Threads.h revision 2148bf0e79c436b8764b9edc4c8f2730cce98a32
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*/); 299 300 // 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 392protected: 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_l(const SortedVector<int> &uids); 408 void getPowerManager_l(); 409 void setEffectSuspended_l(const effect_uuid_t *type, 410 bool suspend, 411 audio_session_t sessionId); 412 // updated mSuspendedSessions when an effect suspended or restored 413 void updateSuspendedSessions_l(const effect_uuid_t *type, 414 bool suspend, 415 audio_session_t sessionId); 416 // check if some effects must be suspended when an effect chain is added 417 void checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain); 418 419 String16 getWakeLockTag(); 420 421 virtual void preExit() { } 422 virtual void setMasterMono_l(bool mono __unused) { } 423 virtual bool requireMonoBlend() { return false; } 424 425 friend class AudioFlinger; // for mEffectChains 426 427 const type_t mType; 428 429 // Used by parameters, config events, addTrack_l, exit 430 Condition mWaitWorkCV; 431 432 const sp<AudioFlinger> mAudioFlinger; 433 434 // updated by PlaybackThread::readOutputParameters_l() or 435 // RecordThread::readInputParameters_l() 436 uint32_t mSampleRate; 437 size_t mFrameCount; // output HAL, direct output, record 438 audio_channel_mask_t mChannelMask; 439 uint32_t mChannelCount; 440 size_t mFrameSize; 441 // not HAL frame size, this is for output sink (to pipe to fast mixer) 442 audio_format_t mFormat; // Source format for Recording and 443 // Sink format for Playback. 444 // Sink format may be different than 445 // HAL format if Fastmixer is used. 446 audio_format_t mHALFormat; 447 size_t mBufferSize; // HAL buffer size for read() or write() 448 449 Vector< sp<ConfigEvent> > mConfigEvents; 450 Vector< sp<ConfigEvent> > mPendingConfigEvents; // events awaiting system ready 451 452 // These fields are written and read by thread itself without lock or barrier, 453 // and read by other threads without lock or barrier via standby(), outDevice() 454 // and inDevice(). 455 // Because of the absence of a lock or barrier, any other thread that reads 456 // these fields must use the information in isolation, or be prepared to deal 457 // with possibility that it might be inconsistent with other information. 458 bool mStandby; // Whether thread is currently in standby. 459 audio_devices_t mOutDevice; // output device 460 audio_devices_t mInDevice; // input device 461 audio_devices_t mPrevOutDevice; // previous output device 462 audio_devices_t mPrevInDevice; // previous input device 463 struct audio_patch mPatch; 464 audio_source_t mAudioSource; 465 466 const audio_io_handle_t mId; 467 Vector< sp<EffectChain> > mEffectChains; 468 469 static const int kThreadNameLength = 16; // prctl(PR_SET_NAME) limit 470 char mThreadName[kThreadNameLength]; // guaranteed NUL-terminated 471 sp<IPowerManager> mPowerManager; 472 sp<IBinder> mWakeLockToken; 473 const sp<PMDeathRecipient> mDeathRecipient; 474 // list of suspended effects per session and per type. The first (outer) vector is 475 // keyed by session ID, the second (inner) by type UUID timeLow field 476 KeyedVector< audio_session_t, KeyedVector< int, sp<SuspendedSessionDesc> > > 477 mSuspendedSessions; 478 static const size_t kLogSize = 4 * 1024; 479 sp<NBLog::Writer> mNBLogWriter; 480 bool mSystemReady; 481 bool mNotifiedBatteryStart; 482 ExtendedTimestamp mTimestamp; 483}; 484 485// --- PlaybackThread --- 486class PlaybackThread : public ThreadBase, public StreamOutHalInterfaceCallback { 487public: 488 489#include "PlaybackTracks.h" 490 491 enum mixer_state { 492 MIXER_IDLE, // no active tracks 493 MIXER_TRACKS_ENABLED, // at least one active track, but no track has any data ready 494 MIXER_TRACKS_READY, // at least one active track, and at least one track has data 495 MIXER_DRAIN_TRACK, // drain currently playing track 496 MIXER_DRAIN_ALL, // fully drain the hardware 497 // standby mode does not have an enum value 498 // suspend by audio policy manager is orthogonal to mixer state 499 }; 500 501 // retry count before removing active track in case of underrun on offloaded thread: 502 // we need to make sure that AudioTrack client has enough time to send large buffers 503 //FIXME may be more appropriate if expressed in time units. Need to revise how underrun is 504 // handled for offloaded tracks 505 static const int8_t kMaxTrackRetriesOffload = 20; 506 static const int8_t kMaxTrackStartupRetriesOffload = 100; 507 static const int8_t kMaxTrackStopRetriesOffload = 2; 508 // 14 tracks max per client allows for 2 misbehaving application leaving 4 available tracks. 509 static const uint32_t kMaxTracksPerUid = 14; 510 511 PlaybackThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 512 audio_io_handle_t id, audio_devices_t device, type_t type, bool systemReady); 513 virtual ~PlaybackThread(); 514 515 void dump(int fd, const Vector<String16>& args); 516 517 // Thread virtuals 518 virtual bool threadLoop(); 519 520 // RefBase 521 virtual void onFirstRef(); 522 523 virtual status_t checkEffectCompatibility_l(const effect_descriptor_t *desc, 524 audio_session_t sessionId); 525 526protected: 527 // Code snippets that were lifted up out of threadLoop() 528 virtual void threadLoop_mix() = 0; 529 virtual void threadLoop_sleepTime() = 0; 530 virtual ssize_t threadLoop_write(); 531 virtual void threadLoop_drain(); 532 virtual void threadLoop_standby(); 533 virtual void threadLoop_exit(); 534 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 535 536 // prepareTracks_l reads and writes mActiveTracks, and returns 537 // the pending set of tracks to remove via Vector 'tracksToRemove'. The caller 538 // is responsible for clearing or destroying this Vector later on, when it 539 // is safe to do so. That will drop the final ref count and destroy the tracks. 540 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove) = 0; 541 void removeTracks_l(const Vector< sp<Track> >& tracksToRemove); 542 543 // StreamOutHalInterfaceCallback implementation 544 virtual void onWriteReady(); 545 virtual void onDrainReady(); 546 virtual void onError(); 547 548 void resetWriteBlocked(uint32_t sequence); 549 void resetDraining(uint32_t sequence); 550 551 virtual bool waitingAsyncCallback(); 552 virtual bool waitingAsyncCallback_l(); 553 virtual bool shouldStandby_l(); 554 virtual void onAddNewTrack_l(); 555 void onAsyncError(); // error reported by AsyncCallbackThread 556 557 // ThreadBase virtuals 558 virtual void preExit(); 559 560 virtual bool keepWakeLock() const { return true; } 561 562public: 563 564 virtual status_t initCheck() const { return (mOutput == NULL) ? NO_INIT : NO_ERROR; } 565 566 // return estimated latency in milliseconds, as reported by HAL 567 uint32_t latency() const; 568 // same, but lock must already be held 569 uint32_t latency_l() const; 570 571 void setMasterVolume(float value); 572 void setMasterMute(bool muted); 573 574 void setStreamVolume(audio_stream_type_t stream, float value); 575 void setStreamMute(audio_stream_type_t stream, bool muted); 576 577 float streamVolume(audio_stream_type_t stream) const; 578 579 sp<Track> createTrack_l( 580 const sp<AudioFlinger::Client>& client, 581 audio_stream_type_t streamType, 582 uint32_t sampleRate, 583 audio_format_t format, 584 audio_channel_mask_t channelMask, 585 size_t *pFrameCount, 586 const sp<IMemory>& sharedBuffer, 587 audio_session_t sessionId, 588 audio_output_flags_t *flags, 589 pid_t tid, 590 uid_t uid, 591 status_t *status /*non-NULL*/); 592 593 AudioStreamOut* getOutput() const; 594 AudioStreamOut* clearOutput(); 595 virtual sp<StreamHalInterface> stream() const; 596 597 // a very large number of suspend() will eventually wraparound, but unlikely 598 void suspend() { (void) android_atomic_inc(&mSuspended); } 599 void restore() 600 { 601 // if restore() is done without suspend(), get back into 602 // range so that the next suspend() will operate correctly 603 if (android_atomic_dec(&mSuspended) <= 0) { 604 android_atomic_release_store(0, &mSuspended); 605 } 606 } 607 bool isSuspended() const 608 { return android_atomic_acquire_load(&mSuspended) > 0; } 609 610 virtual String8 getParameters(const String8& keys); 611 virtual void ioConfigChanged(audio_io_config_event event, pid_t pid = 0); 612 status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames); 613 // FIXME rename mixBuffer() to sinkBuffer() and remove int16_t* dependency. 614 // Consider also removing and passing an explicit mMainBuffer initialization 615 // parameter to AF::PlaybackThread::Track::Track(). 616 int16_t *mixBuffer() const { 617 return reinterpret_cast<int16_t *>(mSinkBuffer); }; 618 619 virtual void detachAuxEffect_l(int effectId); 620 status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track>& track, 621 int EffectId); 622 status_t attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track>& track, 623 int EffectId); 624 625 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 626 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 627 virtual uint32_t hasAudioSession_l(audio_session_t sessionId) const; 628 virtual uint32_t getStrategyForSession_l(audio_session_t sessionId); 629 630 631 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 632 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 633 634 // called with AudioFlinger lock held 635 bool invalidateTracks_l(audio_stream_type_t streamType); 636 virtual void invalidateTracks(audio_stream_type_t streamType); 637 638 virtual size_t frameCount() const { return mNormalFrameCount; } 639 640 status_t getTimestamp_l(AudioTimestamp& timestamp); 641 642 void addPatchTrack(const sp<PatchTrack>& track); 643 void deletePatchTrack(const sp<PatchTrack>& track); 644 645 virtual void getAudioPortConfig(struct audio_port_config *config); 646 647protected: 648 // updated by readOutputParameters_l() 649 size_t mNormalFrameCount; // normal mixer and effects 650 651 bool mThreadThrottle; // throttle the thread processing 652 uint32_t mThreadThrottleTimeMs; // throttle time for MIXER threads 653 uint32_t mThreadThrottleEndMs; // notify once per throttling 654 uint32_t mHalfBufferMs; // half the buffer size in milliseconds 655 656 void* mSinkBuffer; // frame size aligned sink buffer 657 658 // TODO: 659 // Rearrange the buffer info into a struct/class with 660 // clear, copy, construction, destruction methods. 661 // 662 // mSinkBuffer also has associated with it: 663 // 664 // mSinkBufferSize: Sink Buffer Size 665 // mFormat: Sink Buffer Format 666 667 // Mixer Buffer (mMixerBuffer*) 668 // 669 // In the case of floating point or multichannel data, which is not in the 670 // sink format, it is required to accumulate in a higher precision or greater channel count 671 // buffer before downmixing or data conversion to the sink buffer. 672 673 // Set to "true" to enable the Mixer Buffer otherwise mixer output goes to sink buffer. 674 bool mMixerBufferEnabled; 675 676 // Storage, 32 byte aligned (may make this alignment a requirement later). 677 // Due to constraints on mNormalFrameCount, the buffer size is a multiple of 16 frames. 678 void* mMixerBuffer; 679 680 // Size of mMixerBuffer in bytes: mNormalFrameCount * #channels * sampsize. 681 size_t mMixerBufferSize; 682 683 // The audio format of mMixerBuffer. Set to AUDIO_FORMAT_PCM_(FLOAT|16_BIT) only. 684 audio_format_t mMixerBufferFormat; 685 686 // An internal flag set to true by MixerThread::prepareTracks_l() 687 // when mMixerBuffer contains valid data after mixing. 688 bool mMixerBufferValid; 689 690 // Effects Buffer (mEffectsBuffer*) 691 // 692 // In the case of effects data, which is not in the sink format, 693 // it is required to accumulate in a different buffer before data conversion 694 // to the sink buffer. 695 696 // Set to "true" to enable the Effects Buffer otherwise effects output goes to sink buffer. 697 bool mEffectBufferEnabled; 698 699 // Storage, 32 byte aligned (may make this alignment a requirement later). 700 // Due to constraints on mNormalFrameCount, the buffer size is a multiple of 16 frames. 701 void* mEffectBuffer; 702 703 // Size of mEffectsBuffer in bytes: mNormalFrameCount * #channels * sampsize. 704 size_t mEffectBufferSize; 705 706 // The audio format of mEffectsBuffer. Set to AUDIO_FORMAT_PCM_16_BIT only. 707 audio_format_t mEffectBufferFormat; 708 709 // An internal flag set to true by MixerThread::prepareTracks_l() 710 // when mEffectsBuffer contains valid data after mixing. 711 // 712 // When this is set, all mixer data is routed into the effects buffer 713 // for any processing (including output processing). 714 bool mEffectBufferValid; 715 716 // suspend count, > 0 means suspended. While suspended, the thread continues to pull from 717 // tracks and mix, but doesn't write to HAL. A2DP and SCO HAL implementations can't handle 718 // concurrent use of both of them, so Audio Policy Service suspends one of the threads to 719 // workaround that restriction. 720 // 'volatile' means accessed via atomic operations and no lock. 721 volatile int32_t mSuspended; 722 723 int64_t mBytesWritten; 724 int64_t mFramesWritten; // not reset on standby 725 int64_t mSuspendedFrames; // not reset on standby 726private: 727 // mMasterMute is in both PlaybackThread and in AudioFlinger. When a 728 // PlaybackThread needs to find out if master-muted, it checks it's local 729 // copy rather than the one in AudioFlinger. This optimization saves a lock. 730 bool mMasterMute; 731 void setMasterMute_l(bool muted) { mMasterMute = muted; } 732protected: 733 SortedVector< wp<Track> > mActiveTracks; // FIXME check if this could be sp<> 734 SortedVector<int> mWakeLockUids; 735 int mActiveTracksGeneration; 736 wp<Track> mLatestActiveTrack; // latest track added to mActiveTracks 737 738 // Allocate a track name for a given channel mask. 739 // Returns name >= 0 if successful, -1 on failure. 740 virtual int getTrackName_l(audio_channel_mask_t channelMask, audio_format_t format, 741 audio_session_t sessionId, uid_t uid) = 0; 742 virtual void deleteTrackName_l(int name) = 0; 743 744 // Time to sleep between cycles when: 745 virtual uint32_t activeSleepTimeUs() const; // mixer state MIXER_TRACKS_ENABLED 746 virtual uint32_t idleSleepTimeUs() const = 0; // mixer state MIXER_IDLE 747 virtual uint32_t suspendSleepTimeUs() const = 0; // audio policy manager suspended us 748 // No sleep when mixer state == MIXER_TRACKS_READY; relies on audio HAL stream->write() 749 // No sleep in standby mode; waits on a condition 750 751 // Code snippets that are temporarily lifted up out of threadLoop() until the merge 752 void checkSilentMode_l(); 753 754 // Non-trivial for DUPLICATING only 755 virtual void saveOutputTracks() { } 756 virtual void clearOutputTracks() { } 757 758 // Cache various calculated values, at threadLoop() entry and after a parameter change 759 virtual void cacheParameters_l(); 760 761 virtual uint32_t correctLatency_l(uint32_t latency) const; 762 763 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 764 audio_patch_handle_t *handle); 765 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 766 767 bool usesHwAvSync() const { return (mType == DIRECT) && (mOutput != NULL) 768 && mHwSupportsPause 769 && (mOutput->flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC); } 770 771 uint32_t trackCountForUid_l(uid_t uid); 772 773private: 774 775 friend class AudioFlinger; // for numerous 776 777 PlaybackThread& operator = (const PlaybackThread&); 778 779 status_t addTrack_l(const sp<Track>& track); 780 bool destroyTrack_l(const sp<Track>& track); 781 void removeTrack_l(const sp<Track>& track); 782 void broadcast_l(); 783 784 void readOutputParameters_l(); 785 786 virtual void dumpInternals(int fd, const Vector<String16>& args); 787 void dumpTracks(int fd, const Vector<String16>& args); 788 789 SortedVector< sp<Track> > mTracks; 790 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 791 AudioStreamOut *mOutput; 792 793 float mMasterVolume; 794 nsecs_t mLastWriteTime; 795 int mNumWrites; 796 int mNumDelayedWrites; 797 bool mInWrite; 798 799 // FIXME rename these former local variables of threadLoop to standard "m" names 800 nsecs_t mStandbyTimeNs; 801 size_t mSinkBufferSize; 802 803 // cached copies of activeSleepTimeUs() and idleSleepTimeUs() made by cacheParameters_l() 804 uint32_t mActiveSleepTimeUs; 805 uint32_t mIdleSleepTimeUs; 806 807 uint32_t mSleepTimeUs; 808 809 // mixer status returned by prepareTracks_l() 810 mixer_state mMixerStatus; // current cycle 811 // previous cycle when in prepareTracks_l() 812 mixer_state mMixerStatusIgnoringFastTracks; 813 // FIXME or a separate ready state per track 814 815 // FIXME move these declarations into the specific sub-class that needs them 816 // MIXER only 817 uint32_t sleepTimeShift; 818 819 // same as AudioFlinger::mStandbyTimeInNsecs except for DIRECT which uses a shorter value 820 nsecs_t mStandbyDelayNs; 821 822 // MIXER only 823 nsecs_t maxPeriod; 824 825 // DUPLICATING only 826 uint32_t writeFrames; 827 828 size_t mBytesRemaining; 829 size_t mCurrentWriteLength; 830 bool mUseAsyncWrite; 831 // mWriteAckSequence contains current write sequence on bits 31-1. The write sequence is 832 // incremented each time a write(), a flush() or a standby() occurs. 833 // Bit 0 is set when a write blocks and indicates a callback is expected. 834 // Bit 0 is reset by the async callback thread calling resetWriteBlocked(). Out of sequence 835 // callbacks are ignored. 836 uint32_t mWriteAckSequence; 837 // mDrainSequence contains current drain sequence on bits 31-1. The drain sequence is 838 // incremented each time a drain is requested or a flush() or standby() occurs. 839 // Bit 0 is set when the drain() command is called at the HAL and indicates a callback is 840 // expected. 841 // Bit 0 is reset by the async callback thread calling resetDraining(). Out of sequence 842 // callbacks are ignored. 843 uint32_t mDrainSequence; 844 // A condition that must be evaluated by prepareTrack_l() has changed and we must not wait 845 // for async write callback in the thread loop before evaluating it 846 bool mSignalPending; 847 sp<AsyncCallbackThread> mCallbackThread; 848 849private: 850 // The HAL output sink is treated as non-blocking, but current implementation is blocking 851 sp<NBAIO_Sink> mOutputSink; 852 // If a fast mixer is present, the blocking pipe sink, otherwise clear 853 sp<NBAIO_Sink> mPipeSink; 854 // The current sink for the normal mixer to write it's (sub)mix, mOutputSink or mPipeSink 855 sp<NBAIO_Sink> mNormalSink; 856#ifdef TEE_SINK 857 // For dumpsys 858 sp<NBAIO_Sink> mTeeSink; 859 sp<NBAIO_Source> mTeeSource; 860#endif 861 uint32_t mScreenState; // cached copy of gScreenState 862 static const size_t kFastMixerLogSize = 4 * 1024; 863 sp<NBLog::Writer> mFastMixerNBLogWriter; 864 865 // Do not call from a sched_fifo thread as it uses a system time call 866 // and obtains a local mutex. 867 class LocalLog { 868 public: 869 void log(const char *fmt, ...) { 870 va_list val; 871 va_start(val, fmt); 872 873 // format to buffer 874 char buffer[512]; 875 int length = vsnprintf(buffer, sizeof(buffer), fmt, val); 876 if (length >= (signed)sizeof(buffer)) { 877 length = sizeof(buffer) - 1; 878 } 879 880 // strip out trailing newline 881 while (length > 0 && buffer[length - 1] == '\n') { 882 buffer[--length] = 0; 883 } 884 885 // store in circular array 886 AutoMutex _l(mLock); 887 mLog.emplace_back( 888 std::make_pair(systemTime(SYSTEM_TIME_REALTIME), std::string(buffer))); 889 if (mLog.size() > kLogSize) { 890 mLog.pop_front(); 891 } 892 893 va_end(val); 894 } 895 896 void dump(int fd, const Vector<String16>& args, const char *prefix = "") { 897 if (!AudioFlinger::dumpTryLock(mLock)) return; // a local lock, shouldn't happen 898 if (mLog.size() > 0) { 899 bool dumpAll = false; 900 for (const auto &arg : args) { 901 if (arg == String16("--locallog")) { 902 dumpAll = true; 903 } 904 } 905 906 dprintf(fd, "Local Log:\n"); 907 auto it = mLog.begin(); 908 if (!dumpAll && mLog.size() > kLogPrint) { 909 it += (mLog.size() - kLogPrint); 910 } 911 for (; it != mLog.end(); ++it) { 912 const int64_t ns = it->first; 913 const int ns_per_sec = 1000000000; 914 const time_t sec = ns / ns_per_sec; 915 struct tm tm; 916 localtime_r(&sec, &tm); 917 918 dprintf(fd, "%s%02d-%02d %02d:%02d:%02d.%03d %s\n", 919 prefix, 920 tm.tm_mon + 1, // localtime_r uses months in 0 - 11 range 921 tm.tm_mday, tm.tm_hour, tm.tm_min, tm.tm_sec, 922 (int)(ns % ns_per_sec / 1000000), 923 it->second.c_str()); 924 } 925 } 926 mLock.unlock(); 927 } 928 929 private: 930 Mutex mLock; 931 static const size_t kLogSize = 256; // full history 932 static const size_t kLogPrint = 32; // default print history 933 std::deque<std::pair<int64_t, std::string>> mLog; 934 } mLocalLog; 935 936public: 937 virtual bool hasFastMixer() const = 0; 938 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex __unused) const 939 { FastTrackUnderruns dummy; return dummy; } 940 941protected: 942 // accessed by both binder threads and within threadLoop(), lock on mutex needed 943 unsigned mFastTrackAvailMask; // bit i set if fast track [i] is available 944 bool mHwSupportsPause; 945 bool mHwPaused; 946 bool mFlushPending; 947}; 948 949class MixerThread : public PlaybackThread { 950public: 951 MixerThread(const sp<AudioFlinger>& audioFlinger, 952 AudioStreamOut* output, 953 audio_io_handle_t id, 954 audio_devices_t device, 955 bool systemReady, 956 type_t type = MIXER); 957 virtual ~MixerThread(); 958 959 // Thread virtuals 960 961 virtual bool checkForNewParameter_l(const String8& keyValuePair, 962 status_t& status); 963 virtual void dumpInternals(int fd, const Vector<String16>& args); 964 965protected: 966 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 967 virtual int getTrackName_l(audio_channel_mask_t channelMask, audio_format_t format, 968 audio_session_t sessionId, uid_t uid); 969 virtual void deleteTrackName_l(int name); 970 virtual uint32_t idleSleepTimeUs() const; 971 virtual uint32_t suspendSleepTimeUs() const; 972 virtual void cacheParameters_l(); 973 974 virtual void acquireWakeLock_l(int uid = -1) { 975 PlaybackThread::acquireWakeLock_l(uid); 976 if (hasFastMixer()) { 977 mFastMixer->setBoottimeOffset( 978 mTimestamp.mTimebaseOffset[ExtendedTimestamp::TIMEBASE_BOOTTIME]); 979 } 980 } 981 982 // threadLoop snippets 983 virtual ssize_t threadLoop_write(); 984 virtual void threadLoop_standby(); 985 virtual void threadLoop_mix(); 986 virtual void threadLoop_sleepTime(); 987 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 988 virtual uint32_t correctLatency_l(uint32_t latency) const; 989 990 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 991 audio_patch_handle_t *handle); 992 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 993 994 AudioMixer* mAudioMixer; // normal mixer 995private: 996 // one-time initialization, no locks required 997 sp<FastMixer> mFastMixer; // non-0 if there is also a fast mixer 998 sp<AudioWatchdog> mAudioWatchdog; // non-0 if there is an audio watchdog thread 999 1000 // contents are not guaranteed to be consistent, no locks required 1001 FastMixerDumpState mFastMixerDumpState; 1002#ifdef STATE_QUEUE_DUMP 1003 StateQueueObserverDump mStateQueueObserverDump; 1004 StateQueueMutatorDump mStateQueueMutatorDump; 1005#endif 1006 AudioWatchdogDump mAudioWatchdogDump; 1007 1008 // accessible only within the threadLoop(), no locks required 1009 // mFastMixer->sq() // for mutating and pushing state 1010 int32_t mFastMixerFutex; // for cold idle 1011 1012 std::atomic_bool mMasterMono; 1013public: 1014 virtual bool hasFastMixer() const { return mFastMixer != 0; } 1015 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const { 1016 ALOG_ASSERT(fastIndex < FastMixerState::sMaxFastTracks); 1017 return mFastMixerDumpState.mTracks[fastIndex].mUnderruns; 1018 } 1019 1020protected: 1021 virtual void setMasterMono_l(bool mono) { 1022 mMasterMono.store(mono); 1023 if (mFastMixer != nullptr) { /* hasFastMixer() */ 1024 mFastMixer->setMasterMono(mMasterMono); 1025 } 1026 } 1027 // the FastMixer performs mono blend if it exists. 1028 // Blending with limiter is not idempotent, 1029 // and blending without limiter is idempotent but inefficient to do twice. 1030 virtual bool requireMonoBlend() { return mMasterMono.load() && !hasFastMixer(); } 1031}; 1032 1033class DirectOutputThread : public PlaybackThread { 1034public: 1035 1036 DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1037 audio_io_handle_t id, audio_devices_t device, bool systemReady); 1038 virtual ~DirectOutputThread(); 1039 1040 // Thread virtuals 1041 1042 virtual bool checkForNewParameter_l(const String8& keyValuePair, 1043 status_t& status); 1044 virtual void flushHw_l(); 1045 1046protected: 1047 virtual int getTrackName_l(audio_channel_mask_t channelMask, audio_format_t format, 1048 audio_session_t sessionId, uid_t uid); 1049 virtual void deleteTrackName_l(int name); 1050 virtual uint32_t activeSleepTimeUs() const; 1051 virtual uint32_t idleSleepTimeUs() const; 1052 virtual uint32_t suspendSleepTimeUs() const; 1053 virtual void cacheParameters_l(); 1054 1055 // threadLoop snippets 1056 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1057 virtual void threadLoop_mix(); 1058 virtual void threadLoop_sleepTime(); 1059 virtual void threadLoop_exit(); 1060 virtual bool shouldStandby_l(); 1061 1062 virtual void onAddNewTrack_l(); 1063 1064 // volumes last sent to audio HAL with stream->set_volume() 1065 float mLeftVolFloat; 1066 float mRightVolFloat; 1067 1068 DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1069 audio_io_handle_t id, uint32_t device, ThreadBase::type_t type, 1070 bool systemReady); 1071 void processVolume_l(Track *track, bool lastTrack); 1072 1073 // prepareTracks_l() tells threadLoop_mix() the name of the single active track 1074 sp<Track> mActiveTrack; 1075 1076 wp<Track> mPreviousTrack; // used to detect track switch 1077 1078public: 1079 virtual bool hasFastMixer() const { return false; } 1080}; 1081 1082class OffloadThread : public DirectOutputThread { 1083public: 1084 1085 OffloadThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1086 audio_io_handle_t id, uint32_t device, bool systemReady); 1087 virtual ~OffloadThread() {}; 1088 virtual void flushHw_l(); 1089 1090protected: 1091 // threadLoop snippets 1092 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1093 virtual void threadLoop_exit(); 1094 1095 virtual bool waitingAsyncCallback(); 1096 virtual bool waitingAsyncCallback_l(); 1097 virtual void invalidateTracks(audio_stream_type_t streamType); 1098 1099 virtual bool keepWakeLock() const { return (mKeepWakeLock || (mDrainSequence & 1)); } 1100 1101private: 1102 size_t mPausedWriteLength; // length in bytes of write interrupted by pause 1103 size_t mPausedBytesRemaining; // bytes still waiting in mixbuffer after resume 1104 bool mKeepWakeLock; // keep wake lock while waiting for write callback 1105 uint64_t mOffloadUnderrunPosition; // Current frame position for offloaded playback 1106 // used and valid only during underrun. ~0 if 1107 // no underrun has occurred during playback and 1108 // is not reset on standby. 1109}; 1110 1111class AsyncCallbackThread : public Thread { 1112public: 1113 1114 explicit AsyncCallbackThread(const wp<PlaybackThread>& playbackThread); 1115 1116 virtual ~AsyncCallbackThread(); 1117 1118 // Thread virtuals 1119 virtual bool threadLoop(); 1120 1121 // RefBase 1122 virtual void onFirstRef(); 1123 1124 void exit(); 1125 void setWriteBlocked(uint32_t sequence); 1126 void resetWriteBlocked(); 1127 void setDraining(uint32_t sequence); 1128 void resetDraining(); 1129 void setAsyncError(); 1130 1131private: 1132 const wp<PlaybackThread> mPlaybackThread; 1133 // mWriteAckSequence corresponds to the last write sequence passed by the offload thread via 1134 // setWriteBlocked(). The sequence is shifted one bit to the left and the lsb is used 1135 // to indicate that the callback has been received via resetWriteBlocked() 1136 uint32_t mWriteAckSequence; 1137 // mDrainSequence corresponds to the last drain sequence passed by the offload thread via 1138 // setDraining(). The sequence is shifted one bit to the left and the lsb is used 1139 // to indicate that the callback has been received via resetDraining() 1140 uint32_t mDrainSequence; 1141 Condition mWaitWorkCV; 1142 Mutex mLock; 1143 bool mAsyncError; 1144}; 1145 1146class DuplicatingThread : public MixerThread { 1147public: 1148 DuplicatingThread(const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread, 1149 audio_io_handle_t id, bool systemReady); 1150 virtual ~DuplicatingThread(); 1151 1152 // Thread virtuals 1153 void addOutputTrack(MixerThread* thread); 1154 void removeOutputTrack(MixerThread* thread); 1155 uint32_t waitTimeMs() const { return mWaitTimeMs; } 1156protected: 1157 virtual uint32_t activeSleepTimeUs() const; 1158 1159private: 1160 bool outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks); 1161protected: 1162 // threadLoop snippets 1163 virtual void threadLoop_mix(); 1164 virtual void threadLoop_sleepTime(); 1165 virtual ssize_t threadLoop_write(); 1166 virtual void threadLoop_standby(); 1167 virtual void cacheParameters_l(); 1168 1169private: 1170 // called from threadLoop, addOutputTrack, removeOutputTrack 1171 virtual void updateWaitTime_l(); 1172protected: 1173 virtual void saveOutputTracks(); 1174 virtual void clearOutputTracks(); 1175private: 1176 1177 uint32_t mWaitTimeMs; 1178 SortedVector < sp<OutputTrack> > outputTracks; 1179 SortedVector < sp<OutputTrack> > mOutputTracks; 1180public: 1181 virtual bool hasFastMixer() const { return false; } 1182}; 1183 1184 1185// record thread 1186class RecordThread : public ThreadBase 1187{ 1188public: 1189 1190 class RecordTrack; 1191 1192 /* The ResamplerBufferProvider is used to retrieve recorded input data from the 1193 * RecordThread. It maintains local state on the relative position of the read 1194 * position of the RecordTrack compared with the RecordThread. 1195 */ 1196 class ResamplerBufferProvider : public AudioBufferProvider 1197 { 1198 public: 1199 explicit ResamplerBufferProvider(RecordTrack* recordTrack) : 1200 mRecordTrack(recordTrack), 1201 mRsmpInUnrel(0), mRsmpInFront(0) { } 1202 virtual ~ResamplerBufferProvider() { } 1203 1204 // called to set the ResamplerBufferProvider to head of the RecordThread data buffer, 1205 // skipping any previous data read from the hal. 1206 virtual void reset(); 1207 1208 /* Synchronizes RecordTrack position with the RecordThread. 1209 * Calculates available frames and handle overruns if the RecordThread 1210 * has advanced faster than the ResamplerBufferProvider has retrieved data. 1211 * TODO: why not do this for every getNextBuffer? 1212 * 1213 * Parameters 1214 * framesAvailable: pointer to optional output size_t to store record track 1215 * frames available. 1216 * hasOverrun: pointer to optional boolean, returns true if track has overrun. 1217 */ 1218 1219 virtual void sync(size_t *framesAvailable = NULL, bool *hasOverrun = NULL); 1220 1221 // AudioBufferProvider interface 1222 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer); 1223 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 1224 private: 1225 RecordTrack * const mRecordTrack; 1226 size_t mRsmpInUnrel; // unreleased frames remaining from 1227 // most recent getNextBuffer 1228 // for debug only 1229 int32_t mRsmpInFront; // next available frame 1230 // rolling counter that is never cleared 1231 }; 1232 1233 /* The RecordBufferConverter is used for format, channel, and sample rate 1234 * conversion for a RecordTrack. 1235 * 1236 * TODO: Self contained, so move to a separate file later. 1237 * 1238 * RecordBufferConverter uses the convert() method rather than exposing a 1239 * buffer provider interface; this is to save a memory copy. 1240 */ 1241 class RecordBufferConverter 1242 { 1243 public: 1244 RecordBufferConverter( 1245 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 1246 uint32_t srcSampleRate, 1247 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 1248 uint32_t dstSampleRate); 1249 1250 ~RecordBufferConverter(); 1251 1252 /* Converts input data from an AudioBufferProvider by format, channelMask, 1253 * and sampleRate to a destination buffer. 1254 * 1255 * Parameters 1256 * dst: buffer to place the converted data. 1257 * provider: buffer provider to obtain source data. 1258 * frames: number of frames to convert 1259 * 1260 * Returns the number of frames converted. 1261 */ 1262 size_t convert(void *dst, AudioBufferProvider *provider, size_t frames); 1263 1264 // returns NO_ERROR if constructor was successful 1265 status_t initCheck() const { 1266 // mSrcChannelMask set on successful updateParameters 1267 return mSrcChannelMask != AUDIO_CHANNEL_INVALID ? NO_ERROR : NO_INIT; 1268 } 1269 1270 // allows dynamic reconfigure of all parameters 1271 status_t updateParameters( 1272 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 1273 uint32_t srcSampleRate, 1274 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 1275 uint32_t dstSampleRate); 1276 1277 // called to reset resampler buffers on record track discontinuity 1278 void reset() { 1279 if (mResampler != NULL) { 1280 mResampler->reset(); 1281 } 1282 } 1283 1284 private: 1285 // format conversion when not using resampler 1286 void convertNoResampler(void *dst, const void *src, size_t frames); 1287 1288 // format conversion when using resampler; modifies src in-place 1289 void convertResampler(void *dst, /*not-a-const*/ void *src, size_t frames); 1290 1291 // user provided information 1292 audio_channel_mask_t mSrcChannelMask; 1293 audio_format_t mSrcFormat; 1294 uint32_t mSrcSampleRate; 1295 audio_channel_mask_t mDstChannelMask; 1296 audio_format_t mDstFormat; 1297 uint32_t mDstSampleRate; 1298 1299 // derived information 1300 uint32_t mSrcChannelCount; 1301 uint32_t mDstChannelCount; 1302 size_t mDstFrameSize; 1303 1304 // format conversion buffer 1305 void *mBuf; 1306 size_t mBufFrames; 1307 size_t mBufFrameSize; 1308 1309 // resampler info 1310 AudioResampler *mResampler; 1311 1312 bool mIsLegacyDownmix; // legacy stereo to mono conversion needed 1313 bool mIsLegacyUpmix; // legacy mono to stereo conversion needed 1314 bool mRequiresFloat; // data processing requires float (e.g. resampler) 1315 PassthruBufferProvider *mInputConverterProvider; // converts input to float 1316 int8_t mIdxAry[sizeof(uint32_t) * 8]; // used for channel mask conversion 1317 }; 1318 1319#include "RecordTracks.h" 1320 1321 RecordThread(const sp<AudioFlinger>& audioFlinger, 1322 AudioStreamIn *input, 1323 audio_io_handle_t id, 1324 audio_devices_t outDevice, 1325 audio_devices_t inDevice, 1326 bool systemReady 1327#ifdef TEE_SINK 1328 , const sp<NBAIO_Sink>& teeSink 1329#endif 1330 ); 1331 virtual ~RecordThread(); 1332 1333 // no addTrack_l ? 1334 void destroyTrack_l(const sp<RecordTrack>& track); 1335 void removeTrack_l(const sp<RecordTrack>& track); 1336 1337 void dumpInternals(int fd, const Vector<String16>& args); 1338 void dumpTracks(int fd, const Vector<String16>& args); 1339 1340 // Thread virtuals 1341 virtual bool threadLoop(); 1342 1343 // RefBase 1344 virtual void onFirstRef(); 1345 1346 virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; } 1347 1348 virtual sp<MemoryDealer> readOnlyHeap() const { return mReadOnlyHeap; } 1349 1350 virtual sp<IMemory> pipeMemory() const { return mPipeMemory; } 1351 1352 sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l( 1353 const sp<AudioFlinger::Client>& client, 1354 uint32_t sampleRate, 1355 audio_format_t format, 1356 audio_channel_mask_t channelMask, 1357 size_t *pFrameCount, 1358 audio_session_t sessionId, 1359 size_t *notificationFrames, 1360 uid_t uid, 1361 audio_input_flags_t *flags, 1362 pid_t tid, 1363 status_t *status /*non-NULL*/); 1364 1365 status_t start(RecordTrack* recordTrack, 1366 AudioSystem::sync_event_t event, 1367 audio_session_t triggerSession); 1368 1369 // ask the thread to stop the specified track, and 1370 // return true if the caller should then do it's part of the stopping process 1371 bool stop(RecordTrack* recordTrack); 1372 1373 void dump(int fd, const Vector<String16>& args); 1374 AudioStreamIn* clearInput(); 1375 virtual sp<StreamHalInterface> stream() const; 1376 1377 1378 virtual bool checkForNewParameter_l(const String8& keyValuePair, 1379 status_t& status); 1380 virtual void cacheParameters_l() {} 1381 virtual String8 getParameters(const String8& keys); 1382 virtual void ioConfigChanged(audio_io_config_event event, pid_t pid = 0); 1383 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 1384 audio_patch_handle_t *handle); 1385 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 1386 1387 void addPatchRecord(const sp<PatchRecord>& record); 1388 void deletePatchRecord(const sp<PatchRecord>& record); 1389 1390 void readInputParameters_l(); 1391 virtual uint32_t getInputFramesLost(); 1392 1393 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1394 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1395 virtual uint32_t hasAudioSession_l(audio_session_t sessionId) const; 1396 1397 // Return the set of unique session IDs across all tracks. 1398 // The keys are the session IDs, and the associated values are meaningless. 1399 // FIXME replace by Set [and implement Bag/Multiset for other uses]. 1400 KeyedVector<audio_session_t, bool> sessionIds() const; 1401 1402 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 1403 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 1404 1405 static void syncStartEventCallback(const wp<SyncEvent>& event); 1406 1407 virtual size_t frameCount() const { return mFrameCount; } 1408 bool hasFastCapture() const { return mFastCapture != 0; } 1409 virtual void getAudioPortConfig(struct audio_port_config *config); 1410 1411 virtual status_t checkEffectCompatibility_l(const effect_descriptor_t *desc, 1412 audio_session_t sessionId); 1413 1414private: 1415 // Enter standby if not already in standby, and set mStandby flag 1416 void standbyIfNotAlreadyInStandby(); 1417 1418 // Call the HAL standby method unconditionally, and don't change mStandby flag 1419 void inputStandBy(); 1420 1421 AudioStreamIn *mInput; 1422 SortedVector < sp<RecordTrack> > mTracks; 1423 // mActiveTracks has dual roles: it indicates the current active track(s), and 1424 // is used together with mStartStopCond to indicate start()/stop() progress 1425 SortedVector< sp<RecordTrack> > mActiveTracks; 1426 // generation counter for mActiveTracks 1427 int mActiveTracksGen; 1428 Condition mStartStopCond; 1429 1430 // resampler converts input at HAL Hz to output at AudioRecord client Hz 1431 void *mRsmpInBuffer; // size = mRsmpInFramesOA 1432 size_t mRsmpInFrames; // size of resampler input in frames 1433 size_t mRsmpInFramesP2;// size rounded up to a power-of-2 1434 size_t mRsmpInFramesOA;// mRsmpInFramesP2 + over-allocation 1435 1436 // rolling index that is never cleared 1437 int32_t mRsmpInRear; // last filled frame + 1 1438 1439 // For dumpsys 1440 const sp<NBAIO_Sink> mTeeSink; 1441 1442 const sp<MemoryDealer> mReadOnlyHeap; 1443 1444 // one-time initialization, no locks required 1445 sp<FastCapture> mFastCapture; // non-0 if there is also 1446 // a fast capture 1447 1448 // FIXME audio watchdog thread 1449 1450 // contents are not guaranteed to be consistent, no locks required 1451 FastCaptureDumpState mFastCaptureDumpState; 1452#ifdef STATE_QUEUE_DUMP 1453 // FIXME StateQueue observer and mutator dump fields 1454#endif 1455 // FIXME audio watchdog dump 1456 1457 // accessible only within the threadLoop(), no locks required 1458 // mFastCapture->sq() // for mutating and pushing state 1459 int32_t mFastCaptureFutex; // for cold idle 1460 1461 // The HAL input source is treated as non-blocking, 1462 // but current implementation is blocking 1463 sp<NBAIO_Source> mInputSource; 1464 // The source for the normal capture thread to read from: mInputSource or mPipeSource 1465 sp<NBAIO_Source> mNormalSource; 1466 // If a fast capture is present, the non-blocking pipe sink written to by fast capture, 1467 // otherwise clear 1468 sp<NBAIO_Sink> mPipeSink; 1469 // If a fast capture is present, the non-blocking pipe source read by normal thread, 1470 // otherwise clear 1471 sp<NBAIO_Source> mPipeSource; 1472 // Depth of pipe from fast capture to normal thread and fast clients, always power of 2 1473 size_t mPipeFramesP2; 1474 // If a fast capture is present, the Pipe as IMemory, otherwise clear 1475 sp<IMemory> mPipeMemory; 1476 1477 static const size_t kFastCaptureLogSize = 4 * 1024; 1478 sp<NBLog::Writer> mFastCaptureNBLogWriter; 1479 1480 bool mFastTrackAvail; // true if fast track available 1481}; 1482