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 22 class ThreadBase : public Thread { 23 public: 24 25 #include "TrackBase.h" 26 27 enum type_t { 28 MIXER, // Thread class is MixerThread 29 DIRECT, // Thread class is DirectOutputThread 30 DUPLICATING, // Thread class is DuplicatingThread 31 RECORD, // Thread class is RecordThread 32 OFFLOAD // Thread class is OffloadThread 33 }; 34 35 static const char *threadTypeToString(type_t type); 36 37 ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 38 audio_devices_t outDevice, audio_devices_t inDevice, type_t type, 39 bool systemReady); 40 virtual ~ThreadBase(); 41 42 virtual status_t readyToRun(); 43 44 void dumpBase(int fd, const Vector<String16>& args); 45 void dumpEffectChains(int fd, const Vector<String16>& args); 46 47 void clearPowerManager(); 48 49 // base for record and playback 50 enum { 51 CFG_EVENT_IO, 52 CFG_EVENT_PRIO, 53 CFG_EVENT_SET_PARAMETER, 54 CFG_EVENT_CREATE_AUDIO_PATCH, 55 CFG_EVENT_RELEASE_AUDIO_PATCH, 56 }; 57 58 class ConfigEventData: public RefBase { 59 public: 60 virtual ~ConfigEventData() {} 61 62 virtual void dump(char *buffer, size_t size) = 0; 63 protected: 64 ConfigEventData() {} 65 }; 66 67 // Config event sequence by client if status needed (e.g binder thread calling setParameters()): 68 // 1. create SetParameterConfigEvent. This sets mWaitStatus in config event 69 // 2. Lock mLock 70 // 3. Call sendConfigEvent_l(): Append to mConfigEvents and mWaitWorkCV.signal 71 // 4. sendConfigEvent_l() reads status from event->mStatus; 72 // 5. sendConfigEvent_l() returns status 73 // 6. Unlock 74 // 75 // Parameter sequence by server: threadLoop calling processConfigEvents_l(): 76 // 1. Lock mLock 77 // 2. If there is an entry in mConfigEvents proceed ... 78 // 3. Read first entry in mConfigEvents 79 // 4. Remove first entry from mConfigEvents 80 // 5. Process 81 // 6. Set event->mStatus 82 // 7. event->mCond.signal 83 // 8. Unlock 84 85 class ConfigEvent: public RefBase { 86 public: 87 virtual ~ConfigEvent() {} 88 89 void dump(char *buffer, size_t size) { mData->dump(buffer, size); } 90 91 const int mType; // event type e.g. CFG_EVENT_IO 92 Mutex mLock; // mutex associated with mCond 93 Condition mCond; // condition for status return 94 status_t mStatus; // status communicated to sender 95 bool mWaitStatus; // true if sender is waiting for status 96 bool mRequiresSystemReady; // true if must wait for system ready to enter event queue 97 sp<ConfigEventData> mData; // event specific parameter data 98 99 protected: 100 ConfigEvent(int type, bool requiresSystemReady = false) : 101 mType(type), mStatus(NO_ERROR), mWaitStatus(false), 102 mRequiresSystemReady(requiresSystemReady), mData(NULL) {} 103 }; 104 105 class IoConfigEventData : public ConfigEventData { 106 public: 107 IoConfigEventData(audio_io_config_event event, pid_t pid) : 108 mEvent(event), mPid(pid) {} 109 110 virtual void dump(char *buffer, size_t size) { 111 snprintf(buffer, size, "IO event: event %d\n", mEvent); 112 } 113 114 const audio_io_config_event mEvent; 115 const pid_t mPid; 116 }; 117 118 class IoConfigEvent : public ConfigEvent { 119 public: 120 IoConfigEvent(audio_io_config_event event, pid_t pid) : 121 ConfigEvent(CFG_EVENT_IO) { 122 mData = new IoConfigEventData(event, pid); 123 } 124 virtual ~IoConfigEvent() {} 125 }; 126 127 class PrioConfigEventData : public ConfigEventData { 128 public: 129 PrioConfigEventData(pid_t pid, pid_t tid, int32_t prio) : 130 mPid(pid), mTid(tid), mPrio(prio) {} 131 132 virtual void dump(char *buffer, size_t size) { 133 snprintf(buffer, size, "Prio event: pid %d, tid %d, prio %d\n", mPid, mTid, mPrio); 134 } 135 136 const pid_t mPid; 137 const pid_t mTid; 138 const int32_t mPrio; 139 }; 140 141 class PrioConfigEvent : public ConfigEvent { 142 public: 143 PrioConfigEvent(pid_t pid, pid_t tid, int32_t prio) : 144 ConfigEvent(CFG_EVENT_PRIO, true) { 145 mData = new PrioConfigEventData(pid, tid, prio); 146 } 147 virtual ~PrioConfigEvent() {} 148 }; 149 150 class SetParameterConfigEventData : public ConfigEventData { 151 public: 152 SetParameterConfigEventData(String8 keyValuePairs) : 153 mKeyValuePairs(keyValuePairs) {} 154 155 virtual void dump(char *buffer, size_t size) { 156 snprintf(buffer, size, "KeyValue: %s\n", mKeyValuePairs.string()); 157 } 158 159 const String8 mKeyValuePairs; 160 }; 161 162 class SetParameterConfigEvent : public ConfigEvent { 163 public: 164 SetParameterConfigEvent(String8 keyValuePairs) : 165 ConfigEvent(CFG_EVENT_SET_PARAMETER) { 166 mData = new SetParameterConfigEventData(keyValuePairs); 167 mWaitStatus = true; 168 } 169 virtual ~SetParameterConfigEvent() {} 170 }; 171 172 class CreateAudioPatchConfigEventData : public ConfigEventData { 173 public: 174 CreateAudioPatchConfigEventData(const struct audio_patch patch, 175 audio_patch_handle_t handle) : 176 mPatch(patch), mHandle(handle) {} 177 178 virtual void dump(char *buffer, size_t size) { 179 snprintf(buffer, size, "Patch handle: %u\n", mHandle); 180 } 181 182 const struct audio_patch mPatch; 183 audio_patch_handle_t mHandle; 184 }; 185 186 class CreateAudioPatchConfigEvent : public ConfigEvent { 187 public: 188 CreateAudioPatchConfigEvent(const struct audio_patch patch, 189 audio_patch_handle_t handle) : 190 ConfigEvent(CFG_EVENT_CREATE_AUDIO_PATCH) { 191 mData = new CreateAudioPatchConfigEventData(patch, handle); 192 mWaitStatus = true; 193 } 194 virtual ~CreateAudioPatchConfigEvent() {} 195 }; 196 197 class ReleaseAudioPatchConfigEventData : public ConfigEventData { 198 public: 199 ReleaseAudioPatchConfigEventData(const audio_patch_handle_t handle) : 200 mHandle(handle) {} 201 202 virtual void dump(char *buffer, size_t size) { 203 snprintf(buffer, size, "Patch handle: %u\n", mHandle); 204 } 205 206 audio_patch_handle_t mHandle; 207 }; 208 209 class ReleaseAudioPatchConfigEvent : public ConfigEvent { 210 public: 211 ReleaseAudioPatchConfigEvent(const audio_patch_handle_t handle) : 212 ConfigEvent(CFG_EVENT_RELEASE_AUDIO_PATCH) { 213 mData = new ReleaseAudioPatchConfigEventData(handle); 214 mWaitStatus = true; 215 } 216 virtual ~ReleaseAudioPatchConfigEvent() {} 217 }; 218 219 class PMDeathRecipient : public IBinder::DeathRecipient { 220 public: 221 PMDeathRecipient(const wp<ThreadBase>& thread) : mThread(thread) {} 222 virtual ~PMDeathRecipient() {} 223 224 // IBinder::DeathRecipient 225 virtual void binderDied(const wp<IBinder>& who); 226 227 private: 228 PMDeathRecipient(const PMDeathRecipient&); 229 PMDeathRecipient& operator = (const PMDeathRecipient&); 230 231 wp<ThreadBase> mThread; 232 }; 233 234 virtual status_t initCheck() const = 0; 235 236 // static externally-visible 237 type_t type() const { return mType; } 238 bool isDuplicating() const { return (mType == DUPLICATING); } 239 240 audio_io_handle_t id() const { return mId;} 241 242 // dynamic externally-visible 243 uint32_t sampleRate() const { return mSampleRate; } 244 audio_channel_mask_t channelMask() const { return mChannelMask; } 245 audio_format_t format() const { return mHALFormat; } 246 uint32_t channelCount() const { return mChannelCount; } 247 // Called by AudioFlinger::frameCount(audio_io_handle_t output) and effects, 248 // and returns the [normal mix] buffer's frame count. 249 virtual size_t frameCount() const = 0; 250 251 // Return's the HAL's frame count i.e. fast mixer buffer size. 252 size_t frameCountHAL() const { return mFrameCount; } 253 254 size_t frameSize() const { return mFrameSize; } 255 256 // Should be "virtual status_t requestExitAndWait()" and override same 257 // method in Thread, but Thread::requestExitAndWait() is not yet virtual. 258 void exit(); 259 virtual bool checkForNewParameter_l(const String8& keyValuePair, 260 status_t& status) = 0; 261 virtual status_t setParameters(const String8& keyValuePairs); 262 virtual String8 getParameters(const String8& keys) = 0; 263 virtual void ioConfigChanged(audio_io_config_event event, pid_t pid = 0) = 0; 264 // sendConfigEvent_l() must be called with ThreadBase::mLock held 265 // Can temporarily release the lock if waiting for a reply from 266 // processConfigEvents_l(). 267 status_t sendConfigEvent_l(sp<ConfigEvent>& event); 268 void sendIoConfigEvent(audio_io_config_event event, pid_t pid = 0); 269 void sendIoConfigEvent_l(audio_io_config_event event, pid_t pid = 0); 270 void sendPrioConfigEvent(pid_t pid, pid_t tid, int32_t prio); 271 void sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio); 272 status_t sendSetParameterConfigEvent_l(const String8& keyValuePair); 273 status_t sendCreateAudioPatchConfigEvent(const struct audio_patch *patch, 274 audio_patch_handle_t *handle); 275 status_t sendReleaseAudioPatchConfigEvent(audio_patch_handle_t handle); 276 void processConfigEvents_l(); 277 virtual void cacheParameters_l() = 0; 278 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 279 audio_patch_handle_t *handle) = 0; 280 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle) = 0; 281 virtual void getAudioPortConfig(struct audio_port_config *config) = 0; 282 283 284 // see note at declaration of mStandby, mOutDevice and mInDevice 285 bool standby() const { return mStandby; } 286 audio_devices_t outDevice() const { return mOutDevice; } 287 audio_devices_t inDevice() const { return mInDevice; } 288 289 virtual audio_stream_t* stream() const = 0; 290 291 sp<EffectHandle> createEffect_l( 292 const sp<AudioFlinger::Client>& client, 293 const sp<IEffectClient>& effectClient, 294 int32_t priority, 295 audio_session_t sessionId, 296 effect_descriptor_t *desc, 297 int *enabled, 298 status_t *status /*non-NULL*/); 299 300 // return values for hasAudioSession (bit field) 301 enum effect_state { 302 EFFECT_SESSION = 0x1, // the audio session corresponds to at least one 303 // effect 304 TRACK_SESSION = 0x2, // the audio session corresponds to at least one 305 // track 306 FAST_SESSION = 0x4 // the audio session corresponds to at least one 307 // fast track 308 }; 309 310 // get effect chain corresponding to session Id. 311 sp<EffectChain> getEffectChain(audio_session_t sessionId); 312 // same as getEffectChain() but must be called with ThreadBase mutex locked 313 sp<EffectChain> getEffectChain_l(audio_session_t sessionId) const; 314 // add an effect chain to the chain list (mEffectChains) 315 virtual status_t addEffectChain_l(const sp<EffectChain>& chain) = 0; 316 // remove an effect chain from the chain list (mEffectChains) 317 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain) = 0; 318 // lock all effect chains Mutexes. Must be called before releasing the 319 // ThreadBase mutex before processing the mixer and effects. This guarantees the 320 // integrity of the chains during the process. 321 // Also sets the parameter 'effectChains' to current value of mEffectChains. 322 void lockEffectChains_l(Vector< sp<EffectChain> >& effectChains); 323 // unlock effect chains after process 324 void unlockEffectChains(const Vector< sp<EffectChain> >& effectChains); 325 // get a copy of mEffectChains vector 326 Vector< sp<EffectChain> > getEffectChains_l() const { return mEffectChains; }; 327 // set audio mode to all effect chains 328 void setMode(audio_mode_t mode); 329 // get effect module with corresponding ID on specified audio session 330 sp<AudioFlinger::EffectModule> getEffect(audio_session_t sessionId, int effectId); 331 sp<AudioFlinger::EffectModule> getEffect_l(audio_session_t sessionId, int effectId); 332 // add and effect module. Also creates the effect chain is none exists for 333 // the effects audio session 334 status_t addEffect_l(const sp< EffectModule>& effect); 335 // remove and effect module. Also removes the effect chain is this was the last 336 // effect 337 void removeEffect_l(const sp< EffectModule>& effect); 338 // detach all tracks connected to an auxiliary effect 339 virtual void detachAuxEffect_l(int effectId __unused) {} 340 // returns a combination of: 341 // - EFFECT_SESSION if effects on this audio session exist in one chain 342 // - TRACK_SESSION if tracks on this audio session exist 343 // - FAST_SESSION if fast tracks on this audio session exist 344 virtual uint32_t hasAudioSession_l(audio_session_t sessionId) const = 0; 345 uint32_t hasAudioSession(audio_session_t sessionId) const { 346 Mutex::Autolock _l(mLock); 347 return hasAudioSession_l(sessionId); 348 } 349 350 // the value returned by default implementation is not important as the 351 // strategy is only meaningful for PlaybackThread which implements this method 352 virtual uint32_t getStrategyForSession_l(audio_session_t sessionId __unused) 353 { return 0; } 354 355 // suspend or restore effect according to the type of effect passed. a NULL 356 // type pointer means suspend all effects in the session 357 void setEffectSuspended(const effect_uuid_t *type, 358 bool suspend, 359 audio_session_t sessionId = AUDIO_SESSION_OUTPUT_MIX); 360 // check if some effects must be suspended/restored when an effect is enabled 361 // or disabled 362 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 363 bool enabled, 364 audio_session_t sessionId = 365 AUDIO_SESSION_OUTPUT_MIX); 366 void checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 367 bool enabled, 368 audio_session_t sessionId = 369 AUDIO_SESSION_OUTPUT_MIX); 370 371 virtual status_t setSyncEvent(const sp<SyncEvent>& event) = 0; 372 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const = 0; 373 374 // Return a reference to a per-thread heap which can be used to allocate IMemory 375 // objects that will be read-only to client processes, read/write to mediaserver, 376 // and shared by all client processes of the thread. 377 // The heap is per-thread rather than common across all threads, because 378 // clients can't be trusted not to modify the offset of the IMemory they receive. 379 // If a thread does not have such a heap, this method returns 0. 380 virtual sp<MemoryDealer> readOnlyHeap() const { return 0; } 381 382 virtual sp<IMemory> pipeMemory() const { return 0; } 383 384 void systemReady(); 385 386 // checkEffectCompatibility_l() must be called with ThreadBase::mLock held 387 virtual status_t checkEffectCompatibility_l(const effect_descriptor_t *desc, 388 audio_session_t sessionId) = 0; 389 390 mutable Mutex mLock; 391 392 protected: 393 394 // entry describing an effect being suspended in mSuspendedSessions keyed vector 395 class SuspendedSessionDesc : public RefBase { 396 public: 397 SuspendedSessionDesc() : mRefCount(0) {} 398 399 int mRefCount; // number of active suspend requests 400 effect_uuid_t mType; // effect type UUID 401 }; 402 403 void acquireWakeLock(int uid = -1); 404 virtual void acquireWakeLock_l(int uid = -1); 405 void releaseWakeLock(); 406 void releaseWakeLock_l(); 407 void updateWakeLockUids(const SortedVector<int> &uids); 408 void updateWakeLockUids_l(const SortedVector<int> &uids); 409 void getPowerManager_l(); 410 void setEffectSuspended_l(const effect_uuid_t *type, 411 bool suspend, 412 audio_session_t sessionId); 413 // updated mSuspendedSessions when an effect suspended or restored 414 void updateSuspendedSessions_l(const effect_uuid_t *type, 415 bool suspend, 416 audio_session_t sessionId); 417 // check if some effects must be suspended when an effect chain is added 418 void checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain); 419 420 String16 getWakeLockTag(); 421 422 virtual void preExit() { } 423 virtual void setMasterMono_l(bool mono __unused) { } 424 virtual bool requireMonoBlend() { return false; } 425 426 friend class AudioFlinger; // for mEffectChains 427 428 const type_t mType; 429 430 // Used by parameters, config events, addTrack_l, exit 431 Condition mWaitWorkCV; 432 433 const sp<AudioFlinger> mAudioFlinger; 434 435 // updated by PlaybackThread::readOutputParameters_l() or 436 // RecordThread::readInputParameters_l() 437 uint32_t mSampleRate; 438 size_t mFrameCount; // output HAL, direct output, record 439 audio_channel_mask_t mChannelMask; 440 uint32_t mChannelCount; 441 size_t mFrameSize; 442 // not HAL frame size, this is for output sink (to pipe to fast mixer) 443 audio_format_t mFormat; // Source format for Recording and 444 // Sink format for Playback. 445 // Sink format may be different than 446 // HAL format if Fastmixer is used. 447 audio_format_t mHALFormat; 448 size_t mBufferSize; // HAL buffer size for read() or write() 449 450 Vector< sp<ConfigEvent> > mConfigEvents; 451 Vector< sp<ConfigEvent> > mPendingConfigEvents; // events awaiting system ready 452 453 // These fields are written and read by thread itself without lock or barrier, 454 // and read by other threads without lock or barrier via standby(), outDevice() 455 // and inDevice(). 456 // Because of the absence of a lock or barrier, any other thread that reads 457 // these fields must use the information in isolation, or be prepared to deal 458 // with possibility that it might be inconsistent with other information. 459 bool mStandby; // Whether thread is currently in standby. 460 audio_devices_t mOutDevice; // output device 461 audio_devices_t mInDevice; // input device 462 audio_devices_t mPrevOutDevice; // previous output device 463 audio_devices_t mPrevInDevice; // previous input device 464 struct audio_patch mPatch; 465 audio_source_t mAudioSource; 466 467 const audio_io_handle_t mId; 468 Vector< sp<EffectChain> > mEffectChains; 469 470 static const int kThreadNameLength = 16; // prctl(PR_SET_NAME) limit 471 char mThreadName[kThreadNameLength]; // guaranteed NUL-terminated 472 sp<IPowerManager> mPowerManager; 473 sp<IBinder> mWakeLockToken; 474 const sp<PMDeathRecipient> mDeathRecipient; 475 // list of suspended effects per session and per type. The first (outer) vector is 476 // keyed by session ID, the second (inner) by type UUID timeLow field 477 KeyedVector< audio_session_t, KeyedVector< int, sp<SuspendedSessionDesc> > > 478 mSuspendedSessions; 479 static const size_t kLogSize = 4 * 1024; 480 sp<NBLog::Writer> mNBLogWriter; 481 bool mSystemReady; 482 bool mNotifiedBatteryStart; 483 ExtendedTimestamp mTimestamp; 484 }; 485 486 // --- PlaybackThread --- 487 class PlaybackThread : public ThreadBase { 488 public: 489 490 #include "PlaybackTracks.h" 491 492 enum mixer_state { 493 MIXER_IDLE, // no active tracks 494 MIXER_TRACKS_ENABLED, // at least one active track, but no track has any data ready 495 MIXER_TRACKS_READY, // at least one active track, and at least one track has data 496 MIXER_DRAIN_TRACK, // drain currently playing track 497 MIXER_DRAIN_ALL, // fully drain the hardware 498 // standby mode does not have an enum value 499 // suspend by audio policy manager is orthogonal to mixer state 500 }; 501 502 // retry count before removing active track in case of underrun on offloaded thread: 503 // we need to make sure that AudioTrack client has enough time to send large buffers 504 //FIXME may be more appropriate if expressed in time units. Need to revise how underrun is 505 // handled for offloaded tracks 506 static const int8_t kMaxTrackRetriesOffload = 20; 507 static const int8_t kMaxTrackStartupRetriesOffload = 100; 508 static const int8_t kMaxTrackStopRetriesOffload = 2; 509 // 14 tracks max per client allows for 2 misbehaving application leaving 4 available tracks. 510 static const uint32_t kMaxTracksPerUid = 14; 511 512 PlaybackThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 513 audio_io_handle_t id, audio_devices_t device, type_t type, bool systemReady); 514 virtual ~PlaybackThread(); 515 516 void dump(int fd, const Vector<String16>& args); 517 518 // Thread virtuals 519 virtual bool threadLoop(); 520 521 // RefBase 522 virtual void onFirstRef(); 523 524 virtual status_t checkEffectCompatibility_l(const effect_descriptor_t *desc, 525 audio_session_t sessionId); 526 527 protected: 528 // Code snippets that were lifted up out of threadLoop() 529 virtual void threadLoop_mix() = 0; 530 virtual void threadLoop_sleepTime() = 0; 531 virtual ssize_t threadLoop_write(); 532 virtual void threadLoop_drain(); 533 virtual void threadLoop_standby(); 534 virtual void threadLoop_exit(); 535 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 536 537 // prepareTracks_l reads and writes mActiveTracks, and returns 538 // the pending set of tracks to remove via Vector 'tracksToRemove'. The caller 539 // is responsible for clearing or destroying this Vector later on, when it 540 // is safe to do so. That will drop the final ref count and destroy the tracks. 541 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove) = 0; 542 void removeTracks_l(const Vector< sp<Track> >& tracksToRemove); 543 544 void writeCallback(); 545 void resetWriteBlocked(uint32_t sequence); 546 void drainCallback(); 547 void resetDraining(uint32_t sequence); 548 void errorCallback(); 549 550 static int asyncCallback(stream_callback_event_t event, void *param, void *cookie); 551 552 virtual bool waitingAsyncCallback(); 553 virtual bool waitingAsyncCallback_l(); 554 virtual bool shouldStandby_l(); 555 virtual void onAddNewTrack_l(); 556 void onAsyncError(); // error reported by AsyncCallbackThread 557 558 // ThreadBase virtuals 559 virtual void preExit(); 560 561 virtual bool keepWakeLock() const { return true; } 562 563 public: 564 565 virtual status_t initCheck() const { return (mOutput == NULL) ? NO_INIT : NO_ERROR; } 566 567 // return estimated latency in milliseconds, as reported by HAL 568 uint32_t latency() const; 569 // same, but lock must already be held 570 uint32_t latency_l() const; 571 572 void setMasterVolume(float value); 573 void setMasterMute(bool muted); 574 575 void setStreamVolume(audio_stream_type_t stream, float value); 576 void setStreamMute(audio_stream_type_t stream, bool muted); 577 578 float streamVolume(audio_stream_type_t stream) const; 579 580 sp<Track> createTrack_l( 581 const sp<AudioFlinger::Client>& client, 582 audio_stream_type_t streamType, 583 uint32_t sampleRate, 584 audio_format_t format, 585 audio_channel_mask_t channelMask, 586 size_t *pFrameCount, 587 const sp<IMemory>& sharedBuffer, 588 audio_session_t sessionId, 589 audio_output_flags_t *flags, 590 pid_t tid, 591 int uid, 592 status_t *status /*non-NULL*/); 593 594 AudioStreamOut* getOutput() const; 595 AudioStreamOut* clearOutput(); 596 virtual audio_stream_t* stream() const; 597 598 // a very large number of suspend() will eventually wraparound, but unlikely 599 void suspend() { (void) android_atomic_inc(&mSuspended); } 600 void restore() 601 { 602 // if restore() is done without suspend(), get back into 603 // range so that the next suspend() will operate correctly 604 if (android_atomic_dec(&mSuspended) <= 0) { 605 android_atomic_release_store(0, &mSuspended); 606 } 607 } 608 bool isSuspended() const 609 { return android_atomic_acquire_load(&mSuspended) > 0; } 610 611 virtual String8 getParameters(const String8& keys); 612 virtual void ioConfigChanged(audio_io_config_event event, pid_t pid = 0); 613 status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames); 614 // FIXME rename mixBuffer() to sinkBuffer() and remove int16_t* dependency. 615 // Consider also removing and passing an explicit mMainBuffer initialization 616 // parameter to AF::PlaybackThread::Track::Track(). 617 int16_t *mixBuffer() const { 618 return reinterpret_cast<int16_t *>(mSinkBuffer); }; 619 620 virtual void detachAuxEffect_l(int effectId); 621 status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track> track, 622 int EffectId); 623 status_t attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track> track, 624 int EffectId); 625 626 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 627 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 628 virtual uint32_t hasAudioSession_l(audio_session_t sessionId) const; 629 virtual uint32_t getStrategyForSession_l(audio_session_t sessionId); 630 631 632 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 633 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 634 635 // called with AudioFlinger lock held 636 bool invalidateTracks_l(audio_stream_type_t streamType); 637 virtual void invalidateTracks(audio_stream_type_t streamType); 638 639 virtual size_t frameCount() const { return mNormalFrameCount; } 640 641 status_t getTimestamp_l(AudioTimestamp& timestamp); 642 643 void addPatchTrack(const sp<PatchTrack>& track); 644 void deletePatchTrack(const sp<PatchTrack>& track); 645 646 virtual void getAudioPortConfig(struct audio_port_config *config); 647 648 protected: 649 // updated by readOutputParameters_l() 650 size_t mNormalFrameCount; // normal mixer and effects 651 652 bool mThreadThrottle; // throttle the thread processing 653 uint32_t mThreadThrottleTimeMs; // throttle time for MIXER threads 654 uint32_t mThreadThrottleEndMs; // notify once per throttling 655 uint32_t mHalfBufferMs; // half the buffer size in milliseconds 656 657 void* mSinkBuffer; // frame size aligned sink buffer 658 659 // TODO: 660 // Rearrange the buffer info into a struct/class with 661 // clear, copy, construction, destruction methods. 662 // 663 // mSinkBuffer also has associated with it: 664 // 665 // mSinkBufferSize: Sink Buffer Size 666 // mFormat: Sink Buffer Format 667 668 // Mixer Buffer (mMixerBuffer*) 669 // 670 // In the case of floating point or multichannel data, which is not in the 671 // sink format, it is required to accumulate in a higher precision or greater channel count 672 // buffer before downmixing or data conversion to the sink buffer. 673 674 // Set to "true" to enable the Mixer Buffer otherwise mixer output goes to sink buffer. 675 bool mMixerBufferEnabled; 676 677 // Storage, 32 byte aligned (may make this alignment a requirement later). 678 // Due to constraints on mNormalFrameCount, the buffer size is a multiple of 16 frames. 679 void* mMixerBuffer; 680 681 // Size of mMixerBuffer in bytes: mNormalFrameCount * #channels * sampsize. 682 size_t mMixerBufferSize; 683 684 // The audio format of mMixerBuffer. Set to AUDIO_FORMAT_PCM_(FLOAT|16_BIT) only. 685 audio_format_t mMixerBufferFormat; 686 687 // An internal flag set to true by MixerThread::prepareTracks_l() 688 // when mMixerBuffer contains valid data after mixing. 689 bool mMixerBufferValid; 690 691 // Effects Buffer (mEffectsBuffer*) 692 // 693 // In the case of effects data, which is not in the sink format, 694 // it is required to accumulate in a different buffer before data conversion 695 // to the sink buffer. 696 697 // Set to "true" to enable the Effects Buffer otherwise effects output goes to sink buffer. 698 bool mEffectBufferEnabled; 699 700 // Storage, 32 byte aligned (may make this alignment a requirement later). 701 // Due to constraints on mNormalFrameCount, the buffer size is a multiple of 16 frames. 702 void* mEffectBuffer; 703 704 // Size of mEffectsBuffer in bytes: mNormalFrameCount * #channels * sampsize. 705 size_t mEffectBufferSize; 706 707 // The audio format of mEffectsBuffer. Set to AUDIO_FORMAT_PCM_16_BIT only. 708 audio_format_t mEffectBufferFormat; 709 710 // An internal flag set to true by MixerThread::prepareTracks_l() 711 // when mEffectsBuffer contains valid data after mixing. 712 // 713 // When this is set, all mixer data is routed into the effects buffer 714 // for any processing (including output processing). 715 bool mEffectBufferValid; 716 717 // suspend count, > 0 means suspended. While suspended, the thread continues to pull from 718 // tracks and mix, but doesn't write to HAL. A2DP and SCO HAL implementations can't handle 719 // concurrent use of both of them, so Audio Policy Service suspends one of the threads to 720 // workaround that restriction. 721 // 'volatile' means accessed via atomic operations and no lock. 722 volatile int32_t mSuspended; 723 724 int64_t mBytesWritten; 725 int64_t mFramesWritten; // not reset on standby 726 int64_t mSuspendedFrames; // not reset on standby 727 private: 728 // mMasterMute is in both PlaybackThread and in AudioFlinger. When a 729 // PlaybackThread needs to find out if master-muted, it checks it's local 730 // copy rather than the one in AudioFlinger. This optimization saves a lock. 731 bool mMasterMute; 732 void setMasterMute_l(bool muted) { mMasterMute = muted; } 733 protected: 734 SortedVector< wp<Track> > mActiveTracks; // FIXME check if this could be sp<> 735 SortedVector<int> mWakeLockUids; 736 int mActiveTracksGeneration; 737 wp<Track> mLatestActiveTrack; // latest track added to mActiveTracks 738 739 // Allocate a track name for a given channel mask. 740 // Returns name >= 0 if successful, -1 on failure. 741 virtual int getTrackName_l(audio_channel_mask_t channelMask, audio_format_t format, 742 audio_session_t sessionId, uid_t uid) = 0; 743 virtual void deleteTrackName_l(int name) = 0; 744 745 // Time to sleep between cycles when: 746 virtual uint32_t activeSleepTimeUs() const; // mixer state MIXER_TRACKS_ENABLED 747 virtual uint32_t idleSleepTimeUs() const = 0; // mixer state MIXER_IDLE 748 virtual uint32_t suspendSleepTimeUs() const = 0; // audio policy manager suspended us 749 // No sleep when mixer state == MIXER_TRACKS_READY; relies on audio HAL stream->write() 750 // No sleep in standby mode; waits on a condition 751 752 // Code snippets that are temporarily lifted up out of threadLoop() until the merge 753 void checkSilentMode_l(); 754 755 // Non-trivial for DUPLICATING only 756 virtual void saveOutputTracks() { } 757 virtual void clearOutputTracks() { } 758 759 // Cache various calculated values, at threadLoop() entry and after a parameter change 760 virtual void cacheParameters_l(); 761 762 virtual uint32_t correctLatency_l(uint32_t latency) const; 763 764 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 765 audio_patch_handle_t *handle); 766 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 767 768 bool usesHwAvSync() const { return (mType == DIRECT) && (mOutput != NULL) 769 && mHwSupportsPause 770 && (mOutput->flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC); } 771 772 uint32_t trackCountForUid_l(uid_t uid); 773 774 private: 775 776 friend class AudioFlinger; // for numerous 777 778 PlaybackThread& operator = (const PlaybackThread&); 779 780 status_t addTrack_l(const sp<Track>& track); 781 bool destroyTrack_l(const sp<Track>& track); 782 void removeTrack_l(const sp<Track>& track); 783 void broadcast_l(); 784 785 void readOutputParameters_l(); 786 787 virtual void dumpInternals(int fd, const Vector<String16>& args); 788 void dumpTracks(int fd, const Vector<String16>& args); 789 790 SortedVector< sp<Track> > mTracks; 791 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 792 AudioStreamOut *mOutput; 793 794 float mMasterVolume; 795 nsecs_t mLastWriteTime; 796 int mNumWrites; 797 int mNumDelayedWrites; 798 bool mInWrite; 799 800 // FIXME rename these former local variables of threadLoop to standard "m" names 801 nsecs_t mStandbyTimeNs; 802 size_t mSinkBufferSize; 803 804 // cached copies of activeSleepTimeUs() and idleSleepTimeUs() made by cacheParameters_l() 805 uint32_t mActiveSleepTimeUs; 806 uint32_t mIdleSleepTimeUs; 807 808 uint32_t mSleepTimeUs; 809 810 // mixer status returned by prepareTracks_l() 811 mixer_state mMixerStatus; // current cycle 812 // previous cycle when in prepareTracks_l() 813 mixer_state mMixerStatusIgnoringFastTracks; 814 // FIXME or a separate ready state per track 815 816 // FIXME move these declarations into the specific sub-class that needs them 817 // MIXER only 818 uint32_t sleepTimeShift; 819 820 // same as AudioFlinger::mStandbyTimeInNsecs except for DIRECT which uses a shorter value 821 nsecs_t mStandbyDelayNs; 822 823 // MIXER only 824 nsecs_t maxPeriod; 825 826 // DUPLICATING only 827 uint32_t writeFrames; 828 829 size_t mBytesRemaining; 830 size_t mCurrentWriteLength; 831 bool mUseAsyncWrite; 832 // mWriteAckSequence contains current write sequence on bits 31-1. The write sequence is 833 // incremented each time a write(), a flush() or a standby() occurs. 834 // Bit 0 is set when a write blocks and indicates a callback is expected. 835 // Bit 0 is reset by the async callback thread calling resetWriteBlocked(). Out of sequence 836 // callbacks are ignored. 837 uint32_t mWriteAckSequence; 838 // mDrainSequence contains current drain sequence on bits 31-1. The drain sequence is 839 // incremented each time a drain is requested or a flush() or standby() occurs. 840 // Bit 0 is set when the drain() command is called at the HAL and indicates a callback is 841 // expected. 842 // Bit 0 is reset by the async callback thread calling resetDraining(). Out of sequence 843 // callbacks are ignored. 844 uint32_t mDrainSequence; 845 // A condition that must be evaluated by prepareTrack_l() has changed and we must not wait 846 // for async write callback in the thread loop before evaluating it 847 bool mSignalPending; 848 sp<AsyncCallbackThread> mCallbackThread; 849 850 private: 851 // The HAL output sink is treated as non-blocking, but current implementation is blocking 852 sp<NBAIO_Sink> mOutputSink; 853 // If a fast mixer is present, the blocking pipe sink, otherwise clear 854 sp<NBAIO_Sink> mPipeSink; 855 // The current sink for the normal mixer to write it's (sub)mix, mOutputSink or mPipeSink 856 sp<NBAIO_Sink> mNormalSink; 857 #ifdef TEE_SINK 858 // For dumpsys 859 sp<NBAIO_Sink> mTeeSink; 860 sp<NBAIO_Source> mTeeSource; 861 #endif 862 uint32_t mScreenState; // cached copy of gScreenState 863 static const size_t kFastMixerLogSize = 4 * 1024; 864 sp<NBLog::Writer> mFastMixerNBLogWriter; 865 public: 866 virtual bool hasFastMixer() const = 0; 867 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex __unused) const 868 { FastTrackUnderruns dummy; return dummy; } 869 870 protected: 871 // accessed by both binder threads and within threadLoop(), lock on mutex needed 872 unsigned mFastTrackAvailMask; // bit i set if fast track [i] is available 873 bool mHwSupportsPause; 874 bool mHwPaused; 875 bool mFlushPending; 876 }; 877 878 class MixerThread : public PlaybackThread { 879 public: 880 MixerThread(const sp<AudioFlinger>& audioFlinger, 881 AudioStreamOut* output, 882 audio_io_handle_t id, 883 audio_devices_t device, 884 bool systemReady, 885 type_t type = MIXER); 886 virtual ~MixerThread(); 887 888 // Thread virtuals 889 890 virtual bool checkForNewParameter_l(const String8& keyValuePair, 891 status_t& status); 892 virtual void dumpInternals(int fd, const Vector<String16>& args); 893 894 protected: 895 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 896 virtual int getTrackName_l(audio_channel_mask_t channelMask, audio_format_t format, 897 audio_session_t sessionId, uid_t uid); 898 virtual void deleteTrackName_l(int name); 899 virtual uint32_t idleSleepTimeUs() const; 900 virtual uint32_t suspendSleepTimeUs() const; 901 virtual void cacheParameters_l(); 902 903 virtual void acquireWakeLock_l(int uid = -1) { 904 PlaybackThread::acquireWakeLock_l(uid); 905 if (hasFastMixer()) { 906 mFastMixer->setBoottimeOffset( 907 mTimestamp.mTimebaseOffset[ExtendedTimestamp::TIMEBASE_BOOTTIME]); 908 } 909 } 910 911 // threadLoop snippets 912 virtual ssize_t threadLoop_write(); 913 virtual void threadLoop_standby(); 914 virtual void threadLoop_mix(); 915 virtual void threadLoop_sleepTime(); 916 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 917 virtual uint32_t correctLatency_l(uint32_t latency) const; 918 919 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 920 audio_patch_handle_t *handle); 921 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 922 923 AudioMixer* mAudioMixer; // normal mixer 924 private: 925 // one-time initialization, no locks required 926 sp<FastMixer> mFastMixer; // non-0 if there is also a fast mixer 927 sp<AudioWatchdog> mAudioWatchdog; // non-0 if there is an audio watchdog thread 928 929 // contents are not guaranteed to be consistent, no locks required 930 FastMixerDumpState mFastMixerDumpState; 931 #ifdef STATE_QUEUE_DUMP 932 StateQueueObserverDump mStateQueueObserverDump; 933 StateQueueMutatorDump mStateQueueMutatorDump; 934 #endif 935 AudioWatchdogDump mAudioWatchdogDump; 936 937 // accessible only within the threadLoop(), no locks required 938 // mFastMixer->sq() // for mutating and pushing state 939 int32_t mFastMixerFutex; // for cold idle 940 941 std::atomic_bool mMasterMono; 942 public: 943 virtual bool hasFastMixer() const { return mFastMixer != 0; } 944 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const { 945 ALOG_ASSERT(fastIndex < FastMixerState::sMaxFastTracks); 946 return mFastMixerDumpState.mTracks[fastIndex].mUnderruns; 947 } 948 949 protected: 950 virtual void setMasterMono_l(bool mono) { 951 mMasterMono.store(mono); 952 if (mFastMixer != nullptr) { /* hasFastMixer() */ 953 mFastMixer->setMasterMono(mMasterMono); 954 } 955 } 956 // the FastMixer performs mono blend if it exists. 957 // Blending with limiter is not idempotent, 958 // and blending without limiter is idempotent but inefficient to do twice. 959 virtual bool requireMonoBlend() { return mMasterMono.load() && !hasFastMixer(); } 960 }; 961 962 class DirectOutputThread : public PlaybackThread { 963 public: 964 965 DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 966 audio_io_handle_t id, audio_devices_t device, bool systemReady); 967 virtual ~DirectOutputThread(); 968 969 // Thread virtuals 970 971 virtual bool checkForNewParameter_l(const String8& keyValuePair, 972 status_t& status); 973 virtual void flushHw_l(); 974 975 protected: 976 virtual int getTrackName_l(audio_channel_mask_t channelMask, audio_format_t format, 977 audio_session_t sessionId, uid_t uid); 978 virtual void deleteTrackName_l(int name); 979 virtual uint32_t activeSleepTimeUs() const; 980 virtual uint32_t idleSleepTimeUs() const; 981 virtual uint32_t suspendSleepTimeUs() const; 982 virtual void cacheParameters_l(); 983 984 // threadLoop snippets 985 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 986 virtual void threadLoop_mix(); 987 virtual void threadLoop_sleepTime(); 988 virtual void threadLoop_exit(); 989 virtual bool shouldStandby_l(); 990 991 virtual void onAddNewTrack_l(); 992 993 // volumes last sent to audio HAL with stream->set_volume() 994 float mLeftVolFloat; 995 float mRightVolFloat; 996 997 DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 998 audio_io_handle_t id, uint32_t device, ThreadBase::type_t type, 999 bool systemReady); 1000 void processVolume_l(Track *track, bool lastTrack); 1001 1002 // prepareTracks_l() tells threadLoop_mix() the name of the single active track 1003 sp<Track> mActiveTrack; 1004 1005 wp<Track> mPreviousTrack; // used to detect track switch 1006 1007 public: 1008 virtual bool hasFastMixer() const { return false; } 1009 }; 1010 1011 class OffloadThread : public DirectOutputThread { 1012 public: 1013 1014 OffloadThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1015 audio_io_handle_t id, uint32_t device, bool systemReady); 1016 virtual ~OffloadThread() {}; 1017 virtual void flushHw_l(); 1018 1019 protected: 1020 // threadLoop snippets 1021 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1022 virtual void threadLoop_exit(); 1023 1024 virtual bool waitingAsyncCallback(); 1025 virtual bool waitingAsyncCallback_l(); 1026 virtual void invalidateTracks(audio_stream_type_t streamType); 1027 1028 virtual bool keepWakeLock() const { return (mKeepWakeLock || (mDrainSequence & 1)); } 1029 1030 private: 1031 size_t mPausedWriteLength; // length in bytes of write interrupted by pause 1032 size_t mPausedBytesRemaining; // bytes still waiting in mixbuffer after resume 1033 bool mKeepWakeLock; // keep wake lock while waiting for write callback 1034 uint64_t mOffloadUnderrunPosition; // Current frame position for offloaded playback 1035 // used and valid only during underrun. ~0 if 1036 // no underrun has occurred during playback and 1037 // is not reset on standby. 1038 }; 1039 1040 class AsyncCallbackThread : public Thread { 1041 public: 1042 1043 AsyncCallbackThread(const wp<PlaybackThread>& playbackThread); 1044 1045 virtual ~AsyncCallbackThread(); 1046 1047 // Thread virtuals 1048 virtual bool threadLoop(); 1049 1050 // RefBase 1051 virtual void onFirstRef(); 1052 1053 void exit(); 1054 void setWriteBlocked(uint32_t sequence); 1055 void resetWriteBlocked(); 1056 void setDraining(uint32_t sequence); 1057 void resetDraining(); 1058 void setAsyncError(); 1059 1060 private: 1061 const wp<PlaybackThread> mPlaybackThread; 1062 // mWriteAckSequence corresponds to the last write sequence passed by the offload thread via 1063 // setWriteBlocked(). The sequence is shifted one bit to the left and the lsb is used 1064 // to indicate that the callback has been received via resetWriteBlocked() 1065 uint32_t mWriteAckSequence; 1066 // mDrainSequence corresponds to the last drain sequence passed by the offload thread via 1067 // setDraining(). The sequence is shifted one bit to the left and the lsb is used 1068 // to indicate that the callback has been received via resetDraining() 1069 uint32_t mDrainSequence; 1070 Condition mWaitWorkCV; 1071 Mutex mLock; 1072 bool mAsyncError; 1073 }; 1074 1075 class DuplicatingThread : public MixerThread { 1076 public: 1077 DuplicatingThread(const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread, 1078 audio_io_handle_t id, bool systemReady); 1079 virtual ~DuplicatingThread(); 1080 1081 // Thread virtuals 1082 void addOutputTrack(MixerThread* thread); 1083 void removeOutputTrack(MixerThread* thread); 1084 uint32_t waitTimeMs() const { return mWaitTimeMs; } 1085 protected: 1086 virtual uint32_t activeSleepTimeUs() const; 1087 1088 private: 1089 bool outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks); 1090 protected: 1091 // threadLoop snippets 1092 virtual void threadLoop_mix(); 1093 virtual void threadLoop_sleepTime(); 1094 virtual ssize_t threadLoop_write(); 1095 virtual void threadLoop_standby(); 1096 virtual void cacheParameters_l(); 1097 1098 private: 1099 // called from threadLoop, addOutputTrack, removeOutputTrack 1100 virtual void updateWaitTime_l(); 1101 protected: 1102 virtual void saveOutputTracks(); 1103 virtual void clearOutputTracks(); 1104 private: 1105 1106 uint32_t mWaitTimeMs; 1107 SortedVector < sp<OutputTrack> > outputTracks; 1108 SortedVector < sp<OutputTrack> > mOutputTracks; 1109 public: 1110 virtual bool hasFastMixer() const { return false; } 1111 }; 1112 1113 1114 // record thread 1115 class RecordThread : public ThreadBase 1116 { 1117 public: 1118 1119 class RecordTrack; 1120 1121 /* The ResamplerBufferProvider is used to retrieve recorded input data from the 1122 * RecordThread. It maintains local state on the relative position of the read 1123 * position of the RecordTrack compared with the RecordThread. 1124 */ 1125 class ResamplerBufferProvider : public AudioBufferProvider 1126 { 1127 public: 1128 ResamplerBufferProvider(RecordTrack* recordTrack) : 1129 mRecordTrack(recordTrack), 1130 mRsmpInUnrel(0), mRsmpInFront(0) { } 1131 virtual ~ResamplerBufferProvider() { } 1132 1133 // called to set the ResamplerBufferProvider to head of the RecordThread data buffer, 1134 // skipping any previous data read from the hal. 1135 virtual void reset(); 1136 1137 /* Synchronizes RecordTrack position with the RecordThread. 1138 * Calculates available frames and handle overruns if the RecordThread 1139 * has advanced faster than the ResamplerBufferProvider has retrieved data. 1140 * TODO: why not do this for every getNextBuffer? 1141 * 1142 * Parameters 1143 * framesAvailable: pointer to optional output size_t to store record track 1144 * frames available. 1145 * hasOverrun: pointer to optional boolean, returns true if track has overrun. 1146 */ 1147 1148 virtual void sync(size_t *framesAvailable = NULL, bool *hasOverrun = NULL); 1149 1150 // AudioBufferProvider interface 1151 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer); 1152 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 1153 private: 1154 RecordTrack * const mRecordTrack; 1155 size_t mRsmpInUnrel; // unreleased frames remaining from 1156 // most recent getNextBuffer 1157 // for debug only 1158 int32_t mRsmpInFront; // next available frame 1159 // rolling counter that is never cleared 1160 }; 1161 1162 /* The RecordBufferConverter is used for format, channel, and sample rate 1163 * conversion for a RecordTrack. 1164 * 1165 * TODO: Self contained, so move to a separate file later. 1166 * 1167 * RecordBufferConverter uses the convert() method rather than exposing a 1168 * buffer provider interface; this is to save a memory copy. 1169 */ 1170 class RecordBufferConverter 1171 { 1172 public: 1173 RecordBufferConverter( 1174 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 1175 uint32_t srcSampleRate, 1176 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 1177 uint32_t dstSampleRate); 1178 1179 ~RecordBufferConverter(); 1180 1181 /* Converts input data from an AudioBufferProvider by format, channelMask, 1182 * and sampleRate to a destination buffer. 1183 * 1184 * Parameters 1185 * dst: buffer to place the converted data. 1186 * provider: buffer provider to obtain source data. 1187 * frames: number of frames to convert 1188 * 1189 * Returns the number of frames converted. 1190 */ 1191 size_t convert(void *dst, AudioBufferProvider *provider, size_t frames); 1192 1193 // returns NO_ERROR if constructor was successful 1194 status_t initCheck() const { 1195 // mSrcChannelMask set on successful updateParameters 1196 return mSrcChannelMask != AUDIO_CHANNEL_INVALID ? NO_ERROR : NO_INIT; 1197 } 1198 1199 // allows dynamic reconfigure of all parameters 1200 status_t updateParameters( 1201 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 1202 uint32_t srcSampleRate, 1203 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 1204 uint32_t dstSampleRate); 1205 1206 // called to reset resampler buffers on record track discontinuity 1207 void reset() { 1208 if (mResampler != NULL) { 1209 mResampler->reset(); 1210 } 1211 } 1212 1213 private: 1214 // format conversion when not using resampler 1215 void convertNoResampler(void *dst, const void *src, size_t frames); 1216 1217 // format conversion when using resampler; modifies src in-place 1218 void convertResampler(void *dst, /*not-a-const*/ void *src, size_t frames); 1219 1220 // user provided information 1221 audio_channel_mask_t mSrcChannelMask; 1222 audio_format_t mSrcFormat; 1223 uint32_t mSrcSampleRate; 1224 audio_channel_mask_t mDstChannelMask; 1225 audio_format_t mDstFormat; 1226 uint32_t mDstSampleRate; 1227 1228 // derived information 1229 uint32_t mSrcChannelCount; 1230 uint32_t mDstChannelCount; 1231 size_t mDstFrameSize; 1232 1233 // format conversion buffer 1234 void *mBuf; 1235 size_t mBufFrames; 1236 size_t mBufFrameSize; 1237 1238 // resampler info 1239 AudioResampler *mResampler; 1240 1241 bool mIsLegacyDownmix; // legacy stereo to mono conversion needed 1242 bool mIsLegacyUpmix; // legacy mono to stereo conversion needed 1243 bool mRequiresFloat; // data processing requires float (e.g. resampler) 1244 PassthruBufferProvider *mInputConverterProvider; // converts input to float 1245 int8_t mIdxAry[sizeof(uint32_t) * 8]; // used for channel mask conversion 1246 }; 1247 1248 #include "RecordTracks.h" 1249 1250 RecordThread(const sp<AudioFlinger>& audioFlinger, 1251 AudioStreamIn *input, 1252 audio_io_handle_t id, 1253 audio_devices_t outDevice, 1254 audio_devices_t inDevice, 1255 bool systemReady 1256 #ifdef TEE_SINK 1257 , const sp<NBAIO_Sink>& teeSink 1258 #endif 1259 ); 1260 virtual ~RecordThread(); 1261 1262 // no addTrack_l ? 1263 void destroyTrack_l(const sp<RecordTrack>& track); 1264 void removeTrack_l(const sp<RecordTrack>& track); 1265 1266 void dumpInternals(int fd, const Vector<String16>& args); 1267 void dumpTracks(int fd, const Vector<String16>& args); 1268 1269 // Thread virtuals 1270 virtual bool threadLoop(); 1271 1272 // RefBase 1273 virtual void onFirstRef(); 1274 1275 virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; } 1276 1277 virtual sp<MemoryDealer> readOnlyHeap() const { return mReadOnlyHeap; } 1278 1279 virtual sp<IMemory> pipeMemory() const { return mPipeMemory; } 1280 1281 sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l( 1282 const sp<AudioFlinger::Client>& client, 1283 uint32_t sampleRate, 1284 audio_format_t format, 1285 audio_channel_mask_t channelMask, 1286 size_t *pFrameCount, 1287 audio_session_t sessionId, 1288 size_t *notificationFrames, 1289 int uid, 1290 audio_input_flags_t *flags, 1291 pid_t tid, 1292 status_t *status /*non-NULL*/); 1293 1294 status_t start(RecordTrack* recordTrack, 1295 AudioSystem::sync_event_t event, 1296 audio_session_t triggerSession); 1297 1298 // ask the thread to stop the specified track, and 1299 // return true if the caller should then do it's part of the stopping process 1300 bool stop(RecordTrack* recordTrack); 1301 1302 void dump(int fd, const Vector<String16>& args); 1303 AudioStreamIn* clearInput(); 1304 virtual audio_stream_t* stream() const; 1305 1306 1307 virtual bool checkForNewParameter_l(const String8& keyValuePair, 1308 status_t& status); 1309 virtual void cacheParameters_l() {} 1310 virtual String8 getParameters(const String8& keys); 1311 virtual void ioConfigChanged(audio_io_config_event event, pid_t pid = 0); 1312 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 1313 audio_patch_handle_t *handle); 1314 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 1315 1316 void addPatchRecord(const sp<PatchRecord>& record); 1317 void deletePatchRecord(const sp<PatchRecord>& record); 1318 1319 void readInputParameters_l(); 1320 virtual uint32_t getInputFramesLost(); 1321 1322 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1323 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1324 virtual uint32_t hasAudioSession_l(audio_session_t sessionId) const; 1325 1326 // Return the set of unique session IDs across all tracks. 1327 // The keys are the session IDs, and the associated values are meaningless. 1328 // FIXME replace by Set [and implement Bag/Multiset for other uses]. 1329 KeyedVector<audio_session_t, bool> sessionIds() const; 1330 1331 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 1332 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 1333 1334 static void syncStartEventCallback(const wp<SyncEvent>& event); 1335 1336 virtual size_t frameCount() const { return mFrameCount; } 1337 bool hasFastCapture() const { return mFastCapture != 0; } 1338 virtual void getAudioPortConfig(struct audio_port_config *config); 1339 1340 virtual status_t checkEffectCompatibility_l(const effect_descriptor_t *desc, 1341 audio_session_t sessionId); 1342 1343 private: 1344 // Enter standby if not already in standby, and set mStandby flag 1345 void standbyIfNotAlreadyInStandby(); 1346 1347 // Call the HAL standby method unconditionally, and don't change mStandby flag 1348 void inputStandBy(); 1349 1350 AudioStreamIn *mInput; 1351 SortedVector < sp<RecordTrack> > mTracks; 1352 // mActiveTracks has dual roles: it indicates the current active track(s), and 1353 // is used together with mStartStopCond to indicate start()/stop() progress 1354 SortedVector< sp<RecordTrack> > mActiveTracks; 1355 // generation counter for mActiveTracks 1356 int mActiveTracksGen; 1357 Condition mStartStopCond; 1358 1359 // resampler converts input at HAL Hz to output at AudioRecord client Hz 1360 void *mRsmpInBuffer; // 1361 size_t mRsmpInFrames; // size of resampler input in frames 1362 size_t mRsmpInFramesP2;// size rounded up to a power-of-2 1363 1364 // rolling index that is never cleared 1365 int32_t mRsmpInRear; // last filled frame + 1 1366 1367 // For dumpsys 1368 const sp<NBAIO_Sink> mTeeSink; 1369 1370 const sp<MemoryDealer> mReadOnlyHeap; 1371 1372 // one-time initialization, no locks required 1373 sp<FastCapture> mFastCapture; // non-0 if there is also 1374 // a fast capture 1375 1376 // FIXME audio watchdog thread 1377 1378 // contents are not guaranteed to be consistent, no locks required 1379 FastCaptureDumpState mFastCaptureDumpState; 1380 #ifdef STATE_QUEUE_DUMP 1381 // FIXME StateQueue observer and mutator dump fields 1382 #endif 1383 // FIXME audio watchdog dump 1384 1385 // accessible only within the threadLoop(), no locks required 1386 // mFastCapture->sq() // for mutating and pushing state 1387 int32_t mFastCaptureFutex; // for cold idle 1388 1389 // The HAL input source is treated as non-blocking, 1390 // but current implementation is blocking 1391 sp<NBAIO_Source> mInputSource; 1392 // The source for the normal capture thread to read from: mInputSource or mPipeSource 1393 sp<NBAIO_Source> mNormalSource; 1394 // If a fast capture is present, the non-blocking pipe sink written to by fast capture, 1395 // otherwise clear 1396 sp<NBAIO_Sink> mPipeSink; 1397 // If a fast capture is present, the non-blocking pipe source read by normal thread, 1398 // otherwise clear 1399 sp<NBAIO_Source> mPipeSource; 1400 // Depth of pipe from fast capture to normal thread and fast clients, always power of 2 1401 size_t mPipeFramesP2; 1402 // If a fast capture is present, the Pipe as IMemory, otherwise clear 1403 sp<IMemory> mPipeMemory; 1404 1405 static const size_t kFastCaptureLogSize = 4 * 1024; 1406 sp<NBLog::Writer> mFastCaptureNBLogWriter; 1407 1408 bool mFastTrackAvail; // true if fast track available 1409 }; 1410
