void removeTracks_l(const Vector< sp<Track> >& tracksToRemove); 518 519 void writeCallback(); 520 void resetWriteBlocked(uint32_t sequence); 521 void drainCallback(); 522 void resetDraining(uint32_t sequence); 523 524 static int asyncCallback(stream_callback_event_t event, void *param, void *cookie); 525 526 virtual bool waitingAsyncCallback(); 527 virtual bool waitingAsyncCallback_l(); 528 virtual bool shouldStandby_l(); 529 virtual void onAddNewTrack_l(); 530 531 // ThreadBase virtuals 532 virtual void preExit(); 533 534 virtual bool keepWakeLock() const { return true; } 535 536public: 537 538 virtual status_t initCheck() const { return (mOutput == NULL) ? NO_INIT : NO_ERROR; } 539 540 // return estimated latency in milliseconds, as reported by HAL 541 uint32_t latency() const; 542 // same, but lock must already be held 543 uint32_t latency_l() const; 544 545 void setMasterVolume(float value); 546 void setMasterMute(bool muted); 547 548 void setStreamVolume(audio_stream_type_t stream, float value); 549 void setStreamMute(audio_stream_type_t stream, bool muted); 550 551 float streamVolume(audio_stream_type_t stream) const; 552 553 sp<Track> createTrack_l( 554 const sp<AudioFlinger::Client>& client, 555 audio_stream_type_t streamType, 556 uint32_t sampleRate, 557 audio_format_t format, 558 audio_channel_mask_t channelMask, 559 size_t *pFrameCount, 560 const sp<IMemory>& sharedBuffer, 561 audio_session_t sessionId, 562 IAudioFlinger::track_flags_t *flags, 563 pid_t tid, 564 int uid, 565 status_t *status /*non-NULL*/); 566 567 AudioStreamOut* getOutput() const; 568 AudioStreamOut* clearOutput(); 569 virtual audio_stream_t* stream() const; 570 571 // a very large number of suspend() will eventually wraparound, but unlikely 572 void suspend() { (void) android_atomic_inc(&mSuspended); } 573 void restore() 574 { 575 // if restore() is done without suspend(), get back into 576 // range so that the next suspend() will operate correctly 577 if (android_atomic_dec(&mSuspended) <= 0) { 578 android_atomic_release_store(0, &mSuspended); 579 } 580 } 581 bool isSuspended() const 582 { return android_atomic_acquire_load(&mSuspended) > 0; } 583 584 virtual String8 getParameters(const String8& keys); 585 virtual void ioConfigChanged(audio_io_config_event event, pid_t pid = 0); 586 status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames); 587 // FIXME rename mixBuffer() to sinkBuffer() and remove int16_t* dependency. 588 // Consider also removing and passing an explicit mMainBuffer initialization 589 // parameter to AF::PlaybackThread::Track::Track(). 590 int16_t *mixBuffer() const { 591 return reinterpret_cast<int16_t *>(mSinkBuffer); }; 592 593 virtual void detachAuxEffect_l(int effectId); 594 status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track> track, 595 int EffectId); 596 status_t attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track> track, 597 int EffectId); 598 599 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 600 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 601 virtual uint32_t hasAudioSession(audio_session_t sessionId) const; 602 virtual uint32_t getStrategyForSession_l(audio_session_t sessionId); 603 604 605 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 606 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 607 608 // called with AudioFlinger lock held 609 void invalidateTracks(audio_stream_type_t streamType); 610 611 virtual size_t frameCount() const { return mNormalFrameCount; } 612 613 status_t getTimestamp_l(AudioTimestamp& timestamp); 614 615 void addPatchTrack(const sp<PatchTrack>& track); 616 void deletePatchTrack(const sp<PatchTrack>& track); 617 618 virtual void getAudioPortConfig(struct audio_port_config *config); 619 620protected: 621 // updated by readOutputParameters_l() 622 size_t mNormalFrameCount; // normal mixer and effects 623 624 bool mThreadThrottle; // throttle the thread processing 625 uint32_t mThreadThrottleTimeMs; // throttle time for MIXER threads 626 uint32_t mThreadThrottleEndMs; // notify once per throttling 627 uint32_t mHalfBufferMs; // half the buffer size in milliseconds 628 629 void* mSinkBuffer; // frame size aligned sink buffer 630 631 // TODO: 632 // Rearrange the buffer info into a struct/class with 633 // clear, copy, construction, destruction methods. 634 // 635 // mSinkBuffer also has associated with it: 636 // 637 // mSinkBufferSize: Sink Buffer Size 638 // mFormat: Sink Buffer Format 639 640 // Mixer Buffer (mMixerBuffer*) 641 // 642 // In the case of floating point or multichannel data, which is not in the 643 // sink format, it is required to accumulate in a higher precision or greater channel count 644 // buffer before downmixing or data conversion to the sink buffer. 645 646 // Set to "true" to enable the Mixer Buffer otherwise mixer output goes to sink buffer. 647 bool mMixerBufferEnabled; 648 649 // Storage, 32 byte aligned (may make this alignment a requirement later). 650 // Due to constraints on mNormalFrameCount, the buffer size is a multiple of 16 frames. 651 void* mMixerBuffer; 652 653 // Size of mMixerBuffer in bytes: mNormalFrameCount * #channels * sampsize. 654 size_t mMixerBufferSize; 655 656 // The audio format of mMixerBuffer. Set to AUDIO_FORMAT_PCM_(FLOAT|16_BIT) only. 657 audio_format_t mMixerBufferFormat; 658 659 // An internal flag set to true by MixerThread::prepareTracks_l() 660 // when mMixerBuffer contains valid data after mixing. 661 bool mMixerBufferValid; 662 663 // Effects Buffer (mEffectsBuffer*) 664 // 665 // In the case of effects data, which is not in the sink format, 666 // it is required to accumulate in a different buffer before data conversion 667 // to the sink buffer. 668 669 // Set to "true" to enable the Effects Buffer otherwise effects output goes to sink buffer. 670 bool mEffectBufferEnabled; 671 672 // Storage, 32 byte aligned (may make this alignment a requirement later). 673 // Due to constraints on mNormalFrameCount, the buffer size is a multiple of 16 frames. 674 void* mEffectBuffer; 675 676 // Size of mEffectsBuffer in bytes: mNormalFrameCount * #channels * sampsize. 677 size_t mEffectBufferSize; 678 679 // The audio format of mEffectsBuffer. Set to AUDIO_FORMAT_PCM_16_BIT only. 680 audio_format_t mEffectBufferFormat; 681 682 // An internal flag set to true by MixerThread::prepareTracks_l() 683 // when mEffectsBuffer contains valid data after mixing. 684 // 685 // When this is set, all mixer data is routed into the effects buffer 686 // for any processing (including output processing). 687 bool mEffectBufferValid; 688 689 // suspend count, > 0 means suspended. While suspended, the thread continues to pull from 690 // tracks and mix, but doesn't write to HAL. A2DP and SCO HAL implementations can't handle 691 // concurrent use of both of them, so Audio Policy Service suspends one of the threads to 692 // workaround that restriction. 693 // 'volatile' means accessed via atomic operations and no lock. 694 volatile int32_t mSuspended; 695 696 int64_t mBytesWritten; 697 int64_t mFramesWritten; // not reset on standby 698private: 699 // mMasterMute is in both PlaybackThread and in AudioFlinger. When a 700 // PlaybackThread needs to find out if master-muted, it checks it's local 701 // copy rather than the one in AudioFlinger. This optimization saves a lock. 702 bool mMasterMute; 703 void setMasterMute_l(bool muted) { mMasterMute = muted; } 704protected: 705 SortedVector< wp<Track> > mActiveTracks; // FIXME check if this could be sp<> 706 SortedVector<int> mWakeLockUids; 707 int mActiveTracksGeneration; 708 wp<Track> mLatestActiveTrack; // latest track added to mActiveTracks 709 710 // Allocate a track name for a given channel mask. 711 // Returns name >= 0 if successful, -1 on failure. 712 virtual int getTrackName_l(audio_channel_mask_t channelMask, 713 audio_format_t format, audio_session_t sessionId) = 0; 714 virtual void deleteTrackName_l(int name) = 0; 715 716 // Time to sleep between cycles when: 717 virtual uint32_t activeSleepTimeUs() const; // mixer state MIXER_TRACKS_ENABLED 718 virtual uint32_t idleSleepTimeUs() const = 0; // mixer state MIXER_IDLE 719 virtual uint32_t suspendSleepTimeUs() const = 0; // audio policy manager suspended us 720 // No sleep when mixer state == MIXER_TRACKS_READY; relies on audio HAL stream->write() 721 // No sleep in standby mode; waits on a condition 722 723 // Code snippets that are temporarily lifted up out of threadLoop() until the merge 724 void checkSilentMode_l(); 725 726 // Non-trivial for DUPLICATING only 727 virtual void saveOutputTracks() { } 728 virtual void clearOutputTracks() { } 729 730 // Cache various calculated values, at threadLoop() entry and after a parameter change 731 virtual void cacheParameters_l(); 732 733 virtual uint32_t correctLatency_l(uint32_t latency) const; 734 735 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 736 audio_patch_handle_t *handle); 737 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 738 739 bool usesHwAvSync() const { return (mType == DIRECT) && (mOutput != NULL) 740 && mHwSupportsPause 741 && (mOutput->flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC); } 742 743private: 744 745 friend class AudioFlinger; // for numerous 746 747 PlaybackThread& operator = (const PlaybackThread&); 748 749 status_t addTrack_l(const sp<Track>& track); 750 bool destroyTrack_l(const sp<Track>& track); 751 void removeTrack_l(const sp<Track>& track); 752 void broadcast_l(); 753 754 void readOutputParameters_l(); 755 756 virtual void dumpInternals(int fd, const Vector<String16>& args); 757 void dumpTracks(int fd, const Vector<String16>& args); 758 759 SortedVector< sp<Track> > mTracks; 760 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 761 AudioStreamOut *mOutput; 762 763 float mMasterVolume; 764 nsecs_t mLastWriteTime; 765 int mNumWrites; 766 int mNumDelayedWrites; 767 bool mInWrite; 768 769 // FIXME rename these former local variables of threadLoop to standard "m" names 770 nsecs_t mStandbyTimeNs; 771 size_t mSinkBufferSize; 772 773 // cached copies of activeSleepTimeUs() and idleSleepTimeUs() made by cacheParameters_l() 774 uint32_t mActiveSleepTimeUs; 775 uint32_t mIdleSleepTimeUs; 776 777 uint32_t mSleepTimeUs; 778 779 // mixer status returned by prepareTracks_l() 780 mixer_state mMixerStatus; // current cycle 781 // previous cycle when in prepareTracks_l() 782 mixer_state mMixerStatusIgnoringFastTracks; 783 // FIXME or a separate ready state per track 784 785 // FIXME move these declarations into the specific sub-class that needs them 786 // MIXER only 787 uint32_t sleepTimeShift; 788 789 // same as AudioFlinger::mStandbyTimeInNsecs except for DIRECT which uses a shorter value 790 nsecs_t mStandbyDelayNs; 791 792 // MIXER only 793 nsecs_t maxPeriod; 794 795 // DUPLICATING only 796 uint32_t writeFrames; 797 798 size_t mBytesRemaining; 799 size_t mCurrentWriteLength; 800 bool mUseAsyncWrite; 801 // mWriteAckSequence contains current write sequence on bits 31-1. The write sequence is 802 // incremented each time a write(), a flush() or a standby() occurs. 803 // Bit 0 is set when a write blocks and indicates a callback is expected. 804 // Bit 0 is reset by the async callback thread calling resetWriteBlocked(). Out of sequence 805 // callbacks are ignored. 806 uint32_t mWriteAckSequence; 807 // mDrainSequence contains current drain sequence on bits 31-1. The drain sequence is 808 // incremented each time a drain is requested or a flush() or standby() occurs. 809 // Bit 0 is set when the drain() command is called at the HAL and indicates a callback is 810 // expected. 811 // Bit 0 is reset by the async callback thread calling resetDraining(). Out of sequence 812 // callbacks are ignored. 813 uint32_t mDrainSequence; 814 // A condition that must be evaluated by prepareTrack_l() has changed and we must not wait 815 // for async write callback in the thread loop before evaluating it 816 bool mSignalPending; 817 sp<AsyncCallbackThread> mCallbackThread; 818 819private: 820 // The HAL output sink is treated as non-blocking, but current implementation is blocking 821 sp<NBAIO_Sink> mOutputSink; 822 // If a fast mixer is present, the blocking pipe sink, otherwise clear 823 sp<NBAIO_Sink> mPipeSink; 824 // The current sink for the normal mixer to write it's (sub)mix, mOutputSink or mPipeSink 825 sp<NBAIO_Sink> mNormalSink; 826#ifdef TEE_SINK 827 // For dumpsys 828 sp<NBAIO_Sink> mTeeSink; 829 sp<NBAIO_Source> mTeeSource; 830#endif 831 uint32_t mScreenState; // cached copy of gScreenState 832 static const size_t kFastMixerLogSize = 4 * 1024; 833 sp<NBLog::Writer> mFastMixerNBLogWriter; 834public: 835 virtual bool hasFastMixer() const = 0; 836 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex __unused) const 837 { FastTrackUnderruns dummy; return dummy; } 838 839protected: 840 // accessed by both binder threads and within threadLoop(), lock on mutex needed 841 unsigned mFastTrackAvailMask; // bit i set if fast track [i] is available 842 bool mHwSupportsPause; 843 bool mHwPaused; 844 bool mFlushPending; 845 uint32_t mBufferDurationUs; // estimated duration of an audio HAL buffer 846 // based on initial bit rate (offload only) 847}; 848 849class MixerThread : public PlaybackThread { 850public: 851 MixerThread(const sp<AudioFlinger>& audioFlinger, 852 AudioStreamOut* output, 853 audio_io_handle_t id, 854 audio_devices_t device, 855 bool systemReady, 856 type_t type = MIXER); 857 virtual ~MixerThread(); 858 859 // Thread virtuals 860 861 virtual bool checkForNewParameter_l(const String8& keyValuePair, 862 status_t& status); 863 virtual void dumpInternals(int fd, const Vector<String16>& args); 864 865protected: 866 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 867 virtual int getTrackName_l(audio_channel_mask_t channelMask, 868 audio_format_t format, audio_session_t sessionId); 869 virtual void deleteTrackName_l(int name); 870 virtual uint32_t idleSleepTimeUs() const; 871 virtual uint32_t suspendSleepTimeUs() const; 872 virtual void cacheParameters_l(); 873 874 virtual void acquireWakeLock_l(int uid = -1) { 875 PlaybackThread::acquireWakeLock_l(uid); 876 if (hasFastMixer()) { 877 mFastMixer->setBoottimeOffset( 878 mTimestamp.mTimebaseOffset[ExtendedTimestamp::TIMEBASE_BOOTTIME]); 879 } 880 } 881 882 // threadLoop snippets 883 virtual ssize_t threadLoop_write(); 884 virtual void threadLoop_standby(); 885 virtual void threadLoop_mix(); 886 virtual void threadLoop_sleepTime(); 887 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 888 virtual uint32_t correctLatency_l(uint32_t latency) const; 889 890 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 891 audio_patch_handle_t *handle); 892 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 893 894 AudioMixer* mAudioMixer; // normal mixer 895private: 896 // one-time initialization, no locks required 897 sp<FastMixer> mFastMixer; // non-0 if there is also a fast mixer 898 sp<AudioWatchdog> mAudioWatchdog; // non-0 if there is an audio watchdog thread 899 900 // contents are not guaranteed to be consistent, no locks required 901 FastMixerDumpState mFastMixerDumpState; 902#ifdef STATE_QUEUE_DUMP 903 StateQueueObserverDump mStateQueueObserverDump; 904 StateQueueMutatorDump mStateQueueMutatorDump; 905#endif 906 AudioWatchdogDump mAudioWatchdogDump; 907 908 // accessible only within the threadLoop(), no locks required 909 // mFastMixer->sq() // for mutating and pushing state 910 int32_t mFastMixerFutex; // for cold idle 911 912 std::atomic_bool mMasterMono; 913public: 914 virtual bool hasFastMixer() const { return mFastMixer != 0; } 915 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const { 916 ALOG_ASSERT(fastIndex < FastMixerState::kMaxFastTracks); 917 return mFastMixerDumpState.mTracks[fastIndex].mUnderruns; 918 } 919 920protected: 921 virtual void setMasterMono_l(bool mono) { 922 mMasterMono.store(mono); 923 if (mFastMixer != nullptr) { /* hasFastMixer() */ 924 mFastMixer->setMasterMono(mMasterMono); 925 } 926 } 927 // the FastMixer performs mono blend if it exists. 928 // Blending with limiter is not idempotent, 929 // and blending without limiter is idempotent but inefficient to do twice. 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 uint32_t bitRate = 0); 939 virtual ~DirectOutputThread(); 940 941 // Thread virtuals 942 943 virtual bool checkForNewParameter_l(const String8& keyValuePair, 944 status_t& status); 945 virtual void flushHw_l(); 946 947protected: 948 virtual int getTrackName_l(audio_channel_mask_t channelMask, 949 audio_format_t format, audio_session_t sessionId); 950 virtual void deleteTrackName_l(int name); 951 virtual uint32_t activeSleepTimeUs() const; 952 virtual uint32_t idleSleepTimeUs() const; 953 virtual uint32_t suspendSleepTimeUs() const; 954 virtual void cacheParameters_l(); 955 956 // threadLoop snippets 957 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 958 virtual void threadLoop_mix(); 959 virtual void threadLoop_sleepTime(); 960 virtual void threadLoop_exit(); 961 virtual bool shouldStandby_l(); 962 963 virtual void onAddNewTrack_l(); 964 965 // volumes last sent to audio HAL with stream->set_volume() 966 float mLeftVolFloat; 967 float mRightVolFloat; 968 969 DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 970 audio_io_handle_t id, uint32_t device, ThreadBase::type_t type, 971 bool systemReady, uint32_t bitRate = 0); 972 void processVolume_l(Track *track, bool lastTrack); 973 974 // prepareTracks_l() tells threadLoop_mix() the name of the single active track 975 sp<Track> mActiveTrack; 976 977 wp<Track> mPreviousTrack; // used to detect track switch 978 979public: 980 virtual bool hasFastMixer() const { return false; } 981}; 982 983class OffloadThread : public DirectOutputThread { 984public: 985 986 OffloadThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 987 audio_io_handle_t id, uint32_t device, 988 bool systemReady, uint32_t bitRate); 989 virtual ~OffloadThread() {}; 990 virtual void flushHw_l(); 991 992protected: 993 // threadLoop snippets 994 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 995 virtual void threadLoop_exit(); 996 997 virtual uint32_t activeSleepTimeUs() const; 998 999 virtual bool waitingAsyncCallback(); 1000 virtual bool waitingAsyncCallback_l(); 1001 1002 virtual bool keepWakeLock() const { return mKeepWakeLock; } 1003 1004private: 1005 size_t mPausedWriteLength; // length in bytes of write interrupted by pause 1006 size_t mPausedBytesRemaining; // bytes still waiting in mixbuffer after resume 1007 bool mKeepWakeLock; // keep wake lock while waiting for write callback 1008}; 1009 1010class AsyncCallbackThread : public Thread { 1011public: 1012 1013 AsyncCallbackThread(const wp<PlaybackThread>& playbackThread); 1014 1015 virtual ~AsyncCallbackThread(); 1016 1017 // Thread virtuals 1018 virtual bool threadLoop(); 1019 1020 // RefBase 1021 virtual void onFirstRef(); 1022 1023 void exit(); 1024 void setWriteBlocked(uint32_t sequence); 1025 void resetWriteBlocked(); 1026 void setDraining(uint32_t sequence); 1027 void resetDraining(); 1028 1029private: 1030 const wp<PlaybackThread> mPlaybackThread; 1031 // mWriteAckSequence corresponds to the last write sequence passed by the offload thread via 1032 // setWriteBlocked(). The sequence is shifted one bit to the left and the lsb is used 1033 // to indicate that the callback has been received via resetWriteBlocked() 1034 uint32_t mWriteAckSequence; 1035 // mDrainSequence corresponds to the last drain sequence passed by the offload thread via 1036 // setDraining(). The sequence is shifted one bit to the left and the lsb is used 1037 // to indicate that the callback has been received via resetDraining() 1038 uint32_t mDrainSequence; 1039 Condition mWaitWorkCV; 1040 Mutex mLock; 1041}; 1042 1043class DuplicatingThread : public MixerThread { 1044public: 1045 DuplicatingThread(const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread, 1046 audio_io_handle_t id, bool systemReady); 1047 virtual ~DuplicatingThread(); 1048 1049 // Thread virtuals 1050 void addOutputTrack(MixerThread* thread); 1051 void removeOutputTrack(MixerThread* thread); 1052 uint32_t waitTimeMs() const { return mWaitTimeMs; } 1053protected: 1054 virtual uint32_t activeSleepTimeUs() const; 1055 1056private: 1057 bool outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks); 1058protected: 1059 // threadLoop snippets 1060 virtual void threadLoop_mix(); 1061 virtual void threadLoop_sleepTime(); 1062 virtual ssize_t threadLoop_write(); 1063 virtual void threadLoop_standby(); 1064 virtual void cacheParameters_l(); 1065 1066private: 1067 // called from threadLoop, addOutputTrack, removeOutputTrack 1068 virtual void updateWaitTime_l(); 1069protected: 1070 virtual void saveOutputTracks(); 1071 virtual void clearOutputTracks(); 1072private: 1073 1074 uint32_t mWaitTimeMs; 1075 SortedVector < sp<OutputTrack> > outputTracks; 1076 SortedVector < sp<OutputTrack> > mOutputTracks; 1077public: 1078 virtual bool hasFastMixer() const { return false; } 1079}; 1080 1081 1082// record thread 1083class RecordThread : public ThreadBase 1084{ 1085public: 1086 1087 class RecordTrack; 1088 1089 /* The ResamplerBufferProvider is used to retrieve recorded input data from the 1090 * RecordThread. It maintains local state on the relative position of the read 1091 * position of the RecordTrack compared with the RecordThread. 1092 */ 1093 class ResamplerBufferProvider : public AudioBufferProvider 1094 { 1095 public: 1096 ResamplerBufferProvider(RecordTrack* recordTrack) : 1097 mRecordTrack(recordTrack), 1098 mRsmpInUnrel(0), mRsmpInFront(0) { } 1099 virtual ~ResamplerBufferProvider() { } 1100 1101 // called to set the ResamplerBufferProvider to head of the RecordThread data buffer, 1102 // skipping any previous data read from the hal. 1103 virtual void reset(); 1104 1105 /* Synchronizes RecordTrack position with the RecordThread. 1106 * Calculates available frames and handle overruns if the RecordThread 1107 * has advanced faster than the ResamplerBufferProvider has retrieved data. 1108 * TODO: why not do this for every getNextBuffer? 1109 * 1110 * Parameters 1111 * framesAvailable: pointer to optional output size_t to store record track 1112 * frames available. 1113 * hasOverrun: pointer to optional boolean, returns true if track has overrun. 1114 */ 1115 1116 virtual void sync(size_t *framesAvailable = NULL, bool *hasOverrun = NULL); 1117 1118 // AudioBufferProvider interface 1119 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer); 1120 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 1121 private: 1122 RecordTrack * const mRecordTrack; 1123 size_t mRsmpInUnrel; // unreleased frames remaining from 1124 // most recent getNextBuffer 1125 // for debug only 1126 int32_t mRsmpInFront; // next available frame 1127 // rolling counter that is never cleared 1128 }; 1129 1130 /* The RecordBufferConverter is used for format, channel, and sample rate 1131 * conversion for a RecordTrack. 1132 * 1133 * TODO: Self contained, so move to a separate file later. 1134 * 1135 * RecordBufferConverter uses the convert() method rather than exposing a 1136 * buffer provider interface; this is to save a memory copy. 1137 */ 1138 class RecordBufferConverter 1139 { 1140 public: 1141 RecordBufferConverter( 1142 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 1143 uint32_t srcSampleRate, 1144 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 1145 uint32_t dstSampleRate); 1146 1147 ~RecordBufferConverter(); 1148 1149 /* Converts input data from an AudioBufferProvider by format, channelMask, 1150 * and sampleRate to a destination buffer. 1151 * 1152 * Parameters 1153 * dst: buffer to place the converted data. 1154 * provider: buffer provider to obtain source data. 1155 * frames: number of frames to convert 1156 * 1157 * Returns the number of frames converted. 1158 */ 1159 size_t convert(void *dst, AudioBufferProvider *provider, size_t frames); 1160 1161 // returns NO_ERROR if constructor was successful 1162 status_t initCheck() const { 1163 // mSrcChannelMask set on successful updateParameters 1164 return mSrcChannelMask != AUDIO_CHANNEL_INVALID ? NO_ERROR : NO_INIT; 1165 } 1166 1167 // allows dynamic reconfigure of all parameters 1168 status_t updateParameters( 1169 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 1170 uint32_t srcSampleRate, 1171 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 1172 uint32_t dstSampleRate); 1173 1174 // called to reset resampler buffers on record track discontinuity 1175 void reset() { 1176 if (mResampler != NULL) { 1177 mResampler->reset(); 1178 } 1179 } 1180 1181 private: 1182 // format conversion when not using resampler 1183 void convertNoResampler(void *dst, const void *src, size_t frames); 1184 1185 // format conversion when using resampler; modifies src in-place 1186 void convertResampler(void *dst, /*not-a-const*/ void *src, size_t frames); 1187 1188 // user provided information 1189 audio_channel_mask_t mSrcChannelMask; 1190 audio_format_t mSrcFormat; 1191 uint32_t mSrcSampleRate; 1192 audio_channel_mask_t mDstChannelMask; 1193 audio_format_t mDstFormat; 1194 uint32_t mDstSampleRate; 1195 1196 // derived information 1197 uint32_t mSrcChannelCount; 1198 uint32_t mDstChannelCount; 1199 size_t mDstFrameSize; 1200 1201 // format conversion buffer 1202 void *mBuf; 1203 size_t mBufFrames; 1204 size_t mBufFrameSize; 1205 1206 // resampler info 1207 AudioResampler *mResampler; 1208 1209 bool mIsLegacyDownmix; // legacy stereo to mono conversion needed 1210 bool mIsLegacyUpmix; // legacy mono to stereo conversion needed 1211 bool mRequiresFloat; // data processing requires float (e.g. resampler) 1212 PassthruBufferProvider *mInputConverterProvider; // converts input to float 1213 int8_t mIdxAry[sizeof(uint32_t) * 8]; // used for channel mask conversion 1214 }; 1215 1216#include "RecordTracks.h" 1217 1218 RecordThread(const sp<AudioFlinger>& audioFlinger, 1219 AudioStreamIn *input, 1220 audio_io_handle_t id, 1221 audio_devices_t outDevice, 1222 audio_devices_t inDevice, 1223 bool systemReady 1224#ifdef TEE_SINK 1225 , const sp<NBAIO_Sink>& teeSink 1226#endif 1227 ); 1228 virtual ~RecordThread(); 1229 1230 // no addTrack_l ? 1231 void destroyTrack_l(const sp<RecordTrack>& track); 1232 void removeTrack_l(const sp<RecordTrack>& track); 1233 1234 void dumpInternals(int fd, const Vector<String16>& args); 1235 void dumpTracks(int fd, const Vector<String16>& args); 1236 1237 // Thread virtuals 1238 virtual bool threadLoop(); 1239 1240 // RefBase 1241 virtual void onFirstRef(); 1242 1243 virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; } 1244 1245 virtual sp<MemoryDealer> readOnlyHeap() const { return mReadOnlyHeap; } 1246 1247 virtual sp<IMemory> pipeMemory() const { return mPipeMemory; } 1248 1249 sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l( 1250 const sp<AudioFlinger::Client>& client, 1251 uint32_t sampleRate, 1252 audio_format_t format, 1253 audio_channel_mask_t channelMask, 1254 size_t *pFrameCount, 1255 audio_session_t sessionId, 1256 size_t *notificationFrames, 1257 int uid, 1258 IAudioFlinger::track_flags_t *flags, 1259 pid_t tid, 1260 status_t *status /*non-NULL*/); 1261 1262 status_t start(RecordTrack* recordTrack, 1263 AudioSystem::sync_event_t event, 1264 audio_session_t triggerSession); 1265 1266 // ask the thread to stop the specified track, and 1267 // return true if the caller should then do it's part of the stopping process 1268 bool stop(RecordTrack* recordTrack); 1269 1270 void dump(int fd, const Vector<String16>& args); 1271 AudioStreamIn* clearInput(); 1272 virtual audio_stream_t* stream() const; 1273 1274 1275 virtual bool checkForNewParameter_l(const String8& keyValuePair, 1276 status_t& status); 1277 virtual void cacheParameters_l() {} 1278 virtual String8 getParameters(const String8& keys); 1279 virtual void ioConfigChanged(audio_io_config_event event, pid_t pid = 0); 1280 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 1281 audio_patch_handle_t *handle); 1282 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 1283 1284 void addPatchRecord(const sp<PatchRecord>& record); 1285 void deletePatchRecord(const sp<PatchRecord>& record); 1286 1287 void readInputParameters_l(); 1288 virtual uint32_t getInputFramesLost(); 1289 1290 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1291 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1292 virtual uint32_t hasAudioSession(audio_session_t sessionId) const; 1293 1294 // Return the set of unique session IDs across all tracks. 1295 // The keys are the session IDs, and the associated values are meaningless. 1296 // FIXME replace by Set [and implement Bag/Multiset for other uses]. 1297 KeyedVector<audio_session_t, bool> sessionIds() const; 1298 1299 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 1300 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 1301 1302 static void syncStartEventCallback(const wp<SyncEvent>& event); 1303 1304 virtual size_t frameCount() const { return mFrameCount; } 1305 bool hasFastCapture() const { return mFastCapture != 0; } 1306 virtual void getAudioPortConfig(struct audio_port_config *config); 1307 1308private: 1309 // Enter standby if not already in standby, and set mStandby flag 1310 void standbyIfNotAlreadyInStandby(); 1311 1312 // Call the HAL standby method unconditionally, and don't change mStandby flag 1313 void inputStandBy(); 1314 1315 AudioStreamIn *mInput; 1316 SortedVector < sp<RecordTrack> > mTracks; 1317 // mActiveTracks has dual roles: it indicates the current active track(s), and 1318 // is used together with mStartStopCond to indicate start()/stop() progress 1319 SortedVector< sp<RecordTrack> > mActiveTracks; 1320 // generation counter for mActiveTracks 1321 int mActiveTracksGen; 1322 Condition mStartStopCond; 1323 1324 // resampler converts input at HAL Hz to output at AudioRecord client Hz 1325 void *mRsmpInBuffer; // 1326 size_t mRsmpInFrames; // size of resampler input in frames 1327 size_t mRsmpInFramesP2;// size rounded up to a power-of-2 1328 1329 // rolling index that is never cleared 1330 int32_t mRsmpInRear; // last filled frame + 1 1331 1332 // For dumpsys 1333 const sp<NBAIO_Sink> mTeeSink; 1334 1335 const sp<MemoryDealer> mReadOnlyHeap; 1336 1337 // one-time initialization, no locks required 1338 sp<FastCapture> mFastCapture; // non-0 if there is also 1339 // a fast capture 1340 1341 // FIXME audio watchdog thread 1342 1343 // contents are not guaranteed to be consistent, no locks required 1344 FastCaptureDumpState mFastCaptureDumpState; 1345#ifdef STATE_QUEUE_DUMP 1346 // FIXME StateQueue observer and mutator dump fields 1347#endif 1348 // FIXME audio watchdog dump 1349 1350 // accessible only within the threadLoop(), no locks required 1351 // mFastCapture->sq() // for mutating and pushing state 1352 int32_t mFastCaptureFutex; // for cold idle 1353 1354 // The HAL input source is treated as non-blocking, 1355 // but current implementation is blocking 1356 sp<NBAIO_Source> mInputSource; 1357 // The source for the normal capture thread to read from: mInputSource or mPipeSource 1358 sp<NBAIO_Source> mNormalSource; 1359 // If a fast capture is present, the non-blocking pipe sink written to by fast capture, 1360 // otherwise clear 1361 sp<NBAIO_Sink> mPipeSink; 1362 // If a fast capture is present, the non-blocking pipe source read by normal thread, 1363 // otherwise clear 1364 sp<NBAIO_Source> mPipeSource; 1365 // Depth of pipe from fast capture to normal thread and fast clients, always power of 2 1366 size_t mPipeFramesP2; 1367 // If a fast capture is present, the Pipe as IMemory, otherwise clear 1368 sp<IMemory> mPipeMemory; 1369 1370 static const size_t kFastCaptureLogSize = 4 * 1024; 1371 sp<NBLog::Writer> mFastCaptureNBLogWriter; 1372 1373 bool mFastTrackAvail; // true if fast track available 1374}; 1375