Threads.h revision 20b9ef0b55c9150ae11057ab997ae61be2d496ef
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 audio_port_handle_t portId); 596 597 AudioStreamOut* getOutput() const; 598 AudioStreamOut* clearOutput(); 599 virtual sp<StreamHalInterface> stream() const; 600 601 // a very large number of suspend() will eventually wraparound, but unlikely 602 void suspend() { (void) android_atomic_inc(&mSuspended); } 603 void restore() 604 { 605 // if restore() is done without suspend(), get back into 606 // range so that the next suspend() will operate correctly 607 if (android_atomic_dec(&mSuspended) <= 0) { 608 android_atomic_release_store(0, &mSuspended); 609 } 610 } 611 bool isSuspended() const 612 { return android_atomic_acquire_load(&mSuspended) > 0; } 613 614 virtual String8 getParameters(const String8& keys); 615 virtual void ioConfigChanged(audio_io_config_event event, pid_t pid = 0); 616 status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames); 617 // FIXME rename mixBuffer() to sinkBuffer() and remove int16_t* dependency. 618 // Consider also removing and passing an explicit mMainBuffer initialization 619 // parameter to AF::PlaybackThread::Track::Track(). 620 int16_t *mixBuffer() const { 621 return reinterpret_cast<int16_t *>(mSinkBuffer); }; 622 623 virtual void detachAuxEffect_l(int effectId); 624 status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track>& track, 625 int EffectId); 626 status_t attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track>& track, 627 int EffectId); 628 629 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 630 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 631 virtual uint32_t hasAudioSession_l(audio_session_t sessionId) const; 632 virtual uint32_t getStrategyForSession_l(audio_session_t sessionId); 633 634 635 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 636 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 637 638 // called with AudioFlinger lock held 639 bool invalidateTracks_l(audio_stream_type_t streamType); 640 virtual void invalidateTracks(audio_stream_type_t streamType); 641 642 virtual size_t frameCount() const { return mNormalFrameCount; } 643 644 status_t getTimestamp_l(AudioTimestamp& timestamp); 645 646 void addPatchTrack(const sp<PatchTrack>& track); 647 void deletePatchTrack(const sp<PatchTrack>& track); 648 649 virtual void getAudioPortConfig(struct audio_port_config *config); 650 651protected: 652 // updated by readOutputParameters_l() 653 size_t mNormalFrameCount; // normal mixer and effects 654 655 bool mThreadThrottle; // throttle the thread processing 656 uint32_t mThreadThrottleTimeMs; // throttle time for MIXER threads 657 uint32_t mThreadThrottleEndMs; // notify once per throttling 658 uint32_t mHalfBufferMs; // half the buffer size in milliseconds 659 660 void* mSinkBuffer; // frame size aligned sink buffer 661 662 // TODO: 663 // Rearrange the buffer info into a struct/class with 664 // clear, copy, construction, destruction methods. 665 // 666 // mSinkBuffer also has associated with it: 667 // 668 // mSinkBufferSize: Sink Buffer Size 669 // mFormat: Sink Buffer Format 670 671 // Mixer Buffer (mMixerBuffer*) 672 // 673 // In the case of floating point or multichannel data, which is not in the 674 // sink format, it is required to accumulate in a higher precision or greater channel count 675 // buffer before downmixing or data conversion to the sink buffer. 676 677 // Set to "true" to enable the Mixer Buffer otherwise mixer output goes to sink buffer. 678 bool mMixerBufferEnabled; 679 680 // Storage, 32 byte aligned (may make this alignment a requirement later). 681 // Due to constraints on mNormalFrameCount, the buffer size is a multiple of 16 frames. 682 void* mMixerBuffer; 683 684 // Size of mMixerBuffer in bytes: mNormalFrameCount * #channels * sampsize. 685 size_t mMixerBufferSize; 686 687 // The audio format of mMixerBuffer. Set to AUDIO_FORMAT_PCM_(FLOAT|16_BIT) only. 688 audio_format_t mMixerBufferFormat; 689 690 // An internal flag set to true by MixerThread::prepareTracks_l() 691 // when mMixerBuffer contains valid data after mixing. 692 bool mMixerBufferValid; 693 694 // Effects Buffer (mEffectsBuffer*) 695 // 696 // In the case of effects data, which is not in the sink format, 697 // it is required to accumulate in a different buffer before data conversion 698 // to the sink buffer. 699 700 // Set to "true" to enable the Effects Buffer otherwise effects output goes to sink buffer. 701 bool mEffectBufferEnabled; 702 703 // Storage, 32 byte aligned (may make this alignment a requirement later). 704 // Due to constraints on mNormalFrameCount, the buffer size is a multiple of 16 frames. 705 void* mEffectBuffer; 706 707 // Size of mEffectsBuffer in bytes: mNormalFrameCount * #channels * sampsize. 708 size_t mEffectBufferSize; 709 710 // The audio format of mEffectsBuffer. Set to AUDIO_FORMAT_PCM_16_BIT only. 711 audio_format_t mEffectBufferFormat; 712 713 // An internal flag set to true by MixerThread::prepareTracks_l() 714 // when mEffectsBuffer contains valid data after mixing. 715 // 716 // When this is set, all mixer data is routed into the effects buffer 717 // for any processing (including output processing). 718 bool mEffectBufferValid; 719 720 // suspend count, > 0 means suspended. While suspended, the thread continues to pull from 721 // tracks and mix, but doesn't write to HAL. A2DP and SCO HAL implementations can't handle 722 // concurrent use of both of them, so Audio Policy Service suspends one of the threads to 723 // workaround that restriction. 724 // 'volatile' means accessed via atomic operations and no lock. 725 volatile int32_t mSuspended; 726 727 int64_t mBytesWritten; 728 int64_t mFramesWritten; // not reset on standby 729 int64_t mSuspendedFrames; // not reset on standby 730private: 731 // mMasterMute is in both PlaybackThread and in AudioFlinger. When a 732 // PlaybackThread needs to find out if master-muted, it checks it's local 733 // copy rather than the one in AudioFlinger. This optimization saves a lock. 734 bool mMasterMute; 735 void setMasterMute_l(bool muted) { mMasterMute = muted; } 736protected: 737 SortedVector< wp<Track> > mActiveTracks; // FIXME check if this could be sp<> 738 SortedVector<int> mWakeLockUids; 739 int mActiveTracksGeneration; 740 wp<Track> mLatestActiveTrack; // latest track added to mActiveTracks 741 742 // Allocate a track name for a given channel mask. 743 // Returns name >= 0 if successful, -1 on failure. 744 virtual int getTrackName_l(audio_channel_mask_t channelMask, audio_format_t format, 745 audio_session_t sessionId, uid_t uid) = 0; 746 virtual void deleteTrackName_l(int name) = 0; 747 748 // Time to sleep between cycles when: 749 virtual uint32_t activeSleepTimeUs() const; // mixer state MIXER_TRACKS_ENABLED 750 virtual uint32_t idleSleepTimeUs() const = 0; // mixer state MIXER_IDLE 751 virtual uint32_t suspendSleepTimeUs() const = 0; // audio policy manager suspended us 752 // No sleep when mixer state == MIXER_TRACKS_READY; relies on audio HAL stream->write() 753 // No sleep in standby mode; waits on a condition 754 755 // Code snippets that are temporarily lifted up out of threadLoop() until the merge 756 void checkSilentMode_l(); 757 758 // Non-trivial for DUPLICATING only 759 virtual void saveOutputTracks() { } 760 virtual void clearOutputTracks() { } 761 762 // Cache various calculated values, at threadLoop() entry and after a parameter change 763 virtual void cacheParameters_l(); 764 765 virtual uint32_t correctLatency_l(uint32_t latency) const; 766 767 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 768 audio_patch_handle_t *handle); 769 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 770 771 bool usesHwAvSync() const { return (mType == DIRECT) && (mOutput != NULL) 772 && mHwSupportsPause 773 && (mOutput->flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC); } 774 775 uint32_t trackCountForUid_l(uid_t uid); 776 777private: 778 779 friend class AudioFlinger; // for numerous 780 781 PlaybackThread& operator = (const PlaybackThread&); 782 783 status_t addTrack_l(const sp<Track>& track); 784 bool destroyTrack_l(const sp<Track>& track); 785 void removeTrack_l(const sp<Track>& track); 786 void broadcast_l(); 787 788 void readOutputParameters_l(); 789 790 virtual void dumpInternals(int fd, const Vector<String16>& args); 791 void dumpTracks(int fd, const Vector<String16>& args); 792 793 SortedVector< sp<Track> > mTracks; 794 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 795 AudioStreamOut *mOutput; 796 797 float mMasterVolume; 798 nsecs_t mLastWriteTime; 799 int mNumWrites; 800 int mNumDelayedWrites; 801 bool mInWrite; 802 803 // FIXME rename these former local variables of threadLoop to standard "m" names 804 nsecs_t mStandbyTimeNs; 805 size_t mSinkBufferSize; 806 807 // cached copies of activeSleepTimeUs() and idleSleepTimeUs() made by cacheParameters_l() 808 uint32_t mActiveSleepTimeUs; 809 uint32_t mIdleSleepTimeUs; 810 811 uint32_t mSleepTimeUs; 812 813 // mixer status returned by prepareTracks_l() 814 mixer_state mMixerStatus; // current cycle 815 // previous cycle when in prepareTracks_l() 816 mixer_state mMixerStatusIgnoringFastTracks; 817 // FIXME or a separate ready state per track 818 819 // FIXME move these declarations into the specific sub-class that needs them 820 // MIXER only 821 uint32_t sleepTimeShift; 822 823 // same as AudioFlinger::mStandbyTimeInNsecs except for DIRECT which uses a shorter value 824 nsecs_t mStandbyDelayNs; 825 826 // MIXER only 827 nsecs_t maxPeriod; 828 829 // DUPLICATING only 830 uint32_t writeFrames; 831 832 size_t mBytesRemaining; 833 size_t mCurrentWriteLength; 834 bool mUseAsyncWrite; 835 // mWriteAckSequence contains current write sequence on bits 31-1. The write sequence is 836 // incremented each time a write(), a flush() or a standby() occurs. 837 // Bit 0 is set when a write blocks and indicates a callback is expected. 838 // Bit 0 is reset by the async callback thread calling resetWriteBlocked(). Out of sequence 839 // callbacks are ignored. 840 uint32_t mWriteAckSequence; 841 // mDrainSequence contains current drain sequence on bits 31-1. The drain sequence is 842 // incremented each time a drain is requested or a flush() or standby() occurs. 843 // Bit 0 is set when the drain() command is called at the HAL and indicates a callback is 844 // expected. 845 // Bit 0 is reset by the async callback thread calling resetDraining(). Out of sequence 846 // callbacks are ignored. 847 uint32_t mDrainSequence; 848 // A condition that must be evaluated by prepareTrack_l() has changed and we must not wait 849 // for async write callback in the thread loop before evaluating it 850 bool mSignalPending; 851 sp<AsyncCallbackThread> mCallbackThread; 852 853private: 854 // The HAL output sink is treated as non-blocking, but current implementation is blocking 855 sp<NBAIO_Sink> mOutputSink; 856 // If a fast mixer is present, the blocking pipe sink, otherwise clear 857 sp<NBAIO_Sink> mPipeSink; 858 // The current sink for the normal mixer to write it's (sub)mix, mOutputSink or mPipeSink 859 sp<NBAIO_Sink> mNormalSink; 860#ifdef TEE_SINK 861 // For dumpsys 862 sp<NBAIO_Sink> mTeeSink; 863 sp<NBAIO_Source> mTeeSource; 864#endif 865 uint32_t mScreenState; // cached copy of gScreenState 866 static const size_t kFastMixerLogSize = 4 * 1024; 867 sp<NBLog::Writer> mFastMixerNBLogWriter; 868 869 // Do not call from a sched_fifo thread as it uses a system time call 870 // and obtains a local mutex. 871 class LocalLog { 872 public: 873 void log(const char *fmt, ...) { 874 va_list val; 875 va_start(val, fmt); 876 877 // format to buffer 878 char buffer[512]; 879 int length = vsnprintf(buffer, sizeof(buffer), fmt, val); 880 if (length >= (signed)sizeof(buffer)) { 881 length = sizeof(buffer) - 1; 882 } 883 884 // strip out trailing newline 885 while (length > 0 && buffer[length - 1] == '\n') { 886 buffer[--length] = 0; 887 } 888 889 // store in circular array 890 AutoMutex _l(mLock); 891 mLog.emplace_back( 892 std::make_pair(systemTime(SYSTEM_TIME_REALTIME), std::string(buffer))); 893 if (mLog.size() > kLogSize) { 894 mLog.pop_front(); 895 } 896 897 va_end(val); 898 } 899 900 void dump(int fd, const Vector<String16>& args, const char *prefix = "") { 901 if (!AudioFlinger::dumpTryLock(mLock)) return; // a local lock, shouldn't happen 902 if (mLog.size() > 0) { 903 bool dumpAll = false; 904 for (const auto &arg : args) { 905 if (arg == String16("--locallog")) { 906 dumpAll = true; 907 } 908 } 909 910 dprintf(fd, "Local Log:\n"); 911 auto it = mLog.begin(); 912 if (!dumpAll && mLog.size() > kLogPrint) { 913 it += (mLog.size() - kLogPrint); 914 } 915 for (; it != mLog.end(); ++it) { 916 const int64_t ns = it->first; 917 const int ns_per_sec = 1000000000; 918 const time_t sec = ns / ns_per_sec; 919 struct tm tm; 920 localtime_r(&sec, &tm); 921 922 dprintf(fd, "%s%02d-%02d %02d:%02d:%02d.%03d %s\n", 923 prefix, 924 tm.tm_mon + 1, // localtime_r uses months in 0 - 11 range 925 tm.tm_mday, tm.tm_hour, tm.tm_min, tm.tm_sec, 926 (int)(ns % ns_per_sec / 1000000), 927 it->second.c_str()); 928 } 929 } 930 mLock.unlock(); 931 } 932 933 private: 934 Mutex mLock; 935 static const size_t kLogSize = 256; // full history 936 static const size_t kLogPrint = 32; // default print history 937 std::deque<std::pair<int64_t, std::string>> mLog; 938 } mLocalLog; 939 940public: 941 virtual bool hasFastMixer() const = 0; 942 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex __unused) const 943 { FastTrackUnderruns dummy; return dummy; } 944 945protected: 946 // accessed by both binder threads and within threadLoop(), lock on mutex needed 947 unsigned mFastTrackAvailMask; // bit i set if fast track [i] is available 948 bool mHwSupportsPause; 949 bool mHwPaused; 950 bool mFlushPending; 951}; 952 953class MixerThread : public PlaybackThread { 954public: 955 MixerThread(const sp<AudioFlinger>& audioFlinger, 956 AudioStreamOut* output, 957 audio_io_handle_t id, 958 audio_devices_t device, 959 bool systemReady, 960 type_t type = MIXER); 961 virtual ~MixerThread(); 962 963 // Thread virtuals 964 965 virtual bool checkForNewParameter_l(const String8& keyValuePair, 966 status_t& status); 967 virtual void dumpInternals(int fd, const Vector<String16>& args); 968 969protected: 970 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 971 virtual int getTrackName_l(audio_channel_mask_t channelMask, audio_format_t format, 972 audio_session_t sessionId, uid_t uid); 973 virtual void deleteTrackName_l(int name); 974 virtual uint32_t idleSleepTimeUs() const; 975 virtual uint32_t suspendSleepTimeUs() const; 976 virtual void cacheParameters_l(); 977 978 virtual void acquireWakeLock_l(int uid = -1) { 979 PlaybackThread::acquireWakeLock_l(uid); 980 if (hasFastMixer()) { 981 mFastMixer->setBoottimeOffset( 982 mTimestamp.mTimebaseOffset[ExtendedTimestamp::TIMEBASE_BOOTTIME]); 983 } 984 } 985 986 // threadLoop snippets 987 virtual ssize_t threadLoop_write(); 988 virtual void threadLoop_standby(); 989 virtual void threadLoop_mix(); 990 virtual void threadLoop_sleepTime(); 991 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 992 virtual uint32_t correctLatency_l(uint32_t latency) const; 993 994 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 995 audio_patch_handle_t *handle); 996 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 997 998 AudioMixer* mAudioMixer; // normal mixer 999private: 1000 // one-time initialization, no locks required 1001 sp<FastMixer> mFastMixer; // non-0 if there is also a fast mixer 1002 sp<AudioWatchdog> mAudioWatchdog; // non-0 if there is an audio watchdog thread 1003 1004 // contents are not guaranteed to be consistent, no locks required 1005 FastMixerDumpState mFastMixerDumpState; 1006#ifdef STATE_QUEUE_DUMP 1007 StateQueueObserverDump mStateQueueObserverDump; 1008 StateQueueMutatorDump mStateQueueMutatorDump; 1009#endif 1010 AudioWatchdogDump mAudioWatchdogDump; 1011 1012 // accessible only within the threadLoop(), no locks required 1013 // mFastMixer->sq() // for mutating and pushing state 1014 int32_t mFastMixerFutex; // for cold idle 1015 1016 std::atomic_bool mMasterMono; 1017public: 1018 virtual bool hasFastMixer() const { return mFastMixer != 0; } 1019 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const { 1020 ALOG_ASSERT(fastIndex < FastMixerState::sMaxFastTracks); 1021 return mFastMixerDumpState.mTracks[fastIndex].mUnderruns; 1022 } 1023 1024protected: 1025 virtual void setMasterMono_l(bool mono) { 1026 mMasterMono.store(mono); 1027 if (mFastMixer != nullptr) { /* hasFastMixer() */ 1028 mFastMixer->setMasterMono(mMasterMono); 1029 } 1030 } 1031 // the FastMixer performs mono blend if it exists. 1032 // Blending with limiter is not idempotent, 1033 // and blending without limiter is idempotent but inefficient to do twice. 1034 virtual bool requireMonoBlend() { return mMasterMono.load() && !hasFastMixer(); } 1035}; 1036 1037class DirectOutputThread : public PlaybackThread { 1038public: 1039 1040 DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1041 audio_io_handle_t id, audio_devices_t device, bool systemReady); 1042 virtual ~DirectOutputThread(); 1043 1044 // Thread virtuals 1045 1046 virtual bool checkForNewParameter_l(const String8& keyValuePair, 1047 status_t& status); 1048 virtual void flushHw_l(); 1049 1050protected: 1051 virtual int getTrackName_l(audio_channel_mask_t channelMask, audio_format_t format, 1052 audio_session_t sessionId, uid_t uid); 1053 virtual void deleteTrackName_l(int name); 1054 virtual uint32_t activeSleepTimeUs() const; 1055 virtual uint32_t idleSleepTimeUs() const; 1056 virtual uint32_t suspendSleepTimeUs() const; 1057 virtual void cacheParameters_l(); 1058 1059 // threadLoop snippets 1060 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1061 virtual void threadLoop_mix(); 1062 virtual void threadLoop_sleepTime(); 1063 virtual void threadLoop_exit(); 1064 virtual bool shouldStandby_l(); 1065 1066 virtual void onAddNewTrack_l(); 1067 1068 // volumes last sent to audio HAL with stream->set_volume() 1069 float mLeftVolFloat; 1070 float mRightVolFloat; 1071 1072 DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1073 audio_io_handle_t id, uint32_t device, ThreadBase::type_t type, 1074 bool systemReady); 1075 void processVolume_l(Track *track, bool lastTrack); 1076 1077 // prepareTracks_l() tells threadLoop_mix() the name of the single active track 1078 sp<Track> mActiveTrack; 1079 1080 wp<Track> mPreviousTrack; // used to detect track switch 1081 1082public: 1083 virtual bool hasFastMixer() const { return false; } 1084}; 1085 1086class OffloadThread : public DirectOutputThread { 1087public: 1088 1089 OffloadThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1090 audio_io_handle_t id, uint32_t device, bool systemReady); 1091 virtual ~OffloadThread() {}; 1092 virtual void flushHw_l(); 1093 1094protected: 1095 // threadLoop snippets 1096 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1097 virtual void threadLoop_exit(); 1098 1099 virtual bool waitingAsyncCallback(); 1100 virtual bool waitingAsyncCallback_l(); 1101 virtual void invalidateTracks(audio_stream_type_t streamType); 1102 1103 virtual bool keepWakeLock() const { return (mKeepWakeLock || (mDrainSequence & 1)); } 1104 1105private: 1106 size_t mPausedWriteLength; // length in bytes of write interrupted by pause 1107 size_t mPausedBytesRemaining; // bytes still waiting in mixbuffer after resume 1108 bool mKeepWakeLock; // keep wake lock while waiting for write callback 1109 uint64_t mOffloadUnderrunPosition; // Current frame position for offloaded playback 1110 // used and valid only during underrun. ~0 if 1111 // no underrun has occurred during playback and 1112 // is not reset on standby. 1113}; 1114 1115class AsyncCallbackThread : public Thread { 1116public: 1117 1118 explicit AsyncCallbackThread(const wp<PlaybackThread>& playbackThread); 1119 1120 virtual ~AsyncCallbackThread(); 1121 1122 // Thread virtuals 1123 virtual bool threadLoop(); 1124 1125 // RefBase 1126 virtual void onFirstRef(); 1127 1128 void exit(); 1129 void setWriteBlocked(uint32_t sequence); 1130 void resetWriteBlocked(); 1131 void setDraining(uint32_t sequence); 1132 void resetDraining(); 1133 void setAsyncError(); 1134 1135private: 1136 const wp<PlaybackThread> mPlaybackThread; 1137 // mWriteAckSequence corresponds to the last write sequence passed by the offload thread via 1138 // setWriteBlocked(). The sequence is shifted one bit to the left and the lsb is used 1139 // to indicate that the callback has been received via resetWriteBlocked() 1140 uint32_t mWriteAckSequence; 1141 // mDrainSequence corresponds to the last drain sequence passed by the offload thread via 1142 // setDraining(). The sequence is shifted one bit to the left and the lsb is used 1143 // to indicate that the callback has been received via resetDraining() 1144 uint32_t mDrainSequence; 1145 Condition mWaitWorkCV; 1146 Mutex mLock; 1147 bool mAsyncError; 1148}; 1149 1150class DuplicatingThread : public MixerThread { 1151public: 1152 DuplicatingThread(const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread, 1153 audio_io_handle_t id, bool systemReady); 1154 virtual ~DuplicatingThread(); 1155 1156 // Thread virtuals 1157 void addOutputTrack(MixerThread* thread); 1158 void removeOutputTrack(MixerThread* thread); 1159 uint32_t waitTimeMs() const { return mWaitTimeMs; } 1160protected: 1161 virtual uint32_t activeSleepTimeUs() const; 1162 1163private: 1164 bool outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks); 1165protected: 1166 // threadLoop snippets 1167 virtual void threadLoop_mix(); 1168 virtual void threadLoop_sleepTime(); 1169 virtual ssize_t threadLoop_write(); 1170 virtual void threadLoop_standby(); 1171 virtual void cacheParameters_l(); 1172 1173private: 1174 // called from threadLoop, addOutputTrack, removeOutputTrack 1175 virtual void updateWaitTime_l(); 1176protected: 1177 virtual void saveOutputTracks(); 1178 virtual void clearOutputTracks(); 1179private: 1180 1181 uint32_t mWaitTimeMs; 1182 SortedVector < sp<OutputTrack> > outputTracks; 1183 SortedVector < sp<OutputTrack> > mOutputTracks; 1184public: 1185 virtual bool hasFastMixer() const { return false; } 1186}; 1187 1188 1189// record thread 1190class RecordThread : public ThreadBase 1191{ 1192public: 1193 1194 class RecordTrack; 1195 1196 /* The ResamplerBufferProvider is used to retrieve recorded input data from the 1197 * RecordThread. It maintains local state on the relative position of the read 1198 * position of the RecordTrack compared with the RecordThread. 1199 */ 1200 class ResamplerBufferProvider : public AudioBufferProvider 1201 { 1202 public: 1203 explicit ResamplerBufferProvider(RecordTrack* recordTrack) : 1204 mRecordTrack(recordTrack), 1205 mRsmpInUnrel(0), mRsmpInFront(0) { } 1206 virtual ~ResamplerBufferProvider() { } 1207 1208 // called to set the ResamplerBufferProvider to head of the RecordThread data buffer, 1209 // skipping any previous data read from the hal. 1210 virtual void reset(); 1211 1212 /* Synchronizes RecordTrack position with the RecordThread. 1213 * Calculates available frames and handle overruns if the RecordThread 1214 * has advanced faster than the ResamplerBufferProvider has retrieved data. 1215 * TODO: why not do this for every getNextBuffer? 1216 * 1217 * Parameters 1218 * framesAvailable: pointer to optional output size_t to store record track 1219 * frames available. 1220 * hasOverrun: pointer to optional boolean, returns true if track has overrun. 1221 */ 1222 1223 virtual void sync(size_t *framesAvailable = NULL, bool *hasOverrun = NULL); 1224 1225 // AudioBufferProvider interface 1226 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer); 1227 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 1228 private: 1229 RecordTrack * const mRecordTrack; 1230 size_t mRsmpInUnrel; // unreleased frames remaining from 1231 // most recent getNextBuffer 1232 // for debug only 1233 int32_t mRsmpInFront; // next available frame 1234 // rolling counter that is never cleared 1235 }; 1236 1237 /* The RecordBufferConverter is used for format, channel, and sample rate 1238 * conversion for a RecordTrack. 1239 * 1240 * TODO: Self contained, so move to a separate file later. 1241 * 1242 * RecordBufferConverter uses the convert() method rather than exposing a 1243 * buffer provider interface; this is to save a memory copy. 1244 */ 1245 class RecordBufferConverter 1246 { 1247 public: 1248 RecordBufferConverter( 1249 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 1250 uint32_t srcSampleRate, 1251 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 1252 uint32_t dstSampleRate); 1253 1254 ~RecordBufferConverter(); 1255 1256 /* Converts input data from an AudioBufferProvider by format, channelMask, 1257 * and sampleRate to a destination buffer. 1258 * 1259 * Parameters 1260 * dst: buffer to place the converted data. 1261 * provider: buffer provider to obtain source data. 1262 * frames: number of frames to convert 1263 * 1264 * Returns the number of frames converted. 1265 */ 1266 size_t convert(void *dst, AudioBufferProvider *provider, size_t frames); 1267 1268 // returns NO_ERROR if constructor was successful 1269 status_t initCheck() const { 1270 // mSrcChannelMask set on successful updateParameters 1271 return mSrcChannelMask != AUDIO_CHANNEL_INVALID ? NO_ERROR : NO_INIT; 1272 } 1273 1274 // allows dynamic reconfigure of all parameters 1275 status_t updateParameters( 1276 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 1277 uint32_t srcSampleRate, 1278 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 1279 uint32_t dstSampleRate); 1280 1281 // called to reset resampler buffers on record track discontinuity 1282 void reset() { 1283 if (mResampler != NULL) { 1284 mResampler->reset(); 1285 } 1286 } 1287 1288 private: 1289 // format conversion when not using resampler 1290 void convertNoResampler(void *dst, const void *src, size_t frames); 1291 1292 // format conversion when using resampler; modifies src in-place 1293 void convertResampler(void *dst, /*not-a-const*/ void *src, size_t frames); 1294 1295 // user provided information 1296 audio_channel_mask_t mSrcChannelMask; 1297 audio_format_t mSrcFormat; 1298 uint32_t mSrcSampleRate; 1299 audio_channel_mask_t mDstChannelMask; 1300 audio_format_t mDstFormat; 1301 uint32_t mDstSampleRate; 1302 1303 // derived information 1304 uint32_t mSrcChannelCount; 1305 uint32_t mDstChannelCount; 1306 size_t mDstFrameSize; 1307 1308 // format conversion buffer 1309 void *mBuf; 1310 size_t mBufFrames; 1311 size_t mBufFrameSize; 1312 1313 // resampler info 1314 AudioResampler *mResampler; 1315 1316 bool mIsLegacyDownmix; // legacy stereo to mono conversion needed 1317 bool mIsLegacyUpmix; // legacy mono to stereo conversion needed 1318 bool mRequiresFloat; // data processing requires float (e.g. resampler) 1319 PassthruBufferProvider *mInputConverterProvider; // converts input to float 1320 int8_t mIdxAry[sizeof(uint32_t) * 8]; // used for channel mask conversion 1321 }; 1322 1323#include "RecordTracks.h" 1324 1325 RecordThread(const sp<AudioFlinger>& audioFlinger, 1326 AudioStreamIn *input, 1327 audio_io_handle_t id, 1328 audio_devices_t outDevice, 1329 audio_devices_t inDevice, 1330 bool systemReady 1331#ifdef TEE_SINK 1332 , const sp<NBAIO_Sink>& teeSink 1333#endif 1334 ); 1335 virtual ~RecordThread(); 1336 1337 // no addTrack_l ? 1338 void destroyTrack_l(const sp<RecordTrack>& track); 1339 void removeTrack_l(const sp<RecordTrack>& track); 1340 1341 void dumpInternals(int fd, const Vector<String16>& args); 1342 void dumpTracks(int fd, const Vector<String16>& args); 1343 1344 // Thread virtuals 1345 virtual bool threadLoop(); 1346 1347 // RefBase 1348 virtual void onFirstRef(); 1349 1350 virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; } 1351 1352 virtual sp<MemoryDealer> readOnlyHeap() const { return mReadOnlyHeap; } 1353 1354 virtual sp<IMemory> pipeMemory() const { return mPipeMemory; } 1355 1356 sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l( 1357 const sp<AudioFlinger::Client>& client, 1358 uint32_t sampleRate, 1359 audio_format_t format, 1360 audio_channel_mask_t channelMask, 1361 size_t *pFrameCount, 1362 audio_session_t sessionId, 1363 size_t *notificationFrames, 1364 uid_t uid, 1365 audio_input_flags_t *flags, 1366 pid_t tid, 1367 status_t *status /*non-NULL*/, 1368 audio_port_handle_t portId); 1369 1370 status_t start(RecordTrack* recordTrack, 1371 AudioSystem::sync_event_t event, 1372 audio_session_t triggerSession); 1373 1374 // ask the thread to stop the specified track, and 1375 // return true if the caller should then do it's part of the stopping process 1376 bool stop(RecordTrack* recordTrack); 1377 1378 void dump(int fd, const Vector<String16>& args); 1379 AudioStreamIn* clearInput(); 1380 virtual sp<StreamHalInterface> stream() const; 1381 1382 1383 virtual bool checkForNewParameter_l(const String8& keyValuePair, 1384 status_t& status); 1385 virtual void cacheParameters_l() {} 1386 virtual String8 getParameters(const String8& keys); 1387 virtual void ioConfigChanged(audio_io_config_event event, pid_t pid = 0); 1388 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 1389 audio_patch_handle_t *handle); 1390 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 1391 1392 void addPatchRecord(const sp<PatchRecord>& record); 1393 void deletePatchRecord(const sp<PatchRecord>& record); 1394 1395 void readInputParameters_l(); 1396 virtual uint32_t getInputFramesLost(); 1397 1398 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1399 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1400 virtual uint32_t hasAudioSession_l(audio_session_t sessionId) const; 1401 1402 // Return the set of unique session IDs across all tracks. 1403 // The keys are the session IDs, and the associated values are meaningless. 1404 // FIXME replace by Set [and implement Bag/Multiset for other uses]. 1405 KeyedVector<audio_session_t, bool> sessionIds() const; 1406 1407 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 1408 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 1409 1410 static void syncStartEventCallback(const wp<SyncEvent>& event); 1411 1412 virtual size_t frameCount() const { return mFrameCount; } 1413 bool hasFastCapture() const { return mFastCapture != 0; } 1414 virtual void getAudioPortConfig(struct audio_port_config *config); 1415 1416 virtual status_t checkEffectCompatibility_l(const effect_descriptor_t *desc, 1417 audio_session_t sessionId); 1418 1419private: 1420 // Enter standby if not already in standby, and set mStandby flag 1421 void standbyIfNotAlreadyInStandby(); 1422 1423 // Call the HAL standby method unconditionally, and don't change mStandby flag 1424 void inputStandBy(); 1425 1426 AudioStreamIn *mInput; 1427 SortedVector < sp<RecordTrack> > mTracks; 1428 // mActiveTracks has dual roles: it indicates the current active track(s), and 1429 // is used together with mStartStopCond to indicate start()/stop() progress 1430 SortedVector< sp<RecordTrack> > mActiveTracks; 1431 // generation counter for mActiveTracks 1432 int mActiveTracksGen; 1433 Condition mStartStopCond; 1434 1435 // resampler converts input at HAL Hz to output at AudioRecord client Hz 1436 void *mRsmpInBuffer; // size = mRsmpInFramesOA 1437 size_t mRsmpInFrames; // size of resampler input in frames 1438 size_t mRsmpInFramesP2;// size rounded up to a power-of-2 1439 size_t mRsmpInFramesOA;// mRsmpInFramesP2 + over-allocation 1440 1441 // rolling index that is never cleared 1442 int32_t mRsmpInRear; // last filled frame + 1 1443 1444 // For dumpsys 1445 const sp<NBAIO_Sink> mTeeSink; 1446 1447 const sp<MemoryDealer> mReadOnlyHeap; 1448 1449 // one-time initialization, no locks required 1450 sp<FastCapture> mFastCapture; // non-0 if there is also 1451 // a fast capture 1452 1453 // FIXME audio watchdog thread 1454 1455 // contents are not guaranteed to be consistent, no locks required 1456 FastCaptureDumpState mFastCaptureDumpState; 1457#ifdef STATE_QUEUE_DUMP 1458 // FIXME StateQueue observer and mutator dump fields 1459#endif 1460 // FIXME audio watchdog dump 1461 1462 // accessible only within the threadLoop(), no locks required 1463 // mFastCapture->sq() // for mutating and pushing state 1464 int32_t mFastCaptureFutex; // for cold idle 1465 1466 // The HAL input source is treated as non-blocking, 1467 // but current implementation is blocking 1468 sp<NBAIO_Source> mInputSource; 1469 // The source for the normal capture thread to read from: mInputSource or mPipeSource 1470 sp<NBAIO_Source> mNormalSource; 1471 // If a fast capture is present, the non-blocking pipe sink written to by fast capture, 1472 // otherwise clear 1473 sp<NBAIO_Sink> mPipeSink; 1474 // If a fast capture is present, the non-blocking pipe source read by normal thread, 1475 // otherwise clear 1476 sp<NBAIO_Source> mPipeSource; 1477 // Depth of pipe from fast capture to normal thread and fast clients, always power of 2 1478 size_t mPipeFramesP2; 1479 // If a fast capture is present, the Pipe as IMemory, otherwise clear 1480 sp<IMemory> mPipeMemory; 1481 1482 static const size_t kFastCaptureLogSize = 4 * 1024; 1483 sp<NBLog::Writer> mFastCaptureNBLogWriter; 1484 1485 bool mFastTrackAvail; // true if fast track available 1486}; 1487