Threads.h revision 517161856d74f5fe39cce131f29b977bc1745991
1/* 2** 3** Copyright 2012, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18#ifndef INCLUDING_FROM_AUDIOFLINGER_H 19 #error This header file should only be included from AudioFlinger.h 20#endif 21 22class ThreadBase : public Thread { 23public: 24 25#include "TrackBase.h" 26 27 enum type_t { 28 MIXER, // Thread class is MixerThread 29 DIRECT, // Thread class is DirectOutputThread 30 DUPLICATING, // Thread class is DuplicatingThread 31 RECORD, // Thread class is RecordThread 32 OFFLOAD // Thread class is OffloadThread 33 }; 34 35 static const char *threadTypeToString(type_t type); 36 37 ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 38 audio_devices_t outDevice, audio_devices_t inDevice, type_t type, 39 bool systemReady); 40 virtual ~ThreadBase(); 41 42 virtual status_t readyToRun(); 43 44 void dumpBase(int fd, const Vector<String16>& args); 45 void dumpEffectChains(int fd, const Vector<String16>& args); 46 47 void clearPowerManager(); 48 49 // base for record and playback 50 enum { 51 CFG_EVENT_IO, 52 CFG_EVENT_PRIO, 53 CFG_EVENT_SET_PARAMETER, 54 CFG_EVENT_CREATE_AUDIO_PATCH, 55 CFG_EVENT_RELEASE_AUDIO_PATCH, 56 }; 57 58 class ConfigEventData: public RefBase { 59 public: 60 virtual ~ConfigEventData() {} 61 62 virtual void dump(char *buffer, size_t size) = 0; 63 protected: 64 ConfigEventData() {} 65 }; 66 67 // Config event sequence by client if status needed (e.g binder thread calling setParameters()): 68 // 1. create SetParameterConfigEvent. This sets mWaitStatus in config event 69 // 2. Lock mLock 70 // 3. Call sendConfigEvent_l(): Append to mConfigEvents and mWaitWorkCV.signal 71 // 4. sendConfigEvent_l() reads status from event->mStatus; 72 // 5. sendConfigEvent_l() returns status 73 // 6. Unlock 74 // 75 // Parameter sequence by server: threadLoop calling processConfigEvents_l(): 76 // 1. Lock mLock 77 // 2. If there is an entry in mConfigEvents proceed ... 78 // 3. Read first entry in mConfigEvents 79 // 4. Remove first entry from mConfigEvents 80 // 5. Process 81 // 6. Set event->mStatus 82 // 7. event->mCond.signal 83 // 8. Unlock 84 85 class ConfigEvent: public RefBase { 86 public: 87 virtual ~ConfigEvent() {} 88 89 void dump(char *buffer, size_t size) { mData->dump(buffer, size); } 90 91 const int mType; // event type e.g. CFG_EVENT_IO 92 Mutex mLock; // mutex associated with mCond 93 Condition mCond; // condition for status return 94 status_t mStatus; // status communicated to sender 95 bool mWaitStatus; // true if sender is waiting for status 96 bool mRequiresSystemReady; // true if must wait for system ready to enter event queue 97 sp<ConfigEventData> mData; // event specific parameter data 98 99 protected: 100 ConfigEvent(int type, bool requiresSystemReady = false) : 101 mType(type), mStatus(NO_ERROR), mWaitStatus(false), 102 mRequiresSystemReady(requiresSystemReady), mData(NULL) {} 103 }; 104 105 class IoConfigEventData : public ConfigEventData { 106 public: 107 IoConfigEventData(audio_io_config_event event, pid_t pid) : 108 mEvent(event), mPid(pid) {} 109 110 virtual void dump(char *buffer, size_t size) { 111 snprintf(buffer, size, "IO event: event %d\n", mEvent); 112 } 113 114 const audio_io_config_event mEvent; 115 const pid_t mPid; 116 }; 117 118 class IoConfigEvent : public ConfigEvent { 119 public: 120 IoConfigEvent(audio_io_config_event event, pid_t pid) : 121 ConfigEvent(CFG_EVENT_IO) { 122 mData = new IoConfigEventData(event, pid); 123 } 124 virtual ~IoConfigEvent() {} 125 }; 126 127 class PrioConfigEventData : public ConfigEventData { 128 public: 129 PrioConfigEventData(pid_t pid, pid_t tid, int32_t prio) : 130 mPid(pid), mTid(tid), mPrio(prio) {} 131 132 virtual void dump(char *buffer, size_t size) { 133 snprintf(buffer, size, "Prio event: pid %d, tid %d, prio %d\n", mPid, mTid, mPrio); 134 } 135 136 const pid_t mPid; 137 const pid_t mTid; 138 const int32_t mPrio; 139 }; 140 141 class PrioConfigEvent : public ConfigEvent { 142 public: 143 PrioConfigEvent(pid_t pid, pid_t tid, int32_t prio) : 144 ConfigEvent(CFG_EVENT_PRIO, true) { 145 mData = new PrioConfigEventData(pid, tid, prio); 146 } 147 virtual ~PrioConfigEvent() {} 148 }; 149 150 class SetParameterConfigEventData : public ConfigEventData { 151 public: 152 SetParameterConfigEventData(String8 keyValuePairs) : 153 mKeyValuePairs(keyValuePairs) {} 154 155 virtual void dump(char *buffer, size_t size) { 156 snprintf(buffer, size, "KeyValue: %s\n", mKeyValuePairs.string()); 157 } 158 159 const String8 mKeyValuePairs; 160 }; 161 162 class SetParameterConfigEvent : public ConfigEvent { 163 public: 164 SetParameterConfigEvent(String8 keyValuePairs) : 165 ConfigEvent(CFG_EVENT_SET_PARAMETER) { 166 mData = new SetParameterConfigEventData(keyValuePairs); 167 mWaitStatus = true; 168 } 169 virtual ~SetParameterConfigEvent() {} 170 }; 171 172 class CreateAudioPatchConfigEventData : public ConfigEventData { 173 public: 174 CreateAudioPatchConfigEventData(const struct audio_patch patch, 175 audio_patch_handle_t handle) : 176 mPatch(patch), mHandle(handle) {} 177 178 virtual void dump(char *buffer, size_t size) { 179 snprintf(buffer, size, "Patch handle: %u\n", mHandle); 180 } 181 182 const struct audio_patch mPatch; 183 audio_patch_handle_t mHandle; 184 }; 185 186 class CreateAudioPatchConfigEvent : public ConfigEvent { 187 public: 188 CreateAudioPatchConfigEvent(const struct audio_patch patch, 189 audio_patch_handle_t handle) : 190 ConfigEvent(CFG_EVENT_CREATE_AUDIO_PATCH) { 191 mData = new CreateAudioPatchConfigEventData(patch, handle); 192 mWaitStatus = true; 193 } 194 virtual ~CreateAudioPatchConfigEvent() {} 195 }; 196 197 class ReleaseAudioPatchConfigEventData : public ConfigEventData { 198 public: 199 ReleaseAudioPatchConfigEventData(const audio_patch_handle_t handle) : 200 mHandle(handle) {} 201 202 virtual void dump(char *buffer, size_t size) { 203 snprintf(buffer, size, "Patch handle: %u\n", mHandle); 204 } 205 206 audio_patch_handle_t mHandle; 207 }; 208 209 class ReleaseAudioPatchConfigEvent : public ConfigEvent { 210 public: 211 ReleaseAudioPatchConfigEvent(const audio_patch_handle_t handle) : 212 ConfigEvent(CFG_EVENT_RELEASE_AUDIO_PATCH) { 213 mData = new ReleaseAudioPatchConfigEventData(handle); 214 mWaitStatus = true; 215 } 216 virtual ~ReleaseAudioPatchConfigEvent() {} 217 }; 218 219 class PMDeathRecipient : public IBinder::DeathRecipient { 220 public: 221 PMDeathRecipient(const wp<ThreadBase>& thread) : mThread(thread) {} 222 virtual ~PMDeathRecipient() {} 223 224 // IBinder::DeathRecipient 225 virtual void binderDied(const wp<IBinder>& who); 226 227 private: 228 PMDeathRecipient(const PMDeathRecipient&); 229 PMDeathRecipient& operator = (const PMDeathRecipient&); 230 231 wp<ThreadBase> mThread; 232 }; 233 234 virtual status_t initCheck() const = 0; 235 236 // static externally-visible 237 type_t type() const { return mType; } 238 bool isDuplicating() const { return (mType == DUPLICATING); } 239 240 audio_io_handle_t id() const { return mId;} 241 242 // dynamic externally-visible 243 uint32_t sampleRate() const { return mSampleRate; } 244 audio_channel_mask_t channelMask() const { return mChannelMask; } 245 audio_format_t format() const { return mHALFormat; } 246 uint32_t channelCount() const { return mChannelCount; } 247 // Called by AudioFlinger::frameCount(audio_io_handle_t output) and effects, 248 // and returns the [normal mix] buffer's frame count. 249 virtual size_t frameCount() const = 0; 250 size_t frameSize() const { return mFrameSize; } 251 252 // Should be "virtual status_t requestExitAndWait()" and override same 253 // method in Thread, but Thread::requestExitAndWait() is not yet virtual. 254 void exit(); 255 virtual bool checkForNewParameter_l(const String8& keyValuePair, 256 status_t& status) = 0; 257 virtual status_t setParameters(const String8& keyValuePairs); 258 virtual String8 getParameters(const String8& keys) = 0; 259 virtual void ioConfigChanged(audio_io_config_event event, pid_t pid = 0) = 0; 260 // sendConfigEvent_l() must be called with ThreadBase::mLock held 261 // Can temporarily release the lock if waiting for a reply from 262 // processConfigEvents_l(). 263 status_t sendConfigEvent_l(sp<ConfigEvent>& event); 264 void sendIoConfigEvent(audio_io_config_event event, pid_t pid = 0); 265 void sendIoConfigEvent_l(audio_io_config_event event, pid_t pid = 0); 266 void sendPrioConfigEvent(pid_t pid, pid_t tid, int32_t prio); 267 void sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio); 268 status_t sendSetParameterConfigEvent_l(const String8& keyValuePair); 269 status_t sendCreateAudioPatchConfigEvent(const struct audio_patch *patch, 270 audio_patch_handle_t *handle); 271 status_t sendReleaseAudioPatchConfigEvent(audio_patch_handle_t handle); 272 void processConfigEvents_l(); 273 virtual void cacheParameters_l() = 0; 274 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 275 audio_patch_handle_t *handle) = 0; 276 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle) = 0; 277 virtual void getAudioPortConfig(struct audio_port_config *config) = 0; 278 279 280 // see note at declaration of mStandby, mOutDevice and mInDevice 281 bool standby() const { return mStandby; } 282 audio_devices_t outDevice() const { return mOutDevice; } 283 audio_devices_t inDevice() const { return mInDevice; } 284 285 virtual audio_stream_t* stream() const = 0; 286 287 sp<EffectHandle> createEffect_l( 288 const sp<AudioFlinger::Client>& client, 289 const sp<IEffectClient>& effectClient, 290 int32_t priority, 291 int sessionId, 292 effect_descriptor_t *desc, 293 int *enabled, 294 status_t *status /*non-NULL*/); 295 296 // return values for hasAudioSession (bit field) 297 enum effect_state { 298 EFFECT_SESSION = 0x1, // the audio session corresponds to at least one 299 // effect 300 TRACK_SESSION = 0x2 // the audio session corresponds to at least one 301 // track 302 }; 303 304 // get effect chain corresponding to session Id. 305 sp<EffectChain> getEffectChain(int sessionId); 306 // same as getEffectChain() but must be called with ThreadBase mutex locked 307 sp<EffectChain> getEffectChain_l(int sessionId) const; 308 // add an effect chain to the chain list (mEffectChains) 309 virtual status_t addEffectChain_l(const sp<EffectChain>& chain) = 0; 310 // remove an effect chain from the chain list (mEffectChains) 311 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain) = 0; 312 // lock all effect chains Mutexes. Must be called before releasing the 313 // ThreadBase mutex before processing the mixer and effects. This guarantees the 314 // integrity of the chains during the process. 315 // Also sets the parameter 'effectChains' to current value of mEffectChains. 316 void lockEffectChains_l(Vector< sp<EffectChain> >& effectChains); 317 // unlock effect chains after process 318 void unlockEffectChains(const Vector< sp<EffectChain> >& effectChains); 319 // get a copy of mEffectChains vector 320 Vector< sp<EffectChain> > getEffectChains_l() const { return mEffectChains; }; 321 // set audio mode to all effect chains 322 void setMode(audio_mode_t mode); 323 // get effect module with corresponding ID on specified audio session 324 sp<AudioFlinger::EffectModule> getEffect(int sessionId, int effectId); 325 sp<AudioFlinger::EffectModule> getEffect_l(int sessionId, int effectId); 326 // add and effect module. Also creates the effect chain is none exists for 327 // the effects audio session 328 status_t addEffect_l(const sp< EffectModule>& effect); 329 // remove and effect module. Also removes the effect chain is this was the last 330 // effect 331 void removeEffect_l(const sp< EffectModule>& effect); 332 // detach all tracks connected to an auxiliary effect 333 virtual void detachAuxEffect_l(int effectId __unused) {} 334 // returns either EFFECT_SESSION if effects on this audio session exist in one 335 // chain, or TRACK_SESSION if tracks on this audio session exist, or both 336 virtual uint32_t hasAudioSession(int sessionId) const = 0; 337 // the value returned by default implementation is not important as the 338 // strategy is only meaningful for PlaybackThread which implements this method 339 virtual uint32_t getStrategyForSession_l(int sessionId __unused) { return 0; } 340 341 // suspend or restore effect according to the type of effect passed. a NULL 342 // type pointer means suspend all effects in the session 343 void setEffectSuspended(const effect_uuid_t *type, 344 bool suspend, 345 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 346 // check if some effects must be suspended/restored when an effect is enabled 347 // or disabled 348 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 349 bool enabled, 350 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 351 void checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 352 bool enabled, 353 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 354 355 virtual status_t setSyncEvent(const sp<SyncEvent>& event) = 0; 356 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const = 0; 357 358 // Return a reference to a per-thread heap which can be used to allocate IMemory 359 // objects that will be read-only to client processes, read/write to mediaserver, 360 // and shared by all client processes of the thread. 361 // The heap is per-thread rather than common across all threads, because 362 // clients can't be trusted not to modify the offset of the IMemory they receive. 363 // If a thread does not have such a heap, this method returns 0. 364 virtual sp<MemoryDealer> readOnlyHeap() const { return 0; } 365 366 virtual sp<IMemory> pipeMemory() const { return 0; } 367 368 void systemReady(); 369 370 mutable Mutex mLock; 371 372protected: 373 374 // entry describing an effect being suspended in mSuspendedSessions keyed vector 375 class SuspendedSessionDesc : public RefBase { 376 public: 377 SuspendedSessionDesc() : mRefCount(0) {} 378 379 int mRefCount; // number of active suspend requests 380 effect_uuid_t mType; // effect type UUID 381 }; 382 383 void acquireWakeLock(int uid = -1); 384 virtual void acquireWakeLock_l(int uid = -1); 385 void releaseWakeLock(); 386 void releaseWakeLock_l(); 387 void updateWakeLockUids(const SortedVector<int> &uids); 388 void updateWakeLockUids_l(const SortedVector<int> &uids); 389 void getPowerManager_l(); 390 void setEffectSuspended_l(const effect_uuid_t *type, 391 bool suspend, 392 int sessionId); 393 // updated mSuspendedSessions when an effect suspended or restored 394 void updateSuspendedSessions_l(const effect_uuid_t *type, 395 bool suspend, 396 int sessionId); 397 // check if some effects must be suspended when an effect chain is added 398 void checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain); 399 400 String16 getWakeLockTag(); 401 402 virtual void preExit() { } 403 virtual void setMasterMono_l(bool mono __unused) { } 404 virtual bool requireMonoBlend() { return false; } 405 406 friend class AudioFlinger; // for mEffectChains 407 408 const type_t mType; 409 410 // Used by parameters, config events, addTrack_l, exit 411 Condition mWaitWorkCV; 412 413 const sp<AudioFlinger> mAudioFlinger; 414 415 // updated by PlaybackThread::readOutputParameters_l() or 416 // RecordThread::readInputParameters_l() 417 uint32_t mSampleRate; 418 size_t mFrameCount; // output HAL, direct output, record 419 audio_channel_mask_t mChannelMask; 420 uint32_t mChannelCount; 421 size_t mFrameSize; 422 // not HAL frame size, this is for output sink (to pipe to fast mixer) 423 audio_format_t mFormat; // Source format for Recording and 424 // Sink format for Playback. 425 // Sink format may be different than 426 // HAL format if Fastmixer is used. 427 audio_format_t mHALFormat; 428 size_t mBufferSize; // HAL buffer size for read() or write() 429 430 Vector< sp<ConfigEvent> > mConfigEvents; 431 Vector< sp<ConfigEvent> > mPendingConfigEvents; // events awaiting system ready 432 433 // These fields are written and read by thread itself without lock or barrier, 434 // and read by other threads without lock or barrier via standby(), outDevice() 435 // and inDevice(). 436 // Because of the absence of a lock or barrier, any other thread that reads 437 // these fields must use the information in isolation, or be prepared to deal 438 // with possibility that it might be inconsistent with other information. 439 bool mStandby; // Whether thread is currently in standby. 440 audio_devices_t mOutDevice; // output device 441 audio_devices_t mInDevice; // input device 442 audio_devices_t mPrevOutDevice; // previous output device 443 audio_devices_t mPrevInDevice; // previous input device 444 struct audio_patch mPatch; 445 audio_source_t mAudioSource; 446 447 const audio_io_handle_t mId; 448 Vector< sp<EffectChain> > mEffectChains; 449 450 static const int kThreadNameLength = 16; // prctl(PR_SET_NAME) limit 451 char mThreadName[kThreadNameLength]; // guaranteed NUL-terminated 452 sp<IPowerManager> mPowerManager; 453 sp<IBinder> mWakeLockToken; 454 const sp<PMDeathRecipient> mDeathRecipient; 455 // list of suspended effects per session and per type. The first vector is 456 // keyed by session ID, the second by type UUID timeLow field 457 KeyedVector< int, KeyedVector< int, sp<SuspendedSessionDesc> > > 458 mSuspendedSessions; 459 static const size_t kLogSize = 4 * 1024; 460 sp<NBLog::Writer> mNBLogWriter; 461 bool mSystemReady; 462 bool mNotifiedBatteryStart; 463 ExtendedTimestamp mTimestamp; 464}; 465 466// --- PlaybackThread --- 467class PlaybackThread : public ThreadBase { 468public: 469 470#include "PlaybackTracks.h" 471 472 enum mixer_state { 473 MIXER_IDLE, // no active tracks 474 MIXER_TRACKS_ENABLED, // at least one active track, but no track has any data ready 475 MIXER_TRACKS_READY, // at least one active track, and at least one track has data 476 MIXER_DRAIN_TRACK, // drain currently playing track 477 MIXER_DRAIN_ALL, // fully drain the hardware 478 // standby mode does not have an enum value 479 // suspend by audio policy manager is orthogonal to mixer state 480 }; 481 482 PlaybackThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 483 audio_io_handle_t id, audio_devices_t device, type_t type, bool systemReady, 484 uint32_t bitRate = 0); 485 virtual ~PlaybackThread(); 486 487 void dump(int fd, const Vector<String16>& args); 488 489 // Thread virtuals 490 virtual bool threadLoop(); 491 492 // RefBase 493 virtual void onFirstRef(); 494 495protected: 496 // Code snippets that were lifted up out of threadLoop() 497 virtual void threadLoop_mix() = 0; 498 virtual void threadLoop_sleepTime() = 0; 499 virtual ssize_t threadLoop_write(); 500 virtual void threadLoop_drain(); 501 virtual void threadLoop_standby(); 502 virtual void threadLoop_exit(); 503 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 504 505 // prepareTracks_l reads and writes mActiveTracks, and returns 506 // the pending set of tracks to remove via Vector 'tracksToRemove'. The caller 507 // is responsible for clearing or destroying this Vector later on, when it 508 // is safe to do so. That will drop the final ref count and destroy the tracks. 509 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove) = 0; 510 void removeTracks_l(const Vector< sp<Track> >& tracksToRemove); 511 512 void writeCallback(); 513 void resetWriteBlocked(uint32_t sequence); 514 void drainCallback(); 515 void resetDraining(uint32_t sequence); 516 517 static int asyncCallback(stream_callback_event_t event, void *param, void *cookie); 518 519 virtual bool waitingAsyncCallback(); 520 virtual bool waitingAsyncCallback_l(); 521 virtual bool shouldStandby_l(); 522 virtual void onAddNewTrack_l(); 523 524 // ThreadBase virtuals 525 virtual void preExit(); 526 527public: 528 529 virtual status_t initCheck() const { return (mOutput == NULL) ? NO_INIT : NO_ERROR; } 530 531 // return estimated latency in milliseconds, as reported by HAL 532 uint32_t latency() const; 533 // same, but lock must already be held 534 uint32_t latency_l() const; 535 536 void setMasterVolume(float value); 537 void setMasterMute(bool muted); 538 539 void setStreamVolume(audio_stream_type_t stream, float value); 540 void setStreamMute(audio_stream_type_t stream, bool muted); 541 542 float streamVolume(audio_stream_type_t stream) const; 543 544 sp<Track> createTrack_l( 545 const sp<AudioFlinger::Client>& client, 546 audio_stream_type_t streamType, 547 uint32_t sampleRate, 548 audio_format_t format, 549 audio_channel_mask_t channelMask, 550 size_t *pFrameCount, 551 const sp<IMemory>& sharedBuffer, 552 int sessionId, 553 IAudioFlinger::track_flags_t *flags, 554 pid_t tid, 555 int uid, 556 status_t *status /*non-NULL*/); 557 558 AudioStreamOut* getOutput() const; 559 AudioStreamOut* clearOutput(); 560 virtual audio_stream_t* stream() const; 561 562 // a very large number of suspend() will eventually wraparound, but unlikely 563 void suspend() { (void) android_atomic_inc(&mSuspended); } 564 void restore() 565 { 566 // if restore() is done without suspend(), get back into 567 // range so that the next suspend() will operate correctly 568 if (android_atomic_dec(&mSuspended) <= 0) { 569 android_atomic_release_store(0, &mSuspended); 570 } 571 } 572 bool isSuspended() const 573 { return android_atomic_acquire_load(&mSuspended) > 0; } 574 575 virtual String8 getParameters(const String8& keys); 576 virtual void ioConfigChanged(audio_io_config_event event, pid_t pid = 0); 577 status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames); 578 // FIXME rename mixBuffer() to sinkBuffer() and remove int16_t* dependency. 579 // Consider also removing and passing an explicit mMainBuffer initialization 580 // parameter to AF::PlaybackThread::Track::Track(). 581 int16_t *mixBuffer() const { 582 return reinterpret_cast<int16_t *>(mSinkBuffer); }; 583 584 virtual void detachAuxEffect_l(int effectId); 585 status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track> track, 586 int EffectId); 587 status_t attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track> track, 588 int EffectId); 589 590 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 591 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 592 virtual uint32_t hasAudioSession(int sessionId) const; 593 virtual uint32_t getStrategyForSession_l(int sessionId); 594 595 596 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 597 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 598 599 // called with AudioFlinger lock held 600 void invalidateTracks(audio_stream_type_t streamType); 601 602 virtual size_t frameCount() const { return mNormalFrameCount; } 603 604 // Return's the HAL's frame count i.e. fast mixer buffer size. 605 size_t frameCountHAL() const { return mFrameCount; } 606 607 status_t getTimestamp_l(AudioTimestamp& timestamp); 608 609 void addPatchTrack(const sp<PatchTrack>& track); 610 void deletePatchTrack(const sp<PatchTrack>& track); 611 612 virtual void getAudioPortConfig(struct audio_port_config *config); 613 614protected: 615 // updated by readOutputParameters_l() 616 size_t mNormalFrameCount; // normal mixer and effects 617 618 bool mThreadThrottle; // throttle the thread processing 619 uint32_t mThreadThrottleTimeMs; // throttle time for MIXER threads 620 uint32_t mThreadThrottleEndMs; // notify once per throttling 621 uint32_t mHalfBufferMs; // half the buffer size in milliseconds 622 623 void* mSinkBuffer; // frame size aligned sink buffer 624 625 // TODO: 626 // Rearrange the buffer info into a struct/class with 627 // clear, copy, construction, destruction methods. 628 // 629 // mSinkBuffer also has associated with it: 630 // 631 // mSinkBufferSize: Sink Buffer Size 632 // mFormat: Sink Buffer Format 633 634 // Mixer Buffer (mMixerBuffer*) 635 // 636 // In the case of floating point or multichannel data, which is not in the 637 // sink format, it is required to accumulate in a higher precision or greater channel count 638 // buffer before downmixing or data conversion to the sink buffer. 639 640 // Set to "true" to enable the Mixer Buffer otherwise mixer output goes to sink buffer. 641 bool mMixerBufferEnabled; 642 643 // Storage, 32 byte aligned (may make this alignment a requirement later). 644 // Due to constraints on mNormalFrameCount, the buffer size is a multiple of 16 frames. 645 void* mMixerBuffer; 646 647 // Size of mMixerBuffer in bytes: mNormalFrameCount * #channels * sampsize. 648 size_t mMixerBufferSize; 649 650 // The audio format of mMixerBuffer. Set to AUDIO_FORMAT_PCM_(FLOAT|16_BIT) only. 651 audio_format_t mMixerBufferFormat; 652 653 // An internal flag set to true by MixerThread::prepareTracks_l() 654 // when mMixerBuffer contains valid data after mixing. 655 bool mMixerBufferValid; 656 657 // Effects Buffer (mEffectsBuffer*) 658 // 659 // In the case of effects data, which is not in the sink format, 660 // it is required to accumulate in a different buffer before data conversion 661 // to the sink buffer. 662 663 // Set to "true" to enable the Effects Buffer otherwise effects output goes to sink buffer. 664 bool mEffectBufferEnabled; 665 666 // Storage, 32 byte aligned (may make this alignment a requirement later). 667 // Due to constraints on mNormalFrameCount, the buffer size is a multiple of 16 frames. 668 void* mEffectBuffer; 669 670 // Size of mEffectsBuffer in bytes: mNormalFrameCount * #channels * sampsize. 671 size_t mEffectBufferSize; 672 673 // The audio format of mEffectsBuffer. Set to AUDIO_FORMAT_PCM_16_BIT only. 674 audio_format_t mEffectBufferFormat; 675 676 // An internal flag set to true by MixerThread::prepareTracks_l() 677 // when mEffectsBuffer contains valid data after mixing. 678 // 679 // When this is set, all mixer data is routed into the effects buffer 680 // for any processing (including output processing). 681 bool mEffectBufferValid; 682 683 // suspend count, > 0 means suspended. While suspended, the thread continues to pull from 684 // tracks and mix, but doesn't write to HAL. A2DP and SCO HAL implementations can't handle 685 // concurrent use of both of them, so Audio Policy Service suspends one of the threads to 686 // workaround that restriction. 687 // 'volatile' means accessed via atomic operations and no lock. 688 volatile int32_t mSuspended; 689 690 int64_t mBytesWritten; 691 int64_t mFramesWritten; // not reset on standby 692private: 693 // mMasterMute is in both PlaybackThread and in AudioFlinger. When a 694 // PlaybackThread needs to find out if master-muted, it checks it's local 695 // copy rather than the one in AudioFlinger. This optimization saves a lock. 696 bool mMasterMute; 697 void setMasterMute_l(bool muted) { mMasterMute = muted; } 698protected: 699 SortedVector< wp<Track> > mActiveTracks; // FIXME check if this could be sp<> 700 SortedVector<int> mWakeLockUids; 701 int mActiveTracksGeneration; 702 wp<Track> mLatestActiveTrack; // latest track added to mActiveTracks 703 704 // Allocate a track name for a given channel mask. 705 // Returns name >= 0 if successful, -1 on failure. 706 virtual int getTrackName_l(audio_channel_mask_t channelMask, 707 audio_format_t format, int sessionId) = 0; 708 virtual void deleteTrackName_l(int name) = 0; 709 710 // Time to sleep between cycles when: 711 virtual uint32_t activeSleepTimeUs() const; // mixer state MIXER_TRACKS_ENABLED 712 virtual uint32_t idleSleepTimeUs() const = 0; // mixer state MIXER_IDLE 713 virtual uint32_t suspendSleepTimeUs() const = 0; // audio policy manager suspended us 714 // No sleep when mixer state == MIXER_TRACKS_READY; relies on audio HAL stream->write() 715 // No sleep in standby mode; waits on a condition 716 717 // Code snippets that are temporarily lifted up out of threadLoop() until the merge 718 void checkSilentMode_l(); 719 720 // Non-trivial for DUPLICATING only 721 virtual void saveOutputTracks() { } 722 virtual void clearOutputTracks() { } 723 724 // Cache various calculated values, at threadLoop() entry and after a parameter change 725 virtual void cacheParameters_l(); 726 727 virtual uint32_t correctLatency_l(uint32_t latency) const; 728 729 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 730 audio_patch_handle_t *handle); 731 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 732 733 bool usesHwAvSync() const { return (mType == DIRECT) && (mOutput != NULL) 734 && mHwSupportsPause 735 && (mOutput->flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC); } 736 737private: 738 739 friend class AudioFlinger; // for numerous 740 741 PlaybackThread& operator = (const PlaybackThread&); 742 743 status_t addTrack_l(const sp<Track>& track); 744 bool destroyTrack_l(const sp<Track>& track); 745 void removeTrack_l(const sp<Track>& track); 746 void broadcast_l(); 747 748 void readOutputParameters_l(); 749 750 virtual void dumpInternals(int fd, const Vector<String16>& args); 751 void dumpTracks(int fd, const Vector<String16>& args); 752 753 SortedVector< sp<Track> > mTracks; 754 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 755 AudioStreamOut *mOutput; 756 757 float mMasterVolume; 758 nsecs_t mLastWriteTime; 759 int mNumWrites; 760 int mNumDelayedWrites; 761 bool mInWrite; 762 763 // FIXME rename these former local variables of threadLoop to standard "m" names 764 nsecs_t mStandbyTimeNs; 765 size_t mSinkBufferSize; 766 767 // cached copies of activeSleepTimeUs() and idleSleepTimeUs() made by cacheParameters_l() 768 uint32_t mActiveSleepTimeUs; 769 uint32_t mIdleSleepTimeUs; 770 771 uint32_t mSleepTimeUs; 772 773 // mixer status returned by prepareTracks_l() 774 mixer_state mMixerStatus; // current cycle 775 // previous cycle when in prepareTracks_l() 776 mixer_state mMixerStatusIgnoringFastTracks; 777 // FIXME or a separate ready state per track 778 779 // FIXME move these declarations into the specific sub-class that needs them 780 // MIXER only 781 uint32_t sleepTimeShift; 782 783 // same as AudioFlinger::mStandbyTimeInNsecs except for DIRECT which uses a shorter value 784 nsecs_t mStandbyDelayNs; 785 786 // MIXER only 787 nsecs_t maxPeriod; 788 789 // DUPLICATING only 790 uint32_t writeFrames; 791 792 size_t mBytesRemaining; 793 size_t mCurrentWriteLength; 794 bool mUseAsyncWrite; 795 // mWriteAckSequence contains current write sequence on bits 31-1. The write sequence is 796 // incremented each time a write(), a flush() or a standby() occurs. 797 // Bit 0 is set when a write blocks and indicates a callback is expected. 798 // Bit 0 is reset by the async callback thread calling resetWriteBlocked(). Out of sequence 799 // callbacks are ignored. 800 uint32_t mWriteAckSequence; 801 // mDrainSequence contains current drain sequence on bits 31-1. The drain sequence is 802 // incremented each time a drain is requested or a flush() or standby() occurs. 803 // Bit 0 is set when the drain() command is called at the HAL and indicates a callback is 804 // expected. 805 // Bit 0 is reset by the async callback thread calling resetDraining(). Out of sequence 806 // callbacks are ignored. 807 uint32_t mDrainSequence; 808 // A condition that must be evaluated by prepareTrack_l() has changed and we must not wait 809 // for async write callback in the thread loop before evaluating it 810 bool mSignalPending; 811 sp<AsyncCallbackThread> mCallbackThread; 812 813private: 814 // The HAL output sink is treated as non-blocking, but current implementation is blocking 815 sp<NBAIO_Sink> mOutputSink; 816 // If a fast mixer is present, the blocking pipe sink, otherwise clear 817 sp<NBAIO_Sink> mPipeSink; 818 // The current sink for the normal mixer to write it's (sub)mix, mOutputSink or mPipeSink 819 sp<NBAIO_Sink> mNormalSink; 820#ifdef TEE_SINK 821 // For dumpsys 822 sp<NBAIO_Sink> mTeeSink; 823 sp<NBAIO_Source> mTeeSource; 824#endif 825 uint32_t mScreenState; // cached copy of gScreenState 826 static const size_t kFastMixerLogSize = 4 * 1024; 827 sp<NBLog::Writer> mFastMixerNBLogWriter; 828public: 829 virtual bool hasFastMixer() const = 0; 830 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex __unused) const 831 { FastTrackUnderruns dummy; return dummy; } 832 833protected: 834 // accessed by both binder threads and within threadLoop(), lock on mutex needed 835 unsigned mFastTrackAvailMask; // bit i set if fast track [i] is available 836 bool mHwSupportsPause; 837 bool mHwPaused; 838 bool mFlushPending; 839 uint32_t mBufferDurationUs; // estimated duration of an audio HAL buffer 840 // based on initial bit rate (offload only) 841}; 842 843class MixerThread : public PlaybackThread { 844public: 845 MixerThread(const sp<AudioFlinger>& audioFlinger, 846 AudioStreamOut* output, 847 audio_io_handle_t id, 848 audio_devices_t device, 849 bool systemReady, 850 type_t type = MIXER); 851 virtual ~MixerThread(); 852 853 // Thread virtuals 854 855 virtual bool checkForNewParameter_l(const String8& keyValuePair, 856 status_t& status); 857 virtual void dumpInternals(int fd, const Vector<String16>& args); 858 859protected: 860 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 861 virtual int getTrackName_l(audio_channel_mask_t channelMask, 862 audio_format_t format, int sessionId); 863 virtual void deleteTrackName_l(int name); 864 virtual uint32_t idleSleepTimeUs() const; 865 virtual uint32_t suspendSleepTimeUs() const; 866 virtual void cacheParameters_l(); 867 868 virtual void acquireWakeLock_l(int uid = -1) { 869 PlaybackThread::acquireWakeLock_l(uid); 870 if (hasFastMixer()) { 871 mFastMixer->setBoottimeOffset( 872 mTimestamp.mTimebaseOffset[ExtendedTimestamp::TIMEBASE_BOOTTIME]); 873 } 874 } 875 876 // threadLoop snippets 877 virtual ssize_t threadLoop_write(); 878 virtual void threadLoop_standby(); 879 virtual void threadLoop_mix(); 880 virtual void threadLoop_sleepTime(); 881 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 882 virtual uint32_t correctLatency_l(uint32_t latency) const; 883 884 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 885 audio_patch_handle_t *handle); 886 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 887 888 AudioMixer* mAudioMixer; // normal mixer 889private: 890 // one-time initialization, no locks required 891 sp<FastMixer> mFastMixer; // non-0 if there is also a fast mixer 892 sp<AudioWatchdog> mAudioWatchdog; // non-0 if there is an audio watchdog thread 893 894 // contents are not guaranteed to be consistent, no locks required 895 FastMixerDumpState mFastMixerDumpState; 896#ifdef STATE_QUEUE_DUMP 897 StateQueueObserverDump mStateQueueObserverDump; 898 StateQueueMutatorDump mStateQueueMutatorDump; 899#endif 900 AudioWatchdogDump mAudioWatchdogDump; 901 902 // accessible only within the threadLoop(), no locks required 903 // mFastMixer->sq() // for mutating and pushing state 904 int32_t mFastMixerFutex; // for cold idle 905 906 std::atomic_bool mMasterMono; 907public: 908 virtual bool hasFastMixer() const { return mFastMixer != 0; } 909 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const { 910 ALOG_ASSERT(fastIndex < FastMixerState::kMaxFastTracks); 911 return mFastMixerDumpState.mTracks[fastIndex].mUnderruns; 912 } 913 914protected: 915 virtual void setMasterMono_l(bool mono) { 916 mMasterMono.store(mono); 917 if (mFastMixer != nullptr) { /* hasFastMixer() */ 918 mFastMixer->setMasterMono(mMasterMono); 919 } 920 } 921 // the FastMixer performs mono blend if it exists. 922 // Blending with limiter is not idempotent, 923 // and blending without limiter is idempotent but inefficient to do twice. 924 virtual bool requireMonoBlend() { return mMasterMono.load() && !hasFastMixer(); } 925}; 926 927class DirectOutputThread : public PlaybackThread { 928public: 929 930 DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 931 audio_io_handle_t id, audio_devices_t device, bool systemReady, 932 uint32_t bitRate = 0); 933 virtual ~DirectOutputThread(); 934 935 // Thread virtuals 936 937 virtual bool checkForNewParameter_l(const String8& keyValuePair, 938 status_t& status); 939 virtual void flushHw_l(); 940 941protected: 942 virtual int getTrackName_l(audio_channel_mask_t channelMask, 943 audio_format_t format, int sessionId); 944 virtual void deleteTrackName_l(int name); 945 virtual uint32_t activeSleepTimeUs() const; 946 virtual uint32_t idleSleepTimeUs() const; 947 virtual uint32_t suspendSleepTimeUs() const; 948 virtual void cacheParameters_l(); 949 950 // threadLoop snippets 951 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 952 virtual void threadLoop_mix(); 953 virtual void threadLoop_sleepTime(); 954 virtual void threadLoop_exit(); 955 virtual bool shouldStandby_l(); 956 957 virtual void onAddNewTrack_l(); 958 959 // volumes last sent to audio HAL with stream->set_volume() 960 float mLeftVolFloat; 961 float mRightVolFloat; 962 963 DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 964 audio_io_handle_t id, uint32_t device, ThreadBase::type_t type, 965 bool systemReady, uint32_t bitRate = 0); 966 void processVolume_l(Track *track, bool lastTrack); 967 968 // prepareTracks_l() tells threadLoop_mix() the name of the single active track 969 sp<Track> mActiveTrack; 970 971 wp<Track> mPreviousTrack; // used to detect track switch 972 973public: 974 virtual bool hasFastMixer() const { return false; } 975}; 976 977class OffloadThread : public DirectOutputThread { 978public: 979 980 OffloadThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 981 audio_io_handle_t id, uint32_t device, 982 bool systemReady, uint32_t bitRate); 983 virtual ~OffloadThread() {}; 984 virtual void flushHw_l(); 985 986protected: 987 // threadLoop snippets 988 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 989 virtual void threadLoop_exit(); 990 991 virtual uint32_t activeSleepTimeUs() const; 992 993 virtual bool waitingAsyncCallback(); 994 virtual bool waitingAsyncCallback_l(); 995 996private: 997 size_t mPausedWriteLength; // length in bytes of write interrupted by pause 998 size_t mPausedBytesRemaining; // bytes still waiting in mixbuffer after resume 999}; 1000 1001class AsyncCallbackThread : public Thread { 1002public: 1003 1004 AsyncCallbackThread(const wp<PlaybackThread>& playbackThread); 1005 1006 virtual ~AsyncCallbackThread(); 1007 1008 // Thread virtuals 1009 virtual bool threadLoop(); 1010 1011 // RefBase 1012 virtual void onFirstRef(); 1013 1014 void exit(); 1015 void setWriteBlocked(uint32_t sequence); 1016 void resetWriteBlocked(); 1017 void setDraining(uint32_t sequence); 1018 void resetDraining(); 1019 1020private: 1021 const wp<PlaybackThread> mPlaybackThread; 1022 // mWriteAckSequence corresponds to the last write sequence passed by the offload thread via 1023 // setWriteBlocked(). The sequence is shifted one bit to the left and the lsb is used 1024 // to indicate that the callback has been received via resetWriteBlocked() 1025 uint32_t mWriteAckSequence; 1026 // mDrainSequence corresponds to the last drain sequence passed by the offload thread via 1027 // setDraining(). The sequence is shifted one bit to the left and the lsb is used 1028 // to indicate that the callback has been received via resetDraining() 1029 uint32_t mDrainSequence; 1030 Condition mWaitWorkCV; 1031 Mutex mLock; 1032}; 1033 1034class DuplicatingThread : public MixerThread { 1035public: 1036 DuplicatingThread(const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread, 1037 audio_io_handle_t id, bool systemReady); 1038 virtual ~DuplicatingThread(); 1039 1040 // Thread virtuals 1041 void addOutputTrack(MixerThread* thread); 1042 void removeOutputTrack(MixerThread* thread); 1043 uint32_t waitTimeMs() const { return mWaitTimeMs; } 1044protected: 1045 virtual uint32_t activeSleepTimeUs() const; 1046 1047private: 1048 bool outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks); 1049protected: 1050 // threadLoop snippets 1051 virtual void threadLoop_mix(); 1052 virtual void threadLoop_sleepTime(); 1053 virtual ssize_t threadLoop_write(); 1054 virtual void threadLoop_standby(); 1055 virtual void cacheParameters_l(); 1056 1057private: 1058 // called from threadLoop, addOutputTrack, removeOutputTrack 1059 virtual void updateWaitTime_l(); 1060protected: 1061 virtual void saveOutputTracks(); 1062 virtual void clearOutputTracks(); 1063private: 1064 1065 uint32_t mWaitTimeMs; 1066 SortedVector < sp<OutputTrack> > outputTracks; 1067 SortedVector < sp<OutputTrack> > mOutputTracks; 1068public: 1069 virtual bool hasFastMixer() const { return false; } 1070}; 1071 1072 1073// record thread 1074class RecordThread : public ThreadBase 1075{ 1076public: 1077 1078 class RecordTrack; 1079 1080 /* The ResamplerBufferProvider is used to retrieve recorded input data from the 1081 * RecordThread. It maintains local state on the relative position of the read 1082 * position of the RecordTrack compared with the RecordThread. 1083 */ 1084 class ResamplerBufferProvider : public AudioBufferProvider 1085 { 1086 public: 1087 ResamplerBufferProvider(RecordTrack* recordTrack) : 1088 mRecordTrack(recordTrack), 1089 mRsmpInUnrel(0), mRsmpInFront(0) { } 1090 virtual ~ResamplerBufferProvider() { } 1091 1092 // called to set the ResamplerBufferProvider to head of the RecordThread data buffer, 1093 // skipping any previous data read from the hal. 1094 virtual void reset(); 1095 1096 /* Synchronizes RecordTrack position with the RecordThread. 1097 * Calculates available frames and handle overruns if the RecordThread 1098 * has advanced faster than the ResamplerBufferProvider has retrieved data. 1099 * TODO: why not do this for every getNextBuffer? 1100 * 1101 * Parameters 1102 * framesAvailable: pointer to optional output size_t to store record track 1103 * frames available. 1104 * hasOverrun: pointer to optional boolean, returns true if track has overrun. 1105 */ 1106 1107 virtual void sync(size_t *framesAvailable = NULL, bool *hasOverrun = NULL); 1108 1109 // AudioBufferProvider interface 1110 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer); 1111 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 1112 private: 1113 RecordTrack * const mRecordTrack; 1114 size_t mRsmpInUnrel; // unreleased frames remaining from 1115 // most recent getNextBuffer 1116 // for debug only 1117 int32_t mRsmpInFront; // next available frame 1118 // rolling counter that is never cleared 1119 }; 1120 1121 /* The RecordBufferConverter is used for format, channel, and sample rate 1122 * conversion for a RecordTrack. 1123 * 1124 * TODO: Self contained, so move to a separate file later. 1125 * 1126 * RecordBufferConverter uses the convert() method rather than exposing a 1127 * buffer provider interface; this is to save a memory copy. 1128 */ 1129 class RecordBufferConverter 1130 { 1131 public: 1132 RecordBufferConverter( 1133 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 1134 uint32_t srcSampleRate, 1135 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 1136 uint32_t dstSampleRate); 1137 1138 ~RecordBufferConverter(); 1139 1140 /* Converts input data from an AudioBufferProvider by format, channelMask, 1141 * and sampleRate to a destination buffer. 1142 * 1143 * Parameters 1144 * dst: buffer to place the converted data. 1145 * provider: buffer provider to obtain source data. 1146 * frames: number of frames to convert 1147 * 1148 * Returns the number of frames converted. 1149 */ 1150 size_t convert(void *dst, AudioBufferProvider *provider, size_t frames); 1151 1152 // returns NO_ERROR if constructor was successful 1153 status_t initCheck() const { 1154 // mSrcChannelMask set on successful updateParameters 1155 return mSrcChannelMask != AUDIO_CHANNEL_INVALID ? NO_ERROR : NO_INIT; 1156 } 1157 1158 // allows dynamic reconfigure of all parameters 1159 status_t updateParameters( 1160 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 1161 uint32_t srcSampleRate, 1162 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 1163 uint32_t dstSampleRate); 1164 1165 // called to reset resampler buffers on record track discontinuity 1166 void reset() { 1167 if (mResampler != NULL) { 1168 mResampler->reset(); 1169 } 1170 } 1171 1172 private: 1173 // format conversion when not using resampler 1174 void convertNoResampler(void *dst, const void *src, size_t frames); 1175 1176 // format conversion when using resampler; modifies src in-place 1177 void convertResampler(void *dst, /*not-a-const*/ void *src, size_t frames); 1178 1179 // user provided information 1180 audio_channel_mask_t mSrcChannelMask; 1181 audio_format_t mSrcFormat; 1182 uint32_t mSrcSampleRate; 1183 audio_channel_mask_t mDstChannelMask; 1184 audio_format_t mDstFormat; 1185 uint32_t mDstSampleRate; 1186 1187 // derived information 1188 uint32_t mSrcChannelCount; 1189 uint32_t mDstChannelCount; 1190 size_t mDstFrameSize; 1191 1192 // format conversion buffer 1193 void *mBuf; 1194 size_t mBufFrames; 1195 size_t mBufFrameSize; 1196 1197 // resampler info 1198 AudioResampler *mResampler; 1199 1200 bool mIsLegacyDownmix; // legacy stereo to mono conversion needed 1201 bool mIsLegacyUpmix; // legacy mono to stereo conversion needed 1202 bool mRequiresFloat; // data processing requires float (e.g. resampler) 1203 PassthruBufferProvider *mInputConverterProvider; // converts input to float 1204 int8_t mIdxAry[sizeof(uint32_t) * 8]; // used for channel mask conversion 1205 }; 1206 1207#include "RecordTracks.h" 1208 1209 RecordThread(const sp<AudioFlinger>& audioFlinger, 1210 AudioStreamIn *input, 1211 audio_io_handle_t id, 1212 audio_devices_t outDevice, 1213 audio_devices_t inDevice, 1214 bool systemReady 1215#ifdef TEE_SINK 1216 , const sp<NBAIO_Sink>& teeSink 1217#endif 1218 ); 1219 virtual ~RecordThread(); 1220 1221 // no addTrack_l ? 1222 void destroyTrack_l(const sp<RecordTrack>& track); 1223 void removeTrack_l(const sp<RecordTrack>& track); 1224 1225 void dumpInternals(int fd, const Vector<String16>& args); 1226 void dumpTracks(int fd, const Vector<String16>& args); 1227 1228 // Thread virtuals 1229 virtual bool threadLoop(); 1230 1231 // RefBase 1232 virtual void onFirstRef(); 1233 1234 virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; } 1235 1236 virtual sp<MemoryDealer> readOnlyHeap() const { return mReadOnlyHeap; } 1237 1238 virtual sp<IMemory> pipeMemory() const { return mPipeMemory; } 1239 1240 sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l( 1241 const sp<AudioFlinger::Client>& client, 1242 uint32_t sampleRate, 1243 audio_format_t format, 1244 audio_channel_mask_t channelMask, 1245 size_t *pFrameCount, 1246 int sessionId, 1247 size_t *notificationFrames, 1248 int uid, 1249 IAudioFlinger::track_flags_t *flags, 1250 pid_t tid, 1251 status_t *status /*non-NULL*/); 1252 1253 status_t start(RecordTrack* recordTrack, 1254 AudioSystem::sync_event_t event, 1255 int triggerSession); 1256 1257 // ask the thread to stop the specified track, and 1258 // return true if the caller should then do it's part of the stopping process 1259 bool stop(RecordTrack* recordTrack); 1260 1261 void dump(int fd, const Vector<String16>& args); 1262 AudioStreamIn* clearInput(); 1263 virtual audio_stream_t* stream() const; 1264 1265 1266 virtual bool checkForNewParameter_l(const String8& keyValuePair, 1267 status_t& status); 1268 virtual void cacheParameters_l() {} 1269 virtual String8 getParameters(const String8& keys); 1270 virtual void ioConfigChanged(audio_io_config_event event, pid_t pid = 0); 1271 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 1272 audio_patch_handle_t *handle); 1273 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 1274 1275 void addPatchRecord(const sp<PatchRecord>& record); 1276 void deletePatchRecord(const sp<PatchRecord>& record); 1277 1278 void readInputParameters_l(); 1279 virtual uint32_t getInputFramesLost(); 1280 1281 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1282 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1283 virtual uint32_t hasAudioSession(int sessionId) const; 1284 1285 // Return the set of unique session IDs across all tracks. 1286 // The keys are the session IDs, and the associated values are meaningless. 1287 // FIXME replace by Set [and implement Bag/Multiset for other uses]. 1288 KeyedVector<int, bool> sessionIds() const; 1289 1290 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 1291 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 1292 1293 static void syncStartEventCallback(const wp<SyncEvent>& event); 1294 1295 virtual size_t frameCount() const { return mFrameCount; } 1296 bool hasFastCapture() const { return mFastCapture != 0; } 1297 virtual void getAudioPortConfig(struct audio_port_config *config); 1298 1299private: 1300 // Enter standby if not already in standby, and set mStandby flag 1301 void standbyIfNotAlreadyInStandby(); 1302 1303 // Call the HAL standby method unconditionally, and don't change mStandby flag 1304 void inputStandBy(); 1305 1306 AudioStreamIn *mInput; 1307 SortedVector < sp<RecordTrack> > mTracks; 1308 // mActiveTracks has dual roles: it indicates the current active track(s), and 1309 // is used together with mStartStopCond to indicate start()/stop() progress 1310 SortedVector< sp<RecordTrack> > mActiveTracks; 1311 // generation counter for mActiveTracks 1312 int mActiveTracksGen; 1313 Condition mStartStopCond; 1314 1315 // resampler converts input at HAL Hz to output at AudioRecord client Hz 1316 void *mRsmpInBuffer; // 1317 size_t mRsmpInFrames; // size of resampler input in frames 1318 size_t mRsmpInFramesP2;// size rounded up to a power-of-2 1319 1320 // rolling index that is never cleared 1321 int32_t mRsmpInRear; // last filled frame + 1 1322 1323 // For dumpsys 1324 const sp<NBAIO_Sink> mTeeSink; 1325 1326 const sp<MemoryDealer> mReadOnlyHeap; 1327 1328 // one-time initialization, no locks required 1329 sp<FastCapture> mFastCapture; // non-0 if there is also 1330 // a fast capture 1331 1332 // FIXME audio watchdog thread 1333 1334 // contents are not guaranteed to be consistent, no locks required 1335 FastCaptureDumpState mFastCaptureDumpState; 1336#ifdef STATE_QUEUE_DUMP 1337 // FIXME StateQueue observer and mutator dump fields 1338#endif 1339 // FIXME audio watchdog dump 1340 1341 // accessible only within the threadLoop(), no locks required 1342 // mFastCapture->sq() // for mutating and pushing state 1343 int32_t mFastCaptureFutex; // for cold idle 1344 1345 // The HAL input source is treated as non-blocking, 1346 // but current implementation is blocking 1347 sp<NBAIO_Source> mInputSource; 1348 // The source for the normal capture thread to read from: mInputSource or mPipeSource 1349 sp<NBAIO_Source> mNormalSource; 1350 // If a fast capture is present, the non-blocking pipe sink written to by fast capture, 1351 // otherwise clear 1352 sp<NBAIO_Sink> mPipeSink; 1353 // If a fast capture is present, the non-blocking pipe source read by normal thread, 1354 // otherwise clear 1355 sp<NBAIO_Source> mPipeSource; 1356 // Depth of pipe from fast capture to normal thread and fast clients, always power of 2 1357 size_t mPipeFramesP2; 1358 // If a fast capture is present, the Pipe as IMemory, otherwise clear 1359 sp<IMemory> mPipeMemory; 1360 1361 static const size_t kFastCaptureLogSize = 4 * 1024; 1362 sp<NBLog::Writer> mFastCaptureNBLogWriter; 1363 1364 bool mFastTrackAvail; // true if fast track available 1365}; 1366