Threads.h revision abf6ff26df459d991cdbc2dca3b78046c97469db
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 explicit 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 explicit 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 explicit 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 explicit 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 explicit 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 explicit 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 sp<StreamHalInterface> 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 bool pinned); 300 301 // return values for hasAudioSession (bit field) 302 enum effect_state { 303 EFFECT_SESSION = 0x1, // the audio session corresponds to at least one 304 // effect 305 TRACK_SESSION = 0x2, // the audio session corresponds to at least one 306 // track 307 FAST_SESSION = 0x4 // the audio session corresponds to at least one 308 // fast track 309 }; 310 311 // get effect chain corresponding to session Id. 312 sp<EffectChain> getEffectChain(audio_session_t sessionId); 313 // same as getEffectChain() but must be called with ThreadBase mutex locked 314 sp<EffectChain> getEffectChain_l(audio_session_t sessionId) const; 315 // add an effect chain to the chain list (mEffectChains) 316 virtual status_t addEffectChain_l(const sp<EffectChain>& chain) = 0; 317 // remove an effect chain from the chain list (mEffectChains) 318 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain) = 0; 319 // lock all effect chains Mutexes. Must be called before releasing the 320 // ThreadBase mutex before processing the mixer and effects. This guarantees the 321 // integrity of the chains during the process. 322 // Also sets the parameter 'effectChains' to current value of mEffectChains. 323 void lockEffectChains_l(Vector< sp<EffectChain> >& effectChains); 324 // unlock effect chains after process 325 void unlockEffectChains(const Vector< sp<EffectChain> >& effectChains); 326 // get a copy of mEffectChains vector 327 Vector< sp<EffectChain> > getEffectChains_l() const { return mEffectChains; }; 328 // set audio mode to all effect chains 329 void setMode(audio_mode_t mode); 330 // get effect module with corresponding ID on specified audio session 331 sp<AudioFlinger::EffectModule> getEffect(audio_session_t sessionId, int effectId); 332 sp<AudioFlinger::EffectModule> getEffect_l(audio_session_t sessionId, int effectId); 333 // add and effect module. Also creates the effect chain is none exists for 334 // the effects audio session 335 status_t addEffect_l(const sp< EffectModule>& effect); 336 // remove and effect module. Also removes the effect chain is this was the last 337 // effect 338 void removeEffect_l(const sp< EffectModule>& effect, bool release = false); 339 // disconnect an effect handle from module and destroy module if last handle 340 void disconnectEffectHandle(EffectHandle *handle, bool unpinIfLast); 341 // detach all tracks connected to an auxiliary effect 342 virtual void detachAuxEffect_l(int effectId __unused) {} 343 // returns a combination of: 344 // - EFFECT_SESSION if effects on this audio session exist in one chain 345 // - TRACK_SESSION if tracks on this audio session exist 346 // - FAST_SESSION if fast tracks on this audio session exist 347 virtual uint32_t hasAudioSession_l(audio_session_t sessionId) const = 0; 348 uint32_t hasAudioSession(audio_session_t sessionId) const { 349 Mutex::Autolock _l(mLock); 350 return hasAudioSession_l(sessionId); 351 } 352 353 // the value returned by default implementation is not important as the 354 // strategy is only meaningful for PlaybackThread which implements this method 355 virtual uint32_t getStrategyForSession_l(audio_session_t sessionId __unused) 356 { return 0; } 357 358 // suspend or restore effect according to the type of effect passed. a NULL 359 // type pointer means suspend all effects in the session 360 void setEffectSuspended(const effect_uuid_t *type, 361 bool suspend, 362 audio_session_t sessionId = AUDIO_SESSION_OUTPUT_MIX); 363 // check if some effects must be suspended/restored when an effect is enabled 364 // or disabled 365 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 366 bool enabled, 367 audio_session_t sessionId = 368 AUDIO_SESSION_OUTPUT_MIX); 369 void checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 370 bool enabled, 371 audio_session_t sessionId = 372 AUDIO_SESSION_OUTPUT_MIX); 373 374 virtual status_t setSyncEvent(const sp<SyncEvent>& event) = 0; 375 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const = 0; 376 377 // Return a reference to a per-thread heap which can be used to allocate IMemory 378 // objects that will be read-only to client processes, read/write to mediaserver, 379 // and shared by all client processes of the thread. 380 // The heap is per-thread rather than common across all threads, because 381 // clients can't be trusted not to modify the offset of the IMemory they receive. 382 // If a thread does not have such a heap, this method returns 0. 383 virtual sp<MemoryDealer> readOnlyHeap() const { return 0; } 384 385 virtual sp<IMemory> pipeMemory() const { return 0; } 386 387 void systemReady(); 388 389 // checkEffectCompatibility_l() must be called with ThreadBase::mLock held 390 virtual status_t checkEffectCompatibility_l(const effect_descriptor_t *desc, 391 audio_session_t sessionId) = 0; 392 393 mutable Mutex mLock; 394 395protected: 396 397 // entry describing an effect being suspended in mSuspendedSessions keyed vector 398 class SuspendedSessionDesc : public RefBase { 399 public: 400 SuspendedSessionDesc() : mRefCount(0) {} 401 402 int mRefCount; // number of active suspend requests 403 effect_uuid_t mType; // effect type UUID 404 }; 405 406 void acquireWakeLock(int uid = -1); 407 virtual void acquireWakeLock_l(int uid = -1); 408 void releaseWakeLock(); 409 void releaseWakeLock_l(); 410 void updateWakeLockUids_l(const SortedVector<int> &uids); 411 void getPowerManager_l(); 412 void setEffectSuspended_l(const effect_uuid_t *type, 413 bool suspend, 414 audio_session_t sessionId); 415 // updated mSuspendedSessions when an effect suspended or restored 416 void updateSuspendedSessions_l(const effect_uuid_t *type, 417 bool suspend, 418 audio_session_t sessionId); 419 // check if some effects must be suspended when an effect chain is added 420 void checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain); 421 422 String16 getWakeLockTag(); 423 424 virtual void preExit() { } 425 virtual void setMasterMono_l(bool mono __unused) { } 426 virtual bool requireMonoBlend() { return false; } 427 428 friend class AudioFlinger; // for mEffectChains 429 430 const type_t mType; 431 432 // Used by parameters, config events, addTrack_l, exit 433 Condition mWaitWorkCV; 434 435 const sp<AudioFlinger> mAudioFlinger; 436 437 // updated by PlaybackThread::readOutputParameters_l() or 438 // RecordThread::readInputParameters_l() 439 uint32_t mSampleRate; 440 size_t mFrameCount; // output HAL, direct output, record 441 audio_channel_mask_t mChannelMask; 442 uint32_t mChannelCount; 443 size_t mFrameSize; 444 // not HAL frame size, this is for output sink (to pipe to fast mixer) 445 audio_format_t mFormat; // Source format for Recording and 446 // Sink format for Playback. 447 // Sink format may be different than 448 // HAL format if Fastmixer is used. 449 audio_format_t mHALFormat; 450 size_t mBufferSize; // HAL buffer size for read() or write() 451 452 Vector< sp<ConfigEvent> > mConfigEvents; 453 Vector< sp<ConfigEvent> > mPendingConfigEvents; // events awaiting system ready 454 455 // These fields are written and read by thread itself without lock or barrier, 456 // and read by other threads without lock or barrier via standby(), outDevice() 457 // and inDevice(). 458 // Because of the absence of a lock or barrier, any other thread that reads 459 // these fields must use the information in isolation, or be prepared to deal 460 // with possibility that it might be inconsistent with other information. 461 bool mStandby; // Whether thread is currently in standby. 462 audio_devices_t mOutDevice; // output device 463 audio_devices_t mInDevice; // input device 464 audio_devices_t mPrevOutDevice; // previous output device 465 audio_devices_t mPrevInDevice; // previous input device 466 struct audio_patch mPatch; 467 audio_source_t mAudioSource; 468 469 const audio_io_handle_t mId; 470 Vector< sp<EffectChain> > mEffectChains; 471 472 static const int kThreadNameLength = 16; // prctl(PR_SET_NAME) limit 473 char mThreadName[kThreadNameLength]; // guaranteed NUL-terminated 474 sp<IPowerManager> mPowerManager; 475 sp<IBinder> mWakeLockToken; 476 const sp<PMDeathRecipient> mDeathRecipient; 477 // list of suspended effects per session and per type. The first (outer) vector is 478 // keyed by session ID, the second (inner) by type UUID timeLow field 479 KeyedVector< audio_session_t, KeyedVector< int, sp<SuspendedSessionDesc> > > 480 mSuspendedSessions; 481 static const size_t kLogSize = 4 * 1024; 482 sp<NBLog::Writer> mNBLogWriter; 483 bool mSystemReady; 484 bool mNotifiedBatteryStart; 485 ExtendedTimestamp mTimestamp; 486}; 487 488// --- PlaybackThread --- 489class PlaybackThread : public ThreadBase, public StreamOutHalInterfaceCallback { 490public: 491 492#include "PlaybackTracks.h" 493 494 enum mixer_state { 495 MIXER_IDLE, // no active tracks 496 MIXER_TRACKS_ENABLED, // at least one active track, but no track has any data ready 497 MIXER_TRACKS_READY, // at least one active track, and at least one track has data 498 MIXER_DRAIN_TRACK, // drain currently playing track 499 MIXER_DRAIN_ALL, // fully drain the hardware 500 // standby mode does not have an enum value 501 // suspend by audio policy manager is orthogonal to mixer state 502 }; 503 504 // retry count before removing active track in case of underrun on offloaded thread: 505 // we need to make sure that AudioTrack client has enough time to send large buffers 506 //FIXME may be more appropriate if expressed in time units. Need to revise how underrun is 507 // handled for offloaded tracks 508 static const int8_t kMaxTrackRetriesOffload = 20; 509 static const int8_t kMaxTrackStartupRetriesOffload = 100; 510 static const int8_t kMaxTrackStopRetriesOffload = 2; 511 // 14 tracks max per client allows for 2 misbehaving application leaving 4 available tracks. 512 static const uint32_t kMaxTracksPerUid = 14; 513 514 PlaybackThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 515 audio_io_handle_t id, audio_devices_t device, type_t type, bool systemReady); 516 virtual ~PlaybackThread(); 517 518 void dump(int fd, const Vector<String16>& args); 519 520 // Thread virtuals 521 virtual bool threadLoop(); 522 523 // RefBase 524 virtual void onFirstRef(); 525 526 virtual status_t checkEffectCompatibility_l(const effect_descriptor_t *desc, 527 audio_session_t sessionId); 528 529protected: 530 // Code snippets that were lifted up out of threadLoop() 531 virtual void threadLoop_mix() = 0; 532 virtual void threadLoop_sleepTime() = 0; 533 virtual ssize_t threadLoop_write(); 534 virtual void threadLoop_drain(); 535 virtual void threadLoop_standby(); 536 virtual void threadLoop_exit(); 537 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 538 539 // prepareTracks_l reads and writes mActiveTracks, and returns 540 // the pending set of tracks to remove via Vector 'tracksToRemove'. The caller 541 // is responsible for clearing or destroying this Vector later on, when it 542 // is safe to do so. That will drop the final ref count and destroy the tracks. 543 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove) = 0; 544 void removeTracks_l(const Vector< sp<Track> >& tracksToRemove); 545 546 // StreamOutHalInterfaceCallback implementation 547 virtual void onWriteReady(); 548 virtual void onDrainReady(); 549 virtual void onError(); 550 551 void resetWriteBlocked(uint32_t sequence); 552 void resetDraining(uint32_t sequence); 553 554 virtual bool waitingAsyncCallback(); 555 virtual bool waitingAsyncCallback_l(); 556 virtual bool shouldStandby_l(); 557 virtual void onAddNewTrack_l(); 558 void onAsyncError(); // error reported by AsyncCallbackThread 559 560 // ThreadBase virtuals 561 virtual void preExit(); 562 563 virtual bool keepWakeLock() const { return true; } 564 565public: 566 567 virtual status_t initCheck() const { return (mOutput == NULL) ? NO_INIT : NO_ERROR; } 568 569 // return estimated latency in milliseconds, as reported by HAL 570 uint32_t latency() const; 571 // same, but lock must already be held 572 uint32_t latency_l() const; 573 574 void setMasterVolume(float value); 575 void setMasterMute(bool muted); 576 577 void setStreamVolume(audio_stream_type_t stream, float value); 578 void setStreamMute(audio_stream_type_t stream, bool muted); 579 580 float streamVolume(audio_stream_type_t stream) const; 581 582 sp<Track> createTrack_l( 583 const sp<AudioFlinger::Client>& client, 584 audio_stream_type_t streamType, 585 uint32_t sampleRate, 586 audio_format_t format, 587 audio_channel_mask_t channelMask, 588 size_t *pFrameCount, 589 const sp<IMemory>& sharedBuffer, 590 audio_session_t sessionId, 591 audio_output_flags_t *flags, 592 pid_t tid, 593 uid_t uid, 594 status_t *status /*non-NULL*/); 595 596 AudioStreamOut* getOutput() const; 597 AudioStreamOut* clearOutput(); 598 virtual sp<StreamHalInterface> stream() const; 599 600 // a very large number of suspend() will eventually wraparound, but unlikely 601 void suspend() { (void) android_atomic_inc(&mSuspended); } 602 void restore() 603 { 604 // if restore() is done without suspend(), get back into 605 // range so that the next suspend() will operate correctly 606 if (android_atomic_dec(&mSuspended) <= 0) { 607 android_atomic_release_store(0, &mSuspended); 608 } 609 } 610 bool isSuspended() const 611 { return android_atomic_acquire_load(&mSuspended) > 0; } 612 613 virtual String8 getParameters(const String8& keys); 614 virtual void ioConfigChanged(audio_io_config_event event, pid_t pid = 0); 615 status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames); 616 // FIXME rename mixBuffer() to sinkBuffer() and remove int16_t* dependency. 617 // Consider also removing and passing an explicit mMainBuffer initialization 618 // parameter to AF::PlaybackThread::Track::Track(). 619 int16_t *mixBuffer() const { 620 return reinterpret_cast<int16_t *>(mSinkBuffer); }; 621 622 virtual void detachAuxEffect_l(int effectId); 623 status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track>& track, 624 int EffectId); 625 status_t attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track>& track, 626 int EffectId); 627 628 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 629 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 630 virtual uint32_t hasAudioSession_l(audio_session_t sessionId) const; 631 virtual uint32_t getStrategyForSession_l(audio_session_t sessionId); 632 633 634 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 635 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 636 637 // called with AudioFlinger lock held 638 bool invalidateTracks_l(audio_stream_type_t streamType); 639 virtual void invalidateTracks(audio_stream_type_t streamType); 640 641 virtual size_t frameCount() const { return mNormalFrameCount; } 642 643 status_t getTimestamp_l(AudioTimestamp& timestamp); 644 645 void addPatchTrack(const sp<PatchTrack>& track); 646 void deletePatchTrack(const sp<PatchTrack>& track); 647 648 virtual void getAudioPortConfig(struct audio_port_config *config); 649 650protected: 651 // updated by readOutputParameters_l() 652 size_t mNormalFrameCount; // normal mixer and effects 653 654 bool mThreadThrottle; // throttle the thread processing 655 uint32_t mThreadThrottleTimeMs; // throttle time for MIXER threads 656 uint32_t mThreadThrottleEndMs; // notify once per throttling 657 uint32_t mHalfBufferMs; // half the buffer size in milliseconds 658 659 void* mSinkBuffer; // frame size aligned sink buffer 660 661 // TODO: 662 // Rearrange the buffer info into a struct/class with 663 // clear, copy, construction, destruction methods. 664 // 665 // mSinkBuffer also has associated with it: 666 // 667 // mSinkBufferSize: Sink Buffer Size 668 // mFormat: Sink Buffer Format 669 670 // Mixer Buffer (mMixerBuffer*) 671 // 672 // In the case of floating point or multichannel data, which is not in the 673 // sink format, it is required to accumulate in a higher precision or greater channel count 674 // buffer before downmixing or data conversion to the sink buffer. 675 676 // Set to "true" to enable the Mixer Buffer otherwise mixer output goes to sink buffer. 677 bool mMixerBufferEnabled; 678 679 // Storage, 32 byte aligned (may make this alignment a requirement later). 680 // Due to constraints on mNormalFrameCount, the buffer size is a multiple of 16 frames. 681 void* mMixerBuffer; 682 683 // Size of mMixerBuffer in bytes: mNormalFrameCount * #channels * sampsize. 684 size_t mMixerBufferSize; 685 686 // The audio format of mMixerBuffer. Set to AUDIO_FORMAT_PCM_(FLOAT|16_BIT) only. 687 audio_format_t mMixerBufferFormat; 688 689 // An internal flag set to true by MixerThread::prepareTracks_l() 690 // when mMixerBuffer contains valid data after mixing. 691 bool mMixerBufferValid; 692 693 // Effects Buffer (mEffectsBuffer*) 694 // 695 // In the case of effects data, which is not in the sink format, 696 // it is required to accumulate in a different buffer before data conversion 697 // to the sink buffer. 698 699 // Set to "true" to enable the Effects Buffer otherwise effects output goes to sink buffer. 700 bool mEffectBufferEnabled; 701 702 // Storage, 32 byte aligned (may make this alignment a requirement later). 703 // Due to constraints on mNormalFrameCount, the buffer size is a multiple of 16 frames. 704 void* mEffectBuffer; 705 706 // Size of mEffectsBuffer in bytes: mNormalFrameCount * #channels * sampsize. 707 size_t mEffectBufferSize; 708 709 // The audio format of mEffectsBuffer. Set to AUDIO_FORMAT_PCM_16_BIT only. 710 audio_format_t mEffectBufferFormat; 711 712 // An internal flag set to true by MixerThread::prepareTracks_l() 713 // when mEffectsBuffer contains valid data after mixing. 714 // 715 // When this is set, all mixer data is routed into the effects buffer 716 // for any processing (including output processing). 717 bool mEffectBufferValid; 718 719 // suspend count, > 0 means suspended. While suspended, the thread continues to pull from 720 // tracks and mix, but doesn't write to HAL. A2DP and SCO HAL implementations can't handle 721 // concurrent use of both of them, so Audio Policy Service suspends one of the threads to 722 // workaround that restriction. 723 // 'volatile' means accessed via atomic operations and no lock. 724 volatile int32_t mSuspended; 725 726 int64_t mBytesWritten; 727 int64_t mFramesWritten; // not reset on standby 728 int64_t mSuspendedFrames; // not reset on standby 729private: 730 // mMasterMute is in both PlaybackThread and in AudioFlinger. When a 731 // PlaybackThread needs to find out if master-muted, it checks it's local 732 // copy rather than the one in AudioFlinger. This optimization saves a lock. 733 bool mMasterMute; 734 void setMasterMute_l(bool muted) { mMasterMute = muted; } 735protected: 736 SortedVector< wp<Track> > mActiveTracks; // FIXME check if this could be sp<> 737 SortedVector<int> mWakeLockUids; 738 int mActiveTracksGeneration; 739 wp<Track> mLatestActiveTrack; // latest track added to mActiveTracks 740 741 // Allocate a track name for a given channel mask. 742 // Returns name >= 0 if successful, -1 on failure. 743 virtual int getTrackName_l(audio_channel_mask_t channelMask, audio_format_t format, 744 audio_session_t sessionId, uid_t uid) = 0; 745 virtual void deleteTrackName_l(int name) = 0; 746 747 // Time to sleep between cycles when: 748 virtual uint32_t activeSleepTimeUs() const; // mixer state MIXER_TRACKS_ENABLED 749 virtual uint32_t idleSleepTimeUs() const = 0; // mixer state MIXER_IDLE 750 virtual uint32_t suspendSleepTimeUs() const = 0; // audio policy manager suspended us 751 // No sleep when mixer state == MIXER_TRACKS_READY; relies on audio HAL stream->write() 752 // No sleep in standby mode; waits on a condition 753 754 // Code snippets that are temporarily lifted up out of threadLoop() until the merge 755 void checkSilentMode_l(); 756 757 // Non-trivial for DUPLICATING only 758 virtual void saveOutputTracks() { } 759 virtual void clearOutputTracks() { } 760 761 // Cache various calculated values, at threadLoop() entry and after a parameter change 762 virtual void cacheParameters_l(); 763 764 virtual uint32_t correctLatency_l(uint32_t latency) const; 765 766 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 767 audio_patch_handle_t *handle); 768 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 769 770 bool usesHwAvSync() const { return (mType == DIRECT) && (mOutput != NULL) 771 && mHwSupportsPause 772 && (mOutput->flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC); } 773 774 uint32_t trackCountForUid_l(uid_t uid); 775 776private: 777 778 friend class AudioFlinger; // for numerous 779 780 PlaybackThread& operator = (const PlaybackThread&); 781 782 status_t addTrack_l(const sp<Track>& track); 783 bool destroyTrack_l(const sp<Track>& track); 784 void removeTrack_l(const sp<Track>& track); 785 void broadcast_l(); 786 787 void readOutputParameters_l(); 788 789 virtual void dumpInternals(int fd, const Vector<String16>& args); 790 void dumpTracks(int fd, const Vector<String16>& args); 791 792 SortedVector< sp<Track> > mTracks; 793 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 794 AudioStreamOut *mOutput; 795 796 float mMasterVolume; 797 nsecs_t mLastWriteTime; 798 int mNumWrites; 799 int mNumDelayedWrites; 800 bool mInWrite; 801 802 // FIXME rename these former local variables of threadLoop to standard "m" names 803 nsecs_t mStandbyTimeNs; 804 size_t mSinkBufferSize; 805 806 // cached copies of activeSleepTimeUs() and idleSleepTimeUs() made by cacheParameters_l() 807 uint32_t mActiveSleepTimeUs; 808 uint32_t mIdleSleepTimeUs; 809 810 uint32_t mSleepTimeUs; 811 812 // mixer status returned by prepareTracks_l() 813 mixer_state mMixerStatus; // current cycle 814 // previous cycle when in prepareTracks_l() 815 mixer_state mMixerStatusIgnoringFastTracks; 816 // FIXME or a separate ready state per track 817 818 // FIXME move these declarations into the specific sub-class that needs them 819 // MIXER only 820 uint32_t sleepTimeShift; 821 822 // same as AudioFlinger::mStandbyTimeInNsecs except for DIRECT which uses a shorter value 823 nsecs_t mStandbyDelayNs; 824 825 // MIXER only 826 nsecs_t maxPeriod; 827 828 // DUPLICATING only 829 uint32_t writeFrames; 830 831 size_t mBytesRemaining; 832 size_t mCurrentWriteLength; 833 bool mUseAsyncWrite; 834 // mWriteAckSequence contains current write sequence on bits 31-1. The write sequence is 835 // incremented each time a write(), a flush() or a standby() occurs. 836 // Bit 0 is set when a write blocks and indicates a callback is expected. 837 // Bit 0 is reset by the async callback thread calling resetWriteBlocked(). Out of sequence 838 // callbacks are ignored. 839 uint32_t mWriteAckSequence; 840 // mDrainSequence contains current drain sequence on bits 31-1. The drain sequence is 841 // incremented each time a drain is requested or a flush() or standby() occurs. 842 // Bit 0 is set when the drain() command is called at the HAL and indicates a callback is 843 // expected. 844 // Bit 0 is reset by the async callback thread calling resetDraining(). Out of sequence 845 // callbacks are ignored. 846 uint32_t mDrainSequence; 847 // A condition that must be evaluated by prepareTrack_l() has changed and we must not wait 848 // for async write callback in the thread loop before evaluating it 849 bool mSignalPending; 850 sp<AsyncCallbackThread> mCallbackThread; 851 852private: 853 // The HAL output sink is treated as non-blocking, but current implementation is blocking 854 sp<NBAIO_Sink> mOutputSink; 855 // If a fast mixer is present, the blocking pipe sink, otherwise clear 856 sp<NBAIO_Sink> mPipeSink; 857 // The current sink for the normal mixer to write it's (sub)mix, mOutputSink or mPipeSink 858 sp<NBAIO_Sink> mNormalSink; 859#ifdef TEE_SINK 860 // For dumpsys 861 sp<NBAIO_Sink> mTeeSink; 862 sp<NBAIO_Source> mTeeSource; 863#endif 864 uint32_t mScreenState; // cached copy of gScreenState 865 static const size_t kFastMixerLogSize = 4 * 1024; 866 sp<NBLog::Writer> mFastMixerNBLogWriter; 867 868 // Do not call from a sched_fifo thread as it uses a system time call 869 // and obtains a local mutex. 870 class LocalLog { 871 public: 872 void log(const char *fmt, ...) { 873 va_list val; 874 va_start(val, fmt); 875 876 // format to buffer 877 char buffer[512]; 878 int length = vsnprintf(buffer, sizeof(buffer), fmt, val); 879 if (length >= (signed)sizeof(buffer)) { 880 length = sizeof(buffer) - 1; 881 } 882 883 // strip out trailing newline 884 while (length > 0 && buffer[length - 1] == '\n') { 885 buffer[--length] = 0; 886 } 887 888 // store in circular array 889 AutoMutex _l(mLock); 890 mLog.emplace_back( 891 std::make_pair(systemTime(SYSTEM_TIME_REALTIME), std::string(buffer))); 892 if (mLog.size() > kLogSize) { 893 mLog.pop_front(); 894 } 895 896 va_end(val); 897 } 898 899 void dump(int fd, const Vector<String16>& args, const char *prefix = "") { 900 if (!AudioFlinger::dumpTryLock(mLock)) return; // a local lock, shouldn't happen 901 if (mLog.size() > 0) { 902 bool dumpAll = false; 903 for (const auto &arg : args) { 904 if (arg == String16("--locallog")) { 905 dumpAll = true; 906 } 907 } 908 909 dprintf(fd, "Local Log:\n"); 910 auto it = mLog.begin(); 911 if (!dumpAll && mLog.size() > kLogPrint) { 912 it += (mLog.size() - kLogPrint); 913 } 914 for (; it != mLog.end(); ++it) { 915 const int64_t ns = it->first; 916 const int ns_per_sec = 1000000000; 917 const time_t sec = ns / ns_per_sec; 918 struct tm tm; 919 localtime_r(&sec, &tm); 920 921 dprintf(fd, "%s%02d-%02d %02d:%02d:%02d.%03d %s\n", 922 prefix, 923 tm.tm_mon + 1, // localtime_r uses months in 0 - 11 range 924 tm.tm_mday, tm.tm_hour, tm.tm_min, tm.tm_sec, 925 (int)(ns % ns_per_sec / 1000000), 926 it->second.c_str()); 927 } 928 } 929 mLock.unlock(); 930 } 931 932 private: 933 Mutex mLock; 934 static const size_t kLogSize = 256; // full history 935 static const size_t kLogPrint = 32; // default print history 936 std::deque<std::pair<int64_t, std::string>> mLog; 937 } mLocalLog; 938 939public: 940 virtual bool hasFastMixer() const = 0; 941 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex __unused) const 942 { FastTrackUnderruns dummy; return dummy; } 943 944protected: 945 // accessed by both binder threads and within threadLoop(), lock on mutex needed 946 unsigned mFastTrackAvailMask; // bit i set if fast track [i] is available 947 bool mHwSupportsPause; 948 bool mHwPaused; 949 bool mFlushPending; 950}; 951 952class MixerThread : public PlaybackThread { 953public: 954 MixerThread(const sp<AudioFlinger>& audioFlinger, 955 AudioStreamOut* output, 956 audio_io_handle_t id, 957 audio_devices_t device, 958 bool systemReady, 959 type_t type = MIXER); 960 virtual ~MixerThread(); 961 962 // Thread virtuals 963 964 virtual bool checkForNewParameter_l(const String8& keyValuePair, 965 status_t& status); 966 virtual void dumpInternals(int fd, const Vector<String16>& args); 967 968protected: 969 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 970 virtual int getTrackName_l(audio_channel_mask_t channelMask, audio_format_t format, 971 audio_session_t sessionId, uid_t uid); 972 virtual void deleteTrackName_l(int name); 973 virtual uint32_t idleSleepTimeUs() const; 974 virtual uint32_t suspendSleepTimeUs() const; 975 virtual void cacheParameters_l(); 976 977 virtual void acquireWakeLock_l(int uid = -1) { 978 PlaybackThread::acquireWakeLock_l(uid); 979 if (hasFastMixer()) { 980 mFastMixer->setBoottimeOffset( 981 mTimestamp.mTimebaseOffset[ExtendedTimestamp::TIMEBASE_BOOTTIME]); 982 } 983 } 984 985 // threadLoop snippets 986 virtual ssize_t threadLoop_write(); 987 virtual void threadLoop_standby(); 988 virtual void threadLoop_mix(); 989 virtual void threadLoop_sleepTime(); 990 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 991 virtual uint32_t correctLatency_l(uint32_t latency) const; 992 993 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 994 audio_patch_handle_t *handle); 995 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 996 997 AudioMixer* mAudioMixer; // normal mixer 998private: 999 // one-time initialization, no locks required 1000 sp<FastMixer> mFastMixer; // non-0 if there is also a fast mixer 1001 sp<AudioWatchdog> mAudioWatchdog; // non-0 if there is an audio watchdog thread 1002 1003 // contents are not guaranteed to be consistent, no locks required 1004 FastMixerDumpState mFastMixerDumpState; 1005#ifdef STATE_QUEUE_DUMP 1006 StateQueueObserverDump mStateQueueObserverDump; 1007 StateQueueMutatorDump mStateQueueMutatorDump; 1008#endif 1009 AudioWatchdogDump mAudioWatchdogDump; 1010 1011 // accessible only within the threadLoop(), no locks required 1012 // mFastMixer->sq() // for mutating and pushing state 1013 int32_t mFastMixerFutex; // for cold idle 1014 1015 std::atomic_bool mMasterMono; 1016public: 1017 virtual bool hasFastMixer() const { return mFastMixer != 0; } 1018 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const { 1019 ALOG_ASSERT(fastIndex < FastMixerState::sMaxFastTracks); 1020 return mFastMixerDumpState.mTracks[fastIndex].mUnderruns; 1021 } 1022 1023protected: 1024 virtual void setMasterMono_l(bool mono) { 1025 mMasterMono.store(mono); 1026 if (mFastMixer != nullptr) { /* hasFastMixer() */ 1027 mFastMixer->setMasterMono(mMasterMono); 1028 } 1029 } 1030 // the FastMixer performs mono blend if it exists. 1031 // Blending with limiter is not idempotent, 1032 // and blending without limiter is idempotent but inefficient to do twice. 1033 virtual bool requireMonoBlend() { return mMasterMono.load() && !hasFastMixer(); } 1034}; 1035 1036class DirectOutputThread : public PlaybackThread { 1037public: 1038 1039 DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1040 audio_io_handle_t id, audio_devices_t device, bool systemReady); 1041 virtual ~DirectOutputThread(); 1042 1043 // Thread virtuals 1044 1045 virtual bool checkForNewParameter_l(const String8& keyValuePair, 1046 status_t& status); 1047 virtual void flushHw_l(); 1048 1049protected: 1050 virtual int getTrackName_l(audio_channel_mask_t channelMask, audio_format_t format, 1051 audio_session_t sessionId, uid_t uid); 1052 virtual void deleteTrackName_l(int name); 1053 virtual uint32_t activeSleepTimeUs() const; 1054 virtual uint32_t idleSleepTimeUs() const; 1055 virtual uint32_t suspendSleepTimeUs() const; 1056 virtual void cacheParameters_l(); 1057 1058 // threadLoop snippets 1059 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1060 virtual void threadLoop_mix(); 1061 virtual void threadLoop_sleepTime(); 1062 virtual void threadLoop_exit(); 1063 virtual bool shouldStandby_l(); 1064 1065 virtual void onAddNewTrack_l(); 1066 1067 // volumes last sent to audio HAL with stream->set_volume() 1068 float mLeftVolFloat; 1069 float mRightVolFloat; 1070 1071 DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1072 audio_io_handle_t id, uint32_t device, ThreadBase::type_t type, 1073 bool systemReady); 1074 void processVolume_l(Track *track, bool lastTrack); 1075 1076 // prepareTracks_l() tells threadLoop_mix() the name of the single active track 1077 sp<Track> mActiveTrack; 1078 1079 wp<Track> mPreviousTrack; // used to detect track switch 1080 1081public: 1082 virtual bool hasFastMixer() const { return false; } 1083}; 1084 1085class OffloadThread : public DirectOutputThread { 1086public: 1087 1088 OffloadThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1089 audio_io_handle_t id, uint32_t device, bool systemReady); 1090 virtual ~OffloadThread() {}; 1091 virtual void flushHw_l(); 1092 1093protected: 1094 // threadLoop snippets 1095 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1096 virtual void threadLoop_exit(); 1097 1098 virtual bool waitingAsyncCallback(); 1099 virtual bool waitingAsyncCallback_l(); 1100 virtual void invalidateTracks(audio_stream_type_t streamType); 1101 1102 virtual bool keepWakeLock() const { return (mKeepWakeLock || (mDrainSequence & 1)); } 1103 1104private: 1105 size_t mPausedWriteLength; // length in bytes of write interrupted by pause 1106 size_t mPausedBytesRemaining; // bytes still waiting in mixbuffer after resume 1107 bool mKeepWakeLock; // keep wake lock while waiting for write callback 1108 uint64_t mOffloadUnderrunPosition; // Current frame position for offloaded playback 1109 // used and valid only during underrun. ~0 if 1110 // no underrun has occurred during playback and 1111 // is not reset on standby. 1112}; 1113 1114class AsyncCallbackThread : public Thread { 1115public: 1116 1117 explicit AsyncCallbackThread(const wp<PlaybackThread>& playbackThread); 1118 1119 virtual ~AsyncCallbackThread(); 1120 1121 // Thread virtuals 1122 virtual bool threadLoop(); 1123 1124 // RefBase 1125 virtual void onFirstRef(); 1126 1127 void exit(); 1128 void setWriteBlocked(uint32_t sequence); 1129 void resetWriteBlocked(); 1130 void setDraining(uint32_t sequence); 1131 void resetDraining(); 1132 void setAsyncError(); 1133 1134private: 1135 const wp<PlaybackThread> mPlaybackThread; 1136 // mWriteAckSequence corresponds to the last write sequence passed by the offload thread via 1137 // setWriteBlocked(). The sequence is shifted one bit to the left and the lsb is used 1138 // to indicate that the callback has been received via resetWriteBlocked() 1139 uint32_t mWriteAckSequence; 1140 // mDrainSequence corresponds to the last drain sequence passed by the offload thread via 1141 // setDraining(). The sequence is shifted one bit to the left and the lsb is used 1142 // to indicate that the callback has been received via resetDraining() 1143 uint32_t mDrainSequence; 1144 Condition mWaitWorkCV; 1145 Mutex mLock; 1146 bool mAsyncError; 1147}; 1148 1149class DuplicatingThread : public MixerThread { 1150public: 1151 DuplicatingThread(const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread, 1152 audio_io_handle_t id, bool systemReady); 1153 virtual ~DuplicatingThread(); 1154 1155 // Thread virtuals 1156 void addOutputTrack(MixerThread* thread); 1157 void removeOutputTrack(MixerThread* thread); 1158 uint32_t waitTimeMs() const { return mWaitTimeMs; } 1159protected: 1160 virtual uint32_t activeSleepTimeUs() const; 1161 1162private: 1163 bool outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks); 1164protected: 1165 // threadLoop snippets 1166 virtual void threadLoop_mix(); 1167 virtual void threadLoop_sleepTime(); 1168 virtual ssize_t threadLoop_write(); 1169 virtual void threadLoop_standby(); 1170 virtual void cacheParameters_l(); 1171 1172private: 1173 // called from threadLoop, addOutputTrack, removeOutputTrack 1174 virtual void updateWaitTime_l(); 1175protected: 1176 virtual void saveOutputTracks(); 1177 virtual void clearOutputTracks(); 1178private: 1179 1180 uint32_t mWaitTimeMs; 1181 SortedVector < sp<OutputTrack> > outputTracks; 1182 SortedVector < sp<OutputTrack> > mOutputTracks; 1183public: 1184 virtual bool hasFastMixer() const { return false; } 1185}; 1186 1187 1188// record thread 1189class RecordThread : public ThreadBase 1190{ 1191public: 1192 1193 class RecordTrack; 1194 1195 /* The ResamplerBufferProvider is used to retrieve recorded input data from the 1196 * RecordThread. It maintains local state on the relative position of the read 1197 * position of the RecordTrack compared with the RecordThread. 1198 */ 1199 class ResamplerBufferProvider : public AudioBufferProvider 1200 { 1201 public: 1202 explicit ResamplerBufferProvider(RecordTrack* recordTrack) : 1203 mRecordTrack(recordTrack), 1204 mRsmpInUnrel(0), mRsmpInFront(0) { } 1205 virtual ~ResamplerBufferProvider() { } 1206 1207 // called to set the ResamplerBufferProvider to head of the RecordThread data buffer, 1208 // skipping any previous data read from the hal. 1209 virtual void reset(); 1210 1211 /* Synchronizes RecordTrack position with the RecordThread. 1212 * Calculates available frames and handle overruns if the RecordThread 1213 * has advanced faster than the ResamplerBufferProvider has retrieved data. 1214 * TODO: why not do this for every getNextBuffer? 1215 * 1216 * Parameters 1217 * framesAvailable: pointer to optional output size_t to store record track 1218 * frames available. 1219 * hasOverrun: pointer to optional boolean, returns true if track has overrun. 1220 */ 1221 1222 virtual void sync(size_t *framesAvailable = NULL, bool *hasOverrun = NULL); 1223 1224 // AudioBufferProvider interface 1225 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer); 1226 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 1227 private: 1228 RecordTrack * const mRecordTrack; 1229 size_t mRsmpInUnrel; // unreleased frames remaining from 1230 // most recent getNextBuffer 1231 // for debug only 1232 int32_t mRsmpInFront; // next available frame 1233 // rolling counter that is never cleared 1234 }; 1235 1236 /* The RecordBufferConverter is used for format, channel, and sample rate 1237 * conversion for a RecordTrack. 1238 * 1239 * TODO: Self contained, so move to a separate file later. 1240 * 1241 * RecordBufferConverter uses the convert() method rather than exposing a 1242 * buffer provider interface; this is to save a memory copy. 1243 */ 1244 class RecordBufferConverter 1245 { 1246 public: 1247 RecordBufferConverter( 1248 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 1249 uint32_t srcSampleRate, 1250 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 1251 uint32_t dstSampleRate); 1252 1253 ~RecordBufferConverter(); 1254 1255 /* Converts input data from an AudioBufferProvider by format, channelMask, 1256 * and sampleRate to a destination buffer. 1257 * 1258 * Parameters 1259 * dst: buffer to place the converted data. 1260 * provider: buffer provider to obtain source data. 1261 * frames: number of frames to convert 1262 * 1263 * Returns the number of frames converted. 1264 */ 1265 size_t convert(void *dst, AudioBufferProvider *provider, size_t frames); 1266 1267 // returns NO_ERROR if constructor was successful 1268 status_t initCheck() const { 1269 // mSrcChannelMask set on successful updateParameters 1270 return mSrcChannelMask != AUDIO_CHANNEL_INVALID ? NO_ERROR : NO_INIT; 1271 } 1272 1273 // allows dynamic reconfigure of all parameters 1274 status_t updateParameters( 1275 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 1276 uint32_t srcSampleRate, 1277 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 1278 uint32_t dstSampleRate); 1279 1280 // called to reset resampler buffers on record track discontinuity 1281 void reset() { 1282 if (mResampler != NULL) { 1283 mResampler->reset(); 1284 } 1285 } 1286 1287 private: 1288 // format conversion when not using resampler 1289 void convertNoResampler(void *dst, const void *src, size_t frames); 1290 1291 // format conversion when using resampler; modifies src in-place 1292 void convertResampler(void *dst, /*not-a-const*/ void *src, size_t frames); 1293 1294 // user provided information 1295 audio_channel_mask_t mSrcChannelMask; 1296 audio_format_t mSrcFormat; 1297 uint32_t mSrcSampleRate; 1298 audio_channel_mask_t mDstChannelMask; 1299 audio_format_t mDstFormat; 1300 uint32_t mDstSampleRate; 1301 1302 // derived information 1303 uint32_t mSrcChannelCount; 1304 uint32_t mDstChannelCount; 1305 size_t mDstFrameSize; 1306 1307 // format conversion buffer 1308 void *mBuf; 1309 size_t mBufFrames; 1310 size_t mBufFrameSize; 1311 1312 // resampler info 1313 AudioResampler *mResampler; 1314 1315 bool mIsLegacyDownmix; // legacy stereo to mono conversion needed 1316 bool mIsLegacyUpmix; // legacy mono to stereo conversion needed 1317 bool mRequiresFloat; // data processing requires float (e.g. resampler) 1318 PassthruBufferProvider *mInputConverterProvider; // converts input to float 1319 int8_t mIdxAry[sizeof(uint32_t) * 8]; // used for channel mask conversion 1320 }; 1321 1322#include "RecordTracks.h" 1323 1324 RecordThread(const sp<AudioFlinger>& audioFlinger, 1325 AudioStreamIn *input, 1326 audio_io_handle_t id, 1327 audio_devices_t outDevice, 1328 audio_devices_t inDevice, 1329 bool systemReady 1330#ifdef TEE_SINK 1331 , const sp<NBAIO_Sink>& teeSink 1332#endif 1333 ); 1334 virtual ~RecordThread(); 1335 1336 // no addTrack_l ? 1337 void destroyTrack_l(const sp<RecordTrack>& track); 1338 void removeTrack_l(const sp<RecordTrack>& track); 1339 1340 void dumpInternals(int fd, const Vector<String16>& args); 1341 void dumpTracks(int fd, const Vector<String16>& args); 1342 1343 // Thread virtuals 1344 virtual bool threadLoop(); 1345 1346 // RefBase 1347 virtual void onFirstRef(); 1348 1349 virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; } 1350 1351 virtual sp<MemoryDealer> readOnlyHeap() const { return mReadOnlyHeap; } 1352 1353 virtual sp<IMemory> pipeMemory() const { return mPipeMemory; } 1354 1355 sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l( 1356 const sp<AudioFlinger::Client>& client, 1357 uint32_t sampleRate, 1358 audio_format_t format, 1359 audio_channel_mask_t channelMask, 1360 size_t *pFrameCount, 1361 audio_session_t sessionId, 1362 size_t *notificationFrames, 1363 uid_t uid, 1364 audio_input_flags_t *flags, 1365 pid_t tid, 1366 status_t *status /*non-NULL*/); 1367 1368 status_t start(RecordTrack* recordTrack, 1369 AudioSystem::sync_event_t event, 1370 audio_session_t triggerSession); 1371 1372 // ask the thread to stop the specified track, and 1373 // return true if the caller should then do it's part of the stopping process 1374 bool stop(RecordTrack* recordTrack); 1375 1376 void dump(int fd, const Vector<String16>& args); 1377 AudioStreamIn* clearInput(); 1378 virtual sp<StreamHalInterface> stream() const; 1379 1380 1381 virtual bool checkForNewParameter_l(const String8& keyValuePair, 1382 status_t& status); 1383 virtual void cacheParameters_l() {} 1384 virtual String8 getParameters(const String8& keys); 1385 virtual void ioConfigChanged(audio_io_config_event event, pid_t pid = 0); 1386 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 1387 audio_patch_handle_t *handle); 1388 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 1389 1390 void addPatchRecord(const sp<PatchRecord>& record); 1391 void deletePatchRecord(const sp<PatchRecord>& record); 1392 1393 void readInputParameters_l(); 1394 virtual uint32_t getInputFramesLost(); 1395 1396 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1397 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1398 virtual uint32_t hasAudioSession_l(audio_session_t sessionId) const; 1399 1400 // Return the set of unique session IDs across all tracks. 1401 // The keys are the session IDs, and the associated values are meaningless. 1402 // FIXME replace by Set [and implement Bag/Multiset for other uses]. 1403 KeyedVector<audio_session_t, bool> sessionIds() const; 1404 1405 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 1406 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 1407 1408 static void syncStartEventCallback(const wp<SyncEvent>& event); 1409 1410 virtual size_t frameCount() const { return mFrameCount; } 1411 bool hasFastCapture() const { return mFastCapture != 0; } 1412 virtual void getAudioPortConfig(struct audio_port_config *config); 1413 1414 virtual status_t checkEffectCompatibility_l(const effect_descriptor_t *desc, 1415 audio_session_t sessionId); 1416 1417private: 1418 // Enter standby if not already in standby, and set mStandby flag 1419 void standbyIfNotAlreadyInStandby(); 1420 1421 // Call the HAL standby method unconditionally, and don't change mStandby flag 1422 void inputStandBy(); 1423 1424 AudioStreamIn *mInput; 1425 SortedVector < sp<RecordTrack> > mTracks; 1426 // mActiveTracks has dual roles: it indicates the current active track(s), and 1427 // is used together with mStartStopCond to indicate start()/stop() progress 1428 SortedVector< sp<RecordTrack> > mActiveTracks; 1429 // generation counter for mActiveTracks 1430 int mActiveTracksGen; 1431 Condition mStartStopCond; 1432 1433 // resampler converts input at HAL Hz to output at AudioRecord client Hz 1434 void *mRsmpInBuffer; // size = mRsmpInFramesOA 1435 size_t mRsmpInFrames; // size of resampler input in frames 1436 size_t mRsmpInFramesP2;// size rounded up to a power-of-2 1437 size_t mRsmpInFramesOA;// mRsmpInFramesP2 + over-allocation 1438 1439 // rolling index that is never cleared 1440 int32_t mRsmpInRear; // last filled frame + 1 1441 1442 // For dumpsys 1443 const sp<NBAIO_Sink> mTeeSink; 1444 1445 const sp<MemoryDealer> mReadOnlyHeap; 1446 1447 // one-time initialization, no locks required 1448 sp<FastCapture> mFastCapture; // non-0 if there is also 1449 // a fast capture 1450 1451 // FIXME audio watchdog thread 1452 1453 // contents are not guaranteed to be consistent, no locks required 1454 FastCaptureDumpState mFastCaptureDumpState; 1455#ifdef STATE_QUEUE_DUMP 1456 // FIXME StateQueue observer and mutator dump fields 1457#endif 1458 // FIXME audio watchdog dump 1459 1460 // accessible only within the threadLoop(), no locks required 1461 // mFastCapture->sq() // for mutating and pushing state 1462 int32_t mFastCaptureFutex; // for cold idle 1463 1464 // The HAL input source is treated as non-blocking, 1465 // but current implementation is blocking 1466 sp<NBAIO_Source> mInputSource; 1467 // The source for the normal capture thread to read from: mInputSource or mPipeSource 1468 sp<NBAIO_Source> mNormalSource; 1469 // If a fast capture is present, the non-blocking pipe sink written to by fast capture, 1470 // otherwise clear 1471 sp<NBAIO_Sink> mPipeSink; 1472 // If a fast capture is present, the non-blocking pipe source read by normal thread, 1473 // otherwise clear 1474 sp<NBAIO_Source> mPipeSource; 1475 // Depth of pipe from fast capture to normal thread and fast clients, always power of 2 1476 size_t mPipeFramesP2; 1477 // If a fast capture is present, the Pipe as IMemory, otherwise clear 1478 sp<IMemory> mPipeMemory; 1479 1480 static const size_t kFastCaptureLogSize = 4 * 1024; 1481 sp<NBLog::Writer> mFastCaptureNBLogWriter; 1482 1483 bool mFastTrackAvail; // true if fast track available 1484}; 1485