Threads.h revision 3f273d10817ddb2f792ae043de692efcdf1988ae
1/* 2** 3** Copyright 2012, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18#ifndef INCLUDING_FROM_AUDIOFLINGER_H 19 #error This header file should only be included from AudioFlinger.h 20#endif 21 22class ThreadBase : public Thread { 23public: 24 25#include "TrackBase.h" 26 27 enum type_t { 28 MIXER, // Thread class is MixerThread 29 DIRECT, // Thread class is DirectOutputThread 30 DUPLICATING, // Thread class is DuplicatingThread 31 RECORD, // Thread class is RecordThread 32 OFFLOAD // Thread class is OffloadThread 33 }; 34 35 static const char *threadTypeToString(type_t type); 36 37 ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 38 audio_devices_t outDevice, audio_devices_t inDevice, type_t type, 39 bool systemReady); 40 virtual ~ThreadBase(); 41 42 virtual status_t readyToRun(); 43 44 void dumpBase(int fd, const Vector<String16>& args); 45 void dumpEffectChains(int fd, const Vector<String16>& args); 46 47 void clearPowerManager(); 48 49 // base for record and playback 50 enum { 51 CFG_EVENT_IO, 52 CFG_EVENT_PRIO, 53 CFG_EVENT_SET_PARAMETER, 54 CFG_EVENT_CREATE_AUDIO_PATCH, 55 CFG_EVENT_RELEASE_AUDIO_PATCH, 56 }; 57 58 class ConfigEventData: public RefBase { 59 public: 60 virtual ~ConfigEventData() {} 61 62 virtual void dump(char *buffer, size_t size) = 0; 63 protected: 64 ConfigEventData() {} 65 }; 66 67 // Config event sequence by client if status needed (e.g binder thread calling setParameters()): 68 // 1. create SetParameterConfigEvent. This sets mWaitStatus in config event 69 // 2. Lock mLock 70 // 3. Call sendConfigEvent_l(): Append to mConfigEvents and mWaitWorkCV.signal 71 // 4. sendConfigEvent_l() reads status from event->mStatus; 72 // 5. sendConfigEvent_l() returns status 73 // 6. Unlock 74 // 75 // Parameter sequence by server: threadLoop calling processConfigEvents_l(): 76 // 1. Lock mLock 77 // 2. If there is an entry in mConfigEvents proceed ... 78 // 3. Read first entry in mConfigEvents 79 // 4. Remove first entry from mConfigEvents 80 // 5. Process 81 // 6. Set event->mStatus 82 // 7. event->mCond.signal 83 // 8. Unlock 84 85 class ConfigEvent: public RefBase { 86 public: 87 virtual ~ConfigEvent() {} 88 89 void dump(char *buffer, size_t size) { mData->dump(buffer, size); } 90 91 const int mType; // event type e.g. CFG_EVENT_IO 92 Mutex mLock; // mutex associated with mCond 93 Condition mCond; // condition for status return 94 status_t mStatus; // status communicated to sender 95 bool mWaitStatus; // true if sender is waiting for status 96 bool mRequiresSystemReady; // true if must wait for system ready to enter event queue 97 sp<ConfigEventData> mData; // event specific parameter data 98 99 protected: 100 ConfigEvent(int type, bool requiresSystemReady = false) : 101 mType(type), mStatus(NO_ERROR), mWaitStatus(false), 102 mRequiresSystemReady(requiresSystemReady), mData(NULL) {} 103 }; 104 105 class IoConfigEventData : public ConfigEventData { 106 public: 107 IoConfigEventData(audio_io_config_event event, pid_t pid) : 108 mEvent(event), mPid(pid) {} 109 110 virtual void dump(char *buffer, size_t size) { 111 snprintf(buffer, size, "IO event: event %d\n", mEvent); 112 } 113 114 const audio_io_config_event mEvent; 115 const pid_t mPid; 116 }; 117 118 class IoConfigEvent : public ConfigEvent { 119 public: 120 IoConfigEvent(audio_io_config_event event, pid_t pid) : 121 ConfigEvent(CFG_EVENT_IO) { 122 mData = new IoConfigEventData(event, pid); 123 } 124 virtual ~IoConfigEvent() {} 125 }; 126 127 class PrioConfigEventData : public ConfigEventData { 128 public: 129 PrioConfigEventData(pid_t pid, pid_t tid, int32_t prio) : 130 mPid(pid), mTid(tid), mPrio(prio) {} 131 132 virtual void dump(char *buffer, size_t size) { 133 snprintf(buffer, size, "Prio event: pid %d, tid %d, prio %d\n", mPid, mTid, mPrio); 134 } 135 136 const pid_t mPid; 137 const pid_t mTid; 138 const int32_t mPrio; 139 }; 140 141 class PrioConfigEvent : public ConfigEvent { 142 public: 143 PrioConfigEvent(pid_t pid, pid_t tid, int32_t prio) : 144 ConfigEvent(CFG_EVENT_PRIO, true) { 145 mData = new PrioConfigEventData(pid, tid, prio); 146 } 147 virtual ~PrioConfigEvent() {} 148 }; 149 150 class SetParameterConfigEventData : public ConfigEventData { 151 public: 152 SetParameterConfigEventData(String8 keyValuePairs) : 153 mKeyValuePairs(keyValuePairs) {} 154 155 virtual void dump(char *buffer, size_t size) { 156 snprintf(buffer, size, "KeyValue: %s\n", mKeyValuePairs.string()); 157 } 158 159 const String8 mKeyValuePairs; 160 }; 161 162 class SetParameterConfigEvent : public ConfigEvent { 163 public: 164 SetParameterConfigEvent(String8 keyValuePairs) : 165 ConfigEvent(CFG_EVENT_SET_PARAMETER) { 166 mData = new SetParameterConfigEventData(keyValuePairs); 167 mWaitStatus = true; 168 } 169 virtual ~SetParameterConfigEvent() {} 170 }; 171 172 class CreateAudioPatchConfigEventData : public ConfigEventData { 173 public: 174 CreateAudioPatchConfigEventData(const struct audio_patch patch, 175 audio_patch_handle_t handle) : 176 mPatch(patch), mHandle(handle) {} 177 178 virtual void dump(char *buffer, size_t size) { 179 snprintf(buffer, size, "Patch handle: %u\n", mHandle); 180 } 181 182 const struct audio_patch mPatch; 183 audio_patch_handle_t mHandle; 184 }; 185 186 class CreateAudioPatchConfigEvent : public ConfigEvent { 187 public: 188 CreateAudioPatchConfigEvent(const struct audio_patch patch, 189 audio_patch_handle_t handle) : 190 ConfigEvent(CFG_EVENT_CREATE_AUDIO_PATCH) { 191 mData = new CreateAudioPatchConfigEventData(patch, handle); 192 mWaitStatus = true; 193 } 194 virtual ~CreateAudioPatchConfigEvent() {} 195 }; 196 197 class ReleaseAudioPatchConfigEventData : public ConfigEventData { 198 public: 199 ReleaseAudioPatchConfigEventData(const audio_patch_handle_t handle) : 200 mHandle(handle) {} 201 202 virtual void dump(char *buffer, size_t size) { 203 snprintf(buffer, size, "Patch handle: %u\n", mHandle); 204 } 205 206 audio_patch_handle_t mHandle; 207 }; 208 209 class ReleaseAudioPatchConfigEvent : public ConfigEvent { 210 public: 211 ReleaseAudioPatchConfigEvent(const audio_patch_handle_t handle) : 212 ConfigEvent(CFG_EVENT_RELEASE_AUDIO_PATCH) { 213 mData = new ReleaseAudioPatchConfigEventData(handle); 214 mWaitStatus = true; 215 } 216 virtual ~ReleaseAudioPatchConfigEvent() {} 217 }; 218 219 class PMDeathRecipient : public IBinder::DeathRecipient { 220 public: 221 PMDeathRecipient(const wp<ThreadBase>& thread) : mThread(thread) {} 222 virtual ~PMDeathRecipient() {} 223 224 // IBinder::DeathRecipient 225 virtual void binderDied(const wp<IBinder>& who); 226 227 private: 228 PMDeathRecipient(const PMDeathRecipient&); 229 PMDeathRecipient& operator = (const PMDeathRecipient&); 230 231 wp<ThreadBase> mThread; 232 }; 233 234 virtual status_t initCheck() const = 0; 235 236 // static externally-visible 237 type_t type() const { return mType; } 238 bool isDuplicating() const { return (mType == DUPLICATING); } 239 240 audio_io_handle_t id() const { return mId;} 241 242 // dynamic externally-visible 243 uint32_t sampleRate() const { return mSampleRate; } 244 audio_channel_mask_t channelMask() const { return mChannelMask; } 245 audio_format_t format() const { return mHALFormat; } 246 uint32_t channelCount() const { return mChannelCount; } 247 // Called by AudioFlinger::frameCount(audio_io_handle_t output) and effects, 248 // and returns the [normal mix] buffer's frame count. 249 virtual size_t frameCount() const = 0; 250 size_t frameSize() const { return mFrameSize; } 251 252 // Should be "virtual status_t requestExitAndWait()" and override same 253 // method in Thread, but Thread::requestExitAndWait() is not yet virtual. 254 void exit(); 255 virtual bool checkForNewParameter_l(const String8& keyValuePair, 256 status_t& status) = 0; 257 virtual status_t setParameters(const String8& keyValuePairs); 258 virtual String8 getParameters(const String8& keys) = 0; 259 virtual void ioConfigChanged(audio_io_config_event event, pid_t pid = 0) = 0; 260 // sendConfigEvent_l() must be called with ThreadBase::mLock held 261 // Can temporarily release the lock if waiting for a reply from 262 // processConfigEvents_l(). 263 status_t sendConfigEvent_l(sp<ConfigEvent>& event); 264 void sendIoConfigEvent(audio_io_config_event event, pid_t pid = 0); 265 void sendIoConfigEvent_l(audio_io_config_event event, pid_t pid = 0); 266 void sendPrioConfigEvent(pid_t pid, pid_t tid, int32_t prio); 267 void sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio); 268 status_t sendSetParameterConfigEvent_l(const String8& keyValuePair); 269 status_t sendCreateAudioPatchConfigEvent(const struct audio_patch *patch, 270 audio_patch_handle_t *handle); 271 status_t sendReleaseAudioPatchConfigEvent(audio_patch_handle_t handle); 272 void processConfigEvents_l(); 273 virtual void cacheParameters_l() = 0; 274 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 275 audio_patch_handle_t *handle) = 0; 276 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle) = 0; 277 virtual void getAudioPortConfig(struct audio_port_config *config) = 0; 278 279 280 // see note at declaration of mStandby, mOutDevice and mInDevice 281 bool standby() const { return mStandby; } 282 audio_devices_t outDevice() const { return mOutDevice; } 283 audio_devices_t inDevice() const { return mInDevice; } 284 285 virtual audio_stream_t* stream() const = 0; 286 287 sp<EffectHandle> createEffect_l( 288 const sp<AudioFlinger::Client>& client, 289 const sp<IEffectClient>& effectClient, 290 int32_t priority, 291 int sessionId, 292 effect_descriptor_t *desc, 293 int *enabled, 294 status_t *status /*non-NULL*/); 295 296 // return values for hasAudioSession (bit field) 297 enum effect_state { 298 EFFECT_SESSION = 0x1, // the audio session corresponds to at least one 299 // effect 300 TRACK_SESSION = 0x2 // the audio session corresponds to at least one 301 // track 302 }; 303 304 // get effect chain corresponding to session Id. 305 sp<EffectChain> getEffectChain(int sessionId); 306 // same as getEffectChain() but must be called with ThreadBase mutex locked 307 sp<EffectChain> getEffectChain_l(int sessionId) const; 308 // add an effect chain to the chain list (mEffectChains) 309 virtual status_t addEffectChain_l(const sp<EffectChain>& chain) = 0; 310 // remove an effect chain from the chain list (mEffectChains) 311 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain) = 0; 312 // lock all effect chains Mutexes. Must be called before releasing the 313 // ThreadBase mutex before processing the mixer and effects. This guarantees the 314 // integrity of the chains during the process. 315 // Also sets the parameter 'effectChains' to current value of mEffectChains. 316 void lockEffectChains_l(Vector< sp<EffectChain> >& effectChains); 317 // unlock effect chains after process 318 void unlockEffectChains(const Vector< sp<EffectChain> >& effectChains); 319 // get a copy of mEffectChains vector 320 Vector< sp<EffectChain> > getEffectChains_l() const { return mEffectChains; }; 321 // set audio mode to all effect chains 322 void setMode(audio_mode_t mode); 323 // get effect module with corresponding ID on specified audio session 324 sp<AudioFlinger::EffectModule> getEffect(int sessionId, int effectId); 325 sp<AudioFlinger::EffectModule> getEffect_l(int sessionId, int effectId); 326 // add and effect module. Also creates the effect chain is none exists for 327 // the effects audio session 328 status_t addEffect_l(const sp< EffectModule>& effect); 329 // remove and effect module. Also removes the effect chain is this was the last 330 // effect 331 void removeEffect_l(const sp< EffectModule>& effect); 332 // detach all tracks connected to an auxiliary effect 333 virtual void detachAuxEffect_l(int effectId __unused) {} 334 // returns either EFFECT_SESSION if effects on this audio session exist in one 335 // chain, or TRACK_SESSION if tracks on this audio session exist, or both 336 virtual uint32_t hasAudioSession(int sessionId) const = 0; 337 // the value returned by default implementation is not important as the 338 // strategy is only meaningful for PlaybackThread which implements this method 339 virtual uint32_t getStrategyForSession_l(int sessionId __unused) { return 0; } 340 341 // suspend or restore effect according to the type of effect passed. a NULL 342 // type pointer means suspend all effects in the session 343 void setEffectSuspended(const effect_uuid_t *type, 344 bool suspend, 345 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 346 // check if some effects must be suspended/restored when an effect is enabled 347 // or disabled 348 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 349 bool enabled, 350 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 351 void checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 352 bool enabled, 353 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 354 355 virtual status_t setSyncEvent(const sp<SyncEvent>& event) = 0; 356 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const = 0; 357 358 // Return a reference to a per-thread heap which can be used to allocate IMemory 359 // objects that will be read-only to client processes, read/write to mediaserver, 360 // and shared by all client processes of the thread. 361 // The heap is per-thread rather than common across all threads, because 362 // clients can't be trusted not to modify the offset of the IMemory they receive. 363 // If a thread does not have such a heap, this method returns 0. 364 virtual sp<MemoryDealer> readOnlyHeap() const { return 0; } 365 366 virtual sp<IMemory> pipeMemory() const { return 0; } 367 368 void systemReady(); 369 370 mutable Mutex mLock; 371 372protected: 373 374 // entry describing an effect being suspended in mSuspendedSessions keyed vector 375 class SuspendedSessionDesc : public RefBase { 376 public: 377 SuspendedSessionDesc() : mRefCount(0) {} 378 379 int mRefCount; // number of active suspend requests 380 effect_uuid_t mType; // effect type UUID 381 }; 382 383 void acquireWakeLock(int uid = -1); 384 void acquireWakeLock_l(int uid = -1); 385 void releaseWakeLock(); 386 void releaseWakeLock_l(); 387 void updateWakeLockUids(const SortedVector<int> &uids); 388 void updateWakeLockUids_l(const SortedVector<int> &uids); 389 void getPowerManager_l(); 390 void setEffectSuspended_l(const effect_uuid_t *type, 391 bool suspend, 392 int sessionId); 393 // updated mSuspendedSessions when an effect suspended or restored 394 void updateSuspendedSessions_l(const effect_uuid_t *type, 395 bool suspend, 396 int sessionId); 397 // check if some effects must be suspended when an effect chain is added 398 void checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain); 399 400 String16 getWakeLockTag(); 401 402 virtual void preExit() { } 403 404 friend class AudioFlinger; // for mEffectChains 405 406 const type_t mType; 407 408 // Used by parameters, config events, addTrack_l, exit 409 Condition mWaitWorkCV; 410 411 const sp<AudioFlinger> mAudioFlinger; 412 413 // updated by PlaybackThread::readOutputParameters_l() or 414 // RecordThread::readInputParameters_l() 415 uint32_t mSampleRate; 416 size_t mFrameCount; // output HAL, direct output, record 417 audio_channel_mask_t mChannelMask; 418 uint32_t mChannelCount; 419 size_t mFrameSize; 420 // not HAL frame size, this is for output sink (to pipe to fast mixer) 421 audio_format_t mFormat; // Source format for Recording and 422 // Sink format for Playback. 423 // Sink format may be different than 424 // HAL format if Fastmixer is used. 425 audio_format_t mHALFormat; 426 size_t mBufferSize; // HAL buffer size for read() or write() 427 428 Vector< sp<ConfigEvent> > mConfigEvents; 429 Vector< sp<ConfigEvent> > mPendingConfigEvents; // events awaiting system ready 430 431 // These fields are written and read by thread itself without lock or barrier, 432 // and read by other threads without lock or barrier via standby(), outDevice() 433 // and inDevice(). 434 // Because of the absence of a lock or barrier, any other thread that reads 435 // these fields must use the information in isolation, or be prepared to deal 436 // with possibility that it might be inconsistent with other information. 437 bool mStandby; // Whether thread is currently in standby. 438 audio_devices_t mOutDevice; // output device 439 audio_devices_t mInDevice; // input device 440 audio_devices_t mPrevOutDevice; // previous output device 441 audio_devices_t mPrevInDevice; // previous input device 442 struct audio_patch mPatch; 443 audio_source_t mAudioSource; 444 445 const audio_io_handle_t mId; 446 Vector< sp<EffectChain> > mEffectChains; 447 448 static const int kThreadNameLength = 16; // prctl(PR_SET_NAME) limit 449 char mThreadName[kThreadNameLength]; // guaranteed NUL-terminated 450 sp<IPowerManager> mPowerManager; 451 sp<IBinder> mWakeLockToken; 452 const sp<PMDeathRecipient> mDeathRecipient; 453 // list of suspended effects per session and per type. The first vector is 454 // keyed by session ID, the second by type UUID timeLow field 455 KeyedVector< int, KeyedVector< int, sp<SuspendedSessionDesc> > > 456 mSuspendedSessions; 457 static const size_t kLogSize = 4 * 1024; 458 sp<NBLog::Writer> mNBLogWriter; 459 bool mSystemReady; 460 bool mNotifiedBatteryStart; 461}; 462 463// --- PlaybackThread --- 464class PlaybackThread : public ThreadBase { 465public: 466 467#include "PlaybackTracks.h" 468 469 enum mixer_state { 470 MIXER_IDLE, // no active tracks 471 MIXER_TRACKS_ENABLED, // at least one active track, but no track has any data ready 472 MIXER_TRACKS_READY, // at least one active track, and at least one track has data 473 MIXER_DRAIN_TRACK, // drain currently playing track 474 MIXER_DRAIN_ALL, // fully drain the hardware 475 // standby mode does not have an enum value 476 // suspend by audio policy manager is orthogonal to mixer state 477 }; 478 479 // retry count before removing active track in case of underrun on offloaded thread: 480 // we need to make sure that AudioTrack client has enough time to send large buffers 481//FIXME may be more appropriate if expressed in time units. Need to revise how underrun is handled 482 // for offloaded tracks 483 static const int8_t kMaxTrackRetriesOffload = 20; 484 485 PlaybackThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 486 audio_io_handle_t id, audio_devices_t device, type_t type, bool systemReady); 487 virtual ~PlaybackThread(); 488 489 void dump(int fd, const Vector<String16>& args); 490 491 // Thread virtuals 492 virtual bool threadLoop(); 493 494 // RefBase 495 virtual void onFirstRef(); 496 497protected: 498 // Code snippets that were lifted up out of threadLoop() 499 virtual void threadLoop_mix() = 0; 500 virtual void threadLoop_sleepTime() = 0; 501 virtual ssize_t threadLoop_write(); 502 virtual void threadLoop_drain(); 503 virtual void threadLoop_standby(); 504 virtual void threadLoop_exit(); 505 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 506 507 // prepareTracks_l reads and writes mActiveTracks, and returns 508 // the pending set of tracks to remove via Vector 'tracksToRemove'. The caller 509 // is responsible for clearing or destroying this Vector later on, when it 510 // is safe to do so. That will drop the final ref count and destroy the tracks. 511 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove) = 0; 512 void removeTracks_l(const Vector< sp<Track> >& tracksToRemove); 513 514 void writeCallback(); 515 void resetWriteBlocked(uint32_t sequence); 516 void drainCallback(); 517 void resetDraining(uint32_t sequence); 518 519 static int asyncCallback(stream_callback_event_t event, void *param, void *cookie); 520 521 virtual bool waitingAsyncCallback(); 522 virtual bool waitingAsyncCallback_l(); 523 virtual bool shouldStandby_l(); 524 virtual void onAddNewTrack_l(); 525 526 // ThreadBase virtuals 527 virtual void preExit(); 528 529public: 530 531 virtual status_t initCheck() const { return (mOutput == NULL) ? NO_INIT : NO_ERROR; } 532 533 // return estimated latency in milliseconds, as reported by HAL 534 uint32_t latency() const; 535 // same, but lock must already be held 536 uint32_t latency_l() const; 537 538 void setMasterVolume(float value); 539 void setMasterMute(bool muted); 540 541 void setStreamVolume(audio_stream_type_t stream, float value); 542 void setStreamMute(audio_stream_type_t stream, bool muted); 543 544 float streamVolume(audio_stream_type_t stream) const; 545 546 sp<Track> createTrack_l( 547 const sp<AudioFlinger::Client>& client, 548 audio_stream_type_t streamType, 549 uint32_t sampleRate, 550 audio_format_t format, 551 audio_channel_mask_t channelMask, 552 size_t *pFrameCount, 553 const sp<IMemory>& sharedBuffer, 554 int sessionId, 555 IAudioFlinger::track_flags_t *flags, 556 pid_t tid, 557 int uid, 558 status_t *status /*non-NULL*/); 559 560 AudioStreamOut* getOutput() const; 561 AudioStreamOut* clearOutput(); 562 virtual audio_stream_t* stream() const; 563 564 // a very large number of suspend() will eventually wraparound, but unlikely 565 void suspend() { (void) android_atomic_inc(&mSuspended); } 566 void restore() 567 { 568 // if restore() is done without suspend(), get back into 569 // range so that the next suspend() will operate correctly 570 if (android_atomic_dec(&mSuspended) <= 0) { 571 android_atomic_release_store(0, &mSuspended); 572 } 573 } 574 bool isSuspended() const 575 { return android_atomic_acquire_load(&mSuspended) > 0; } 576 577 virtual String8 getParameters(const String8& keys); 578 virtual void ioConfigChanged(audio_io_config_event event, pid_t pid = 0); 579 status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames); 580 // FIXME rename mixBuffer() to sinkBuffer() and remove int16_t* dependency. 581 // Consider also removing and passing an explicit mMainBuffer initialization 582 // parameter to AF::PlaybackThread::Track::Track(). 583 int16_t *mixBuffer() const { 584 return reinterpret_cast<int16_t *>(mSinkBuffer); }; 585 586 virtual void detachAuxEffect_l(int effectId); 587 status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track> track, 588 int EffectId); 589 status_t attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track> track, 590 int EffectId); 591 592 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 593 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 594 virtual uint32_t hasAudioSession(int sessionId) const; 595 virtual uint32_t getStrategyForSession_l(int sessionId); 596 597 598 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 599 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 600 601 // called with AudioFlinger lock held 602 void invalidateTracks(audio_stream_type_t streamType); 603 604 virtual size_t frameCount() const { return mNormalFrameCount; } 605 606 // Return's the HAL's frame count i.e. fast mixer buffer size. 607 size_t frameCountHAL() const { return mFrameCount; } 608 609 status_t getTimestamp_l(AudioTimestamp& timestamp); 610 611 void addPatchTrack(const sp<PatchTrack>& track); 612 void deletePatchTrack(const sp<PatchTrack>& track); 613 614 virtual void getAudioPortConfig(struct audio_port_config *config); 615 616protected: 617 // updated by readOutputParameters_l() 618 size_t mNormalFrameCount; // normal mixer and effects 619 620 bool mThreadThrottle; // throttle the thread processing 621 uint32_t mThreadThrottleTimeMs; // throttle time for MIXER threads 622 uint32_t mThreadThrottleEndMs; // notify once per throttling 623 uint32_t mHalfBufferMs; // half the buffer size in milliseconds 624 625 void* mSinkBuffer; // frame size aligned sink buffer 626 627 // TODO: 628 // Rearrange the buffer info into a struct/class with 629 // clear, copy, construction, destruction methods. 630 // 631 // mSinkBuffer also has associated with it: 632 // 633 // mSinkBufferSize: Sink Buffer Size 634 // mFormat: Sink Buffer Format 635 636 // Mixer Buffer (mMixerBuffer*) 637 // 638 // In the case of floating point or multichannel data, which is not in the 639 // sink format, it is required to accumulate in a higher precision or greater channel count 640 // buffer before downmixing or data conversion to the sink buffer. 641 642 // Set to "true" to enable the Mixer Buffer otherwise mixer output goes to sink buffer. 643 bool mMixerBufferEnabled; 644 645 // Storage, 32 byte aligned (may make this alignment a requirement later). 646 // Due to constraints on mNormalFrameCount, the buffer size is a multiple of 16 frames. 647 void* mMixerBuffer; 648 649 // Size of mMixerBuffer in bytes: mNormalFrameCount * #channels * sampsize. 650 size_t mMixerBufferSize; 651 652 // The audio format of mMixerBuffer. Set to AUDIO_FORMAT_PCM_(FLOAT|16_BIT) only. 653 audio_format_t mMixerBufferFormat; 654 655 // An internal flag set to true by MixerThread::prepareTracks_l() 656 // when mMixerBuffer contains valid data after mixing. 657 bool mMixerBufferValid; 658 659 // Effects Buffer (mEffectsBuffer*) 660 // 661 // In the case of effects data, which is not in the sink format, 662 // it is required to accumulate in a different buffer before data conversion 663 // to the sink buffer. 664 665 // Set to "true" to enable the Effects Buffer otherwise effects output goes to sink buffer. 666 bool mEffectBufferEnabled; 667 668 // Storage, 32 byte aligned (may make this alignment a requirement later). 669 // Due to constraints on mNormalFrameCount, the buffer size is a multiple of 16 frames. 670 void* mEffectBuffer; 671 672 // Size of mEffectsBuffer in bytes: mNormalFrameCount * #channels * sampsize. 673 size_t mEffectBufferSize; 674 675 // The audio format of mEffectsBuffer. Set to AUDIO_FORMAT_PCM_16_BIT only. 676 audio_format_t mEffectBufferFormat; 677 678 // An internal flag set to true by MixerThread::prepareTracks_l() 679 // when mEffectsBuffer contains valid data after mixing. 680 // 681 // When this is set, all mixer data is routed into the effects buffer 682 // for any processing (including output processing). 683 bool mEffectBufferValid; 684 685 // suspend count, > 0 means suspended. While suspended, the thread continues to pull from 686 // tracks and mix, but doesn't write to HAL. A2DP and SCO HAL implementations can't handle 687 // concurrent use of both of them, so Audio Policy Service suspends one of the threads to 688 // workaround that restriction. 689 // 'volatile' means accessed via atomic operations and no lock. 690 volatile int32_t mSuspended; 691 692 // FIXME overflows every 6+ hours at 44.1 kHz stereo 16-bit samples 693 // mFramesWritten would be better, or 64-bit even better 694 size_t mBytesWritten; 695private: 696 // mMasterMute is in both PlaybackThread and in AudioFlinger. When a 697 // PlaybackThread needs to find out if master-muted, it checks it's local 698 // copy rather than the one in AudioFlinger. This optimization saves a lock. 699 bool mMasterMute; 700 void setMasterMute_l(bool muted) { mMasterMute = muted; } 701protected: 702 SortedVector< wp<Track> > mActiveTracks; // FIXME check if this could be sp<> 703 SortedVector<int> mWakeLockUids; 704 int mActiveTracksGeneration; 705 wp<Track> mLatestActiveTrack; // latest track added to mActiveTracks 706 707 // Allocate a track name for a given channel mask. 708 // Returns name >= 0 if successful, -1 on failure. 709 virtual int getTrackName_l(audio_channel_mask_t channelMask, 710 audio_format_t format, int sessionId) = 0; 711 virtual void deleteTrackName_l(int name) = 0; 712 713 // Time to sleep between cycles when: 714 virtual uint32_t activeSleepTimeUs() const; // mixer state MIXER_TRACKS_ENABLED 715 virtual uint32_t idleSleepTimeUs() const = 0; // mixer state MIXER_IDLE 716 virtual uint32_t suspendSleepTimeUs() const = 0; // audio policy manager suspended us 717 // No sleep when mixer state == MIXER_TRACKS_READY; relies on audio HAL stream->write() 718 // No sleep in standby mode; waits on a condition 719 720 // Code snippets that are temporarily lifted up out of threadLoop() until the merge 721 void checkSilentMode_l(); 722 723 // Non-trivial for DUPLICATING only 724 virtual void saveOutputTracks() { } 725 virtual void clearOutputTracks() { } 726 727 // Cache various calculated values, at threadLoop() entry and after a parameter change 728 virtual void cacheParameters_l(); 729 730 virtual uint32_t correctLatency_l(uint32_t latency) const; 731 732 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 733 audio_patch_handle_t *handle); 734 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 735 736 bool usesHwAvSync() const { return (mType == DIRECT) && (mOutput != NULL) 737 && mHwSupportsPause 738 && (mOutput->flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC); } 739 740private: 741 742 friend class AudioFlinger; // for numerous 743 744 PlaybackThread& operator = (const PlaybackThread&); 745 746 status_t addTrack_l(const sp<Track>& track); 747 bool destroyTrack_l(const sp<Track>& track); 748 void removeTrack_l(const sp<Track>& track); 749 void broadcast_l(); 750 751 void readOutputParameters_l(); 752 753 virtual void dumpInternals(int fd, const Vector<String16>& args); 754 void dumpTracks(int fd, const Vector<String16>& args); 755 756 SortedVector< sp<Track> > mTracks; 757 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 758 AudioStreamOut *mOutput; 759 760 float mMasterVolume; 761 nsecs_t mLastWriteTime; 762 int mNumWrites; 763 int mNumDelayedWrites; 764 bool mInWrite; 765 766 // FIXME rename these former local variables of threadLoop to standard "m" names 767 nsecs_t mStandbyTimeNs; 768 size_t mSinkBufferSize; 769 770 // cached copies of activeSleepTimeUs() and idleSleepTimeUs() made by cacheParameters_l() 771 uint32_t mActiveSleepTimeUs; 772 uint32_t mIdleSleepTimeUs; 773 774 uint32_t mSleepTimeUs; 775 776 // mixer status returned by prepareTracks_l() 777 mixer_state mMixerStatus; // current cycle 778 // previous cycle when in prepareTracks_l() 779 mixer_state mMixerStatusIgnoringFastTracks; 780 // FIXME or a separate ready state per track 781 782 // FIXME move these declarations into the specific sub-class that needs them 783 // MIXER only 784 uint32_t sleepTimeShift; 785 786 // same as AudioFlinger::mStandbyTimeInNsecs except for DIRECT which uses a shorter value 787 nsecs_t mStandbyDelayNs; 788 789 // MIXER only 790 nsecs_t maxPeriod; 791 792 // DUPLICATING only 793 uint32_t writeFrames; 794 795 size_t mBytesRemaining; 796 size_t mCurrentWriteLength; 797 bool mUseAsyncWrite; 798 // mWriteAckSequence contains current write sequence on bits 31-1. The write sequence is 799 // incremented each time a write(), a flush() or a standby() occurs. 800 // Bit 0 is set when a write blocks and indicates a callback is expected. 801 // Bit 0 is reset by the async callback thread calling resetWriteBlocked(). Out of sequence 802 // callbacks are ignored. 803 uint32_t mWriteAckSequence; 804 // mDrainSequence contains current drain sequence on bits 31-1. The drain sequence is 805 // incremented each time a drain is requested or a flush() or standby() occurs. 806 // Bit 0 is set when the drain() command is called at the HAL and indicates a callback is 807 // expected. 808 // Bit 0 is reset by the async callback thread calling resetDraining(). Out of sequence 809 // callbacks are ignored. 810 uint32_t mDrainSequence; 811 // A condition that must be evaluated by prepareTrack_l() has changed and we must not wait 812 // for async write callback in the thread loop before evaluating it 813 bool mSignalPending; 814 sp<AsyncCallbackThread> mCallbackThread; 815 816private: 817 // The HAL output sink is treated as non-blocking, but current implementation is blocking 818 sp<NBAIO_Sink> mOutputSink; 819 // If a fast mixer is present, the blocking pipe sink, otherwise clear 820 sp<NBAIO_Sink> mPipeSink; 821 // The current sink for the normal mixer to write it's (sub)mix, mOutputSink or mPipeSink 822 sp<NBAIO_Sink> mNormalSink; 823#ifdef TEE_SINK 824 // For dumpsys 825 sp<NBAIO_Sink> mTeeSink; 826 sp<NBAIO_Source> mTeeSource; 827#endif 828 uint32_t mScreenState; // cached copy of gScreenState 829 static const size_t kFastMixerLogSize = 4 * 1024; 830 sp<NBLog::Writer> mFastMixerNBLogWriter; 831public: 832 virtual bool hasFastMixer() const = 0; 833 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex __unused) const 834 { FastTrackUnderruns dummy; return dummy; } 835 836protected: 837 // accessed by both binder threads and within threadLoop(), lock on mutex needed 838 unsigned mFastTrackAvailMask; // bit i set if fast track [i] is available 839 bool mHwSupportsPause; 840 bool mHwPaused; 841 bool mFlushPending; 842private: 843 // timestamp latch: 844 // D input is written by threadLoop_write while mutex is unlocked, and read while locked 845 // Q output is written while locked, and read while locked 846 struct { 847 AudioTimestamp mTimestamp; 848 uint32_t mUnpresentedFrames; 849 KeyedVector<Track *, uint32_t> mFramesReleased; 850 } mLatchD, mLatchQ; 851 bool mLatchDValid; // true means mLatchD is valid 852 // (except for mFramesReleased which is filled in later), 853 // and clock it into latch at next opportunity 854 bool mLatchQValid; // true means mLatchQ is valid 855}; 856 857class MixerThread : public PlaybackThread { 858public: 859 MixerThread(const sp<AudioFlinger>& audioFlinger, 860 AudioStreamOut* output, 861 audio_io_handle_t id, 862 audio_devices_t device, 863 bool systemReady, 864 type_t type = MIXER); 865 virtual ~MixerThread(); 866 867 // Thread virtuals 868 869 virtual bool checkForNewParameter_l(const String8& keyValuePair, 870 status_t& status); 871 virtual void dumpInternals(int fd, const Vector<String16>& args); 872 873protected: 874 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 875 virtual int getTrackName_l(audio_channel_mask_t channelMask, 876 audio_format_t format, int sessionId); 877 virtual void deleteTrackName_l(int name); 878 virtual uint32_t idleSleepTimeUs() const; 879 virtual uint32_t suspendSleepTimeUs() const; 880 virtual void cacheParameters_l(); 881 882 // threadLoop snippets 883 virtual ssize_t threadLoop_write(); 884 virtual void threadLoop_standby(); 885 virtual void threadLoop_mix(); 886 virtual void threadLoop_sleepTime(); 887 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 888 virtual uint32_t correctLatency_l(uint32_t latency) const; 889 890 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 891 audio_patch_handle_t *handle); 892 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 893 894 AudioMixer* mAudioMixer; // normal mixer 895private: 896 // one-time initialization, no locks required 897 sp<FastMixer> mFastMixer; // non-0 if there is also a fast mixer 898 sp<AudioWatchdog> mAudioWatchdog; // non-0 if there is an audio watchdog thread 899 900 // contents are not guaranteed to be consistent, no locks required 901 FastMixerDumpState mFastMixerDumpState; 902#ifdef STATE_QUEUE_DUMP 903 StateQueueObserverDump mStateQueueObserverDump; 904 StateQueueMutatorDump mStateQueueMutatorDump; 905#endif 906 AudioWatchdogDump mAudioWatchdogDump; 907 908 // accessible only within the threadLoop(), no locks required 909 // mFastMixer->sq() // for mutating and pushing state 910 int32_t mFastMixerFutex; // for cold idle 911 912public: 913 virtual bool hasFastMixer() const { return mFastMixer != 0; } 914 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const { 915 ALOG_ASSERT(fastIndex < FastMixerState::kMaxFastTracks); 916 return mFastMixerDumpState.mTracks[fastIndex].mUnderruns; 917 } 918 919}; 920 921class DirectOutputThread : public PlaybackThread { 922public: 923 924 DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 925 audio_io_handle_t id, audio_devices_t device, bool systemReady); 926 virtual ~DirectOutputThread(); 927 928 // Thread virtuals 929 930 virtual bool checkForNewParameter_l(const String8& keyValuePair, 931 status_t& status); 932 virtual void flushHw_l(); 933 934protected: 935 virtual int getTrackName_l(audio_channel_mask_t channelMask, 936 audio_format_t format, int sessionId); 937 virtual void deleteTrackName_l(int name); 938 virtual uint32_t activeSleepTimeUs() const; 939 virtual uint32_t idleSleepTimeUs() const; 940 virtual uint32_t suspendSleepTimeUs() const; 941 virtual void cacheParameters_l(); 942 943 // threadLoop snippets 944 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 945 virtual void threadLoop_mix(); 946 virtual void threadLoop_sleepTime(); 947 virtual void threadLoop_exit(); 948 virtual bool shouldStandby_l(); 949 950 virtual void onAddNewTrack_l(); 951 952 // volumes last sent to audio HAL with stream->set_volume() 953 float mLeftVolFloat; 954 float mRightVolFloat; 955 956 DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 957 audio_io_handle_t id, uint32_t device, ThreadBase::type_t type, 958 bool systemReady); 959 void processVolume_l(Track *track, bool lastTrack); 960 961 // prepareTracks_l() tells threadLoop_mix() the name of the single active track 962 sp<Track> mActiveTrack; 963 964 wp<Track> mPreviousTrack; // used to detect track switch 965 966public: 967 virtual bool hasFastMixer() const { return false; } 968}; 969 970class OffloadThread : public DirectOutputThread { 971public: 972 973 OffloadThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 974 audio_io_handle_t id, uint32_t device, bool systemReady); 975 virtual ~OffloadThread() {}; 976 virtual void flushHw_l(); 977 978protected: 979 // threadLoop snippets 980 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 981 virtual void threadLoop_exit(); 982 983 virtual bool waitingAsyncCallback(); 984 virtual bool waitingAsyncCallback_l(); 985 986private: 987 size_t mPausedWriteLength; // length in bytes of write interrupted by pause 988 size_t mPausedBytesRemaining; // bytes still waiting in mixbuffer after resume 989}; 990 991class AsyncCallbackThread : public Thread { 992public: 993 994 AsyncCallbackThread(const wp<PlaybackThread>& playbackThread); 995 996 virtual ~AsyncCallbackThread(); 997 998 // Thread virtuals 999 virtual bool threadLoop(); 1000 1001 // RefBase 1002 virtual void onFirstRef(); 1003 1004 void exit(); 1005 void setWriteBlocked(uint32_t sequence); 1006 void resetWriteBlocked(); 1007 void setDraining(uint32_t sequence); 1008 void resetDraining(); 1009 1010private: 1011 const wp<PlaybackThread> mPlaybackThread; 1012 // mWriteAckSequence corresponds to the last write sequence passed by the offload thread via 1013 // setWriteBlocked(). The sequence is shifted one bit to the left and the lsb is used 1014 // to indicate that the callback has been received via resetWriteBlocked() 1015 uint32_t mWriteAckSequence; 1016 // mDrainSequence corresponds to the last drain sequence passed by the offload thread via 1017 // setDraining(). The sequence is shifted one bit to the left and the lsb is used 1018 // to indicate that the callback has been received via resetDraining() 1019 uint32_t mDrainSequence; 1020 Condition mWaitWorkCV; 1021 Mutex mLock; 1022}; 1023 1024class DuplicatingThread : public MixerThread { 1025public: 1026 DuplicatingThread(const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread, 1027 audio_io_handle_t id, bool systemReady); 1028 virtual ~DuplicatingThread(); 1029 1030 // Thread virtuals 1031 void addOutputTrack(MixerThread* thread); 1032 void removeOutputTrack(MixerThread* thread); 1033 uint32_t waitTimeMs() const { return mWaitTimeMs; } 1034protected: 1035 virtual uint32_t activeSleepTimeUs() const; 1036 1037private: 1038 bool outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks); 1039protected: 1040 // threadLoop snippets 1041 virtual void threadLoop_mix(); 1042 virtual void threadLoop_sleepTime(); 1043 virtual ssize_t threadLoop_write(); 1044 virtual void threadLoop_standby(); 1045 virtual void cacheParameters_l(); 1046 1047private: 1048 // called from threadLoop, addOutputTrack, removeOutputTrack 1049 virtual void updateWaitTime_l(); 1050protected: 1051 virtual void saveOutputTracks(); 1052 virtual void clearOutputTracks(); 1053private: 1054 1055 uint32_t mWaitTimeMs; 1056 SortedVector < sp<OutputTrack> > outputTracks; 1057 SortedVector < sp<OutputTrack> > mOutputTracks; 1058public: 1059 virtual bool hasFastMixer() const { return false; } 1060}; 1061 1062 1063// record thread 1064class RecordThread : public ThreadBase 1065{ 1066public: 1067 1068 class RecordTrack; 1069 1070 /* The ResamplerBufferProvider is used to retrieve recorded input data from the 1071 * RecordThread. It maintains local state on the relative position of the read 1072 * position of the RecordTrack compared with the RecordThread. 1073 */ 1074 class ResamplerBufferProvider : public AudioBufferProvider 1075 { 1076 public: 1077 ResamplerBufferProvider(RecordTrack* recordTrack) : 1078 mRecordTrack(recordTrack), 1079 mRsmpInUnrel(0), mRsmpInFront(0) { } 1080 virtual ~ResamplerBufferProvider() { } 1081 1082 // called to set the ResamplerBufferProvider to head of the RecordThread data buffer, 1083 // skipping any previous data read from the hal. 1084 virtual void reset(); 1085 1086 /* Synchronizes RecordTrack position with the RecordThread. 1087 * Calculates available frames and handle overruns if the RecordThread 1088 * has advanced faster than the ResamplerBufferProvider has retrieved data. 1089 * TODO: why not do this for every getNextBuffer? 1090 * 1091 * Parameters 1092 * framesAvailable: pointer to optional output size_t to store record track 1093 * frames available. 1094 * hasOverrun: pointer to optional boolean, returns true if track has overrun. 1095 */ 1096 1097 virtual void sync(size_t *framesAvailable = NULL, bool *hasOverrun = NULL); 1098 1099 // AudioBufferProvider interface 1100 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts); 1101 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 1102 private: 1103 RecordTrack * const mRecordTrack; 1104 size_t mRsmpInUnrel; // unreleased frames remaining from 1105 // most recent getNextBuffer 1106 // for debug only 1107 int32_t mRsmpInFront; // next available frame 1108 // rolling counter that is never cleared 1109 }; 1110 1111 /* The RecordBufferConverter is used for format, channel, and sample rate 1112 * conversion for a RecordTrack. 1113 * 1114 * TODO: Self contained, so move to a separate file later. 1115 * 1116 * RecordBufferConverter uses the convert() method rather than exposing a 1117 * buffer provider interface; this is to save a memory copy. 1118 */ 1119 class RecordBufferConverter 1120 { 1121 public: 1122 RecordBufferConverter( 1123 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 1124 uint32_t srcSampleRate, 1125 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 1126 uint32_t dstSampleRate); 1127 1128 ~RecordBufferConverter(); 1129 1130 /* Converts input data from an AudioBufferProvider by format, channelMask, 1131 * and sampleRate to a destination buffer. 1132 * 1133 * Parameters 1134 * dst: buffer to place the converted data. 1135 * provider: buffer provider to obtain source data. 1136 * frames: number of frames to convert 1137 * 1138 * Returns the number of frames converted. 1139 */ 1140 size_t convert(void *dst, AudioBufferProvider *provider, size_t frames); 1141 1142 // returns NO_ERROR if constructor was successful 1143 status_t initCheck() const { 1144 // mSrcChannelMask set on successful updateParameters 1145 return mSrcChannelMask != AUDIO_CHANNEL_INVALID ? NO_ERROR : NO_INIT; 1146 } 1147 1148 // allows dynamic reconfigure of all parameters 1149 status_t updateParameters( 1150 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 1151 uint32_t srcSampleRate, 1152 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 1153 uint32_t dstSampleRate); 1154 1155 // called to reset resampler buffers on record track discontinuity 1156 void reset() { 1157 if (mResampler != NULL) { 1158 mResampler->reset(); 1159 } 1160 } 1161 1162 private: 1163 // format conversion when not using resampler 1164 void convertNoResampler(void *dst, const void *src, size_t frames); 1165 1166 // format conversion when using resampler; modifies src in-place 1167 void convertResampler(void *dst, /*not-a-const*/ void *src, size_t frames); 1168 1169 // user provided information 1170 audio_channel_mask_t mSrcChannelMask; 1171 audio_format_t mSrcFormat; 1172 uint32_t mSrcSampleRate; 1173 audio_channel_mask_t mDstChannelMask; 1174 audio_format_t mDstFormat; 1175 uint32_t mDstSampleRate; 1176 1177 // derived information 1178 uint32_t mSrcChannelCount; 1179 uint32_t mDstChannelCount; 1180 size_t mDstFrameSize; 1181 1182 // format conversion buffer 1183 void *mBuf; 1184 size_t mBufFrames; 1185 size_t mBufFrameSize; 1186 1187 // resampler info 1188 AudioResampler *mResampler; 1189 1190 bool mIsLegacyDownmix; // legacy stereo to mono conversion needed 1191 bool mIsLegacyUpmix; // legacy mono to stereo conversion needed 1192 bool mRequiresFloat; // data processing requires float (e.g. resampler) 1193 PassthruBufferProvider *mInputConverterProvider; // converts input to float 1194 int8_t mIdxAry[sizeof(uint32_t) * 8]; // used for channel mask conversion 1195 }; 1196 1197#include "RecordTracks.h" 1198 1199 RecordThread(const sp<AudioFlinger>& audioFlinger, 1200 AudioStreamIn *input, 1201 audio_io_handle_t id, 1202 audio_devices_t outDevice, 1203 audio_devices_t inDevice, 1204 bool systemReady 1205#ifdef TEE_SINK 1206 , const sp<NBAIO_Sink>& teeSink 1207#endif 1208 ); 1209 virtual ~RecordThread(); 1210 1211 // no addTrack_l ? 1212 void destroyTrack_l(const sp<RecordTrack>& track); 1213 void removeTrack_l(const sp<RecordTrack>& track); 1214 1215 void dumpInternals(int fd, const Vector<String16>& args); 1216 void dumpTracks(int fd, const Vector<String16>& args); 1217 1218 // Thread virtuals 1219 virtual bool threadLoop(); 1220 1221 // RefBase 1222 virtual void onFirstRef(); 1223 1224 virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; } 1225 1226 virtual sp<MemoryDealer> readOnlyHeap() const { return mReadOnlyHeap; } 1227 1228 virtual sp<IMemory> pipeMemory() const { return mPipeMemory; } 1229 1230 sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l( 1231 const sp<AudioFlinger::Client>& client, 1232 uint32_t sampleRate, 1233 audio_format_t format, 1234 audio_channel_mask_t channelMask, 1235 size_t *pFrameCount, 1236 int sessionId, 1237 size_t *notificationFrames, 1238 int uid, 1239 IAudioFlinger::track_flags_t *flags, 1240 pid_t tid, 1241 status_t *status /*non-NULL*/); 1242 1243 status_t start(RecordTrack* recordTrack, 1244 AudioSystem::sync_event_t event, 1245 int triggerSession); 1246 1247 // ask the thread to stop the specified track, and 1248 // return true if the caller should then do it's part of the stopping process 1249 bool stop(RecordTrack* recordTrack); 1250 1251 void dump(int fd, const Vector<String16>& args); 1252 AudioStreamIn* clearInput(); 1253 virtual audio_stream_t* stream() const; 1254 1255 1256 virtual bool checkForNewParameter_l(const String8& keyValuePair, 1257 status_t& status); 1258 virtual void cacheParameters_l() {} 1259 virtual String8 getParameters(const String8& keys); 1260 virtual void ioConfigChanged(audio_io_config_event event, pid_t pid = 0); 1261 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 1262 audio_patch_handle_t *handle); 1263 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 1264 1265 void addPatchRecord(const sp<PatchRecord>& record); 1266 void deletePatchRecord(const sp<PatchRecord>& record); 1267 1268 void readInputParameters_l(); 1269 virtual uint32_t getInputFramesLost(); 1270 1271 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1272 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1273 virtual uint32_t hasAudioSession(int sessionId) const; 1274 1275 // Return the set of unique session IDs across all tracks. 1276 // The keys are the session IDs, and the associated values are meaningless. 1277 // FIXME replace by Set [and implement Bag/Multiset for other uses]. 1278 KeyedVector<int, bool> sessionIds() const; 1279 1280 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 1281 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 1282 1283 static void syncStartEventCallback(const wp<SyncEvent>& event); 1284 1285 virtual size_t frameCount() const { return mFrameCount; } 1286 bool hasFastCapture() const { return mFastCapture != 0; } 1287 virtual void getAudioPortConfig(struct audio_port_config *config); 1288 1289private: 1290 // Enter standby if not already in standby, and set mStandby flag 1291 void standbyIfNotAlreadyInStandby(); 1292 1293 // Call the HAL standby method unconditionally, and don't change mStandby flag 1294 void inputStandBy(); 1295 1296 AudioStreamIn *mInput; 1297 SortedVector < sp<RecordTrack> > mTracks; 1298 // mActiveTracks has dual roles: it indicates the current active track(s), and 1299 // is used together with mStartStopCond to indicate start()/stop() progress 1300 SortedVector< sp<RecordTrack> > mActiveTracks; 1301 // generation counter for mActiveTracks 1302 int mActiveTracksGen; 1303 Condition mStartStopCond; 1304 1305 // resampler converts input at HAL Hz to output at AudioRecord client Hz 1306 void *mRsmpInBuffer; // 1307 size_t mRsmpInFrames; // size of resampler input in frames 1308 size_t mRsmpInFramesP2;// size rounded up to a power-of-2 1309 1310 // rolling index that is never cleared 1311 int32_t mRsmpInRear; // last filled frame + 1 1312 1313 // For dumpsys 1314 const sp<NBAIO_Sink> mTeeSink; 1315 1316 const sp<MemoryDealer> mReadOnlyHeap; 1317 1318 // one-time initialization, no locks required 1319 sp<FastCapture> mFastCapture; // non-0 if there is also 1320 // a fast capture 1321 1322 // FIXME audio watchdog thread 1323 1324 // contents are not guaranteed to be consistent, no locks required 1325 FastCaptureDumpState mFastCaptureDumpState; 1326#ifdef STATE_QUEUE_DUMP 1327 // FIXME StateQueue observer and mutator dump fields 1328#endif 1329 // FIXME audio watchdog dump 1330 1331 // accessible only within the threadLoop(), no locks required 1332 // mFastCapture->sq() // for mutating and pushing state 1333 int32_t mFastCaptureFutex; // for cold idle 1334 1335 // The HAL input source is treated as non-blocking, 1336 // but current implementation is blocking 1337 sp<NBAIO_Source> mInputSource; 1338 // The source for the normal capture thread to read from: mInputSource or mPipeSource 1339 sp<NBAIO_Source> mNormalSource; 1340 // If a fast capture is present, the non-blocking pipe sink written to by fast capture, 1341 // otherwise clear 1342 sp<NBAIO_Sink> mPipeSink; 1343 // If a fast capture is present, the non-blocking pipe source read by normal thread, 1344 // otherwise clear 1345 sp<NBAIO_Source> mPipeSource; 1346 // Depth of pipe from fast capture to normal thread and fast clients, always power of 2 1347 size_t mPipeFramesP2; 1348 // If a fast capture is present, the Pipe as IMemory, otherwise clear 1349 sp<IMemory> mPipeMemory; 1350 1351 static const size_t kFastCaptureLogSize = 4 * 1024; 1352 sp<NBLog::Writer> mFastCaptureNBLogWriter; 1353 1354 bool mFastTrackAvail; // true if fast track available 1355}; 1356