Threads.h revision 72e3f39146fce4686bd96f11057c051bea376dfb
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) : 108 mEvent(event) {} 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 }; 116 117 class IoConfigEvent : public ConfigEvent { 118 public: 119 IoConfigEvent(audio_io_config_event event) : 120 ConfigEvent(CFG_EVENT_IO) { 121 mData = new IoConfigEventData(event); 122 } 123 virtual ~IoConfigEvent() {} 124 }; 125 126 class PrioConfigEventData : public ConfigEventData { 127 public: 128 PrioConfigEventData(pid_t pid, pid_t tid, int32_t prio) : 129 mPid(pid), mTid(tid), mPrio(prio) {} 130 131 virtual void dump(char *buffer, size_t size) { 132 snprintf(buffer, size, "Prio event: pid %d, tid %d, prio %d\n", mPid, mTid, mPrio); 133 } 134 135 const pid_t mPid; 136 const pid_t mTid; 137 const int32_t mPrio; 138 }; 139 140 class PrioConfigEvent : public ConfigEvent { 141 public: 142 PrioConfigEvent(pid_t pid, pid_t tid, int32_t prio) : 143 ConfigEvent(CFG_EVENT_PRIO, true) { 144 mData = new PrioConfigEventData(pid, tid, prio); 145 } 146 virtual ~PrioConfigEvent() {} 147 }; 148 149 class SetParameterConfigEventData : public ConfigEventData { 150 public: 151 SetParameterConfigEventData(String8 keyValuePairs) : 152 mKeyValuePairs(keyValuePairs) {} 153 154 virtual void dump(char *buffer, size_t size) { 155 snprintf(buffer, size, "KeyValue: %s\n", mKeyValuePairs.string()); 156 } 157 158 const String8 mKeyValuePairs; 159 }; 160 161 class SetParameterConfigEvent : public ConfigEvent { 162 public: 163 SetParameterConfigEvent(String8 keyValuePairs) : 164 ConfigEvent(CFG_EVENT_SET_PARAMETER) { 165 mData = new SetParameterConfigEventData(keyValuePairs); 166 mWaitStatus = true; 167 } 168 virtual ~SetParameterConfigEvent() {} 169 }; 170 171 class CreateAudioPatchConfigEventData : public ConfigEventData { 172 public: 173 CreateAudioPatchConfigEventData(const struct audio_patch patch, 174 audio_patch_handle_t handle) : 175 mPatch(patch), mHandle(handle) {} 176 177 virtual void dump(char *buffer, size_t size) { 178 snprintf(buffer, size, "Patch handle: %u\n", mHandle); 179 } 180 181 const struct audio_patch mPatch; 182 audio_patch_handle_t mHandle; 183 }; 184 185 class CreateAudioPatchConfigEvent : public ConfigEvent { 186 public: 187 CreateAudioPatchConfigEvent(const struct audio_patch patch, 188 audio_patch_handle_t handle) : 189 ConfigEvent(CFG_EVENT_CREATE_AUDIO_PATCH) { 190 mData = new CreateAudioPatchConfigEventData(patch, handle); 191 mWaitStatus = true; 192 } 193 virtual ~CreateAudioPatchConfigEvent() {} 194 }; 195 196 class ReleaseAudioPatchConfigEventData : public ConfigEventData { 197 public: 198 ReleaseAudioPatchConfigEventData(const audio_patch_handle_t handle) : 199 mHandle(handle) {} 200 201 virtual void dump(char *buffer, size_t size) { 202 snprintf(buffer, size, "Patch handle: %u\n", mHandle); 203 } 204 205 audio_patch_handle_t mHandle; 206 }; 207 208 class ReleaseAudioPatchConfigEvent : public ConfigEvent { 209 public: 210 ReleaseAudioPatchConfigEvent(const audio_patch_handle_t handle) : 211 ConfigEvent(CFG_EVENT_RELEASE_AUDIO_PATCH) { 212 mData = new ReleaseAudioPatchConfigEventData(handle); 213 mWaitStatus = true; 214 } 215 virtual ~ReleaseAudioPatchConfigEvent() {} 216 }; 217 218 class PMDeathRecipient : public IBinder::DeathRecipient { 219 public: 220 PMDeathRecipient(const wp<ThreadBase>& thread) : mThread(thread) {} 221 virtual ~PMDeathRecipient() {} 222 223 // IBinder::DeathRecipient 224 virtual void binderDied(const wp<IBinder>& who); 225 226 private: 227 PMDeathRecipient(const PMDeathRecipient&); 228 PMDeathRecipient& operator = (const PMDeathRecipient&); 229 230 wp<ThreadBase> mThread; 231 }; 232 233 virtual status_t initCheck() const = 0; 234 235 // static externally-visible 236 type_t type() const { return mType; } 237 bool isDuplicating() const { return (mType == DUPLICATING); } 238 239 audio_io_handle_t id() const { return mId;} 240 241 // dynamic externally-visible 242 uint32_t sampleRate() const { return mSampleRate; } 243 audio_channel_mask_t channelMask() const { return mChannelMask; } 244 audio_format_t format() const { return mHALFormat; } 245 uint32_t channelCount() const { return mChannelCount; } 246 // Called by AudioFlinger::frameCount(audio_io_handle_t output) and effects, 247 // and returns the [normal mix] buffer's frame count. 248 virtual size_t frameCount() const = 0; 249 size_t frameSize() const { return mFrameSize; } 250 251 // Should be "virtual status_t requestExitAndWait()" and override same 252 // method in Thread, but Thread::requestExitAndWait() is not yet virtual. 253 void exit(); 254 virtual bool checkForNewParameter_l(const String8& keyValuePair, 255 status_t& status) = 0; 256 virtual status_t setParameters(const String8& keyValuePairs); 257 virtual String8 getParameters(const String8& keys) = 0; 258 virtual void ioConfigChanged(audio_io_config_event event) = 0; 259 // sendConfigEvent_l() must be called with ThreadBase::mLock held 260 // Can temporarily release the lock if waiting for a reply from 261 // processConfigEvents_l(). 262 status_t sendConfigEvent_l(sp<ConfigEvent>& event); 263 void sendIoConfigEvent(audio_io_config_event event); 264 void sendIoConfigEvent_l(audio_io_config_event event); 265 void sendPrioConfigEvent(pid_t pid, pid_t tid, int32_t prio); 266 void sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio); 267 status_t sendSetParameterConfigEvent_l(const String8& keyValuePair); 268 status_t sendCreateAudioPatchConfigEvent(const struct audio_patch *patch, 269 audio_patch_handle_t *handle); 270 status_t sendReleaseAudioPatchConfigEvent(audio_patch_handle_t handle); 271 void processConfigEvents_l(); 272 virtual void cacheParameters_l() = 0; 273 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 274 audio_patch_handle_t *handle) = 0; 275 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle) = 0; 276 virtual void getAudioPortConfig(struct audio_port_config *config) = 0; 277 278 279 // see note at declaration of mStandby, mOutDevice and mInDevice 280 bool standby() const { return mStandby; } 281 audio_devices_t outDevice() const { return mOutDevice; } 282 audio_devices_t inDevice() const { return mInDevice; } 283 284 virtual audio_stream_t* stream() const = 0; 285 286 sp<EffectHandle> createEffect_l( 287 const sp<AudioFlinger::Client>& client, 288 const sp<IEffectClient>& effectClient, 289 int32_t priority, 290 int sessionId, 291 effect_descriptor_t *desc, 292 int *enabled, 293 status_t *status /*non-NULL*/); 294 295 // return values for hasAudioSession (bit field) 296 enum effect_state { 297 EFFECT_SESSION = 0x1, // the audio session corresponds to at least one 298 // effect 299 TRACK_SESSION = 0x2 // the audio session corresponds to at least one 300 // track 301 }; 302 303 // get effect chain corresponding to session Id. 304 sp<EffectChain> getEffectChain(int sessionId); 305 // same as getEffectChain() but must be called with ThreadBase mutex locked 306 sp<EffectChain> getEffectChain_l(int sessionId) const; 307 // add an effect chain to the chain list (mEffectChains) 308 virtual status_t addEffectChain_l(const sp<EffectChain>& chain) = 0; 309 // remove an effect chain from the chain list (mEffectChains) 310 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain) = 0; 311 // lock all effect chains Mutexes. Must be called before releasing the 312 // ThreadBase mutex before processing the mixer and effects. This guarantees the 313 // integrity of the chains during the process. 314 // Also sets the parameter 'effectChains' to current value of mEffectChains. 315 void lockEffectChains_l(Vector< sp<EffectChain> >& effectChains); 316 // unlock effect chains after process 317 void unlockEffectChains(const Vector< sp<EffectChain> >& effectChains); 318 // get a copy of mEffectChains vector 319 Vector< sp<EffectChain> > getEffectChains_l() const { return mEffectChains; }; 320 // set audio mode to all effect chains 321 void setMode(audio_mode_t mode); 322 // get effect module with corresponding ID on specified audio session 323 sp<AudioFlinger::EffectModule> getEffect(int sessionId, int effectId); 324 sp<AudioFlinger::EffectModule> getEffect_l(int sessionId, int effectId); 325 // add and effect module. Also creates the effect chain is none exists for 326 // the effects audio session 327 status_t addEffect_l(const sp< EffectModule>& effect); 328 // remove and effect module. Also removes the effect chain is this was the last 329 // effect 330 void removeEffect_l(const sp< EffectModule>& effect); 331 // detach all tracks connected to an auxiliary effect 332 virtual void detachAuxEffect_l(int effectId __unused) {} 333 // returns either EFFECT_SESSION if effects on this audio session exist in one 334 // chain, or TRACK_SESSION if tracks on this audio session exist, or both 335 virtual uint32_t hasAudioSession(int sessionId) const = 0; 336 // the value returned by default implementation is not important as the 337 // strategy is only meaningful for PlaybackThread which implements this method 338 virtual uint32_t getStrategyForSession_l(int sessionId __unused) { return 0; } 339 340 // suspend or restore effect according to the type of effect passed. a NULL 341 // type pointer means suspend all effects in the session 342 void setEffectSuspended(const effect_uuid_t *type, 343 bool suspend, 344 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 345 // check if some effects must be suspended/restored when an effect is enabled 346 // or disabled 347 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 348 bool enabled, 349 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 350 void checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 351 bool enabled, 352 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 353 354 virtual status_t setSyncEvent(const sp<SyncEvent>& event) = 0; 355 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const = 0; 356 357 // Return a reference to a per-thread heap which can be used to allocate IMemory 358 // objects that will be read-only to client processes, read/write to mediaserver, 359 // and shared by all client processes of the thread. 360 // The heap is per-thread rather than common across all threads, because 361 // clients can't be trusted not to modify the offset of the IMemory they receive. 362 // If a thread does not have such a heap, this method returns 0. 363 virtual sp<MemoryDealer> readOnlyHeap() const { return 0; } 364 365 virtual sp<IMemory> pipeMemory() const { return 0; } 366 367 void systemReady(); 368 369 mutable Mutex mLock; 370 371protected: 372 373 // entry describing an effect being suspended in mSuspendedSessions keyed vector 374 class SuspendedSessionDesc : public RefBase { 375 public: 376 SuspendedSessionDesc() : mRefCount(0) {} 377 378 int mRefCount; // number of active suspend requests 379 effect_uuid_t mType; // effect type UUID 380 }; 381 382 void acquireWakeLock(int uid = -1); 383 void acquireWakeLock_l(int uid = -1); 384 void releaseWakeLock(); 385 void releaseWakeLock_l(); 386 void updateWakeLockUids(const SortedVector<int> &uids); 387 void updateWakeLockUids_l(const SortedVector<int> &uids); 388 void getPowerManager_l(); 389 void setEffectSuspended_l(const effect_uuid_t *type, 390 bool suspend, 391 int sessionId); 392 // updated mSuspendedSessions when an effect suspended or restored 393 void updateSuspendedSessions_l(const effect_uuid_t *type, 394 bool suspend, 395 int sessionId); 396 // check if some effects must be suspended when an effect chain is added 397 void checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain); 398 399 String16 getWakeLockTag(); 400 401 virtual void preExit() { } 402 403 friend class AudioFlinger; // for mEffectChains 404 405 const type_t mType; 406 407 // Used by parameters, config events, addTrack_l, exit 408 Condition mWaitWorkCV; 409 410 const sp<AudioFlinger> mAudioFlinger; 411 412 // updated by PlaybackThread::readOutputParameters_l() or 413 // RecordThread::readInputParameters_l() 414 uint32_t mSampleRate; 415 size_t mFrameCount; // output HAL, direct output, record 416 audio_channel_mask_t mChannelMask; 417 uint32_t mChannelCount; 418 size_t mFrameSize; 419 // not HAL frame size, this is for output sink (to pipe to fast mixer) 420 audio_format_t mFormat; // Source format for Recording and 421 // Sink format for Playback. 422 // Sink format may be different than 423 // HAL format if Fastmixer is used. 424 audio_format_t mHALFormat; 425 size_t mBufferSize; // HAL buffer size for read() or write() 426 427 Vector< sp<ConfigEvent> > mConfigEvents; 428 Vector< sp<ConfigEvent> > mPendingConfigEvents; // events awaiting system ready 429 430 // These fields are written and read by thread itself without lock or barrier, 431 // and read by other threads without lock or barrier via standby(), outDevice() 432 // and inDevice(). 433 // Because of the absence of a lock or barrier, any other thread that reads 434 // these fields must use the information in isolation, or be prepared to deal 435 // with possibility that it might be inconsistent with other information. 436 bool mStandby; // Whether thread is currently in standby. 437 audio_devices_t mOutDevice; // output device 438 audio_devices_t mInDevice; // input device 439 struct audio_patch mPatch; 440 audio_source_t mAudioSource; 441 442 const audio_io_handle_t mId; 443 Vector< sp<EffectChain> > mEffectChains; 444 445 static const int kThreadNameLength = 16; // prctl(PR_SET_NAME) limit 446 char mThreadName[kThreadNameLength]; // guaranteed NUL-terminated 447 sp<IPowerManager> mPowerManager; 448 sp<IBinder> mWakeLockToken; 449 const sp<PMDeathRecipient> mDeathRecipient; 450 // list of suspended effects per session and per type. The first vector is 451 // keyed by session ID, the second by type UUID timeLow field 452 KeyedVector< int, KeyedVector< int, sp<SuspendedSessionDesc> > > 453 mSuspendedSessions; 454 static const size_t kLogSize = 4 * 1024; 455 sp<NBLog::Writer> mNBLogWriter; 456 bool mSystemReady; 457}; 458 459// --- PlaybackThread --- 460class PlaybackThread : public ThreadBase { 461public: 462 463#include "PlaybackTracks.h" 464 465 enum mixer_state { 466 MIXER_IDLE, // no active tracks 467 MIXER_TRACKS_ENABLED, // at least one active track, but no track has any data ready 468 MIXER_TRACKS_READY, // at least one active track, and at least one track has data 469 MIXER_DRAIN_TRACK, // drain currently playing track 470 MIXER_DRAIN_ALL, // fully drain the hardware 471 // standby mode does not have an enum value 472 // suspend by audio policy manager is orthogonal to mixer state 473 }; 474 475 // retry count before removing active track in case of underrun on offloaded thread: 476 // we need to make sure that AudioTrack client has enough time to send large buffers 477//FIXME may be more appropriate if expressed in time units. Need to revise how underrun is handled 478 // for offloaded tracks 479 static const int8_t kMaxTrackRetriesOffload = 20; 480 481 PlaybackThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 482 audio_io_handle_t id, audio_devices_t device, type_t type, bool systemReady); 483 virtual ~PlaybackThread(); 484 485 void dump(int fd, const Vector<String16>& args); 486 487 // Thread virtuals 488 virtual bool threadLoop(); 489 490 // RefBase 491 virtual void onFirstRef(); 492 493protected: 494 // Code snippets that were lifted up out of threadLoop() 495 virtual void threadLoop_mix() = 0; 496 virtual void threadLoop_sleepTime() = 0; 497 virtual ssize_t threadLoop_write(); 498 virtual void threadLoop_drain(); 499 virtual void threadLoop_standby(); 500 virtual void threadLoop_exit(); 501 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 502 503 // prepareTracks_l reads and writes mActiveTracks, and returns 504 // the pending set of tracks to remove via Vector 'tracksToRemove'. The caller 505 // is responsible for clearing or destroying this Vector later on, when it 506 // is safe to do so. That will drop the final ref count and destroy the tracks. 507 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove) = 0; 508 void removeTracks_l(const Vector< sp<Track> >& tracksToRemove); 509 510 void writeCallback(); 511 void resetWriteBlocked(uint32_t sequence); 512 void drainCallback(); 513 void resetDraining(uint32_t sequence); 514 515 static int asyncCallback(stream_callback_event_t event, void *param, void *cookie); 516 517 virtual bool waitingAsyncCallback(); 518 virtual bool waitingAsyncCallback_l(); 519 virtual bool shouldStandby_l(); 520 virtual void onAddNewTrack_l(); 521 522 // ThreadBase virtuals 523 virtual void preExit(); 524 525public: 526 527 virtual status_t initCheck() const { return (mOutput == NULL) ? NO_INIT : NO_ERROR; } 528 529 // return estimated latency in milliseconds, as reported by HAL 530 uint32_t latency() const; 531 // same, but lock must already be held 532 uint32_t latency_l() const; 533 534 void setMasterVolume(float value); 535 void setMasterMute(bool muted); 536 537 void setStreamVolume(audio_stream_type_t stream, float value); 538 void setStreamMute(audio_stream_type_t stream, bool muted); 539 540 float streamVolume(audio_stream_type_t stream) const; 541 542 sp<Track> createTrack_l( 543 const sp<AudioFlinger::Client>& client, 544 audio_stream_type_t streamType, 545 uint32_t sampleRate, 546 audio_format_t format, 547 audio_channel_mask_t channelMask, 548 size_t *pFrameCount, 549 const sp<IMemory>& sharedBuffer, 550 int sessionId, 551 IAudioFlinger::track_flags_t *flags, 552 pid_t tid, 553 int uid, 554 status_t *status /*non-NULL*/); 555 556 AudioStreamOut* getOutput() const; 557 AudioStreamOut* clearOutput(); 558 virtual audio_stream_t* stream() const; 559 560 // a very large number of suspend() will eventually wraparound, but unlikely 561 void suspend() { (void) android_atomic_inc(&mSuspended); } 562 void restore() 563 { 564 // if restore() is done without suspend(), get back into 565 // range so that the next suspend() will operate correctly 566 if (android_atomic_dec(&mSuspended) <= 0) { 567 android_atomic_release_store(0, &mSuspended); 568 } 569 } 570 bool isSuspended() const 571 { return android_atomic_acquire_load(&mSuspended) > 0; } 572 573 virtual String8 getParameters(const String8& keys); 574 virtual void ioConfigChanged(audio_io_config_event event); 575 status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames); 576 // FIXME rename mixBuffer() to sinkBuffer() and remove int16_t* dependency. 577 // Consider also removing and passing an explicit mMainBuffer initialization 578 // parameter to AF::PlaybackThread::Track::Track(). 579 int16_t *mixBuffer() const { 580 return reinterpret_cast<int16_t *>(mSinkBuffer); }; 581 582 virtual void detachAuxEffect_l(int effectId); 583 status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track> track, 584 int EffectId); 585 status_t attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track> track, 586 int EffectId); 587 588 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 589 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 590 virtual uint32_t hasAudioSession(int sessionId) const; 591 virtual uint32_t getStrategyForSession_l(int sessionId); 592 593 594 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 595 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 596 597 // called with AudioFlinger lock held 598 void invalidateTracks(audio_stream_type_t streamType); 599 600 virtual size_t frameCount() const { return mNormalFrameCount; } 601 602 // Return's the HAL's frame count i.e. fast mixer buffer size. 603 size_t frameCountHAL() const { return mFrameCount; } 604 605 status_t getTimestamp_l(AudioTimestamp& timestamp); 606 607 void addPatchTrack(const sp<PatchTrack>& track); 608 void deletePatchTrack(const sp<PatchTrack>& track); 609 610 virtual void getAudioPortConfig(struct audio_port_config *config); 611 612protected: 613 // updated by readOutputParameters_l() 614 size_t mNormalFrameCount; // normal mixer and effects 615 616 void* mSinkBuffer; // frame size aligned sink buffer 617 618 // TODO: 619 // Rearrange the buffer info into a struct/class with 620 // clear, copy, construction, destruction methods. 621 // 622 // mSinkBuffer also has associated with it: 623 // 624 // mSinkBufferSize: Sink Buffer Size 625 // mFormat: Sink Buffer Format 626 627 // Mixer Buffer (mMixerBuffer*) 628 // 629 // In the case of floating point or multichannel data, which is not in the 630 // sink format, it is required to accumulate in a higher precision or greater channel count 631 // buffer before downmixing or data conversion to the sink buffer. 632 633 // Set to "true" to enable the Mixer Buffer otherwise mixer output goes to sink buffer. 634 bool mMixerBufferEnabled; 635 636 // Storage, 32 byte aligned (may make this alignment a requirement later). 637 // Due to constraints on mNormalFrameCount, the buffer size is a multiple of 16 frames. 638 void* mMixerBuffer; 639 640 // Size of mMixerBuffer in bytes: mNormalFrameCount * #channels * sampsize. 641 size_t mMixerBufferSize; 642 643 // The audio format of mMixerBuffer. Set to AUDIO_FORMAT_PCM_(FLOAT|16_BIT) only. 644 audio_format_t mMixerBufferFormat; 645 646 // An internal flag set to true by MixerThread::prepareTracks_l() 647 // when mMixerBuffer contains valid data after mixing. 648 bool mMixerBufferValid; 649 650 // Effects Buffer (mEffectsBuffer*) 651 // 652 // In the case of effects data, which is not in the sink format, 653 // it is required to accumulate in a different buffer before data conversion 654 // to the sink buffer. 655 656 // Set to "true" to enable the Effects Buffer otherwise effects output goes to sink buffer. 657 bool mEffectBufferEnabled; 658 659 // Storage, 32 byte aligned (may make this alignment a requirement later). 660 // Due to constraints on mNormalFrameCount, the buffer size is a multiple of 16 frames. 661 void* mEffectBuffer; 662 663 // Size of mEffectsBuffer in bytes: mNormalFrameCount * #channels * sampsize. 664 size_t mEffectBufferSize; 665 666 // The audio format of mEffectsBuffer. Set to AUDIO_FORMAT_PCM_16_BIT only. 667 audio_format_t mEffectBufferFormat; 668 669 // An internal flag set to true by MixerThread::prepareTracks_l() 670 // when mEffectsBuffer contains valid data after mixing. 671 // 672 // When this is set, all mixer data is routed into the effects buffer 673 // for any processing (including output processing). 674 bool mEffectBufferValid; 675 676 // suspend count, > 0 means suspended. While suspended, the thread continues to pull from 677 // tracks and mix, but doesn't write to HAL. A2DP and SCO HAL implementations can't handle 678 // concurrent use of both of them, so Audio Policy Service suspends one of the threads to 679 // workaround that restriction. 680 // 'volatile' means accessed via atomic operations and no lock. 681 volatile int32_t mSuspended; 682 683 // FIXME overflows every 6+ hours at 44.1 kHz stereo 16-bit samples 684 // mFramesWritten would be better, or 64-bit even better 685 size_t mBytesWritten; 686private: 687 // mMasterMute is in both PlaybackThread and in AudioFlinger. When a 688 // PlaybackThread needs to find out if master-muted, it checks it's local 689 // copy rather than the one in AudioFlinger. This optimization saves a lock. 690 bool mMasterMute; 691 void setMasterMute_l(bool muted) { mMasterMute = muted; } 692protected: 693 SortedVector< wp<Track> > mActiveTracks; // FIXME check if this could be sp<> 694 SortedVector<int> mWakeLockUids; 695 int mActiveTracksGeneration; 696 wp<Track> mLatestActiveTrack; // latest track added to mActiveTracks 697 698 // Allocate a track name for a given channel mask. 699 // Returns name >= 0 if successful, -1 on failure. 700 virtual int getTrackName_l(audio_channel_mask_t channelMask, 701 audio_format_t format, int sessionId) = 0; 702 virtual void deleteTrackName_l(int name) = 0; 703 704 // Time to sleep between cycles when: 705 virtual uint32_t activeSleepTimeUs() const; // mixer state MIXER_TRACKS_ENABLED 706 virtual uint32_t idleSleepTimeUs() const = 0; // mixer state MIXER_IDLE 707 virtual uint32_t suspendSleepTimeUs() const = 0; // audio policy manager suspended us 708 // No sleep when mixer state == MIXER_TRACKS_READY; relies on audio HAL stream->write() 709 // No sleep in standby mode; waits on a condition 710 711 // Code snippets that are temporarily lifted up out of threadLoop() until the merge 712 void checkSilentMode_l(); 713 714 // Non-trivial for DUPLICATING only 715 virtual void saveOutputTracks() { } 716 virtual void clearOutputTracks() { } 717 718 // Cache various calculated values, at threadLoop() entry and after a parameter change 719 virtual void cacheParameters_l(); 720 721 virtual uint32_t correctLatency_l(uint32_t latency) const; 722 723 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 724 audio_patch_handle_t *handle); 725 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 726 727 bool usesHwAvSync() const { return (mType == DIRECT) && (mOutput != NULL) 728 && mHwSupportsPause 729 && (mOutput->flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC); } 730 731private: 732 733 friend class AudioFlinger; // for numerous 734 735 PlaybackThread& operator = (const PlaybackThread&); 736 737 status_t addTrack_l(const sp<Track>& track); 738 bool destroyTrack_l(const sp<Track>& track); 739 void removeTrack_l(const sp<Track>& track); 740 void broadcast_l(); 741 742 void readOutputParameters_l(); 743 744 virtual void dumpInternals(int fd, const Vector<String16>& args); 745 void dumpTracks(int fd, const Vector<String16>& args); 746 747 SortedVector< sp<Track> > mTracks; 748 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 749 AudioStreamOut *mOutput; 750 751 float mMasterVolume; 752 nsecs_t mLastWriteTime; 753 int mNumWrites; 754 int mNumDelayedWrites; 755 bool mInWrite; 756 757 // FIXME rename these former local variables of threadLoop to standard "m" names 758 nsecs_t standbyTime; 759 size_t mSinkBufferSize; 760 761 // cached copies of activeSleepTimeUs() and idleSleepTimeUs() made by cacheParameters_l() 762 uint32_t activeSleepTime; 763 uint32_t idleSleepTime; 764 765 uint32_t sleepTime; 766 767 // mixer status returned by prepareTracks_l() 768 mixer_state mMixerStatus; // current cycle 769 // previous cycle when in prepareTracks_l() 770 mixer_state mMixerStatusIgnoringFastTracks; 771 // FIXME or a separate ready state per track 772 773 // FIXME move these declarations into the specific sub-class that needs them 774 // MIXER only 775 uint32_t sleepTimeShift; 776 777 // same as AudioFlinger::mStandbyTimeInNsecs except for DIRECT which uses a shorter value 778 nsecs_t standbyDelay; 779 780 // MIXER only 781 nsecs_t maxPeriod; 782 783 // DUPLICATING only 784 uint32_t writeFrames; 785 786 size_t mBytesRemaining; 787 size_t mCurrentWriteLength; 788 bool mUseAsyncWrite; 789 // mWriteAckSequence contains current write sequence on bits 31-1. The write sequence is 790 // incremented each time a write(), a flush() or a standby() occurs. 791 // Bit 0 is set when a write blocks and indicates a callback is expected. 792 // Bit 0 is reset by the async callback thread calling resetWriteBlocked(). Out of sequence 793 // callbacks are ignored. 794 uint32_t mWriteAckSequence; 795 // mDrainSequence contains current drain sequence on bits 31-1. The drain sequence is 796 // incremented each time a drain is requested or a flush() or standby() occurs. 797 // Bit 0 is set when the drain() command is called at the HAL and indicates a callback is 798 // expected. 799 // Bit 0 is reset by the async callback thread calling resetDraining(). Out of sequence 800 // callbacks are ignored. 801 uint32_t mDrainSequence; 802 // A condition that must be evaluated by prepareTrack_l() has changed and we must not wait 803 // for async write callback in the thread loop before evaluating it 804 bool mSignalPending; 805 sp<AsyncCallbackThread> mCallbackThread; 806 807private: 808 // The HAL output sink is treated as non-blocking, but current implementation is blocking 809 sp<NBAIO_Sink> mOutputSink; 810 // If a fast mixer is present, the blocking pipe sink, otherwise clear 811 sp<NBAIO_Sink> mPipeSink; 812 // The current sink for the normal mixer to write it's (sub)mix, mOutputSink or mPipeSink 813 sp<NBAIO_Sink> mNormalSink; 814#ifdef TEE_SINK 815 // For dumpsys 816 sp<NBAIO_Sink> mTeeSink; 817 sp<NBAIO_Source> mTeeSource; 818#endif 819 uint32_t mScreenState; // cached copy of gScreenState 820 static const size_t kFastMixerLogSize = 4 * 1024; 821 sp<NBLog::Writer> mFastMixerNBLogWriter; 822public: 823 virtual bool hasFastMixer() const = 0; 824 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex __unused) const 825 { FastTrackUnderruns dummy; return dummy; } 826 827protected: 828 // accessed by both binder threads and within threadLoop(), lock on mutex needed 829 unsigned mFastTrackAvailMask; // bit i set if fast track [i] is available 830 bool mHwSupportsPause; 831 bool mHwPaused; 832 bool mFlushPending; 833private: 834 // timestamp latch: 835 // D input is written by threadLoop_write while mutex is unlocked, and read while locked 836 // Q output is written while locked, and read while locked 837 struct { 838 AudioTimestamp mTimestamp; 839 uint32_t mUnpresentedFrames; 840 KeyedVector<Track *, uint32_t> mFramesReleased; 841 } mLatchD, mLatchQ; 842 bool mLatchDValid; // true means mLatchD is valid 843 // (except for mFramesReleased which is filled in later), 844 // and clock it into latch at next opportunity 845 bool mLatchQValid; // true means mLatchQ is valid 846}; 847 848class MixerThread : public PlaybackThread { 849public: 850 MixerThread(const sp<AudioFlinger>& audioFlinger, 851 AudioStreamOut* output, 852 audio_io_handle_t id, 853 audio_devices_t device, 854 bool systemReady, 855 type_t type = MIXER); 856 virtual ~MixerThread(); 857 858 // Thread virtuals 859 860 virtual bool checkForNewParameter_l(const String8& keyValuePair, 861 status_t& status); 862 virtual void dumpInternals(int fd, const Vector<String16>& args); 863 864protected: 865 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 866 virtual int getTrackName_l(audio_channel_mask_t channelMask, 867 audio_format_t format, int sessionId); 868 virtual void deleteTrackName_l(int name); 869 virtual uint32_t idleSleepTimeUs() const; 870 virtual uint32_t suspendSleepTimeUs() const; 871 virtual void cacheParameters_l(); 872 873 // threadLoop snippets 874 virtual ssize_t threadLoop_write(); 875 virtual void threadLoop_standby(); 876 virtual void threadLoop_mix(); 877 virtual void threadLoop_sleepTime(); 878 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 879 virtual uint32_t correctLatency_l(uint32_t latency) const; 880 881 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 882 audio_patch_handle_t *handle); 883 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 884 885 AudioMixer* mAudioMixer; // normal mixer 886private: 887 // one-time initialization, no locks required 888 sp<FastMixer> mFastMixer; // non-0 if there is also a fast mixer 889 sp<AudioWatchdog> mAudioWatchdog; // non-0 if there is an audio watchdog thread 890 891 // contents are not guaranteed to be consistent, no locks required 892 FastMixerDumpState mFastMixerDumpState; 893#ifdef STATE_QUEUE_DUMP 894 StateQueueObserverDump mStateQueueObserverDump; 895 StateQueueMutatorDump mStateQueueMutatorDump; 896#endif 897 AudioWatchdogDump mAudioWatchdogDump; 898 899 // accessible only within the threadLoop(), no locks required 900 // mFastMixer->sq() // for mutating and pushing state 901 int32_t mFastMixerFutex; // for cold idle 902 903public: 904 virtual bool hasFastMixer() const { return mFastMixer != 0; } 905 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const { 906 ALOG_ASSERT(fastIndex < FastMixerState::kMaxFastTracks); 907 return mFastMixerDumpState.mTracks[fastIndex].mUnderruns; 908 } 909 910}; 911 912class DirectOutputThread : public PlaybackThread { 913public: 914 915 DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 916 audio_io_handle_t id, audio_devices_t device, bool systemReady); 917 virtual ~DirectOutputThread(); 918 919 // Thread virtuals 920 921 virtual bool checkForNewParameter_l(const String8& keyValuePair, 922 status_t& status); 923 virtual void flushHw_l(); 924 925protected: 926 virtual int getTrackName_l(audio_channel_mask_t channelMask, 927 audio_format_t format, int sessionId); 928 virtual void deleteTrackName_l(int name); 929 virtual uint32_t activeSleepTimeUs() const; 930 virtual uint32_t idleSleepTimeUs() const; 931 virtual uint32_t suspendSleepTimeUs() const; 932 virtual void cacheParameters_l(); 933 934 // threadLoop snippets 935 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 936 virtual void threadLoop_mix(); 937 virtual void threadLoop_sleepTime(); 938 virtual void threadLoop_exit(); 939 virtual bool shouldStandby_l(); 940 941 // volumes last sent to audio HAL with stream->set_volume() 942 float mLeftVolFloat; 943 float mRightVolFloat; 944 945 DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 946 audio_io_handle_t id, uint32_t device, ThreadBase::type_t type, 947 bool systemReady); 948 void processVolume_l(Track *track, bool lastTrack); 949 950 // prepareTracks_l() tells threadLoop_mix() the name of the single active track 951 sp<Track> mActiveTrack; 952public: 953 virtual bool hasFastMixer() const { return false; } 954}; 955 956class OffloadThread : public DirectOutputThread { 957public: 958 959 OffloadThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 960 audio_io_handle_t id, uint32_t device, bool systemReady); 961 virtual ~OffloadThread() {}; 962 virtual void flushHw_l(); 963 964protected: 965 // threadLoop snippets 966 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 967 virtual void threadLoop_exit(); 968 969 virtual bool waitingAsyncCallback(); 970 virtual bool waitingAsyncCallback_l(); 971 virtual void onAddNewTrack_l(); 972 973private: 974 size_t mPausedWriteLength; // length in bytes of write interrupted by pause 975 size_t mPausedBytesRemaining; // bytes still waiting in mixbuffer after resume 976 wp<Track> mPreviousTrack; // used to detect track switch 977}; 978 979class AsyncCallbackThread : public Thread { 980public: 981 982 AsyncCallbackThread(const wp<PlaybackThread>& playbackThread); 983 984 virtual ~AsyncCallbackThread(); 985 986 // Thread virtuals 987 virtual bool threadLoop(); 988 989 // RefBase 990 virtual void onFirstRef(); 991 992 void exit(); 993 void setWriteBlocked(uint32_t sequence); 994 void resetWriteBlocked(); 995 void setDraining(uint32_t sequence); 996 void resetDraining(); 997 998private: 999 const wp<PlaybackThread> mPlaybackThread; 1000 // mWriteAckSequence corresponds to the last write sequence passed by the offload thread via 1001 // setWriteBlocked(). The sequence is shifted one bit to the left and the lsb is used 1002 // to indicate that the callback has been received via resetWriteBlocked() 1003 uint32_t mWriteAckSequence; 1004 // mDrainSequence corresponds to the last drain sequence passed by the offload thread via 1005 // setDraining(). The sequence is shifted one bit to the left and the lsb is used 1006 // to indicate that the callback has been received via resetDraining() 1007 uint32_t mDrainSequence; 1008 Condition mWaitWorkCV; 1009 Mutex mLock; 1010}; 1011 1012class DuplicatingThread : public MixerThread { 1013public: 1014 DuplicatingThread(const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread, 1015 audio_io_handle_t id, bool systemReady); 1016 virtual ~DuplicatingThread(); 1017 1018 // Thread virtuals 1019 void addOutputTrack(MixerThread* thread); 1020 void removeOutputTrack(MixerThread* thread); 1021 uint32_t waitTimeMs() const { return mWaitTimeMs; } 1022protected: 1023 virtual uint32_t activeSleepTimeUs() const; 1024 1025private: 1026 bool outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks); 1027protected: 1028 // threadLoop snippets 1029 virtual void threadLoop_mix(); 1030 virtual void threadLoop_sleepTime(); 1031 virtual ssize_t threadLoop_write(); 1032 virtual void threadLoop_standby(); 1033 virtual void cacheParameters_l(); 1034 1035private: 1036 // called from threadLoop, addOutputTrack, removeOutputTrack 1037 virtual void updateWaitTime_l(); 1038protected: 1039 virtual void saveOutputTracks(); 1040 virtual void clearOutputTracks(); 1041private: 1042 1043 uint32_t mWaitTimeMs; 1044 SortedVector < sp<OutputTrack> > outputTracks; 1045 SortedVector < sp<OutputTrack> > mOutputTracks; 1046public: 1047 virtual bool hasFastMixer() const { return false; } 1048}; 1049 1050 1051// record thread 1052class RecordThread : public ThreadBase 1053{ 1054public: 1055 1056 class RecordTrack; 1057 1058 /* The ResamplerBufferProvider is used to retrieve recorded input data from the 1059 * RecordThread. It maintains local state on the relative position of the read 1060 * position of the RecordTrack compared with the RecordThread. 1061 */ 1062 class ResamplerBufferProvider : public AudioBufferProvider 1063 { 1064 public: 1065 ResamplerBufferProvider(RecordTrack* recordTrack) : 1066 mRecordTrack(recordTrack), 1067 mRsmpInUnrel(0), mRsmpInFront(0) { } 1068 virtual ~ResamplerBufferProvider() { } 1069 1070 // called to set the ResamplerBufferProvider to head of the RecordThread data buffer, 1071 // skipping any previous data read from the hal. 1072 virtual void reset(); 1073 1074 /* Synchronizes RecordTrack position with the RecordThread. 1075 * Calculates available frames and handle overruns if the RecordThread 1076 * has advanced faster than the ResamplerBufferProvider has retrieved data. 1077 * TODO: why not do this for every getNextBuffer? 1078 * 1079 * Parameters 1080 * framesAvailable: pointer to optional output size_t to store record track 1081 * frames available. 1082 * hasOverrun: pointer to optional boolean, returns true if track has overrun. 1083 */ 1084 1085 virtual void sync(size_t *framesAvailable = NULL, bool *hasOverrun = NULL); 1086 1087 // AudioBufferProvider interface 1088 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts); 1089 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 1090 private: 1091 RecordTrack * const mRecordTrack; 1092 size_t mRsmpInUnrel; // unreleased frames remaining from 1093 // most recent getNextBuffer 1094 // for debug only 1095 int32_t mRsmpInFront; // next available frame 1096 // rolling counter that is never cleared 1097 }; 1098 1099 /* The RecordBufferConverter is used for format, channel, and sample rate 1100 * conversion for a RecordTrack. 1101 * 1102 * TODO: Self contained, so move to a separate file later. 1103 * 1104 * RecordBufferConverter uses the convert() method rather than exposing a 1105 * buffer provider interface; this is to save a memory copy. 1106 */ 1107 class RecordBufferConverter 1108 { 1109 public: 1110 RecordBufferConverter( 1111 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 1112 uint32_t srcSampleRate, 1113 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 1114 uint32_t dstSampleRate); 1115 1116 ~RecordBufferConverter(); 1117 1118 /* Converts input data from an AudioBufferProvider by format, channelMask, 1119 * and sampleRate to a destination buffer. 1120 * 1121 * Parameters 1122 * dst: buffer to place the converted data. 1123 * provider: buffer provider to obtain source data. 1124 * frames: number of frames to convert 1125 * 1126 * Returns the number of frames converted. 1127 */ 1128 size_t convert(void *dst, AudioBufferProvider *provider, size_t frames); 1129 1130 // returns NO_ERROR if constructor was successful 1131 status_t initCheck() const { 1132 // mSrcChannelMask set on successful updateParameters 1133 return mSrcChannelMask != AUDIO_CHANNEL_INVALID ? NO_ERROR : NO_INIT; 1134 } 1135 1136 // allows dynamic reconfigure of all parameters 1137 status_t updateParameters( 1138 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 1139 uint32_t srcSampleRate, 1140 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 1141 uint32_t dstSampleRate); 1142 1143 // called to reset resampler buffers on record track discontinuity 1144 void reset() { 1145 if (mResampler != NULL) { 1146 mResampler->reset(); 1147 } 1148 } 1149 1150 private: 1151 // format conversion when not using resampler 1152 void convertNoResampler(void *dst, const void *src, size_t frames); 1153 1154 // format conversion when using resampler; modifies src in-place 1155 void convertResampler(void *dst, /*not-a-const*/ void *src, size_t frames); 1156 1157 // user provided information 1158 audio_channel_mask_t mSrcChannelMask; 1159 audio_format_t mSrcFormat; 1160 uint32_t mSrcSampleRate; 1161 audio_channel_mask_t mDstChannelMask; 1162 audio_format_t mDstFormat; 1163 uint32_t mDstSampleRate; 1164 1165 // derived information 1166 uint32_t mSrcChannelCount; 1167 uint32_t mDstChannelCount; 1168 size_t mDstFrameSize; 1169 1170 // format conversion buffer 1171 void *mBuf; 1172 size_t mBufFrames; 1173 size_t mBufFrameSize; 1174 1175 // resampler info 1176 AudioResampler *mResampler; 1177 1178 bool mIsLegacyDownmix; // legacy stereo to mono conversion needed 1179 bool mIsLegacyUpmix; // legacy mono to stereo conversion needed 1180 bool mRequiresFloat; // data processing requires float (e.g. resampler) 1181 PassthruBufferProvider *mInputConverterProvider; // converts input to float 1182 int8_t mIdxAry[sizeof(uint32_t) * 8]; // used for channel mask conversion 1183 }; 1184 1185#include "RecordTracks.h" 1186 1187 RecordThread(const sp<AudioFlinger>& audioFlinger, 1188 AudioStreamIn *input, 1189 audio_io_handle_t id, 1190 audio_devices_t outDevice, 1191 audio_devices_t inDevice, 1192 bool systemReady 1193#ifdef TEE_SINK 1194 , const sp<NBAIO_Sink>& teeSink 1195#endif 1196 ); 1197 virtual ~RecordThread(); 1198 1199 // no addTrack_l ? 1200 void destroyTrack_l(const sp<RecordTrack>& track); 1201 void removeTrack_l(const sp<RecordTrack>& track); 1202 1203 void dumpInternals(int fd, const Vector<String16>& args); 1204 void dumpTracks(int fd, const Vector<String16>& args); 1205 1206 // Thread virtuals 1207 virtual bool threadLoop(); 1208 1209 // RefBase 1210 virtual void onFirstRef(); 1211 1212 virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; } 1213 1214 virtual sp<MemoryDealer> readOnlyHeap() const { return mReadOnlyHeap; } 1215 1216 virtual sp<IMemory> pipeMemory() const { return mPipeMemory; } 1217 1218 sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l( 1219 const sp<AudioFlinger::Client>& client, 1220 uint32_t sampleRate, 1221 audio_format_t format, 1222 audio_channel_mask_t channelMask, 1223 size_t *pFrameCount, 1224 int sessionId, 1225 size_t *notificationFrames, 1226 int uid, 1227 IAudioFlinger::track_flags_t *flags, 1228 pid_t tid, 1229 status_t *status /*non-NULL*/); 1230 1231 status_t start(RecordTrack* recordTrack, 1232 AudioSystem::sync_event_t event, 1233 int triggerSession); 1234 1235 // ask the thread to stop the specified track, and 1236 // return true if the caller should then do it's part of the stopping process 1237 bool stop(RecordTrack* recordTrack); 1238 1239 void dump(int fd, const Vector<String16>& args); 1240 AudioStreamIn* clearInput(); 1241 virtual audio_stream_t* stream() const; 1242 1243 1244 virtual bool checkForNewParameter_l(const String8& keyValuePair, 1245 status_t& status); 1246 virtual void cacheParameters_l() {} 1247 virtual String8 getParameters(const String8& keys); 1248 virtual void ioConfigChanged(audio_io_config_event event); 1249 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 1250 audio_patch_handle_t *handle); 1251 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 1252 1253 void addPatchRecord(const sp<PatchRecord>& record); 1254 void deletePatchRecord(const sp<PatchRecord>& record); 1255 1256 void readInputParameters_l(); 1257 virtual uint32_t getInputFramesLost(); 1258 1259 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1260 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1261 virtual uint32_t hasAudioSession(int sessionId) const; 1262 1263 // Return the set of unique session IDs across all tracks. 1264 // The keys are the session IDs, and the associated values are meaningless. 1265 // FIXME replace by Set [and implement Bag/Multiset for other uses]. 1266 KeyedVector<int, bool> sessionIds() const; 1267 1268 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 1269 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 1270 1271 static void syncStartEventCallback(const wp<SyncEvent>& event); 1272 1273 virtual size_t frameCount() const { return mFrameCount; } 1274 bool hasFastCapture() const { return mFastCapture != 0; } 1275 virtual void getAudioPortConfig(struct audio_port_config *config); 1276 1277private: 1278 // Enter standby if not already in standby, and set mStandby flag 1279 void standbyIfNotAlreadyInStandby(); 1280 1281 // Call the HAL standby method unconditionally, and don't change mStandby flag 1282 void inputStandBy(); 1283 1284 AudioStreamIn *mInput; 1285 SortedVector < sp<RecordTrack> > mTracks; 1286 // mActiveTracks has dual roles: it indicates the current active track(s), and 1287 // is used together with mStartStopCond to indicate start()/stop() progress 1288 SortedVector< sp<RecordTrack> > mActiveTracks; 1289 // generation counter for mActiveTracks 1290 int mActiveTracksGen; 1291 Condition mStartStopCond; 1292 1293 // resampler converts input at HAL Hz to output at AudioRecord client Hz 1294 void *mRsmpInBuffer; // 1295 size_t mRsmpInFrames; // size of resampler input in frames 1296 size_t mRsmpInFramesP2;// size rounded up to a power-of-2 1297 1298 // rolling index that is never cleared 1299 int32_t mRsmpInRear; // last filled frame + 1 1300 1301 // For dumpsys 1302 const sp<NBAIO_Sink> mTeeSink; 1303 1304 const sp<MemoryDealer> mReadOnlyHeap; 1305 1306 // one-time initialization, no locks required 1307 sp<FastCapture> mFastCapture; // non-0 if there is also 1308 // a fast capture 1309 1310 // FIXME audio watchdog thread 1311 1312 // contents are not guaranteed to be consistent, no locks required 1313 FastCaptureDumpState mFastCaptureDumpState; 1314#ifdef STATE_QUEUE_DUMP 1315 // FIXME StateQueue observer and mutator dump fields 1316#endif 1317 // FIXME audio watchdog dump 1318 1319 // accessible only within the threadLoop(), no locks required 1320 // mFastCapture->sq() // for mutating and pushing state 1321 int32_t mFastCaptureFutex; // for cold idle 1322 1323 // The HAL input source is treated as non-blocking, 1324 // but current implementation is blocking 1325 sp<NBAIO_Source> mInputSource; 1326 // The source for the normal capture thread to read from: mInputSource or mPipeSource 1327 sp<NBAIO_Source> mNormalSource; 1328 // If a fast capture is present, the non-blocking pipe sink written to by fast capture, 1329 // otherwise clear 1330 sp<NBAIO_Sink> mPipeSink; 1331 // If a fast capture is present, the non-blocking pipe source read by normal thread, 1332 // otherwise clear 1333 sp<NBAIO_Source> mPipeSource; 1334 // Depth of pipe from fast capture to normal thread and fast clients, always power of 2 1335 size_t mPipeFramesP2; 1336 // If a fast capture is present, the Pipe as IMemory, otherwise clear 1337 sp<IMemory> mPipeMemory; 1338 1339 static const size_t kFastCaptureLogSize = 4 * 1024; 1340 sp<NBLog::Writer> mFastCaptureNBLogWriter; 1341 1342 bool mFastTrackAvail; // true if fast track available 1343}; 1344