Threads.h revision 0d5a2ed0a05a2bf337c68edb54f24c60e18032c1
1/* 2** 3** Copyright 2012, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18#ifndef INCLUDING_FROM_AUDIOFLINGER_H 19 #error This header file should only be included from AudioFlinger.h 20#endif 21 22class ThreadBase : public Thread { 23public: 24 25#include "TrackBase.h" 26 27 enum type_t { 28 MIXER, // Thread class is MixerThread 29 DIRECT, // Thread class is DirectOutputThread 30 DUPLICATING, // Thread class is DuplicatingThread 31 RECORD, // Thread class is RecordThread 32 OFFLOAD // Thread class is OffloadThread 33 }; 34 35 static const char *threadTypeToString(type_t type); 36 37 ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 38 audio_devices_t outDevice, audio_devices_t inDevice, type_t type, 39 bool systemReady); 40 virtual ~ThreadBase(); 41 42 virtual status_t readyToRun(); 43 44 void dumpBase(int fd, const Vector<String16>& args); 45 void dumpEffectChains(int fd, const Vector<String16>& args); 46 47 void clearPowerManager(); 48 49 // base for record and playback 50 enum { 51 CFG_EVENT_IO, 52 CFG_EVENT_PRIO, 53 CFG_EVENT_SET_PARAMETER, 54 CFG_EVENT_CREATE_AUDIO_PATCH, 55 CFG_EVENT_RELEASE_AUDIO_PATCH, 56 }; 57 58 class ConfigEventData: public RefBase { 59 public: 60 virtual ~ConfigEventData() {} 61 62 virtual void dump(char *buffer, size_t size) = 0; 63 protected: 64 ConfigEventData() {} 65 }; 66 67 // Config event sequence by client if status needed (e.g binder thread calling setParameters()): 68 // 1. create SetParameterConfigEvent. This sets mWaitStatus in config event 69 // 2. Lock mLock 70 // 3. Call sendConfigEvent_l(): Append to mConfigEvents and mWaitWorkCV.signal 71 // 4. sendConfigEvent_l() reads status from event->mStatus; 72 // 5. sendConfigEvent_l() returns status 73 // 6. Unlock 74 // 75 // Parameter sequence by server: threadLoop calling processConfigEvents_l(): 76 // 1. Lock mLock 77 // 2. If there is an entry in mConfigEvents proceed ... 78 // 3. Read first entry in mConfigEvents 79 // 4. Remove first entry from mConfigEvents 80 // 5. Process 81 // 6. Set event->mStatus 82 // 7. event->mCond.signal 83 // 8. Unlock 84 85 class ConfigEvent: public RefBase { 86 public: 87 virtual ~ConfigEvent() {} 88 89 void dump(char *buffer, size_t size) { mData->dump(buffer, size); } 90 91 const int mType; // event type e.g. CFG_EVENT_IO 92 Mutex mLock; // mutex associated with mCond 93 Condition mCond; // condition for status return 94 status_t mStatus; // status communicated to sender 95 bool mWaitStatus; // true if sender is waiting for status 96 bool mRequiresSystemReady; // true if must wait for system ready to enter event queue 97 sp<ConfigEventData> mData; // event specific parameter data 98 99 protected: 100 explicit ConfigEvent(int type, bool requiresSystemReady = false) : 101 mType(type), mStatus(NO_ERROR), mWaitStatus(false), 102 mRequiresSystemReady(requiresSystemReady), mData(NULL) {} 103 }; 104 105 class IoConfigEventData : public ConfigEventData { 106 public: 107 IoConfigEventData(audio_io_config_event event, pid_t pid) : 108 mEvent(event), mPid(pid) {} 109 110 virtual void dump(char *buffer, size_t size) { 111 snprintf(buffer, size, "IO event: event %d\n", mEvent); 112 } 113 114 const audio_io_config_event mEvent; 115 const pid_t mPid; 116 }; 117 118 class IoConfigEvent : public ConfigEvent { 119 public: 120 IoConfigEvent(audio_io_config_event event, pid_t pid) : 121 ConfigEvent(CFG_EVENT_IO) { 122 mData = new IoConfigEventData(event, pid); 123 } 124 virtual ~IoConfigEvent() {} 125 }; 126 127 class PrioConfigEventData : public ConfigEventData { 128 public: 129 PrioConfigEventData(pid_t pid, pid_t tid, int32_t prio) : 130 mPid(pid), mTid(tid), mPrio(prio) {} 131 132 virtual void dump(char *buffer, size_t size) { 133 snprintf(buffer, size, "Prio event: pid %d, tid %d, prio %d\n", mPid, mTid, mPrio); 134 } 135 136 const pid_t mPid; 137 const pid_t mTid; 138 const int32_t mPrio; 139 }; 140 141 class PrioConfigEvent : public ConfigEvent { 142 public: 143 PrioConfigEvent(pid_t pid, pid_t tid, int32_t prio) : 144 ConfigEvent(CFG_EVENT_PRIO, true) { 145 mData = new PrioConfigEventData(pid, tid, prio); 146 } 147 virtual ~PrioConfigEvent() {} 148 }; 149 150 class SetParameterConfigEventData : public ConfigEventData { 151 public: 152 explicit SetParameterConfigEventData(String8 keyValuePairs) : 153 mKeyValuePairs(keyValuePairs) {} 154 155 virtual void dump(char *buffer, size_t size) { 156 snprintf(buffer, size, "KeyValue: %s\n", mKeyValuePairs.string()); 157 } 158 159 const String8 mKeyValuePairs; 160 }; 161 162 class SetParameterConfigEvent : public ConfigEvent { 163 public: 164 explicit SetParameterConfigEvent(String8 keyValuePairs) : 165 ConfigEvent(CFG_EVENT_SET_PARAMETER) { 166 mData = new SetParameterConfigEventData(keyValuePairs); 167 mWaitStatus = true; 168 } 169 virtual ~SetParameterConfigEvent() {} 170 }; 171 172 class CreateAudioPatchConfigEventData : public ConfigEventData { 173 public: 174 CreateAudioPatchConfigEventData(const struct audio_patch patch, 175 audio_patch_handle_t handle) : 176 mPatch(patch), mHandle(handle) {} 177 178 virtual void dump(char *buffer, size_t size) { 179 snprintf(buffer, size, "Patch handle: %u\n", mHandle); 180 } 181 182 const struct audio_patch mPatch; 183 audio_patch_handle_t mHandle; 184 }; 185 186 class CreateAudioPatchConfigEvent : public ConfigEvent { 187 public: 188 CreateAudioPatchConfigEvent(const struct audio_patch patch, 189 audio_patch_handle_t handle) : 190 ConfigEvent(CFG_EVENT_CREATE_AUDIO_PATCH) { 191 mData = new CreateAudioPatchConfigEventData(patch, handle); 192 mWaitStatus = true; 193 } 194 virtual ~CreateAudioPatchConfigEvent() {} 195 }; 196 197 class ReleaseAudioPatchConfigEventData : public ConfigEventData { 198 public: 199 explicit ReleaseAudioPatchConfigEventData(const audio_patch_handle_t handle) : 200 mHandle(handle) {} 201 202 virtual void dump(char *buffer, size_t size) { 203 snprintf(buffer, size, "Patch handle: %u\n", mHandle); 204 } 205 206 audio_patch_handle_t mHandle; 207 }; 208 209 class ReleaseAudioPatchConfigEvent : public ConfigEvent { 210 public: 211 explicit ReleaseAudioPatchConfigEvent(const audio_patch_handle_t handle) : 212 ConfigEvent(CFG_EVENT_RELEASE_AUDIO_PATCH) { 213 mData = new ReleaseAudioPatchConfigEventData(handle); 214 mWaitStatus = true; 215 } 216 virtual ~ReleaseAudioPatchConfigEvent() {} 217 }; 218 219 class PMDeathRecipient : public IBinder::DeathRecipient { 220 public: 221 explicit PMDeathRecipient(const wp<ThreadBase>& thread) : mThread(thread) {} 222 virtual ~PMDeathRecipient() {} 223 224 // IBinder::DeathRecipient 225 virtual void binderDied(const wp<IBinder>& who); 226 227 private: 228 PMDeathRecipient(const PMDeathRecipient&); 229 PMDeathRecipient& operator = (const PMDeathRecipient&); 230 231 wp<ThreadBase> mThread; 232 }; 233 234 virtual status_t initCheck() const = 0; 235 236 // static externally-visible 237 type_t type() const { return mType; } 238 bool isDuplicating() const { return (mType == DUPLICATING); } 239 240 audio_io_handle_t id() const { return mId;} 241 242 // dynamic externally-visible 243 uint32_t sampleRate() const { return mSampleRate; } 244 audio_channel_mask_t channelMask() const { return mChannelMask; } 245 audio_format_t format() const { return mHALFormat; } 246 uint32_t channelCount() const { return mChannelCount; } 247 // Called by AudioFlinger::frameCount(audio_io_handle_t output) and effects, 248 // and returns the [normal mix] buffer's frame count. 249 virtual size_t frameCount() const = 0; 250 251 // Return's the HAL's frame count i.e. fast mixer buffer size. 252 size_t frameCountHAL() const { return mFrameCount; } 253 254 size_t frameSize() const { return mFrameSize; } 255 256 // Should be "virtual status_t requestExitAndWait()" and override same 257 // method in Thread, but Thread::requestExitAndWait() is not yet virtual. 258 void exit(); 259 virtual bool checkForNewParameter_l(const String8& keyValuePair, 260 status_t& status) = 0; 261 virtual status_t setParameters(const String8& keyValuePairs); 262 virtual String8 getParameters(const String8& keys) = 0; 263 virtual void ioConfigChanged(audio_io_config_event event, pid_t pid = 0) = 0; 264 // sendConfigEvent_l() must be called with ThreadBase::mLock held 265 // Can temporarily release the lock if waiting for a reply from 266 // processConfigEvents_l(). 267 status_t sendConfigEvent_l(sp<ConfigEvent>& event); 268 void sendIoConfigEvent(audio_io_config_event event, pid_t pid = 0); 269 void sendIoConfigEvent_l(audio_io_config_event event, pid_t pid = 0); 270 void sendPrioConfigEvent(pid_t pid, pid_t tid, int32_t prio); 271 void sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio); 272 status_t sendSetParameterConfigEvent_l(const String8& keyValuePair); 273 status_t sendCreateAudioPatchConfigEvent(const struct audio_patch *patch, 274 audio_patch_handle_t *handle); 275 status_t sendReleaseAudioPatchConfigEvent(audio_patch_handle_t handle); 276 void processConfigEvents_l(); 277 virtual void cacheParameters_l() = 0; 278 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 279 audio_patch_handle_t *handle) = 0; 280 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle) = 0; 281 virtual void getAudioPortConfig(struct audio_port_config *config) = 0; 282 283 284 // see note at declaration of mStandby, mOutDevice and mInDevice 285 bool standby() const { return mStandby; } 286 audio_devices_t outDevice() const { return mOutDevice; } 287 audio_devices_t inDevice() const { return mInDevice; } 288 289 virtual sp<StreamHalInterface> stream() const = 0; 290 291 sp<EffectHandle> createEffect_l( 292 const sp<AudioFlinger::Client>& client, 293 const sp<IEffectClient>& effectClient, 294 int32_t priority, 295 audio_session_t sessionId, 296 effect_descriptor_t *desc, 297 int *enabled, 298 status_t *status /*non-NULL*/, 299 bool pinned); 300 301 // return values for hasAudioSession (bit field) 302 enum effect_state { 303 EFFECT_SESSION = 0x1, // the audio session corresponds to at least one 304 // effect 305 TRACK_SESSION = 0x2, // the audio session corresponds to at least one 306 // track 307 FAST_SESSION = 0x4 // the audio session corresponds to at least one 308 // fast track 309 }; 310 311 // get effect chain corresponding to session Id. 312 sp<EffectChain> getEffectChain(audio_session_t sessionId); 313 // same as getEffectChain() but must be called with ThreadBase mutex locked 314 sp<EffectChain> getEffectChain_l(audio_session_t sessionId) const; 315 // add an effect chain to the chain list (mEffectChains) 316 virtual status_t addEffectChain_l(const sp<EffectChain>& chain) = 0; 317 // remove an effect chain from the chain list (mEffectChains) 318 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain) = 0; 319 // lock all effect chains Mutexes. Must be called before releasing the 320 // ThreadBase mutex before processing the mixer and effects. This guarantees the 321 // integrity of the chains during the process. 322 // Also sets the parameter 'effectChains' to current value of mEffectChains. 323 void lockEffectChains_l(Vector< sp<EffectChain> >& effectChains); 324 // unlock effect chains after process 325 void unlockEffectChains(const Vector< sp<EffectChain> >& effectChains); 326 // get a copy of mEffectChains vector 327 Vector< sp<EffectChain> > getEffectChains_l() const { return mEffectChains; }; 328 // set audio mode to all effect chains 329 void setMode(audio_mode_t mode); 330 // get effect module with corresponding ID on specified audio session 331 sp<AudioFlinger::EffectModule> getEffect(audio_session_t sessionId, int effectId); 332 sp<AudioFlinger::EffectModule> getEffect_l(audio_session_t sessionId, int effectId); 333 // add and effect module. Also creates the effect chain is none exists for 334 // the effects audio session 335 status_t addEffect_l(const sp< EffectModule>& effect); 336 // remove and effect module. Also removes the effect chain is this was the last 337 // effect 338 void removeEffect_l(const sp< EffectModule>& effect, bool release = false); 339 // disconnect an effect handle from module and destroy module if last handle 340 void disconnectEffectHandle(EffectHandle *handle, bool unpinIfLast); 341 // detach all tracks connected to an auxiliary effect 342 virtual void detachAuxEffect_l(int effectId __unused) {} 343 // returns a combination of: 344 // - EFFECT_SESSION if effects on this audio session exist in one chain 345 // - TRACK_SESSION if tracks on this audio session exist 346 // - FAST_SESSION if fast tracks on this audio session exist 347 virtual uint32_t hasAudioSession_l(audio_session_t sessionId) const = 0; 348 uint32_t hasAudioSession(audio_session_t sessionId) const { 349 Mutex::Autolock _l(mLock); 350 return hasAudioSession_l(sessionId); 351 } 352 353 // the value returned by default implementation is not important as the 354 // strategy is only meaningful for PlaybackThread which implements this method 355 virtual uint32_t getStrategyForSession_l(audio_session_t sessionId __unused) 356 { return 0; } 357 358 // suspend or restore effect according to the type of effect passed. a NULL 359 // type pointer means suspend all effects in the session 360 void setEffectSuspended(const effect_uuid_t *type, 361 bool suspend, 362 audio_session_t sessionId = AUDIO_SESSION_OUTPUT_MIX); 363 // check if some effects must be suspended/restored when an effect is enabled 364 // or disabled 365 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 366 bool enabled, 367 audio_session_t sessionId = 368 AUDIO_SESSION_OUTPUT_MIX); 369 void checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 370 bool enabled, 371 audio_session_t sessionId = 372 AUDIO_SESSION_OUTPUT_MIX); 373 374 virtual status_t setSyncEvent(const sp<SyncEvent>& event) = 0; 375 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const = 0; 376 377 // Return a reference to a per-thread heap which can be used to allocate IMemory 378 // objects that will be read-only to client processes, read/write to mediaserver, 379 // and shared by all client processes of the thread. 380 // The heap is per-thread rather than common across all threads, because 381 // clients can't be trusted not to modify the offset of the IMemory they receive. 382 // If a thread does not have such a heap, this method returns 0. 383 virtual sp<MemoryDealer> readOnlyHeap() const { return 0; } 384 385 virtual sp<IMemory> pipeMemory() const { return 0; } 386 387 void systemReady(); 388 389 // checkEffectCompatibility_l() must be called with ThreadBase::mLock held 390 virtual status_t checkEffectCompatibility_l(const effect_descriptor_t *desc, 391 audio_session_t sessionId) = 0; 392 393 mutable Mutex mLock; 394 395protected: 396 397 // entry describing an effect being suspended in mSuspendedSessions keyed vector 398 class SuspendedSessionDesc : public RefBase { 399 public: 400 SuspendedSessionDesc() : mRefCount(0) {} 401 402 int mRefCount; // number of active suspend requests 403 effect_uuid_t mType; // effect type UUID 404 }; 405 406 void acquireWakeLock(int uid = -1); 407 virtual void acquireWakeLock_l(int uid = -1); 408 void releaseWakeLock(); 409 void releaseWakeLock_l(); 410 void updateWakeLockUids_l(const SortedVector<int> &uids); 411 void getPowerManager_l(); 412 void setEffectSuspended_l(const effect_uuid_t *type, 413 bool suspend, 414 audio_session_t sessionId); 415 // updated mSuspendedSessions when an effect suspended or restored 416 void updateSuspendedSessions_l(const effect_uuid_t *type, 417 bool suspend, 418 audio_session_t sessionId); 419 // check if some effects must be suspended when an effect chain is added 420 void checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain); 421 422 String16 getWakeLockTag(); 423 424 virtual void preExit() { } 425 virtual void setMasterMono_l(bool mono __unused) { } 426 virtual bool requireMonoBlend() { return false; } 427 428 friend class AudioFlinger; // for mEffectChains 429 430 const type_t mType; 431 432 // Used by parameters, config events, addTrack_l, exit 433 Condition mWaitWorkCV; 434 435 const sp<AudioFlinger> mAudioFlinger; 436 437 // updated by PlaybackThread::readOutputParameters_l() or 438 // RecordThread::readInputParameters_l() 439 uint32_t mSampleRate; 440 size_t mFrameCount; // output HAL, direct output, record 441 audio_channel_mask_t mChannelMask; 442 uint32_t mChannelCount; 443 size_t mFrameSize; 444 // not HAL frame size, this is for output sink (to pipe to fast mixer) 445 audio_format_t mFormat; // Source format for Recording and 446 // Sink format for Playback. 447 // Sink format may be different than 448 // HAL format if Fastmixer is used. 449 audio_format_t mHALFormat; 450 size_t mBufferSize; // HAL buffer size for read() or write() 451 452 Vector< sp<ConfigEvent> > mConfigEvents; 453 Vector< sp<ConfigEvent> > mPendingConfigEvents; // events awaiting system ready 454 455 // These fields are written and read by thread itself without lock or barrier, 456 // and read by other threads without lock or barrier via standby(), outDevice() 457 // and inDevice(). 458 // Because of the absence of a lock or barrier, any other thread that reads 459 // these fields must use the information in isolation, or be prepared to deal 460 // with possibility that it might be inconsistent with other information. 461 bool mStandby; // Whether thread is currently in standby. 462 audio_devices_t mOutDevice; // output device 463 audio_devices_t mInDevice; // input device 464 audio_devices_t mPrevOutDevice; // previous output device 465 audio_devices_t mPrevInDevice; // previous input device 466 struct audio_patch mPatch; 467 audio_source_t mAudioSource; 468 469 const audio_io_handle_t mId; 470 Vector< sp<EffectChain> > mEffectChains; 471 472 static const int kThreadNameLength = 16; // prctl(PR_SET_NAME) limit 473 char mThreadName[kThreadNameLength]; // guaranteed NUL-terminated 474 sp<IPowerManager> mPowerManager; 475 sp<IBinder> mWakeLockToken; 476 const sp<PMDeathRecipient> mDeathRecipient; 477 // list of suspended effects per session and per type. The first (outer) vector is 478 // keyed by session ID, the second (inner) by type UUID timeLow field 479 KeyedVector< audio_session_t, KeyedVector< int, sp<SuspendedSessionDesc> > > 480 mSuspendedSessions; 481 static const size_t kLogSize = 4 * 1024; 482 sp<NBLog::Writer> mNBLogWriter; 483 bool mSystemReady; 484 bool mNotifiedBatteryStart; 485 ExtendedTimestamp mTimestamp; 486}; 487 488// --- PlaybackThread --- 489class PlaybackThread : public ThreadBase, public StreamOutHalInterfaceCallback { 490public: 491 492#include "PlaybackTracks.h" 493 494 enum mixer_state { 495 MIXER_IDLE, // no active tracks 496 MIXER_TRACKS_ENABLED, // at least one active track, but no track has any data ready 497 MIXER_TRACKS_READY, // at least one active track, and at least one track has data 498 MIXER_DRAIN_TRACK, // drain currently playing track 499 MIXER_DRAIN_ALL, // fully drain the hardware 500 // standby mode does not have an enum value 501 // suspend by audio policy manager is orthogonal to mixer state 502 }; 503 504 // retry count before removing active track in case of underrun on offloaded thread: 505 // we need to make sure that AudioTrack client has enough time to send large buffers 506 //FIXME may be more appropriate if expressed in time units. Need to revise how underrun is 507 // handled for offloaded tracks 508 static const int8_t kMaxTrackRetriesOffload = 20; 509 static const int8_t kMaxTrackStartupRetriesOffload = 100; 510 static const int8_t kMaxTrackStopRetriesOffload = 2; 511 // 14 tracks max per client allows for 2 misbehaving application leaving 4 available tracks. 512 static const uint32_t kMaxTracksPerUid = 14; 513 514 PlaybackThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 515 audio_io_handle_t id, audio_devices_t device, type_t type, bool systemReady); 516 virtual ~PlaybackThread(); 517 518 void dump(int fd, const Vector<String16>& args); 519 520 // Thread virtuals 521 virtual bool threadLoop(); 522 523 // RefBase 524 virtual void onFirstRef(); 525 526 virtual status_t checkEffectCompatibility_l(const effect_descriptor_t *desc, 527 audio_session_t sessionId); 528 529protected: 530 // Code snippets that were lifted up out of threadLoop() 531 virtual void threadLoop_mix() = 0; 532 virtual void threadLoop_sleepTime() = 0; 533 virtual ssize_t threadLoop_write(); 534 virtual void threadLoop_drain(); 535 virtual void threadLoop_standby(); 536 virtual void threadLoop_exit(); 537 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 538 539 // prepareTracks_l reads and writes mActiveTracks, and returns 540 // the pending set of tracks to remove via Vector 'tracksToRemove'. The caller 541 // is responsible for clearing or destroying this Vector later on, when it 542 // is safe to do so. That will drop the final ref count and destroy the tracks. 543 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove) = 0; 544 void removeTracks_l(const Vector< sp<Track> >& tracksToRemove); 545 546 // StreamOutHalInterfaceCallback implementation 547 virtual void onWriteReady(); 548 virtual void onDrainReady(); 549 virtual void onError(); 550 551 void resetWriteBlocked(uint32_t sequence); 552 void resetDraining(uint32_t sequence); 553 554 virtual bool waitingAsyncCallback(); 555 virtual bool waitingAsyncCallback_l(); 556 virtual bool shouldStandby_l(); 557 virtual void onAddNewTrack_l(); 558 void onAsyncError(); // error reported by AsyncCallbackThread 559 560 // ThreadBase virtuals 561 virtual void preExit(); 562 563 virtual bool keepWakeLock() const { return true; } 564 565public: 566 567 virtual status_t initCheck() const { return (mOutput == NULL) ? NO_INIT : NO_ERROR; } 568 569 // return estimated latency in milliseconds, as reported by HAL 570 uint32_t latency() const; 571 // same, but lock must already be held 572 uint32_t latency_l() const; 573 574 void setMasterVolume(float value); 575 void setMasterMute(bool muted); 576 577 void setStreamVolume(audio_stream_type_t stream, float value); 578 void setStreamMute(audio_stream_type_t stream, bool muted); 579 580 float streamVolume(audio_stream_type_t stream) const; 581 582 sp<Track> createTrack_l( 583 const sp<AudioFlinger::Client>& client, 584 audio_stream_type_t streamType, 585 uint32_t sampleRate, 586 audio_format_t format, 587 audio_channel_mask_t channelMask, 588 size_t *pFrameCount, 589 const sp<IMemory>& sharedBuffer, 590 audio_session_t sessionId, 591 audio_output_flags_t *flags, 592 pid_t tid, 593 uid_t uid, 594 status_t *status /*non-NULL*/); 595 596 AudioStreamOut* getOutput() const; 597 AudioStreamOut* clearOutput(); 598 virtual sp<StreamHalInterface> stream() const; 599 600 // a very large number of suspend() will eventually wraparound, but unlikely 601 void suspend() { (void) android_atomic_inc(&mSuspended); } 602 void restore() 603 { 604 // if restore() is done without suspend(), get back into 605 // range so that the next suspend() will operate correctly 606 if (android_atomic_dec(&mSuspended) <= 0) { 607 android_atomic_release_store(0, &mSuspended); 608 } 609 } 610 bool isSuspended() const 611 { return android_atomic_acquire_load(&mSuspended) > 0; } 612 613 virtual String8 getParameters(const String8& keys); 614 virtual void ioConfigChanged(audio_io_config_event event, pid_t pid = 0); 615 status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames); 616 // FIXME rename mixBuffer() to sinkBuffer() and remove int16_t* dependency. 617 // Consider also removing and passing an explicit mMainBuffer initialization 618 // parameter to AF::PlaybackThread::Track::Track(). 619 int16_t *mixBuffer() const { 620 return reinterpret_cast<int16_t *>(mSinkBuffer); }; 621 622 virtual void detachAuxEffect_l(int effectId); 623 status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track>& track, 624 int EffectId); 625 status_t attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track>& track, 626 int EffectId); 627 628 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 629 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 630 virtual uint32_t hasAudioSession_l(audio_session_t sessionId) const; 631 virtual uint32_t getStrategyForSession_l(audio_session_t sessionId); 632 633 634 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 635 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 636 637 // called with AudioFlinger lock held 638 bool invalidateTracks_l(audio_stream_type_t streamType); 639 virtual void invalidateTracks(audio_stream_type_t streamType); 640 641 virtual size_t frameCount() const { return mNormalFrameCount; } 642 643 status_t getTimestamp_l(AudioTimestamp& timestamp); 644 645 void addPatchTrack(const sp<PatchTrack>& track); 646 void deletePatchTrack(const sp<PatchTrack>& track); 647 648 virtual void getAudioPortConfig(struct audio_port_config *config); 649 650protected: 651 // updated by readOutputParameters_l() 652 size_t mNormalFrameCount; // normal mixer and effects 653 654 bool mThreadThrottle; // throttle the thread processing 655 uint32_t mThreadThrottleTimeMs; // throttle time for MIXER threads 656 uint32_t mThreadThrottleEndMs; // notify once per throttling 657 uint32_t mHalfBufferMs; // half the buffer size in milliseconds 658 659 void* mSinkBuffer; // frame size aligned sink buffer 660 661 // TODO: 662 // Rearrange the buffer info into a struct/class with 663 // clear, copy, construction, destruction methods. 664 // 665 // mSinkBuffer also has associated with it: 666 // 667 // mSinkBufferSize: Sink Buffer Size 668 // mFormat: Sink Buffer Format 669 670 // Mixer Buffer (mMixerBuffer*) 671 // 672 // In the case of floating point or multichannel data, which is not in the 673 // sink format, it is required to accumulate in a higher precision or greater channel count 674 // buffer before downmixing or data conversion to the sink buffer. 675 676 // Set to "true" to enable the Mixer Buffer otherwise mixer output goes to sink buffer. 677 bool mMixerBufferEnabled; 678 679 // Storage, 32 byte aligned (may make this alignment a requirement later). 680 // Due to constraints on mNormalFrameCount, the buffer size is a multiple of 16 frames. 681 void* mMixerBuffer; 682 683 // Size of mMixerBuffer in bytes: mNormalFrameCount * #channels * sampsize. 684 size_t mMixerBufferSize; 685 686 // The audio format of mMixerBuffer. Set to AUDIO_FORMAT_PCM_(FLOAT|16_BIT) only. 687 audio_format_t mMixerBufferFormat; 688 689 // An internal flag set to true by MixerThread::prepareTracks_l() 690 // when mMixerBuffer contains valid data after mixing. 691 bool mMixerBufferValid; 692 693 // Effects Buffer (mEffectsBuffer*) 694 // 695 // In the case of effects data, which is not in the sink format, 696 // it is required to accumulate in a different buffer before data conversion 697 // to the sink buffer. 698 699 // Set to "true" to enable the Effects Buffer otherwise effects output goes to sink buffer. 700 bool mEffectBufferEnabled; 701 702 // Storage, 32 byte aligned (may make this alignment a requirement later). 703 // Due to constraints on mNormalFrameCount, the buffer size is a multiple of 16 frames. 704 void* mEffectBuffer; 705 706 // Size of mEffectsBuffer in bytes: mNormalFrameCount * #channels * sampsize. 707 size_t mEffectBufferSize; 708 709 // The audio format of mEffectsBuffer. Set to AUDIO_FORMAT_PCM_16_BIT only. 710 audio_format_t mEffectBufferFormat; 711 712 // An internal flag set to true by MixerThread::prepareTracks_l() 713 // when mEffectsBuffer contains valid data after mixing. 714 // 715 // When this is set, all mixer data is routed into the effects buffer 716 // for any processing (including output processing). 717 bool mEffectBufferValid; 718 719 // suspend count, > 0 means suspended. While suspended, the thread continues to pull from 720 // tracks and mix, but doesn't write to HAL. A2DP and SCO HAL implementations can't handle 721 // concurrent use of both of them, so Audio Policy Service suspends one of the threads to 722 // workaround that restriction. 723 // 'volatile' means accessed via atomic operations and no lock. 724 volatile int32_t mSuspended; 725 726 int64_t mBytesWritten; 727 int64_t mFramesWritten; // not reset on standby 728 int64_t mSuspendedFrames; // not reset on standby 729private: 730 // mMasterMute is in both PlaybackThread and in AudioFlinger. When a 731 // PlaybackThread needs to find out if master-muted, it checks it's local 732 // copy rather than the one in AudioFlinger. This optimization saves a lock. 733 bool mMasterMute; 734 void setMasterMute_l(bool muted) { mMasterMute = muted; } 735protected: 736 SortedVector< wp<Track> > mActiveTracks; // FIXME check if this could be sp<> 737 SortedVector<int> mWakeLockUids; 738 int mActiveTracksGeneration; 739 wp<Track> mLatestActiveTrack; // latest track added to mActiveTracks 740 741 // Allocate a track name for a given channel mask. 742 // Returns name >= 0 if successful, -1 on failure. 743 virtual int getTrackName_l(audio_channel_mask_t channelMask, audio_format_t format, 744 audio_session_t sessionId, uid_t uid) = 0; 745 virtual void deleteTrackName_l(int name) = 0; 746 747 // Time to sleep between cycles when: 748 virtual uint32_t activeSleepTimeUs() const; // mixer state MIXER_TRACKS_ENABLED 749 virtual uint32_t idleSleepTimeUs() const = 0; // mixer state MIXER_IDLE 750 virtual uint32_t suspendSleepTimeUs() const = 0; // audio policy manager suspended us 751 // No sleep when mixer state == MIXER_TRACKS_READY; relies on audio HAL stream->write() 752 // No sleep in standby mode; waits on a condition 753 754 // Code snippets that are temporarily lifted up out of threadLoop() until the merge 755 void checkSilentMode_l(); 756 757 // Non-trivial for DUPLICATING only 758 virtual void saveOutputTracks() { } 759 virtual void clearOutputTracks() { } 760 761 // Cache various calculated values, at threadLoop() entry and after a parameter change 762 virtual void cacheParameters_l(); 763 764 virtual uint32_t correctLatency_l(uint32_t latency) const; 765 766 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 767 audio_patch_handle_t *handle); 768 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 769 770 bool usesHwAvSync() const { return (mType == DIRECT) && (mOutput != NULL) 771 && mHwSupportsPause 772 && (mOutput->flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC); } 773 774 uint32_t trackCountForUid_l(uid_t uid); 775 776private: 777 778 friend class AudioFlinger; // for numerous 779 780 PlaybackThread& operator = (const PlaybackThread&); 781 782 status_t addTrack_l(const sp<Track>& track); 783 bool destroyTrack_l(const sp<Track>& track); 784 void removeTrack_l(const sp<Track>& track); 785 void broadcast_l(); 786 787 void readOutputParameters_l(); 788 789 virtual void dumpInternals(int fd, const Vector<String16>& args); 790 void dumpTracks(int fd, const Vector<String16>& args); 791 792 SortedVector< sp<Track> > mTracks; 793 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 794 AudioStreamOut *mOutput; 795 796 float mMasterVolume; 797 nsecs_t mLastWriteTime; 798 int mNumWrites; 799 int mNumDelayedWrites; 800 bool mInWrite; 801 802 // FIXME rename these former local variables of threadLoop to standard "m" names 803 nsecs_t mStandbyTimeNs; 804 size_t mSinkBufferSize; 805 806 // cached copies of activeSleepTimeUs() and idleSleepTimeUs() made by cacheParameters_l() 807 uint32_t mActiveSleepTimeUs; 808 uint32_t mIdleSleepTimeUs; 809 810 uint32_t mSleepTimeUs; 811 812 // mixer status returned by prepareTracks_l() 813 mixer_state mMixerStatus; // current cycle 814 // previous cycle when in prepareTracks_l() 815 mixer_state mMixerStatusIgnoringFastTracks; 816 // FIXME or a separate ready state per track 817 818 // FIXME move these declarations into the specific sub-class that needs them 819 // MIXER only 820 uint32_t sleepTimeShift; 821 822 // same as AudioFlinger::mStandbyTimeInNsecs except for DIRECT which uses a shorter value 823 nsecs_t mStandbyDelayNs; 824 825 // MIXER only 826 nsecs_t maxPeriod; 827 828 // DUPLICATING only 829 uint32_t writeFrames; 830 831 size_t mBytesRemaining; 832 size_t mCurrentWriteLength; 833 bool mUseAsyncWrite; 834 // mWriteAckSequence contains current write sequence on bits 31-1. The write sequence is 835 // incremented each time a write(), a flush() or a standby() occurs. 836 // Bit 0 is set when a write blocks and indicates a callback is expected. 837 // Bit 0 is reset by the async callback thread calling resetWriteBlocked(). Out of sequence 838 // callbacks are ignored. 839 uint32_t mWriteAckSequence; 840 // mDrainSequence contains current drain sequence on bits 31-1. The drain sequence is 841 // incremented each time a drain is requested or a flush() or standby() occurs. 842 // Bit 0 is set when the drain() command is called at the HAL and indicates a callback is 843 // expected. 844 // Bit 0 is reset by the async callback thread calling resetDraining(). Out of sequence 845 // callbacks are ignored. 846 uint32_t mDrainSequence; 847 // A condition that must be evaluated by prepareTrack_l() has changed and we must not wait 848 // for async write callback in the thread loop before evaluating it 849 bool mSignalPending; 850 sp<AsyncCallbackThread> mCallbackThread; 851 852private: 853 // The HAL output sink is treated as non-blocking, but current implementation is blocking 854 sp<NBAIO_Sink> mOutputSink; 855 // If a fast mixer is present, the blocking pipe sink, otherwise clear 856 sp<NBAIO_Sink> mPipeSink; 857 // The current sink for the normal mixer to write it's (sub)mix, mOutputSink or mPipeSink 858 sp<NBAIO_Sink> mNormalSink; 859#ifdef TEE_SINK 860 // For dumpsys 861 sp<NBAIO_Sink> mTeeSink; 862 sp<NBAIO_Source> mTeeSource; 863#endif 864 uint32_t mScreenState; // cached copy of gScreenState 865 static const size_t kFastMixerLogSize = 4 * 1024; 866 sp<NBLog::Writer> mFastMixerNBLogWriter; 867public: 868 virtual bool hasFastMixer() const = 0; 869 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex __unused) const 870 { FastTrackUnderruns dummy; return dummy; } 871 872protected: 873 // accessed by both binder threads and within threadLoop(), lock on mutex needed 874 unsigned mFastTrackAvailMask; // bit i set if fast track [i] is available 875 bool mHwSupportsPause; 876 bool mHwPaused; 877 bool mFlushPending; 878}; 879 880class MixerThread : public PlaybackThread { 881public: 882 MixerThread(const sp<AudioFlinger>& audioFlinger, 883 AudioStreamOut* output, 884 audio_io_handle_t id, 885 audio_devices_t device, 886 bool systemReady, 887 type_t type = MIXER); 888 virtual ~MixerThread(); 889 890 // Thread virtuals 891 892 virtual bool checkForNewParameter_l(const String8& keyValuePair, 893 status_t& status); 894 virtual void dumpInternals(int fd, const Vector<String16>& args); 895 896protected: 897 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 898 virtual int getTrackName_l(audio_channel_mask_t channelMask, audio_format_t format, 899 audio_session_t sessionId, uid_t uid); 900 virtual void deleteTrackName_l(int name); 901 virtual uint32_t idleSleepTimeUs() const; 902 virtual uint32_t suspendSleepTimeUs() const; 903 virtual void cacheParameters_l(); 904 905 virtual void acquireWakeLock_l(int uid = -1) { 906 PlaybackThread::acquireWakeLock_l(uid); 907 if (hasFastMixer()) { 908 mFastMixer->setBoottimeOffset( 909 mTimestamp.mTimebaseOffset[ExtendedTimestamp::TIMEBASE_BOOTTIME]); 910 } 911 } 912 913 // threadLoop snippets 914 virtual ssize_t threadLoop_write(); 915 virtual void threadLoop_standby(); 916 virtual void threadLoop_mix(); 917 virtual void threadLoop_sleepTime(); 918 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 919 virtual uint32_t correctLatency_l(uint32_t latency) const; 920 921 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 922 audio_patch_handle_t *handle); 923 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 924 925 AudioMixer* mAudioMixer; // normal mixer 926private: 927 // one-time initialization, no locks required 928 sp<FastMixer> mFastMixer; // non-0 if there is also a fast mixer 929 sp<AudioWatchdog> mAudioWatchdog; // non-0 if there is an audio watchdog thread 930 931 // contents are not guaranteed to be consistent, no locks required 932 FastMixerDumpState mFastMixerDumpState; 933#ifdef STATE_QUEUE_DUMP 934 StateQueueObserverDump mStateQueueObserverDump; 935 StateQueueMutatorDump mStateQueueMutatorDump; 936#endif 937 AudioWatchdogDump mAudioWatchdogDump; 938 939 // accessible only within the threadLoop(), no locks required 940 // mFastMixer->sq() // for mutating and pushing state 941 int32_t mFastMixerFutex; // for cold idle 942 943 std::atomic_bool mMasterMono; 944public: 945 virtual bool hasFastMixer() const { return mFastMixer != 0; } 946 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const { 947 ALOG_ASSERT(fastIndex < FastMixerState::sMaxFastTracks); 948 return mFastMixerDumpState.mTracks[fastIndex].mUnderruns; 949 } 950 951protected: 952 virtual void setMasterMono_l(bool mono) { 953 mMasterMono.store(mono); 954 if (mFastMixer != nullptr) { /* hasFastMixer() */ 955 mFastMixer->setMasterMono(mMasterMono); 956 } 957 } 958 // the FastMixer performs mono blend if it exists. 959 // Blending with limiter is not idempotent, 960 // and blending without limiter is idempotent but inefficient to do twice. 961 virtual bool requireMonoBlend() { return mMasterMono.load() && !hasFastMixer(); } 962}; 963 964class DirectOutputThread : public PlaybackThread { 965public: 966 967 DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 968 audio_io_handle_t id, audio_devices_t device, bool systemReady); 969 virtual ~DirectOutputThread(); 970 971 // Thread virtuals 972 973 virtual bool checkForNewParameter_l(const String8& keyValuePair, 974 status_t& status); 975 virtual void flushHw_l(); 976 977protected: 978 virtual int getTrackName_l(audio_channel_mask_t channelMask, audio_format_t format, 979 audio_session_t sessionId, uid_t uid); 980 virtual void deleteTrackName_l(int name); 981 virtual uint32_t activeSleepTimeUs() const; 982 virtual uint32_t idleSleepTimeUs() const; 983 virtual uint32_t suspendSleepTimeUs() const; 984 virtual void cacheParameters_l(); 985 986 // threadLoop snippets 987 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 988 virtual void threadLoop_mix(); 989 virtual void threadLoop_sleepTime(); 990 virtual void threadLoop_exit(); 991 virtual bool shouldStandby_l(); 992 993 virtual void onAddNewTrack_l(); 994 995 // volumes last sent to audio HAL with stream->set_volume() 996 float mLeftVolFloat; 997 float mRightVolFloat; 998 999 DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1000 audio_io_handle_t id, uint32_t device, ThreadBase::type_t type, 1001 bool systemReady); 1002 void processVolume_l(Track *track, bool lastTrack); 1003 1004 // prepareTracks_l() tells threadLoop_mix() the name of the single active track 1005 sp<Track> mActiveTrack; 1006 1007 wp<Track> mPreviousTrack; // used to detect track switch 1008 1009public: 1010 virtual bool hasFastMixer() const { return false; } 1011}; 1012 1013class OffloadThread : public DirectOutputThread { 1014public: 1015 1016 OffloadThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1017 audio_io_handle_t id, uint32_t device, bool systemReady); 1018 virtual ~OffloadThread() {}; 1019 virtual void flushHw_l(); 1020 1021protected: 1022 // threadLoop snippets 1023 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1024 virtual void threadLoop_exit(); 1025 1026 virtual bool waitingAsyncCallback(); 1027 virtual bool waitingAsyncCallback_l(); 1028 virtual void invalidateTracks(audio_stream_type_t streamType); 1029 1030 virtual bool keepWakeLock() const { return (mKeepWakeLock || (mDrainSequence & 1)); } 1031 1032private: 1033 size_t mPausedWriteLength; // length in bytes of write interrupted by pause 1034 size_t mPausedBytesRemaining; // bytes still waiting in mixbuffer after resume 1035 bool mKeepWakeLock; // keep wake lock while waiting for write callback 1036 uint64_t mOffloadUnderrunPosition; // Current frame position for offloaded playback 1037 // used and valid only during underrun. ~0 if 1038 // no underrun has occurred during playback and 1039 // is not reset on standby. 1040}; 1041 1042class AsyncCallbackThread : public Thread { 1043public: 1044 1045 explicit AsyncCallbackThread(const wp<PlaybackThread>& playbackThread); 1046 1047 virtual ~AsyncCallbackThread(); 1048 1049 // Thread virtuals 1050 virtual bool threadLoop(); 1051 1052 // RefBase 1053 virtual void onFirstRef(); 1054 1055 void exit(); 1056 void setWriteBlocked(uint32_t sequence); 1057 void resetWriteBlocked(); 1058 void setDraining(uint32_t sequence); 1059 void resetDraining(); 1060 void setAsyncError(); 1061 1062private: 1063 const wp<PlaybackThread> mPlaybackThread; 1064 // mWriteAckSequence corresponds to the last write sequence passed by the offload thread via 1065 // setWriteBlocked(). The sequence is shifted one bit to the left and the lsb is used 1066 // to indicate that the callback has been received via resetWriteBlocked() 1067 uint32_t mWriteAckSequence; 1068 // mDrainSequence corresponds to the last drain sequence passed by the offload thread via 1069 // setDraining(). The sequence is shifted one bit to the left and the lsb is used 1070 // to indicate that the callback has been received via resetDraining() 1071 uint32_t mDrainSequence; 1072 Condition mWaitWorkCV; 1073 Mutex mLock; 1074 bool mAsyncError; 1075}; 1076 1077class DuplicatingThread : public MixerThread { 1078public: 1079 DuplicatingThread(const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread, 1080 audio_io_handle_t id, bool systemReady); 1081 virtual ~DuplicatingThread(); 1082 1083 // Thread virtuals 1084 void addOutputTrack(MixerThread* thread); 1085 void removeOutputTrack(MixerThread* thread); 1086 uint32_t waitTimeMs() const { return mWaitTimeMs; } 1087protected: 1088 virtual uint32_t activeSleepTimeUs() const; 1089 1090private: 1091 bool outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks); 1092protected: 1093 // threadLoop snippets 1094 virtual void threadLoop_mix(); 1095 virtual void threadLoop_sleepTime(); 1096 virtual ssize_t threadLoop_write(); 1097 virtual void threadLoop_standby(); 1098 virtual void cacheParameters_l(); 1099 1100private: 1101 // called from threadLoop, addOutputTrack, removeOutputTrack 1102 virtual void updateWaitTime_l(); 1103protected: 1104 virtual void saveOutputTracks(); 1105 virtual void clearOutputTracks(); 1106private: 1107 1108 uint32_t mWaitTimeMs; 1109 SortedVector < sp<OutputTrack> > outputTracks; 1110 SortedVector < sp<OutputTrack> > mOutputTracks; 1111public: 1112 virtual bool hasFastMixer() const { return false; } 1113}; 1114 1115 1116// record thread 1117class RecordThread : public ThreadBase 1118{ 1119public: 1120 1121 class RecordTrack; 1122 1123 /* The ResamplerBufferProvider is used to retrieve recorded input data from the 1124 * RecordThread. It maintains local state on the relative position of the read 1125 * position of the RecordTrack compared with the RecordThread. 1126 */ 1127 class ResamplerBufferProvider : public AudioBufferProvider 1128 { 1129 public: 1130 explicit ResamplerBufferProvider(RecordTrack* recordTrack) : 1131 mRecordTrack(recordTrack), 1132 mRsmpInUnrel(0), mRsmpInFront(0) { } 1133 virtual ~ResamplerBufferProvider() { } 1134 1135 // called to set the ResamplerBufferProvider to head of the RecordThread data buffer, 1136 // skipping any previous data read from the hal. 1137 virtual void reset(); 1138 1139 /* Synchronizes RecordTrack position with the RecordThread. 1140 * Calculates available frames and handle overruns if the RecordThread 1141 * has advanced faster than the ResamplerBufferProvider has retrieved data. 1142 * TODO: why not do this for every getNextBuffer? 1143 * 1144 * Parameters 1145 * framesAvailable: pointer to optional output size_t to store record track 1146 * frames available. 1147 * hasOverrun: pointer to optional boolean, returns true if track has overrun. 1148 */ 1149 1150 virtual void sync(size_t *framesAvailable = NULL, bool *hasOverrun = NULL); 1151 1152 // AudioBufferProvider interface 1153 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer); 1154 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 1155 private: 1156 RecordTrack * const mRecordTrack; 1157 size_t mRsmpInUnrel; // unreleased frames remaining from 1158 // most recent getNextBuffer 1159 // for debug only 1160 int32_t mRsmpInFront; // next available frame 1161 // rolling counter that is never cleared 1162 }; 1163 1164 /* The RecordBufferConverter is used for format, channel, and sample rate 1165 * conversion for a RecordTrack. 1166 * 1167 * TODO: Self contained, so move to a separate file later. 1168 * 1169 * RecordBufferConverter uses the convert() method rather than exposing a 1170 * buffer provider interface; this is to save a memory copy. 1171 */ 1172 class RecordBufferConverter 1173 { 1174 public: 1175 RecordBufferConverter( 1176 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 1177 uint32_t srcSampleRate, 1178 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 1179 uint32_t dstSampleRate); 1180 1181 ~RecordBufferConverter(); 1182 1183 /* Converts input data from an AudioBufferProvider by format, channelMask, 1184 * and sampleRate to a destination buffer. 1185 * 1186 * Parameters 1187 * dst: buffer to place the converted data. 1188 * provider: buffer provider to obtain source data. 1189 * frames: number of frames to convert 1190 * 1191 * Returns the number of frames converted. 1192 */ 1193 size_t convert(void *dst, AudioBufferProvider *provider, size_t frames); 1194 1195 // returns NO_ERROR if constructor was successful 1196 status_t initCheck() const { 1197 // mSrcChannelMask set on successful updateParameters 1198 return mSrcChannelMask != AUDIO_CHANNEL_INVALID ? NO_ERROR : NO_INIT; 1199 } 1200 1201 // allows dynamic reconfigure of all parameters 1202 status_t updateParameters( 1203 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 1204 uint32_t srcSampleRate, 1205 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 1206 uint32_t dstSampleRate); 1207 1208 // called to reset resampler buffers on record track discontinuity 1209 void reset() { 1210 if (mResampler != NULL) { 1211 mResampler->reset(); 1212 } 1213 } 1214 1215 private: 1216 // format conversion when not using resampler 1217 void convertNoResampler(void *dst, const void *src, size_t frames); 1218 1219 // format conversion when using resampler; modifies src in-place 1220 void convertResampler(void *dst, /*not-a-const*/ void *src, size_t frames); 1221 1222 // user provided information 1223 audio_channel_mask_t mSrcChannelMask; 1224 audio_format_t mSrcFormat; 1225 uint32_t mSrcSampleRate; 1226 audio_channel_mask_t mDstChannelMask; 1227 audio_format_t mDstFormat; 1228 uint32_t mDstSampleRate; 1229 1230 // derived information 1231 uint32_t mSrcChannelCount; 1232 uint32_t mDstChannelCount; 1233 size_t mDstFrameSize; 1234 1235 // format conversion buffer 1236 void *mBuf; 1237 size_t mBufFrames; 1238 size_t mBufFrameSize; 1239 1240 // resampler info 1241 AudioResampler *mResampler; 1242 1243 bool mIsLegacyDownmix; // legacy stereo to mono conversion needed 1244 bool mIsLegacyUpmix; // legacy mono to stereo conversion needed 1245 bool mRequiresFloat; // data processing requires float (e.g. resampler) 1246 PassthruBufferProvider *mInputConverterProvider; // converts input to float 1247 int8_t mIdxAry[sizeof(uint32_t) * 8]; // used for channel mask conversion 1248 }; 1249 1250#include "RecordTracks.h" 1251 1252 RecordThread(const sp<AudioFlinger>& audioFlinger, 1253 AudioStreamIn *input, 1254 audio_io_handle_t id, 1255 audio_devices_t outDevice, 1256 audio_devices_t inDevice, 1257 bool systemReady 1258#ifdef TEE_SINK 1259 , const sp<NBAIO_Sink>& teeSink 1260#endif 1261 ); 1262 virtual ~RecordThread(); 1263 1264 // no addTrack_l ? 1265 void destroyTrack_l(const sp<RecordTrack>& track); 1266 void removeTrack_l(const sp<RecordTrack>& track); 1267 1268 void dumpInternals(int fd, const Vector<String16>& args); 1269 void dumpTracks(int fd, const Vector<String16>& args); 1270 1271 // Thread virtuals 1272 virtual bool threadLoop(); 1273 1274 // RefBase 1275 virtual void onFirstRef(); 1276 1277 virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; } 1278 1279 virtual sp<MemoryDealer> readOnlyHeap() const { return mReadOnlyHeap; } 1280 1281 virtual sp<IMemory> pipeMemory() const { return mPipeMemory; } 1282 1283 sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l( 1284 const sp<AudioFlinger::Client>& client, 1285 uint32_t sampleRate, 1286 audio_format_t format, 1287 audio_channel_mask_t channelMask, 1288 size_t *pFrameCount, 1289 audio_session_t sessionId, 1290 size_t *notificationFrames, 1291 uid_t uid, 1292 audio_input_flags_t *flags, 1293 pid_t tid, 1294 status_t *status /*non-NULL*/); 1295 1296 status_t start(RecordTrack* recordTrack, 1297 AudioSystem::sync_event_t event, 1298 audio_session_t triggerSession); 1299 1300 // ask the thread to stop the specified track, and 1301 // return true if the caller should then do it's part of the stopping process 1302 bool stop(RecordTrack* recordTrack); 1303 1304 void dump(int fd, const Vector<String16>& args); 1305 AudioStreamIn* clearInput(); 1306 virtual sp<StreamHalInterface> stream() const; 1307 1308 1309 virtual bool checkForNewParameter_l(const String8& keyValuePair, 1310 status_t& status); 1311 virtual void cacheParameters_l() {} 1312 virtual String8 getParameters(const String8& keys); 1313 virtual void ioConfigChanged(audio_io_config_event event, pid_t pid = 0); 1314 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 1315 audio_patch_handle_t *handle); 1316 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 1317 1318 void addPatchRecord(const sp<PatchRecord>& record); 1319 void deletePatchRecord(const sp<PatchRecord>& record); 1320 1321 void readInputParameters_l(); 1322 virtual uint32_t getInputFramesLost(); 1323 1324 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1325 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1326 virtual uint32_t hasAudioSession_l(audio_session_t sessionId) const; 1327 1328 // Return the set of unique session IDs across all tracks. 1329 // The keys are the session IDs, and the associated values are meaningless. 1330 // FIXME replace by Set [and implement Bag/Multiset for other uses]. 1331 KeyedVector<audio_session_t, bool> sessionIds() const; 1332 1333 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 1334 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 1335 1336 static void syncStartEventCallback(const wp<SyncEvent>& event); 1337 1338 virtual size_t frameCount() const { return mFrameCount; } 1339 bool hasFastCapture() const { return mFastCapture != 0; } 1340 virtual void getAudioPortConfig(struct audio_port_config *config); 1341 1342 virtual status_t checkEffectCompatibility_l(const effect_descriptor_t *desc, 1343 audio_session_t sessionId); 1344 1345private: 1346 // Enter standby if not already in standby, and set mStandby flag 1347 void standbyIfNotAlreadyInStandby(); 1348 1349 // Call the HAL standby method unconditionally, and don't change mStandby flag 1350 void inputStandBy(); 1351 1352 AudioStreamIn *mInput; 1353 SortedVector < sp<RecordTrack> > mTracks; 1354 // mActiveTracks has dual roles: it indicates the current active track(s), and 1355 // is used together with mStartStopCond to indicate start()/stop() progress 1356 SortedVector< sp<RecordTrack> > mActiveTracks; 1357 // generation counter for mActiveTracks 1358 int mActiveTracksGen; 1359 Condition mStartStopCond; 1360 1361 // resampler converts input at HAL Hz to output at AudioRecord client Hz 1362 void *mRsmpInBuffer; // size = mRsmpInFramesOA 1363 size_t mRsmpInFrames; // size of resampler input in frames 1364 size_t mRsmpInFramesP2;// size rounded up to a power-of-2 1365 size_t mRsmpInFramesOA;// mRsmpInFramesP2 + over-allocation 1366 1367 // rolling index that is never cleared 1368 int32_t mRsmpInRear; // last filled frame + 1 1369 1370 // For dumpsys 1371 const sp<NBAIO_Sink> mTeeSink; 1372 1373 const sp<MemoryDealer> mReadOnlyHeap; 1374 1375 // one-time initialization, no locks required 1376 sp<FastCapture> mFastCapture; // non-0 if there is also 1377 // a fast capture 1378 1379 // FIXME audio watchdog thread 1380 1381 // contents are not guaranteed to be consistent, no locks required 1382 FastCaptureDumpState mFastCaptureDumpState; 1383#ifdef STATE_QUEUE_DUMP 1384 // FIXME StateQueue observer and mutator dump fields 1385#endif 1386 // FIXME audio watchdog dump 1387 1388 // accessible only within the threadLoop(), no locks required 1389 // mFastCapture->sq() // for mutating and pushing state 1390 int32_t mFastCaptureFutex; // for cold idle 1391 1392 // The HAL input source is treated as non-blocking, 1393 // but current implementation is blocking 1394 sp<NBAIO_Source> mInputSource; 1395 // The source for the normal capture thread to read from: mInputSource or mPipeSource 1396 sp<NBAIO_Source> mNormalSource; 1397 // If a fast capture is present, the non-blocking pipe sink written to by fast capture, 1398 // otherwise clear 1399 sp<NBAIO_Sink> mPipeSink; 1400 // If a fast capture is present, the non-blocking pipe source read by normal thread, 1401 // otherwise clear 1402 sp<NBAIO_Source> mPipeSource; 1403 // Depth of pipe from fast capture to normal thread and fast clients, always power of 2 1404 size_t mPipeFramesP2; 1405 // If a fast capture is present, the Pipe as IMemory, otherwise clear 1406 sp<IMemory> mPipeMemory; 1407 1408 static const size_t kFastCaptureLogSize = 4 * 1024; 1409 sp<NBLog::Writer> mFastCaptureNBLogWriter; 1410 1411 bool mFastTrackAvail; // true if fast track available 1412}; 1413