Threads.h revision 646679779a8f952980a5d4219ad9c6f93efc4b92
1/* 2** 3** Copyright 2012, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18#ifndef INCLUDING_FROM_AUDIOFLINGER_H 19 #error This header file should only be included from AudioFlinger.h 20#endif 21 22class ThreadBase : public Thread { 23public: 24 25#include "TrackBase.h" 26 27 enum type_t { 28 MIXER, // Thread class is MixerThread 29 DIRECT, // Thread class is DirectOutputThread 30 DUPLICATING, // Thread class is DuplicatingThread 31 RECORD, // Thread class is RecordThread 32 OFFLOAD // Thread class is OffloadThread 33 }; 34 35 static const char *threadTypeToString(type_t type); 36 37 ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 38 audio_devices_t outDevice, audio_devices_t inDevice, type_t type, 39 bool systemReady); 40 virtual ~ThreadBase(); 41 42 virtual status_t readyToRun(); 43 44 void dumpBase(int fd, const Vector<String16>& args); 45 void dumpEffectChains(int fd, const Vector<String16>& args); 46 47 void clearPowerManager(); 48 49 // base for record and playback 50 enum { 51 CFG_EVENT_IO, 52 CFG_EVENT_PRIO, 53 CFG_EVENT_SET_PARAMETER, 54 CFG_EVENT_CREATE_AUDIO_PATCH, 55 CFG_EVENT_RELEASE_AUDIO_PATCH, 56 }; 57 58 class ConfigEventData: public RefBase { 59 public: 60 virtual ~ConfigEventData() {} 61 62 virtual void dump(char *buffer, size_t size) = 0; 63 protected: 64 ConfigEventData() {} 65 }; 66 67 // Config event sequence by client if status needed (e.g binder thread calling setParameters()): 68 // 1. create SetParameterConfigEvent. This sets mWaitStatus in config event 69 // 2. Lock mLock 70 // 3. Call sendConfigEvent_l(): Append to mConfigEvents and mWaitWorkCV.signal 71 // 4. sendConfigEvent_l() reads status from event->mStatus; 72 // 5. sendConfigEvent_l() returns status 73 // 6. Unlock 74 // 75 // Parameter sequence by server: threadLoop calling processConfigEvents_l(): 76 // 1. Lock mLock 77 // 2. If there is an entry in mConfigEvents proceed ... 78 // 3. Read first entry in mConfigEvents 79 // 4. Remove first entry from mConfigEvents 80 // 5. Process 81 // 6. Set event->mStatus 82 // 7. event->mCond.signal 83 // 8. Unlock 84 85 class ConfigEvent: public RefBase { 86 public: 87 virtual ~ConfigEvent() {} 88 89 void dump(char *buffer, size_t size) { mData->dump(buffer, size); } 90 91 const int mType; // event type e.g. CFG_EVENT_IO 92 Mutex mLock; // mutex associated with mCond 93 Condition mCond; // condition for status return 94 status_t mStatus; // status communicated to sender 95 bool mWaitStatus; // true if sender is waiting for status 96 bool mRequiresSystemReady; // true if must wait for system ready to enter event queue 97 sp<ConfigEventData> mData; // event specific parameter data 98 99 protected: 100 ConfigEvent(int type, bool requiresSystemReady = false) : 101 mType(type), mStatus(NO_ERROR), mWaitStatus(false), 102 mRequiresSystemReady(requiresSystemReady), mData(NULL) {} 103 }; 104 105 class IoConfigEventData : public ConfigEventData { 106 public: 107 IoConfigEventData(audio_io_config_event event, pid_t pid) : 108 mEvent(event), mPid(pid) {} 109 110 virtual void dump(char *buffer, size_t size) { 111 snprintf(buffer, size, "IO event: event %d\n", mEvent); 112 } 113 114 const audio_io_config_event mEvent; 115 const pid_t mPid; 116 }; 117 118 class IoConfigEvent : public ConfigEvent { 119 public: 120 IoConfigEvent(audio_io_config_event event, pid_t pid) : 121 ConfigEvent(CFG_EVENT_IO) { 122 mData = new IoConfigEventData(event, pid); 123 } 124 virtual ~IoConfigEvent() {} 125 }; 126 127 class PrioConfigEventData : public ConfigEventData { 128 public: 129 PrioConfigEventData(pid_t pid, pid_t tid, int32_t prio) : 130 mPid(pid), mTid(tid), mPrio(prio) {} 131 132 virtual void dump(char *buffer, size_t size) { 133 snprintf(buffer, size, "Prio event: pid %d, tid %d, prio %d\n", mPid, mTid, mPrio); 134 } 135 136 const pid_t mPid; 137 const pid_t mTid; 138 const int32_t mPrio; 139 }; 140 141 class PrioConfigEvent : public ConfigEvent { 142 public: 143 PrioConfigEvent(pid_t pid, pid_t tid, int32_t prio) : 144 ConfigEvent(CFG_EVENT_PRIO, true) { 145 mData = new PrioConfigEventData(pid, tid, prio); 146 } 147 virtual ~PrioConfigEvent() {} 148 }; 149 150 class SetParameterConfigEventData : public ConfigEventData { 151 public: 152 SetParameterConfigEventData(String8 keyValuePairs) : 153 mKeyValuePairs(keyValuePairs) {} 154 155 virtual void dump(char *buffer, size_t size) { 156 snprintf(buffer, size, "KeyValue: %s\n", mKeyValuePairs.string()); 157 } 158 159 const String8 mKeyValuePairs; 160 }; 161 162 class SetParameterConfigEvent : public ConfigEvent { 163 public: 164 SetParameterConfigEvent(String8 keyValuePairs) : 165 ConfigEvent(CFG_EVENT_SET_PARAMETER) { 166 mData = new SetParameterConfigEventData(keyValuePairs); 167 mWaitStatus = true; 168 } 169 virtual ~SetParameterConfigEvent() {} 170 }; 171 172 class CreateAudioPatchConfigEventData : public ConfigEventData { 173 public: 174 CreateAudioPatchConfigEventData(const struct audio_patch patch, 175 audio_patch_handle_t handle) : 176 mPatch(patch), mHandle(handle) {} 177 178 virtual void dump(char *buffer, size_t size) { 179 snprintf(buffer, size, "Patch handle: %u\n", mHandle); 180 } 181 182 const struct audio_patch mPatch; 183 audio_patch_handle_t mHandle; 184 }; 185 186 class CreateAudioPatchConfigEvent : public ConfigEvent { 187 public: 188 CreateAudioPatchConfigEvent(const struct audio_patch patch, 189 audio_patch_handle_t handle) : 190 ConfigEvent(CFG_EVENT_CREATE_AUDIO_PATCH) { 191 mData = new CreateAudioPatchConfigEventData(patch, handle); 192 mWaitStatus = true; 193 } 194 virtual ~CreateAudioPatchConfigEvent() {} 195 }; 196 197 class ReleaseAudioPatchConfigEventData : public ConfigEventData { 198 public: 199 ReleaseAudioPatchConfigEventData(const audio_patch_handle_t handle) : 200 mHandle(handle) {} 201 202 virtual void dump(char *buffer, size_t size) { 203 snprintf(buffer, size, "Patch handle: %u\n", mHandle); 204 } 205 206 audio_patch_handle_t mHandle; 207 }; 208 209 class ReleaseAudioPatchConfigEvent : public ConfigEvent { 210 public: 211 ReleaseAudioPatchConfigEvent(const audio_patch_handle_t handle) : 212 ConfigEvent(CFG_EVENT_RELEASE_AUDIO_PATCH) { 213 mData = new ReleaseAudioPatchConfigEventData(handle); 214 mWaitStatus = true; 215 } 216 virtual ~ReleaseAudioPatchConfigEvent() {} 217 }; 218 219 class PMDeathRecipient : public IBinder::DeathRecipient { 220 public: 221 PMDeathRecipient(const wp<ThreadBase>& thread) : mThread(thread) {} 222 virtual ~PMDeathRecipient() {} 223 224 // IBinder::DeathRecipient 225 virtual void binderDied(const wp<IBinder>& who); 226 227 private: 228 PMDeathRecipient(const PMDeathRecipient&); 229 PMDeathRecipient& operator = (const PMDeathRecipient&); 230 231 wp<ThreadBase> mThread; 232 }; 233 234 virtual status_t initCheck() const = 0; 235 236 // static externally-visible 237 type_t type() const { return mType; } 238 bool isDuplicating() const { return (mType == DUPLICATING); } 239 240 audio_io_handle_t id() const { return mId;} 241 242 // dynamic externally-visible 243 uint32_t sampleRate() const { return mSampleRate; } 244 audio_channel_mask_t channelMask() const { return mChannelMask; } 245 audio_format_t format() const { return mHALFormat; } 246 uint32_t channelCount() const { return mChannelCount; } 247 // Called by AudioFlinger::frameCount(audio_io_handle_t output) and effects, 248 // and returns the [normal mix] buffer's frame count. 249 virtual size_t frameCount() const = 0; 250 size_t frameSize() const { return mFrameSize; } 251 252 // Should be "virtual status_t requestExitAndWait()" and override same 253 // method in Thread, but Thread::requestExitAndWait() is not yet virtual. 254 void exit(); 255 virtual bool checkForNewParameter_l(const String8& keyValuePair, 256 status_t& status) = 0; 257 virtual status_t setParameters(const String8& keyValuePairs); 258 virtual String8 getParameters(const String8& keys) = 0; 259 virtual void ioConfigChanged(audio_io_config_event event, pid_t pid = 0) = 0; 260 // sendConfigEvent_l() must be called with ThreadBase::mLock held 261 // Can temporarily release the lock if waiting for a reply from 262 // processConfigEvents_l(). 263 status_t sendConfigEvent_l(sp<ConfigEvent>& event); 264 void sendIoConfigEvent(audio_io_config_event event, pid_t pid = 0); 265 void sendIoConfigEvent_l(audio_io_config_event event, pid_t pid = 0); 266 void sendPrioConfigEvent(pid_t pid, pid_t tid, int32_t prio); 267 void sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio); 268 status_t sendSetParameterConfigEvent_l(const String8& keyValuePair); 269 status_t sendCreateAudioPatchConfigEvent(const struct audio_patch *patch, 270 audio_patch_handle_t *handle); 271 status_t sendReleaseAudioPatchConfigEvent(audio_patch_handle_t handle); 272 void processConfigEvents_l(); 273 virtual void cacheParameters_l() = 0; 274 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 275 audio_patch_handle_t *handle) = 0; 276 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle) = 0; 277 virtual void getAudioPortConfig(struct audio_port_config *config) = 0; 278 279 280 // see note at declaration of mStandby, mOutDevice and mInDevice 281 bool standby() const { return mStandby; } 282 audio_devices_t outDevice() const { return mOutDevice; } 283 audio_devices_t inDevice() const { return mInDevice; } 284 285 virtual audio_stream_t* stream() const = 0; 286 287 sp<EffectHandle> createEffect_l( 288 const sp<AudioFlinger::Client>& client, 289 const sp<IEffectClient>& effectClient, 290 int32_t priority, 291 audio_session_t sessionId, 292 effect_descriptor_t *desc, 293 int *enabled, 294 status_t *status /*non-NULL*/); 295 296 // return values for hasAudioSession (bit field) 297 enum effect_state { 298 EFFECT_SESSION = 0x1, // the audio session corresponds to at least one 299 // effect 300 TRACK_SESSION = 0x2 // the audio session corresponds to at least one 301 // track 302 }; 303 304 // get effect chain corresponding to session Id. 305 sp<EffectChain> getEffectChain(audio_session_t sessionId); 306 // same as getEffectChain() but must be called with ThreadBase mutex locked 307 sp<EffectChain> getEffectChain_l(audio_session_t sessionId) const; 308 // add an effect chain to the chain list (mEffectChains) 309 virtual status_t addEffectChain_l(const sp<EffectChain>& chain) = 0; 310 // remove an effect chain from the chain list (mEffectChains) 311 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain) = 0; 312 // lock all effect chains Mutexes. Must be called before releasing the 313 // ThreadBase mutex before processing the mixer and effects. This guarantees the 314 // integrity of the chains during the process. 315 // Also sets the parameter 'effectChains' to current value of mEffectChains. 316 void lockEffectChains_l(Vector< sp<EffectChain> >& effectChains); 317 // unlock effect chains after process 318 void unlockEffectChains(const Vector< sp<EffectChain> >& effectChains); 319 // get a copy of mEffectChains vector 320 Vector< sp<EffectChain> > getEffectChains_l() const { return mEffectChains; }; 321 // set audio mode to all effect chains 322 void setMode(audio_mode_t mode); 323 // get effect module with corresponding ID on specified audio session 324 sp<AudioFlinger::EffectModule> getEffect(audio_session_t sessionId, int effectId); 325 sp<AudioFlinger::EffectModule> getEffect_l(audio_session_t sessionId, int effectId); 326 // add and effect module. Also creates the effect chain is none exists for 327 // the effects audio session 328 status_t addEffect_l(const sp< EffectModule>& effect); 329 // remove and effect module. Also removes the effect chain is this was the last 330 // effect 331 void removeEffect_l(const sp< EffectModule>& effect); 332 // detach all tracks connected to an auxiliary effect 333 virtual void detachAuxEffect_l(int effectId __unused) {} 334 // returns either EFFECT_SESSION if effects on this audio session exist in one 335 // chain, or TRACK_SESSION if tracks on this audio session exist, or both 336 virtual uint32_t hasAudioSession(audio_session_t sessionId) const = 0; 337 // the value returned by default implementation is not important as the 338 // strategy is only meaningful for PlaybackThread which implements this method 339 virtual uint32_t getStrategyForSession_l(audio_session_t sessionId __unused) 340 { return 0; } 341 342 // suspend or restore effect according to the type of effect passed. a NULL 343 // type pointer means suspend all effects in the session 344 void setEffectSuspended(const effect_uuid_t *type, 345 bool suspend, 346 audio_session_t sessionId = AUDIO_SESSION_OUTPUT_MIX); 347 // check if some effects must be suspended/restored when an effect is enabled 348 // or disabled 349 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 350 bool enabled, 351 audio_session_t sessionId = 352 AUDIO_SESSION_OUTPUT_MIX); 353 void checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 354 bool enabled, 355 audio_session_t sessionId = 356 AUDIO_SESSION_OUTPUT_MIX); 357 358 virtual status_t setSyncEvent(const sp<SyncEvent>& event) = 0; 359 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const = 0; 360 361 // Return a reference to a per-thread heap which can be used to allocate IMemory 362 // objects that will be read-only to client processes, read/write to mediaserver, 363 // and shared by all client processes of the thread. 364 // The heap is per-thread rather than common across all threads, because 365 // clients can't be trusted not to modify the offset of the IMemory they receive. 366 // If a thread does not have such a heap, this method returns 0. 367 virtual sp<MemoryDealer> readOnlyHeap() const { return 0; } 368 369 virtual sp<IMemory> pipeMemory() const { return 0; } 370 371 void systemReady(); 372 373 mutable Mutex mLock; 374 375protected: 376 377 // entry describing an effect being suspended in mSuspendedSessions keyed vector 378 class SuspendedSessionDesc : public RefBase { 379 public: 380 SuspendedSessionDesc() : mRefCount(0) {} 381 382 int mRefCount; // number of active suspend requests 383 effect_uuid_t mType; // effect type UUID 384 }; 385 386 void acquireWakeLock(int uid = -1); 387 virtual void acquireWakeLock_l(int uid = -1); 388 void releaseWakeLock(); 389 void releaseWakeLock_l(); 390 void updateWakeLockUids(const SortedVector<int> &uids); 391 void updateWakeLockUids_l(const SortedVector<int> &uids); 392 void getPowerManager_l(); 393 void setEffectSuspended_l(const effect_uuid_t *type, 394 bool suspend, 395 audio_session_t sessionId); 396 // updated mSuspendedSessions when an effect suspended or restored 397 void updateSuspendedSessions_l(const effect_uuid_t *type, 398 bool suspend, 399 audio_session_t sessionId); 400 // check if some effects must be suspended when an effect chain is added 401 void checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain); 402 403 String16 getWakeLockTag(); 404 405 virtual void preExit() { } 406 virtual void setMasterMono_l(bool mono __unused) { } 407 virtual bool requireMonoBlend() { return false; } 408 409 friend class AudioFlinger; // for mEffectChains 410 411 const type_t mType; 412 413 // Used by parameters, config events, addTrack_l, exit 414 Condition mWaitWorkCV; 415 416 const sp<AudioFlinger> mAudioFlinger; 417 418 // updated by PlaybackThread::readOutputParameters_l() or 419 // RecordThread::readInputParameters_l() 420 uint32_t mSampleRate; 421 size_t mFrameCount; // output HAL, direct output, record 422 audio_channel_mask_t mChannelMask; 423 uint32_t mChannelCount; 424 size_t mFrameSize; 425 // not HAL frame size, this is for output sink (to pipe to fast mixer) 426 audio_format_t mFormat; // Source format for Recording and 427 // Sink format for Playback. 428 // Sink format may be different than 429 // HAL format if Fastmixer is used. 430 audio_format_t mHALFormat; 431 size_t mBufferSize; // HAL buffer size for read() or write() 432 433 Vector< sp<ConfigEvent> > mConfigEvents; 434 Vector< sp<ConfigEvent> > mPendingConfigEvents; // events awaiting system ready 435 436 // These fields are written and read by thread itself without lock or barrier, 437 // and read by other threads without lock or barrier via standby(), outDevice() 438 // and inDevice(). 439 // Because of the absence of a lock or barrier, any other thread that reads 440 // these fields must use the information in isolation, or be prepared to deal 441 // with possibility that it might be inconsistent with other information. 442 bool mStandby; // Whether thread is currently in standby. 443 audio_devices_t mOutDevice; // output device 444 audio_devices_t mInDevice; // input device 445 audio_devices_t mPrevOutDevice; // previous output device 446 audio_devices_t mPrevInDevice; // previous input device 447 struct audio_patch mPatch; 448 audio_source_t mAudioSource; 449 450 const audio_io_handle_t mId; 451 Vector< sp<EffectChain> > mEffectChains; 452 453 static const int kThreadNameLength = 16; // prctl(PR_SET_NAME) limit 454 char mThreadName[kThreadNameLength]; // guaranteed NUL-terminated 455 sp<IPowerManager> mPowerManager; 456 sp<IBinder> mWakeLockToken; 457 const sp<PMDeathRecipient> mDeathRecipient; 458 // list of suspended effects per session and per type. The first (outer) vector is 459 // keyed by session ID, the second (inner) by type UUID timeLow field 460 KeyedVector< audio_session_t, KeyedVector< int, sp<SuspendedSessionDesc> > > 461 mSuspendedSessions; 462 static const size_t kLogSize = 4 * 1024; 463 sp<NBLog::Writer> mNBLogWriter; 464 bool mSystemReady; 465 bool mNotifiedBatteryStart; 466 ExtendedTimestamp mTimestamp; 467}; 468 469// --- PlaybackThread --- 470class PlaybackThread : public ThreadBase { 471public: 472 473#include "PlaybackTracks.h" 474 475 enum mixer_state { 476 MIXER_IDLE, // no active tracks 477 MIXER_TRACKS_ENABLED, // at least one active track, but no track has any data ready 478 MIXER_TRACKS_READY, // at least one active track, and at least one track has data 479 MIXER_DRAIN_TRACK, // drain currently playing track 480 MIXER_DRAIN_ALL, // fully drain the hardware 481 // standby mode does not have an enum value 482 // suspend by audio policy manager is orthogonal to mixer state 483 }; 484 485 PlaybackThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 486 audio_io_handle_t id, audio_devices_t device, type_t type, bool systemReady, 487 uint32_t bitRate = 0); 488 virtual ~PlaybackThread(); 489 490 void dump(int fd, const Vector<String16>& args); 491 492 // Thread virtuals 493 virtual bool threadLoop(); 494 495 // RefBase 496 virtual void onFirstRef(); 497 498protected: 499 // Code snippets that were lifted up out of threadLoop() 500 virtual void threadLoop_mix() = 0; 501 virtual void threadLoop_sleepTime() = 0; 502 virtual ssize_t threadLoop_write(); 503 virtual void threadLoop_drain(); 504 virtual void threadLoop_standby(); 505 virtual void threadLoop_exit(); 506 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 507 508 // prepareTracks_l reads and writes mActiveTracks, and returns 509 // the pending set of tracks to remove via Vector 'tracksToRemove'. The caller 510 // is responsible for clearing or destroying this Vector later on, when it 511 // is safe to do so. That will drop the final ref count and destroy the tracks. 512 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove) = 0; 513 void removeTracks_l(const Vector< sp<Track> >& tracksToRemove); 514 515 void writeCallback(); 516 void resetWriteBlocked(uint32_t sequence); 517 void drainCallback(); 518 void resetDraining(uint32_t sequence); 519 520 static int asyncCallback(stream_callback_event_t event, void *param, void *cookie); 521 522 virtual bool waitingAsyncCallback(); 523 virtual bool waitingAsyncCallback_l(); 524 virtual bool shouldStandby_l(); 525 virtual void onAddNewTrack_l(); 526 527 // ThreadBase virtuals 528 virtual void preExit(); 529 530 virtual bool keepWakeLock() const { return true; } 531 532public: 533 534 virtual status_t initCheck() const { return (mOutput == NULL) ? NO_INIT : NO_ERROR; } 535 536 // return estimated latency in milliseconds, as reported by HAL 537 uint32_t latency() const; 538 // same, but lock must already be held 539 uint32_t latency_l() const; 540 541 void setMasterVolume(float value); 542 void setMasterMute(bool muted); 543 544 void setStreamVolume(audio_stream_type_t stream, float value); 545 void setStreamMute(audio_stream_type_t stream, bool muted); 546 547 float streamVolume(audio_stream_type_t stream) const; 548 549 sp<Track> createTrack_l( 550 const sp<AudioFlinger::Client>& client, 551 audio_stream_type_t streamType, 552 uint32_t sampleRate, 553 audio_format_t format, 554 audio_channel_mask_t channelMask, 555 size_t *pFrameCount, 556 const sp<IMemory>& sharedBuffer, 557 audio_session_t sessionId, 558 IAudioFlinger::track_flags_t *flags, 559 pid_t tid, 560 int uid, 561 status_t *status /*non-NULL*/); 562 563 AudioStreamOut* getOutput() const; 564 AudioStreamOut* clearOutput(); 565 virtual audio_stream_t* stream() const; 566 567 // a very large number of suspend() will eventually wraparound, but unlikely 568 void suspend() { (void) android_atomic_inc(&mSuspended); } 569 void restore() 570 { 571 // if restore() is done without suspend(), get back into 572 // range so that the next suspend() will operate correctly 573 if (android_atomic_dec(&mSuspended) <= 0) { 574 android_atomic_release_store(0, &mSuspended); 575 } 576 } 577 bool isSuspended() const 578 { return android_atomic_acquire_load(&mSuspended) > 0; } 579 580 virtual String8 getParameters(const String8& keys); 581 virtual void ioConfigChanged(audio_io_config_event event, pid_t pid = 0); 582 status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames); 583 // FIXME rename mixBuffer() to sinkBuffer() and remove int16_t* dependency. 584 // Consider also removing and passing an explicit mMainBuffer initialization 585 // parameter to AF::PlaybackThread::Track::Track(). 586 int16_t *mixBuffer() const { 587 return reinterpret_cast<int16_t *>(mSinkBuffer); }; 588 589 virtual void detachAuxEffect_l(int effectId); 590 status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track> track, 591 int EffectId); 592 status_t attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track> track, 593 int EffectId); 594 595 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 596 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 597 virtual uint32_t hasAudioSession(audio_session_t sessionId) const; 598 virtual uint32_t getStrategyForSession_l(audio_session_t sessionId); 599 600 601 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 602 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 603 604 // called with AudioFlinger lock held 605 void invalidateTracks(audio_stream_type_t streamType); 606 607 virtual size_t frameCount() const { return mNormalFrameCount; } 608 609 // Return's the HAL's frame count i.e. fast mixer buffer size. 610 size_t frameCountHAL() const { return mFrameCount; } 611 612 status_t getTimestamp_l(AudioTimestamp& timestamp); 613 614 void addPatchTrack(const sp<PatchTrack>& track); 615 void deletePatchTrack(const sp<PatchTrack>& track); 616 617 virtual void getAudioPortConfig(struct audio_port_config *config); 618 619protected: 620 // updated by readOutputParameters_l() 621 size_t mNormalFrameCount; // normal mixer and effects 622 623 bool mThreadThrottle; // throttle the thread processing 624 uint32_t mThreadThrottleTimeMs; // throttle time for MIXER threads 625 uint32_t mThreadThrottleEndMs; // notify once per throttling 626 uint32_t mHalfBufferMs; // half the buffer size in milliseconds 627 628 void* mSinkBuffer; // frame size aligned sink buffer 629 630 // TODO: 631 // Rearrange the buffer info into a struct/class with 632 // clear, copy, construction, destruction methods. 633 // 634 // mSinkBuffer also has associated with it: 635 // 636 // mSinkBufferSize: Sink Buffer Size 637 // mFormat: Sink Buffer Format 638 639 // Mixer Buffer (mMixerBuffer*) 640 // 641 // In the case of floating point or multichannel data, which is not in the 642 // sink format, it is required to accumulate in a higher precision or greater channel count 643 // buffer before downmixing or data conversion to the sink buffer. 644 645 // Set to "true" to enable the Mixer Buffer otherwise mixer output goes to sink buffer. 646 bool mMixerBufferEnabled; 647 648 // Storage, 32 byte aligned (may make this alignment a requirement later). 649 // Due to constraints on mNormalFrameCount, the buffer size is a multiple of 16 frames. 650 void* mMixerBuffer; 651 652 // Size of mMixerBuffer in bytes: mNormalFrameCount * #channels * sampsize. 653 size_t mMixerBufferSize; 654 655 // The audio format of mMixerBuffer. Set to AUDIO_FORMAT_PCM_(FLOAT|16_BIT) only. 656 audio_format_t mMixerBufferFormat; 657 658 // An internal flag set to true by MixerThread::prepareTracks_l() 659 // when mMixerBuffer contains valid data after mixing. 660 bool mMixerBufferValid; 661 662 // Effects Buffer (mEffectsBuffer*) 663 // 664 // In the case of effects data, which is not in the sink format, 665 // it is required to accumulate in a different buffer before data conversion 666 // to the sink buffer. 667 668 // Set to "true" to enable the Effects Buffer otherwise effects output goes to sink buffer. 669 bool mEffectBufferEnabled; 670 671 // Storage, 32 byte aligned (may make this alignment a requirement later). 672 // Due to constraints on mNormalFrameCount, the buffer size is a multiple of 16 frames. 673 void* mEffectBuffer; 674 675 // Size of mEffectsBuffer in bytes: mNormalFrameCount * #channels * sampsize. 676 size_t mEffectBufferSize; 677 678 // The audio format of mEffectsBuffer. Set to AUDIO_FORMAT_PCM_16_BIT only. 679 audio_format_t mEffectBufferFormat; 680 681 // An internal flag set to true by MixerThread::prepareTracks_l() 682 // when mEffectsBuffer contains valid data after mixing. 683 // 684 // When this is set, all mixer data is routed into the effects buffer 685 // for any processing (including output processing). 686 bool mEffectBufferValid; 687 688 // suspend count, > 0 means suspended. While suspended, the thread continues to pull from 689 // tracks and mix, but doesn't write to HAL. A2DP and SCO HAL implementations can't handle 690 // concurrent use of both of them, so Audio Policy Service suspends one of the threads to 691 // workaround that restriction. 692 // 'volatile' means accessed via atomic operations and no lock. 693 volatile int32_t mSuspended; 694 695 int64_t mBytesWritten; 696 int64_t mFramesWritten; // not reset on standby 697private: 698 // mMasterMute is in both PlaybackThread and in AudioFlinger. When a 699 // PlaybackThread needs to find out if master-muted, it checks it's local 700 // copy rather than the one in AudioFlinger. This optimization saves a lock. 701 bool mMasterMute; 702 void setMasterMute_l(bool muted) { mMasterMute = muted; } 703protected: 704 SortedVector< wp<Track> > mActiveTracks; // FIXME check if this could be sp<> 705 SortedVector<int> mWakeLockUids; 706 int mActiveTracksGeneration; 707 wp<Track> mLatestActiveTrack; // latest track added to mActiveTracks 708 709 // Allocate a track name for a given channel mask. 710 // Returns name >= 0 if successful, -1 on failure. 711 virtual int getTrackName_l(audio_channel_mask_t channelMask, 712 audio_format_t format, audio_session_t sessionId) = 0; 713 virtual void deleteTrackName_l(int name) = 0; 714 715 // Time to sleep between cycles when: 716 virtual uint32_t activeSleepTimeUs() const; // mixer state MIXER_TRACKS_ENABLED 717 virtual uint32_t idleSleepTimeUs() const = 0; // mixer state MIXER_IDLE 718 virtual uint32_t suspendSleepTimeUs() const = 0; // audio policy manager suspended us 719 // No sleep when mixer state == MIXER_TRACKS_READY; relies on audio HAL stream->write() 720 // No sleep in standby mode; waits on a condition 721 722 // Code snippets that are temporarily lifted up out of threadLoop() until the merge 723 void checkSilentMode_l(); 724 725 // Non-trivial for DUPLICATING only 726 virtual void saveOutputTracks() { } 727 virtual void clearOutputTracks() { } 728 729 // Cache various calculated values, at threadLoop() entry and after a parameter change 730 virtual void cacheParameters_l(); 731 732 virtual uint32_t correctLatency_l(uint32_t latency) const; 733 734 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 735 audio_patch_handle_t *handle); 736 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 737 738 bool usesHwAvSync() const { return (mType == DIRECT) && (mOutput != NULL) 739 && mHwSupportsPause 740 && (mOutput->flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC); } 741 742private: 743 744 friend class AudioFlinger; // for numerous 745 746 PlaybackThread& operator = (const PlaybackThread&); 747 748 status_t addTrack_l(const sp<Track>& track); 749 bool destroyTrack_l(const sp<Track>& track); 750 void removeTrack_l(const sp<Track>& track); 751 void broadcast_l(); 752 753 void readOutputParameters_l(); 754 755 virtual void dumpInternals(int fd, const Vector<String16>& args); 756 void dumpTracks(int fd, const Vector<String16>& args); 757 758 SortedVector< sp<Track> > mTracks; 759 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 760 AudioStreamOut *mOutput; 761 762 float mMasterVolume; 763 nsecs_t mLastWriteTime; 764 int mNumWrites; 765 int mNumDelayedWrites; 766 bool mInWrite; 767 768 // FIXME rename these former local variables of threadLoop to standard "m" names 769 nsecs_t mStandbyTimeNs; 770 size_t mSinkBufferSize; 771 772 // cached copies of activeSleepTimeUs() and idleSleepTimeUs() made by cacheParameters_l() 773 uint32_t mActiveSleepTimeUs; 774 uint32_t mIdleSleepTimeUs; 775 776 uint32_t mSleepTimeUs; 777 778 // mixer status returned by prepareTracks_l() 779 mixer_state mMixerStatus; // current cycle 780 // previous cycle when in prepareTracks_l() 781 mixer_state mMixerStatusIgnoringFastTracks; 782 // FIXME or a separate ready state per track 783 784 // FIXME move these declarations into the specific sub-class that needs them 785 // MIXER only 786 uint32_t sleepTimeShift; 787 788 // same as AudioFlinger::mStandbyTimeInNsecs except for DIRECT which uses a shorter value 789 nsecs_t mStandbyDelayNs; 790 791 // MIXER only 792 nsecs_t maxPeriod; 793 794 // DUPLICATING only 795 uint32_t writeFrames; 796 797 size_t mBytesRemaining; 798 size_t mCurrentWriteLength; 799 bool mUseAsyncWrite; 800 // mWriteAckSequence contains current write sequence on bits 31-1. The write sequence is 801 // incremented each time a write(), a flush() or a standby() occurs. 802 // Bit 0 is set when a write blocks and indicates a callback is expected. 803 // Bit 0 is reset by the async callback thread calling resetWriteBlocked(). Out of sequence 804 // callbacks are ignored. 805 uint32_t mWriteAckSequence; 806 // mDrainSequence contains current drain sequence on bits 31-1. The drain sequence is 807 // incremented each time a drain is requested or a flush() or standby() occurs. 808 // Bit 0 is set when the drain() command is called at the HAL and indicates a callback is 809 // expected. 810 // Bit 0 is reset by the async callback thread calling resetDraining(). Out of sequence 811 // callbacks are ignored. 812 uint32_t mDrainSequence; 813 // A condition that must be evaluated by prepareTrack_l() has changed and we must not wait 814 // for async write callback in the thread loop before evaluating it 815 bool mSignalPending; 816 sp<AsyncCallbackThread> mCallbackThread; 817 818private: 819 // The HAL output sink is treated as non-blocking, but current implementation is blocking 820 sp<NBAIO_Sink> mOutputSink; 821 // If a fast mixer is present, the blocking pipe sink, otherwise clear 822 sp<NBAIO_Sink> mPipeSink; 823 // The current sink for the normal mixer to write it's (sub)mix, mOutputSink or mPipeSink 824 sp<NBAIO_Sink> mNormalSink; 825#ifdef TEE_SINK 826 // For dumpsys 827 sp<NBAIO_Sink> mTeeSink; 828 sp<NBAIO_Source> mTeeSource; 829#endif 830 uint32_t mScreenState; // cached copy of gScreenState 831 static const size_t kFastMixerLogSize = 4 * 1024; 832 sp<NBLog::Writer> mFastMixerNBLogWriter; 833public: 834 virtual bool hasFastMixer() const = 0; 835 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex __unused) const 836 { FastTrackUnderruns dummy; return dummy; } 837 838protected: 839 // accessed by both binder threads and within threadLoop(), lock on mutex needed 840 unsigned mFastTrackAvailMask; // bit i set if fast track [i] is available 841 bool mHwSupportsPause; 842 bool mHwPaused; 843 bool mFlushPending; 844 uint32_t mBufferDurationUs; // estimated duration of an audio HAL buffer 845 // based on initial bit rate (offload only) 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, audio_session_t 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 virtual void acquireWakeLock_l(int uid = -1) { 874 PlaybackThread::acquireWakeLock_l(uid); 875 if (hasFastMixer()) { 876 mFastMixer->setBoottimeOffset( 877 mTimestamp.mTimebaseOffset[ExtendedTimestamp::TIMEBASE_BOOTTIME]); 878 } 879 } 880 881 // threadLoop snippets 882 virtual ssize_t threadLoop_write(); 883 virtual void threadLoop_standby(); 884 virtual void threadLoop_mix(); 885 virtual void threadLoop_sleepTime(); 886 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 887 virtual uint32_t correctLatency_l(uint32_t latency) const; 888 889 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 890 audio_patch_handle_t *handle); 891 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 892 893 AudioMixer* mAudioMixer; // normal mixer 894private: 895 // one-time initialization, no locks required 896 sp<FastMixer> mFastMixer; // non-0 if there is also a fast mixer 897 sp<AudioWatchdog> mAudioWatchdog; // non-0 if there is an audio watchdog thread 898 899 // contents are not guaranteed to be consistent, no locks required 900 FastMixerDumpState mFastMixerDumpState; 901#ifdef STATE_QUEUE_DUMP 902 StateQueueObserverDump mStateQueueObserverDump; 903 StateQueueMutatorDump mStateQueueMutatorDump; 904#endif 905 AudioWatchdogDump mAudioWatchdogDump; 906 907 // accessible only within the threadLoop(), no locks required 908 // mFastMixer->sq() // for mutating and pushing state 909 int32_t mFastMixerFutex; // for cold idle 910 911 std::atomic_bool mMasterMono; 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 919protected: 920 virtual void setMasterMono_l(bool mono) { 921 mMasterMono.store(mono); 922 if (mFastMixer != nullptr) { /* hasFastMixer() */ 923 mFastMixer->setMasterMono(mMasterMono); 924 } 925 } 926 // the FastMixer performs mono blend if it exists. 927 // Blending with limiter is not idempotent, 928 // and blending without limiter is idempotent but inefficient to do twice. 929 virtual bool requireMonoBlend() { return mMasterMono.load() && !hasFastMixer(); } 930}; 931 932class DirectOutputThread : public PlaybackThread { 933public: 934 935 DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 936 audio_io_handle_t id, audio_devices_t device, bool systemReady, 937 uint32_t bitRate = 0); 938 virtual ~DirectOutputThread(); 939 940 // Thread virtuals 941 942 virtual bool checkForNewParameter_l(const String8& keyValuePair, 943 status_t& status); 944 virtual void flushHw_l(); 945 946protected: 947 virtual int getTrackName_l(audio_channel_mask_t channelMask, 948 audio_format_t format, audio_session_t sessionId); 949 virtual void deleteTrackName_l(int name); 950 virtual uint32_t activeSleepTimeUs() const; 951 virtual uint32_t idleSleepTimeUs() const; 952 virtual uint32_t suspendSleepTimeUs() const; 953 virtual void cacheParameters_l(); 954 955 // threadLoop snippets 956 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 957 virtual void threadLoop_mix(); 958 virtual void threadLoop_sleepTime(); 959 virtual void threadLoop_exit(); 960 virtual bool shouldStandby_l(); 961 962 virtual void onAddNewTrack_l(); 963 964 // volumes last sent to audio HAL with stream->set_volume() 965 float mLeftVolFloat; 966 float mRightVolFloat; 967 968 DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 969 audio_io_handle_t id, uint32_t device, ThreadBase::type_t type, 970 bool systemReady, uint32_t bitRate = 0); 971 void processVolume_l(Track *track, bool lastTrack); 972 973 // prepareTracks_l() tells threadLoop_mix() the name of the single active track 974 sp<Track> mActiveTrack; 975 976 wp<Track> mPreviousTrack; // used to detect track switch 977 978public: 979 virtual bool hasFastMixer() const { return false; } 980}; 981 982class OffloadThread : public DirectOutputThread { 983public: 984 985 OffloadThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 986 audio_io_handle_t id, uint32_t device, 987 bool systemReady, uint32_t bitRate); 988 virtual ~OffloadThread() {}; 989 virtual void flushHw_l(); 990 991protected: 992 // threadLoop snippets 993 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 994 virtual void threadLoop_exit(); 995 996 virtual uint32_t activeSleepTimeUs() const; 997 998 virtual bool waitingAsyncCallback(); 999 virtual bool waitingAsyncCallback_l(); 1000 1001 virtual bool keepWakeLock() const { return mKeepWakeLock; } 1002 1003private: 1004 size_t mPausedWriteLength; // length in bytes of write interrupted by pause 1005 size_t mPausedBytesRemaining; // bytes still waiting in mixbuffer after resume 1006 bool mKeepWakeLock; // keep wake lock while waiting for write callback 1007}; 1008 1009class AsyncCallbackThread : public Thread { 1010public: 1011 1012 AsyncCallbackThread(const wp<PlaybackThread>& playbackThread); 1013 1014 virtual ~AsyncCallbackThread(); 1015 1016 // Thread virtuals 1017 virtual bool threadLoop(); 1018 1019 // RefBase 1020 virtual void onFirstRef(); 1021 1022 void exit(); 1023 void setWriteBlocked(uint32_t sequence); 1024 void resetWriteBlocked(); 1025 void setDraining(uint32_t sequence); 1026 void resetDraining(); 1027 1028private: 1029 const wp<PlaybackThread> mPlaybackThread; 1030 // mWriteAckSequence corresponds to the last write sequence passed by the offload thread via 1031 // setWriteBlocked(). The sequence is shifted one bit to the left and the lsb is used 1032 // to indicate that the callback has been received via resetWriteBlocked() 1033 uint32_t mWriteAckSequence; 1034 // mDrainSequence corresponds to the last drain sequence passed by the offload thread via 1035 // setDraining(). The sequence is shifted one bit to the left and the lsb is used 1036 // to indicate that the callback has been received via resetDraining() 1037 uint32_t mDrainSequence; 1038 Condition mWaitWorkCV; 1039 Mutex mLock; 1040}; 1041 1042class DuplicatingThread : public MixerThread { 1043public: 1044 DuplicatingThread(const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread, 1045 audio_io_handle_t id, bool systemReady); 1046 virtual ~DuplicatingThread(); 1047 1048 // Thread virtuals 1049 void addOutputTrack(MixerThread* thread); 1050 void removeOutputTrack(MixerThread* thread); 1051 uint32_t waitTimeMs() const { return mWaitTimeMs; } 1052protected: 1053 virtual uint32_t activeSleepTimeUs() const; 1054 1055private: 1056 bool outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks); 1057protected: 1058 // threadLoop snippets 1059 virtual void threadLoop_mix(); 1060 virtual void threadLoop_sleepTime(); 1061 virtual ssize_t threadLoop_write(); 1062 virtual void threadLoop_standby(); 1063 virtual void cacheParameters_l(); 1064 1065private: 1066 // called from threadLoop, addOutputTrack, removeOutputTrack 1067 virtual void updateWaitTime_l(); 1068protected: 1069 virtual void saveOutputTracks(); 1070 virtual void clearOutputTracks(); 1071private: 1072 1073 uint32_t mWaitTimeMs; 1074 SortedVector < sp<OutputTrack> > outputTracks; 1075 SortedVector < sp<OutputTrack> > mOutputTracks; 1076public: 1077 virtual bool hasFastMixer() const { return false; } 1078}; 1079 1080 1081// record thread 1082class RecordThread : public ThreadBase 1083{ 1084public: 1085 1086 class RecordTrack; 1087 1088 /* The ResamplerBufferProvider is used to retrieve recorded input data from the 1089 * RecordThread. It maintains local state on the relative position of the read 1090 * position of the RecordTrack compared with the RecordThread. 1091 */ 1092 class ResamplerBufferProvider : public AudioBufferProvider 1093 { 1094 public: 1095 ResamplerBufferProvider(RecordTrack* recordTrack) : 1096 mRecordTrack(recordTrack), 1097 mRsmpInUnrel(0), mRsmpInFront(0) { } 1098 virtual ~ResamplerBufferProvider() { } 1099 1100 // called to set the ResamplerBufferProvider to head of the RecordThread data buffer, 1101 // skipping any previous data read from the hal. 1102 virtual void reset(); 1103 1104 /* Synchronizes RecordTrack position with the RecordThread. 1105 * Calculates available frames and handle overruns if the RecordThread 1106 * has advanced faster than the ResamplerBufferProvider has retrieved data. 1107 * TODO: why not do this for every getNextBuffer? 1108 * 1109 * Parameters 1110 * framesAvailable: pointer to optional output size_t to store record track 1111 * frames available. 1112 * hasOverrun: pointer to optional boolean, returns true if track has overrun. 1113 */ 1114 1115 virtual void sync(size_t *framesAvailable = NULL, bool *hasOverrun = NULL); 1116 1117 // AudioBufferProvider interface 1118 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer); 1119 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 1120 private: 1121 RecordTrack * const mRecordTrack; 1122 size_t mRsmpInUnrel; // unreleased frames remaining from 1123 // most recent getNextBuffer 1124 // for debug only 1125 int32_t mRsmpInFront; // next available frame 1126 // rolling counter that is never cleared 1127 }; 1128 1129 /* The RecordBufferConverter is used for format, channel, and sample rate 1130 * conversion for a RecordTrack. 1131 * 1132 * TODO: Self contained, so move to a separate file later. 1133 * 1134 * RecordBufferConverter uses the convert() method rather than exposing a 1135 * buffer provider interface; this is to save a memory copy. 1136 */ 1137 class RecordBufferConverter 1138 { 1139 public: 1140 RecordBufferConverter( 1141 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 1142 uint32_t srcSampleRate, 1143 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 1144 uint32_t dstSampleRate); 1145 1146 ~RecordBufferConverter(); 1147 1148 /* Converts input data from an AudioBufferProvider by format, channelMask, 1149 * and sampleRate to a destination buffer. 1150 * 1151 * Parameters 1152 * dst: buffer to place the converted data. 1153 * provider: buffer provider to obtain source data. 1154 * frames: number of frames to convert 1155 * 1156 * Returns the number of frames converted. 1157 */ 1158 size_t convert(void *dst, AudioBufferProvider *provider, size_t frames); 1159 1160 // returns NO_ERROR if constructor was successful 1161 status_t initCheck() const { 1162 // mSrcChannelMask set on successful updateParameters 1163 return mSrcChannelMask != AUDIO_CHANNEL_INVALID ? NO_ERROR : NO_INIT; 1164 } 1165 1166 // allows dynamic reconfigure of all parameters 1167 status_t updateParameters( 1168 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 1169 uint32_t srcSampleRate, 1170 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 1171 uint32_t dstSampleRate); 1172 1173 // called to reset resampler buffers on record track discontinuity 1174 void reset() { 1175 if (mResampler != NULL) { 1176 mResampler->reset(); 1177 } 1178 } 1179 1180 private: 1181 // format conversion when not using resampler 1182 void convertNoResampler(void *dst, const void *src, size_t frames); 1183 1184 // format conversion when using resampler; modifies src in-place 1185 void convertResampler(void *dst, /*not-a-const*/ void *src, size_t frames); 1186 1187 // user provided information 1188 audio_channel_mask_t mSrcChannelMask; 1189 audio_format_t mSrcFormat; 1190 uint32_t mSrcSampleRate; 1191 audio_channel_mask_t mDstChannelMask; 1192 audio_format_t mDstFormat; 1193 uint32_t mDstSampleRate; 1194 1195 // derived information 1196 uint32_t mSrcChannelCount; 1197 uint32_t mDstChannelCount; 1198 size_t mDstFrameSize; 1199 1200 // format conversion buffer 1201 void *mBuf; 1202 size_t mBufFrames; 1203 size_t mBufFrameSize; 1204 1205 // resampler info 1206 AudioResampler *mResampler; 1207 1208 bool mIsLegacyDownmix; // legacy stereo to mono conversion needed 1209 bool mIsLegacyUpmix; // legacy mono to stereo conversion needed 1210 bool mRequiresFloat; // data processing requires float (e.g. resampler) 1211 PassthruBufferProvider *mInputConverterProvider; // converts input to float 1212 int8_t mIdxAry[sizeof(uint32_t) * 8]; // used for channel mask conversion 1213 }; 1214 1215#include "RecordTracks.h" 1216 1217 RecordThread(const sp<AudioFlinger>& audioFlinger, 1218 AudioStreamIn *input, 1219 audio_io_handle_t id, 1220 audio_devices_t outDevice, 1221 audio_devices_t inDevice, 1222 bool systemReady 1223#ifdef TEE_SINK 1224 , const sp<NBAIO_Sink>& teeSink 1225#endif 1226 ); 1227 virtual ~RecordThread(); 1228 1229 // no addTrack_l ? 1230 void destroyTrack_l(const sp<RecordTrack>& track); 1231 void removeTrack_l(const sp<RecordTrack>& track); 1232 1233 void dumpInternals(int fd, const Vector<String16>& args); 1234 void dumpTracks(int fd, const Vector<String16>& args); 1235 1236 // Thread virtuals 1237 virtual bool threadLoop(); 1238 1239 // RefBase 1240 virtual void onFirstRef(); 1241 1242 virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; } 1243 1244 virtual sp<MemoryDealer> readOnlyHeap() const { return mReadOnlyHeap; } 1245 1246 virtual sp<IMemory> pipeMemory() const { return mPipeMemory; } 1247 1248 sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l( 1249 const sp<AudioFlinger::Client>& client, 1250 uint32_t sampleRate, 1251 audio_format_t format, 1252 audio_channel_mask_t channelMask, 1253 size_t *pFrameCount, 1254 audio_session_t sessionId, 1255 size_t *notificationFrames, 1256 int uid, 1257 IAudioFlinger::track_flags_t *flags, 1258 pid_t tid, 1259 status_t *status /*non-NULL*/); 1260 1261 status_t start(RecordTrack* recordTrack, 1262 AudioSystem::sync_event_t event, 1263 audio_session_t triggerSession); 1264 1265 // ask the thread to stop the specified track, and 1266 // return true if the caller should then do it's part of the stopping process 1267 bool stop(RecordTrack* recordTrack); 1268 1269 void dump(int fd, const Vector<String16>& args); 1270 AudioStreamIn* clearInput(); 1271 virtual audio_stream_t* stream() const; 1272 1273 1274 virtual bool checkForNewParameter_l(const String8& keyValuePair, 1275 status_t& status); 1276 virtual void cacheParameters_l() {} 1277 virtual String8 getParameters(const String8& keys); 1278 virtual void ioConfigChanged(audio_io_config_event event, pid_t pid = 0); 1279 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 1280 audio_patch_handle_t *handle); 1281 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 1282 1283 void addPatchRecord(const sp<PatchRecord>& record); 1284 void deletePatchRecord(const sp<PatchRecord>& record); 1285 1286 void readInputParameters_l(); 1287 virtual uint32_t getInputFramesLost(); 1288 1289 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1290 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1291 virtual uint32_t hasAudioSession(audio_session_t sessionId) const; 1292 1293 // Return the set of unique session IDs across all tracks. 1294 // The keys are the session IDs, and the associated values are meaningless. 1295 // FIXME replace by Set [and implement Bag/Multiset for other uses]. 1296 KeyedVector<audio_session_t, bool> sessionIds() const; 1297 1298 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 1299 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 1300 1301 static void syncStartEventCallback(const wp<SyncEvent>& event); 1302 1303 virtual size_t frameCount() const { return mFrameCount; } 1304 bool hasFastCapture() const { return mFastCapture != 0; } 1305 virtual void getAudioPortConfig(struct audio_port_config *config); 1306 1307private: 1308 // Enter standby if not already in standby, and set mStandby flag 1309 void standbyIfNotAlreadyInStandby(); 1310 1311 // Call the HAL standby method unconditionally, and don't change mStandby flag 1312 void inputStandBy(); 1313 1314 AudioStreamIn *mInput; 1315 SortedVector < sp<RecordTrack> > mTracks; 1316 // mActiveTracks has dual roles: it indicates the current active track(s), and 1317 // is used together with mStartStopCond to indicate start()/stop() progress 1318 SortedVector< sp<RecordTrack> > mActiveTracks; 1319 // generation counter for mActiveTracks 1320 int mActiveTracksGen; 1321 Condition mStartStopCond; 1322 1323 // resampler converts input at HAL Hz to output at AudioRecord client Hz 1324 void *mRsmpInBuffer; // 1325 size_t mRsmpInFrames; // size of resampler input in frames 1326 size_t mRsmpInFramesP2;// size rounded up to a power-of-2 1327 1328 // rolling index that is never cleared 1329 int32_t mRsmpInRear; // last filled frame + 1 1330 1331 // For dumpsys 1332 const sp<NBAIO_Sink> mTeeSink; 1333 1334 const sp<MemoryDealer> mReadOnlyHeap; 1335 1336 // one-time initialization, no locks required 1337 sp<FastCapture> mFastCapture; // non-0 if there is also 1338 // a fast capture 1339 1340 // FIXME audio watchdog thread 1341 1342 // contents are not guaranteed to be consistent, no locks required 1343 FastCaptureDumpState mFastCaptureDumpState; 1344#ifdef STATE_QUEUE_DUMP 1345 // FIXME StateQueue observer and mutator dump fields 1346#endif 1347 // FIXME audio watchdog dump 1348 1349 // accessible only within the threadLoop(), no locks required 1350 // mFastCapture->sq() // for mutating and pushing state 1351 int32_t mFastCaptureFutex; // for cold idle 1352 1353 // The HAL input source is treated as non-blocking, 1354 // but current implementation is blocking 1355 sp<NBAIO_Source> mInputSource; 1356 // The source for the normal capture thread to read from: mInputSource or mPipeSource 1357 sp<NBAIO_Source> mNormalSource; 1358 // If a fast capture is present, the non-blocking pipe sink written to by fast capture, 1359 // otherwise clear 1360 sp<NBAIO_Sink> mPipeSink; 1361 // If a fast capture is present, the non-blocking pipe source read by normal thread, 1362 // otherwise clear 1363 sp<NBAIO_Source> mPipeSource; 1364 // Depth of pipe from fast capture to normal thread and fast clients, always power of 2 1365 size_t mPipeFramesP2; 1366 // If a fast capture is present, the Pipe as IMemory, otherwise clear 1367 sp<IMemory> mPipeMemory; 1368 1369 static const size_t kFastCaptureLogSize = 4 * 1024; 1370 sp<NBLog::Writer> mFastCaptureNBLogWriter; 1371 1372 bool mFastTrackAvail; // true if fast track available 1373}; 1374