Threads.h revision 4c415062ad1bb53e9af8f644d8215837262b79bb
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 ConfigEvent(int type, bool requiresSystemReady = false) : 101 mType(type), mStatus(NO_ERROR), mWaitStatus(false), 102 mRequiresSystemReady(requiresSystemReady), mData(NULL) {} 103 }; 104 105 class IoConfigEventData : public ConfigEventData { 106 public: 107 IoConfigEventData(audio_io_config_event event, pid_t pid) : 108 mEvent(event), mPid(pid) {} 109 110 virtual void dump(char *buffer, size_t size) { 111 snprintf(buffer, size, "IO event: event %d\n", mEvent); 112 } 113 114 const audio_io_config_event mEvent; 115 const pid_t mPid; 116 }; 117 118 class IoConfigEvent : public ConfigEvent { 119 public: 120 IoConfigEvent(audio_io_config_event event, pid_t pid) : 121 ConfigEvent(CFG_EVENT_IO) { 122 mData = new IoConfigEventData(event, pid); 123 } 124 virtual ~IoConfigEvent() {} 125 }; 126 127 class PrioConfigEventData : public ConfigEventData { 128 public: 129 PrioConfigEventData(pid_t pid, pid_t tid, int32_t prio) : 130 mPid(pid), mTid(tid), mPrio(prio) {} 131 132 virtual void dump(char *buffer, size_t size) { 133 snprintf(buffer, size, "Prio event: pid %d, tid %d, prio %d\n", mPid, mTid, mPrio); 134 } 135 136 const pid_t mPid; 137 const pid_t mTid; 138 const int32_t mPrio; 139 }; 140 141 class PrioConfigEvent : public ConfigEvent { 142 public: 143 PrioConfigEvent(pid_t pid, pid_t tid, int32_t prio) : 144 ConfigEvent(CFG_EVENT_PRIO, true) { 145 mData = new PrioConfigEventData(pid, tid, prio); 146 } 147 virtual ~PrioConfigEvent() {} 148 }; 149 150 class SetParameterConfigEventData : public ConfigEventData { 151 public: 152 SetParameterConfigEventData(String8 keyValuePairs) : 153 mKeyValuePairs(keyValuePairs) {} 154 155 virtual void dump(char *buffer, size_t size) { 156 snprintf(buffer, size, "KeyValue: %s\n", mKeyValuePairs.string()); 157 } 158 159 const String8 mKeyValuePairs; 160 }; 161 162 class SetParameterConfigEvent : public ConfigEvent { 163 public: 164 SetParameterConfigEvent(String8 keyValuePairs) : 165 ConfigEvent(CFG_EVENT_SET_PARAMETER) { 166 mData = new SetParameterConfigEventData(keyValuePairs); 167 mWaitStatus = true; 168 } 169 virtual ~SetParameterConfigEvent() {} 170 }; 171 172 class CreateAudioPatchConfigEventData : public ConfigEventData { 173 public: 174 CreateAudioPatchConfigEventData(const struct audio_patch patch, 175 audio_patch_handle_t handle) : 176 mPatch(patch), mHandle(handle) {} 177 178 virtual void dump(char *buffer, size_t size) { 179 snprintf(buffer, size, "Patch handle: %u\n", mHandle); 180 } 181 182 const struct audio_patch mPatch; 183 audio_patch_handle_t mHandle; 184 }; 185 186 class CreateAudioPatchConfigEvent : public ConfigEvent { 187 public: 188 CreateAudioPatchConfigEvent(const struct audio_patch patch, 189 audio_patch_handle_t handle) : 190 ConfigEvent(CFG_EVENT_CREATE_AUDIO_PATCH) { 191 mData = new CreateAudioPatchConfigEventData(patch, handle); 192 mWaitStatus = true; 193 } 194 virtual ~CreateAudioPatchConfigEvent() {} 195 }; 196 197 class ReleaseAudioPatchConfigEventData : public ConfigEventData { 198 public: 199 ReleaseAudioPatchConfigEventData(const audio_patch_handle_t handle) : 200 mHandle(handle) {} 201 202 virtual void dump(char *buffer, size_t size) { 203 snprintf(buffer, size, "Patch handle: %u\n", mHandle); 204 } 205 206 audio_patch_handle_t mHandle; 207 }; 208 209 class ReleaseAudioPatchConfigEvent : public ConfigEvent { 210 public: 211 ReleaseAudioPatchConfigEvent(const audio_patch_handle_t handle) : 212 ConfigEvent(CFG_EVENT_RELEASE_AUDIO_PATCH) { 213 mData = new ReleaseAudioPatchConfigEventData(handle); 214 mWaitStatus = true; 215 } 216 virtual ~ReleaseAudioPatchConfigEvent() {} 217 }; 218 219 class PMDeathRecipient : public IBinder::DeathRecipient { 220 public: 221 PMDeathRecipient(const wp<ThreadBase>& thread) : mThread(thread) {} 222 virtual ~PMDeathRecipient() {} 223 224 // IBinder::DeathRecipient 225 virtual void binderDied(const wp<IBinder>& who); 226 227 private: 228 PMDeathRecipient(const PMDeathRecipient&); 229 PMDeathRecipient& operator = (const PMDeathRecipient&); 230 231 wp<ThreadBase> mThread; 232 }; 233 234 virtual status_t initCheck() const = 0; 235 236 // static externally-visible 237 type_t type() const { return mType; } 238 bool isDuplicating() const { return (mType == DUPLICATING); } 239 240 audio_io_handle_t id() const { return mId;} 241 242 // dynamic externally-visible 243 uint32_t sampleRate() const { return mSampleRate; } 244 audio_channel_mask_t channelMask() const { return mChannelMask; } 245 audio_format_t format() const { return mHALFormat; } 246 uint32_t channelCount() const { return mChannelCount; } 247 // Called by AudioFlinger::frameCount(audio_io_handle_t output) and effects, 248 // and returns the [normal mix] buffer's frame count. 249 virtual size_t frameCount() const = 0; 250 251 // Return's the HAL's frame count i.e. fast mixer buffer size. 252 size_t frameCountHAL() const { return mFrameCount; } 253 254 size_t frameSize() const { return mFrameSize; } 255 256 // Should be "virtual status_t requestExitAndWait()" and override same 257 // method in Thread, but Thread::requestExitAndWait() is not yet virtual. 258 void exit(); 259 virtual bool checkForNewParameter_l(const String8& keyValuePair, 260 status_t& status) = 0; 261 virtual status_t setParameters(const String8& keyValuePairs); 262 virtual String8 getParameters(const String8& keys) = 0; 263 virtual void ioConfigChanged(audio_io_config_event event, pid_t pid = 0) = 0; 264 // sendConfigEvent_l() must be called with ThreadBase::mLock held 265 // Can temporarily release the lock if waiting for a reply from 266 // processConfigEvents_l(). 267 status_t sendConfigEvent_l(sp<ConfigEvent>& event); 268 void sendIoConfigEvent(audio_io_config_event event, pid_t pid = 0); 269 void sendIoConfigEvent_l(audio_io_config_event event, pid_t pid = 0); 270 void sendPrioConfigEvent(pid_t pid, pid_t tid, int32_t prio); 271 void sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio); 272 status_t sendSetParameterConfigEvent_l(const String8& keyValuePair); 273 status_t sendCreateAudioPatchConfigEvent(const struct audio_patch *patch, 274 audio_patch_handle_t *handle); 275 status_t sendReleaseAudioPatchConfigEvent(audio_patch_handle_t handle); 276 void processConfigEvents_l(); 277 virtual void cacheParameters_l() = 0; 278 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 279 audio_patch_handle_t *handle) = 0; 280 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle) = 0; 281 virtual void getAudioPortConfig(struct audio_port_config *config) = 0; 282 283 284 // see note at declaration of mStandby, mOutDevice and mInDevice 285 bool standby() const { return mStandby; } 286 audio_devices_t outDevice() const { return mOutDevice; } 287 audio_devices_t inDevice() const { return mInDevice; } 288 289 virtual audio_stream_t* stream() const = 0; 290 291 sp<EffectHandle> createEffect_l( 292 const sp<AudioFlinger::Client>& client, 293 const sp<IEffectClient>& effectClient, 294 int32_t priority, 295 audio_session_t sessionId, 296 effect_descriptor_t *desc, 297 int *enabled, 298 status_t *status /*non-NULL*/); 299 300 // return values for hasAudioSession (bit field) 301 enum effect_state { 302 EFFECT_SESSION = 0x1, // the audio session corresponds to at least one 303 // effect 304 TRACK_SESSION = 0x2, // the audio session corresponds to at least one 305 // track 306 FAST_SESSION = 0x4 // the audio session corresponds to at least one 307 // fast track 308 }; 309 310 // get effect chain corresponding to session Id. 311 sp<EffectChain> getEffectChain(audio_session_t sessionId); 312 // same as getEffectChain() but must be called with ThreadBase mutex locked 313 sp<EffectChain> getEffectChain_l(audio_session_t sessionId) const; 314 // add an effect chain to the chain list (mEffectChains) 315 virtual status_t addEffectChain_l(const sp<EffectChain>& chain) = 0; 316 // remove an effect chain from the chain list (mEffectChains) 317 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain) = 0; 318 // lock all effect chains Mutexes. Must be called before releasing the 319 // ThreadBase mutex before processing the mixer and effects. This guarantees the 320 // integrity of the chains during the process. 321 // Also sets the parameter 'effectChains' to current value of mEffectChains. 322 void lockEffectChains_l(Vector< sp<EffectChain> >& effectChains); 323 // unlock effect chains after process 324 void unlockEffectChains(const Vector< sp<EffectChain> >& effectChains); 325 // get a copy of mEffectChains vector 326 Vector< sp<EffectChain> > getEffectChains_l() const { return mEffectChains; }; 327 // set audio mode to all effect chains 328 void setMode(audio_mode_t mode); 329 // get effect module with corresponding ID on specified audio session 330 sp<AudioFlinger::EffectModule> getEffect(audio_session_t sessionId, int effectId); 331 sp<AudioFlinger::EffectModule> getEffect_l(audio_session_t sessionId, int effectId); 332 // add and effect module. Also creates the effect chain is none exists for 333 // the effects audio session 334 status_t addEffect_l(const sp< EffectModule>& effect); 335 // remove and effect module. Also removes the effect chain is this was the last 336 // effect 337 void removeEffect_l(const sp< EffectModule>& effect); 338 // detach all tracks connected to an auxiliary effect 339 virtual void detachAuxEffect_l(int effectId __unused) {} 340 // returns a combination of: 341 // - EFFECT_SESSION if effects on this audio session exist in one chain 342 // - TRACK_SESSION if tracks on this audio session exist 343 // - FAST_SESSION if fast tracks on this audio session exist 344 virtual uint32_t hasAudioSession_l(audio_session_t sessionId) const = 0; 345 uint32_t hasAudioSession(audio_session_t sessionId) const { 346 Mutex::Autolock _l(mLock); 347 return hasAudioSession_l(sessionId); 348 } 349 350 // the value returned by default implementation is not important as the 351 // strategy is only meaningful for PlaybackThread which implements this method 352 virtual uint32_t getStrategyForSession_l(audio_session_t sessionId __unused) 353 { return 0; } 354 355 // suspend or restore effect according to the type of effect passed. a NULL 356 // type pointer means suspend all effects in the session 357 void setEffectSuspended(const effect_uuid_t *type, 358 bool suspend, 359 audio_session_t sessionId = AUDIO_SESSION_OUTPUT_MIX); 360 // check if some effects must be suspended/restored when an effect is enabled 361 // or disabled 362 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 363 bool enabled, 364 audio_session_t sessionId = 365 AUDIO_SESSION_OUTPUT_MIX); 366 void checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 367 bool enabled, 368 audio_session_t sessionId = 369 AUDIO_SESSION_OUTPUT_MIX); 370 371 virtual status_t setSyncEvent(const sp<SyncEvent>& event) = 0; 372 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const = 0; 373 374 // Return a reference to a per-thread heap which can be used to allocate IMemory 375 // objects that will be read-only to client processes, read/write to mediaserver, 376 // and shared by all client processes of the thread. 377 // The heap is per-thread rather than common across all threads, because 378 // clients can't be trusted not to modify the offset of the IMemory they receive. 379 // If a thread does not have such a heap, this method returns 0. 380 virtual sp<MemoryDealer> readOnlyHeap() const { return 0; } 381 382 virtual sp<IMemory> pipeMemory() const { return 0; } 383 384 void systemReady(); 385 386 // checkEffectCompatibility_l() must be called with ThreadBase::mLock held 387 virtual status_t checkEffectCompatibility_l(const effect_descriptor_t *desc, 388 audio_session_t sessionId) = 0; 389 390 mutable Mutex mLock; 391 392protected: 393 394 // entry describing an effect being suspended in mSuspendedSessions keyed vector 395 class SuspendedSessionDesc : public RefBase { 396 public: 397 SuspendedSessionDesc() : mRefCount(0) {} 398 399 int mRefCount; // number of active suspend requests 400 effect_uuid_t mType; // effect type UUID 401 }; 402 403 void acquireWakeLock(int uid = -1); 404 virtual void acquireWakeLock_l(int uid = -1); 405 void releaseWakeLock(); 406 void releaseWakeLock_l(); 407 void updateWakeLockUids(const SortedVector<int> &uids); 408 void updateWakeLockUids_l(const SortedVector<int> &uids); 409 void getPowerManager_l(); 410 void setEffectSuspended_l(const effect_uuid_t *type, 411 bool suspend, 412 audio_session_t sessionId); 413 // updated mSuspendedSessions when an effect suspended or restored 414 void updateSuspendedSessions_l(const effect_uuid_t *type, 415 bool suspend, 416 audio_session_t sessionId); 417 // check if some effects must be suspended when an effect chain is added 418 void checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain); 419 420 String16 getWakeLockTag(); 421 422 virtual void preExit() { } 423 virtual void setMasterMono_l(bool mono __unused) { } 424 virtual bool requireMonoBlend() { return false; } 425 426 friend class AudioFlinger; // for mEffectChains 427 428 const type_t mType; 429 430 // Used by parameters, config events, addTrack_l, exit 431 Condition mWaitWorkCV; 432 433 const sp<AudioFlinger> mAudioFlinger; 434 435 // updated by PlaybackThread::readOutputParameters_l() or 436 // RecordThread::readInputParameters_l() 437 uint32_t mSampleRate; 438 size_t mFrameCount; // output HAL, direct output, record 439 audio_channel_mask_t mChannelMask; 440 uint32_t mChannelCount; 441 size_t mFrameSize; 442 // not HAL frame size, this is for output sink (to pipe to fast mixer) 443 audio_format_t mFormat; // Source format for Recording and 444 // Sink format for Playback. 445 // Sink format may be different than 446 // HAL format if Fastmixer is used. 447 audio_format_t mHALFormat; 448 size_t mBufferSize; // HAL buffer size for read() or write() 449 450 Vector< sp<ConfigEvent> > mConfigEvents; 451 Vector< sp<ConfigEvent> > mPendingConfigEvents; // events awaiting system ready 452 453 // These fields are written and read by thread itself without lock or barrier, 454 // and read by other threads without lock or barrier via standby(), outDevice() 455 // and inDevice(). 456 // Because of the absence of a lock or barrier, any other thread that reads 457 // these fields must use the information in isolation, or be prepared to deal 458 // with possibility that it might be inconsistent with other information. 459 bool mStandby; // Whether thread is currently in standby. 460 audio_devices_t mOutDevice; // output device 461 audio_devices_t mInDevice; // input device 462 audio_devices_t mPrevOutDevice; // previous output device 463 audio_devices_t mPrevInDevice; // previous input device 464 struct audio_patch mPatch; 465 audio_source_t mAudioSource; 466 467 const audio_io_handle_t mId; 468 Vector< sp<EffectChain> > mEffectChains; 469 470 static const int kThreadNameLength = 16; // prctl(PR_SET_NAME) limit 471 char mThreadName[kThreadNameLength]; // guaranteed NUL-terminated 472 sp<IPowerManager> mPowerManager; 473 sp<IBinder> mWakeLockToken; 474 const sp<PMDeathRecipient> mDeathRecipient; 475 // list of suspended effects per session and per type. The first (outer) vector is 476 // keyed by session ID, the second (inner) by type UUID timeLow field 477 KeyedVector< audio_session_t, KeyedVector< int, sp<SuspendedSessionDesc> > > 478 mSuspendedSessions; 479 static const size_t kLogSize = 4 * 1024; 480 sp<NBLog::Writer> mNBLogWriter; 481 bool mSystemReady; 482 bool mNotifiedBatteryStart; 483 ExtendedTimestamp mTimestamp; 484}; 485 486// --- PlaybackThread --- 487class PlaybackThread : public ThreadBase { 488public: 489 490#include "PlaybackTracks.h" 491 492 enum mixer_state { 493 MIXER_IDLE, // no active tracks 494 MIXER_TRACKS_ENABLED, // at least one active track, but no track has any data ready 495 MIXER_TRACKS_READY, // at least one active track, and at least one track has data 496 MIXER_DRAIN_TRACK, // drain currently playing track 497 MIXER_DRAIN_ALL, // fully drain the hardware 498 // standby mode does not have an enum value 499 // suspend by audio policy manager is orthogonal to mixer state 500 }; 501 502 // retry count before removing active track in case of underrun on offloaded thread: 503 // we need to make sure that AudioTrack client has enough time to send large buffers 504 //FIXME may be more appropriate if expressed in time units. Need to revise how underrun is 505 // handled for offloaded tracks 506 static const int8_t kMaxTrackRetriesOffload = 20; 507 static const int8_t kMaxTrackStartupRetriesOffload = 100; 508 static const int8_t kMaxTrackStopRetriesOffload = 2; 509 510 PlaybackThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 511 audio_io_handle_t id, audio_devices_t device, type_t type, bool systemReady); 512 virtual ~PlaybackThread(); 513 514 void dump(int fd, const Vector<String16>& args); 515 516 // Thread virtuals 517 virtual bool threadLoop(); 518 519 // RefBase 520 virtual void onFirstRef(); 521 522 virtual status_t checkEffectCompatibility_l(const effect_descriptor_t *desc, 523 audio_session_t sessionId); 524 525protected: 526 // Code snippets that were lifted up out of threadLoop() 527 virtual void threadLoop_mix() = 0; 528 virtual void threadLoop_sleepTime() = 0; 529 virtual ssize_t threadLoop_write(); 530 virtual void threadLoop_drain(); 531 virtual void threadLoop_standby(); 532 virtual void threadLoop_exit(); 533 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 534 535 // prepareTracks_l reads and writes mActiveTracks, and returns 536 // the pending set of tracks to remove via Vector 'tracksToRemove'. The caller 537 // is responsible for clearing or destroying this Vector later on, when it 538 // is safe to do so. That will drop the final ref count and destroy the tracks. 539 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove) = 0; 540 void removeTracks_l(const Vector< sp<Track> >& tracksToRemove); 541 542 void writeCallback(); 543 void resetWriteBlocked(uint32_t sequence); 544 void drainCallback(); 545 void resetDraining(uint32_t sequence); 546 547 static int asyncCallback(stream_callback_event_t event, void *param, void *cookie); 548 549 virtual bool waitingAsyncCallback(); 550 virtual bool waitingAsyncCallback_l(); 551 virtual bool shouldStandby_l(); 552 virtual void onAddNewTrack_l(); 553 554 // ThreadBase virtuals 555 virtual void preExit(); 556 557 virtual bool keepWakeLock() const { return true; } 558 559public: 560 561 virtual status_t initCheck() const { return (mOutput == NULL) ? NO_INIT : NO_ERROR; } 562 563 // return estimated latency in milliseconds, as reported by HAL 564 uint32_t latency() const; 565 // same, but lock must already be held 566 uint32_t latency_l() const; 567 568 void setMasterVolume(float value); 569 void setMasterMute(bool muted); 570 571 void setStreamVolume(audio_stream_type_t stream, float value); 572 void setStreamMute(audio_stream_type_t stream, bool muted); 573 574 float streamVolume(audio_stream_type_t stream) const; 575 576 sp<Track> createTrack_l( 577 const sp<AudioFlinger::Client>& client, 578 audio_stream_type_t streamType, 579 uint32_t sampleRate, 580 audio_format_t format, 581 audio_channel_mask_t channelMask, 582 size_t *pFrameCount, 583 const sp<IMemory>& sharedBuffer, 584 audio_session_t sessionId, 585 audio_output_flags_t *flags, 586 pid_t tid, 587 int uid, 588 status_t *status /*non-NULL*/); 589 590 AudioStreamOut* getOutput() const; 591 AudioStreamOut* clearOutput(); 592 virtual audio_stream_t* stream() const; 593 594 // a very large number of suspend() will eventually wraparound, but unlikely 595 void suspend() { (void) android_atomic_inc(&mSuspended); } 596 void restore() 597 { 598 // if restore() is done without suspend(), get back into 599 // range so that the next suspend() will operate correctly 600 if (android_atomic_dec(&mSuspended) <= 0) { 601 android_atomic_release_store(0, &mSuspended); 602 } 603 } 604 bool isSuspended() const 605 { return android_atomic_acquire_load(&mSuspended) > 0; } 606 607 virtual String8 getParameters(const String8& keys); 608 virtual void ioConfigChanged(audio_io_config_event event, pid_t pid = 0); 609 status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames); 610 // FIXME rename mixBuffer() to sinkBuffer() and remove int16_t* dependency. 611 // Consider also removing and passing an explicit mMainBuffer initialization 612 // parameter to AF::PlaybackThread::Track::Track(). 613 int16_t *mixBuffer() const { 614 return reinterpret_cast<int16_t *>(mSinkBuffer); }; 615 616 virtual void detachAuxEffect_l(int effectId); 617 status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track> track, 618 int EffectId); 619 status_t attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track> track, 620 int EffectId); 621 622 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 623 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 624 virtual uint32_t hasAudioSession_l(audio_session_t sessionId) const; 625 virtual uint32_t getStrategyForSession_l(audio_session_t sessionId); 626 627 628 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 629 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 630 631 // called with AudioFlinger lock held 632 bool invalidateTracks_l(audio_stream_type_t streamType); 633 virtual void invalidateTracks(audio_stream_type_t streamType); 634 635 virtual size_t frameCount() const { return mNormalFrameCount; } 636 637 status_t getTimestamp_l(AudioTimestamp& timestamp); 638 639 void addPatchTrack(const sp<PatchTrack>& track); 640 void deletePatchTrack(const sp<PatchTrack>& track); 641 642 virtual void getAudioPortConfig(struct audio_port_config *config); 643 644protected: 645 // updated by readOutputParameters_l() 646 size_t mNormalFrameCount; // normal mixer and effects 647 648 bool mThreadThrottle; // throttle the thread processing 649 uint32_t mThreadThrottleTimeMs; // throttle time for MIXER threads 650 uint32_t mThreadThrottleEndMs; // notify once per throttling 651 uint32_t mHalfBufferMs; // half the buffer size in milliseconds 652 653 void* mSinkBuffer; // frame size aligned sink buffer 654 655 // TODO: 656 // Rearrange the buffer info into a struct/class with 657 // clear, copy, construction, destruction methods. 658 // 659 // mSinkBuffer also has associated with it: 660 // 661 // mSinkBufferSize: Sink Buffer Size 662 // mFormat: Sink Buffer Format 663 664 // Mixer Buffer (mMixerBuffer*) 665 // 666 // In the case of floating point or multichannel data, which is not in the 667 // sink format, it is required to accumulate in a higher precision or greater channel count 668 // buffer before downmixing or data conversion to the sink buffer. 669 670 // Set to "true" to enable the Mixer Buffer otherwise mixer output goes to sink buffer. 671 bool mMixerBufferEnabled; 672 673 // Storage, 32 byte aligned (may make this alignment a requirement later). 674 // Due to constraints on mNormalFrameCount, the buffer size is a multiple of 16 frames. 675 void* mMixerBuffer; 676 677 // Size of mMixerBuffer in bytes: mNormalFrameCount * #channels * sampsize. 678 size_t mMixerBufferSize; 679 680 // The audio format of mMixerBuffer. Set to AUDIO_FORMAT_PCM_(FLOAT|16_BIT) only. 681 audio_format_t mMixerBufferFormat; 682 683 // An internal flag set to true by MixerThread::prepareTracks_l() 684 // when mMixerBuffer contains valid data after mixing. 685 bool mMixerBufferValid; 686 687 // Effects Buffer (mEffectsBuffer*) 688 // 689 // In the case of effects data, which is not in the sink format, 690 // it is required to accumulate in a different buffer before data conversion 691 // to the sink buffer. 692 693 // Set to "true" to enable the Effects Buffer otherwise effects output goes to sink buffer. 694 bool mEffectBufferEnabled; 695 696 // Storage, 32 byte aligned (may make this alignment a requirement later). 697 // Due to constraints on mNormalFrameCount, the buffer size is a multiple of 16 frames. 698 void* mEffectBuffer; 699 700 // Size of mEffectsBuffer in bytes: mNormalFrameCount * #channels * sampsize. 701 size_t mEffectBufferSize; 702 703 // The audio format of mEffectsBuffer. Set to AUDIO_FORMAT_PCM_16_BIT only. 704 audio_format_t mEffectBufferFormat; 705 706 // An internal flag set to true by MixerThread::prepareTracks_l() 707 // when mEffectsBuffer contains valid data after mixing. 708 // 709 // When this is set, all mixer data is routed into the effects buffer 710 // for any processing (including output processing). 711 bool mEffectBufferValid; 712 713 // suspend count, > 0 means suspended. While suspended, the thread continues to pull from 714 // tracks and mix, but doesn't write to HAL. A2DP and SCO HAL implementations can't handle 715 // concurrent use of both of them, so Audio Policy Service suspends one of the threads to 716 // workaround that restriction. 717 // 'volatile' means accessed via atomic operations and no lock. 718 volatile int32_t mSuspended; 719 720 int64_t mBytesWritten; 721 int64_t mFramesWritten; // not reset on standby 722private: 723 // mMasterMute is in both PlaybackThread and in AudioFlinger. When a 724 // PlaybackThread needs to find out if master-muted, it checks it's local 725 // copy rather than the one in AudioFlinger. This optimization saves a lock. 726 bool mMasterMute; 727 void setMasterMute_l(bool muted) { mMasterMute = muted; } 728protected: 729 SortedVector< wp<Track> > mActiveTracks; // FIXME check if this could be sp<> 730 SortedVector<int> mWakeLockUids; 731 int mActiveTracksGeneration; 732 wp<Track> mLatestActiveTrack; // latest track added to mActiveTracks 733 734 // Allocate a track name for a given channel mask. 735 // Returns name >= 0 if successful, -1 on failure. 736 virtual int getTrackName_l(audio_channel_mask_t channelMask, 737 audio_format_t format, audio_session_t sessionId) = 0; 738 virtual void deleteTrackName_l(int name) = 0; 739 740 // Time to sleep between cycles when: 741 virtual uint32_t activeSleepTimeUs() const; // mixer state MIXER_TRACKS_ENABLED 742 virtual uint32_t idleSleepTimeUs() const = 0; // mixer state MIXER_IDLE 743 virtual uint32_t suspendSleepTimeUs() const = 0; // audio policy manager suspended us 744 // No sleep when mixer state == MIXER_TRACKS_READY; relies on audio HAL stream->write() 745 // No sleep in standby mode; waits on a condition 746 747 // Code snippets that are temporarily lifted up out of threadLoop() until the merge 748 void checkSilentMode_l(); 749 750 // Non-trivial for DUPLICATING only 751 virtual void saveOutputTracks() { } 752 virtual void clearOutputTracks() { } 753 754 // Cache various calculated values, at threadLoop() entry and after a parameter change 755 virtual void cacheParameters_l(); 756 757 virtual uint32_t correctLatency_l(uint32_t latency) const; 758 759 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 760 audio_patch_handle_t *handle); 761 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 762 763 bool usesHwAvSync() const { return (mType == DIRECT) && (mOutput != NULL) 764 && mHwSupportsPause 765 && (mOutput->flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC); } 766 767private: 768 769 friend class AudioFlinger; // for numerous 770 771 PlaybackThread& operator = (const PlaybackThread&); 772 773 status_t addTrack_l(const sp<Track>& track); 774 bool destroyTrack_l(const sp<Track>& track); 775 void removeTrack_l(const sp<Track>& track); 776 void broadcast_l(); 777 778 void readOutputParameters_l(); 779 780 virtual void dumpInternals(int fd, const Vector<String16>& args); 781 void dumpTracks(int fd, const Vector<String16>& args); 782 783 SortedVector< sp<Track> > mTracks; 784 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 785 AudioStreamOut *mOutput; 786 787 float mMasterVolume; 788 nsecs_t mLastWriteTime; 789 int mNumWrites; 790 int mNumDelayedWrites; 791 bool mInWrite; 792 793 // FIXME rename these former local variables of threadLoop to standard "m" names 794 nsecs_t mStandbyTimeNs; 795 size_t mSinkBufferSize; 796 797 // cached copies of activeSleepTimeUs() and idleSleepTimeUs() made by cacheParameters_l() 798 uint32_t mActiveSleepTimeUs; 799 uint32_t mIdleSleepTimeUs; 800 801 uint32_t mSleepTimeUs; 802 803 // mixer status returned by prepareTracks_l() 804 mixer_state mMixerStatus; // current cycle 805 // previous cycle when in prepareTracks_l() 806 mixer_state mMixerStatusIgnoringFastTracks; 807 // FIXME or a separate ready state per track 808 809 // FIXME move these declarations into the specific sub-class that needs them 810 // MIXER only 811 uint32_t sleepTimeShift; 812 813 // same as AudioFlinger::mStandbyTimeInNsecs except for DIRECT which uses a shorter value 814 nsecs_t mStandbyDelayNs; 815 816 // MIXER only 817 nsecs_t maxPeriod; 818 819 // DUPLICATING only 820 uint32_t writeFrames; 821 822 size_t mBytesRemaining; 823 size_t mCurrentWriteLength; 824 bool mUseAsyncWrite; 825 // mWriteAckSequence contains current write sequence on bits 31-1. The write sequence is 826 // incremented each time a write(), a flush() or a standby() occurs. 827 // Bit 0 is set when a write blocks and indicates a callback is expected. 828 // Bit 0 is reset by the async callback thread calling resetWriteBlocked(). Out of sequence 829 // callbacks are ignored. 830 uint32_t mWriteAckSequence; 831 // mDrainSequence contains current drain sequence on bits 31-1. The drain sequence is 832 // incremented each time a drain is requested or a flush() or standby() occurs. 833 // Bit 0 is set when the drain() command is called at the HAL and indicates a callback is 834 // expected. 835 // Bit 0 is reset by the async callback thread calling resetDraining(). Out of sequence 836 // callbacks are ignored. 837 uint32_t mDrainSequence; 838 // A condition that must be evaluated by prepareTrack_l() has changed and we must not wait 839 // for async write callback in the thread loop before evaluating it 840 bool mSignalPending; 841 sp<AsyncCallbackThread> mCallbackThread; 842 843private: 844 // The HAL output sink is treated as non-blocking, but current implementation is blocking 845 sp<NBAIO_Sink> mOutputSink; 846 // If a fast mixer is present, the blocking pipe sink, otherwise clear 847 sp<NBAIO_Sink> mPipeSink; 848 // The current sink for the normal mixer to write it's (sub)mix, mOutputSink or mPipeSink 849 sp<NBAIO_Sink> mNormalSink; 850#ifdef TEE_SINK 851 // For dumpsys 852 sp<NBAIO_Sink> mTeeSink; 853 sp<NBAIO_Source> mTeeSource; 854#endif 855 uint32_t mScreenState; // cached copy of gScreenState 856 static const size_t kFastMixerLogSize = 4 * 1024; 857 sp<NBLog::Writer> mFastMixerNBLogWriter; 858public: 859 virtual bool hasFastMixer() const = 0; 860 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex __unused) const 861 { FastTrackUnderruns dummy; return dummy; } 862 863protected: 864 // accessed by both binder threads and within threadLoop(), lock on mutex needed 865 unsigned mFastTrackAvailMask; // bit i set if fast track [i] is available 866 bool mHwSupportsPause; 867 bool mHwPaused; 868 bool mFlushPending; 869}; 870 871class MixerThread : public PlaybackThread { 872public: 873 MixerThread(const sp<AudioFlinger>& audioFlinger, 874 AudioStreamOut* output, 875 audio_io_handle_t id, 876 audio_devices_t device, 877 bool systemReady, 878 type_t type = MIXER); 879 virtual ~MixerThread(); 880 881 // Thread virtuals 882 883 virtual bool checkForNewParameter_l(const String8& keyValuePair, 884 status_t& status); 885 virtual void dumpInternals(int fd, const Vector<String16>& args); 886 887protected: 888 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 889 virtual int getTrackName_l(audio_channel_mask_t channelMask, 890 audio_format_t format, audio_session_t sessionId); 891 virtual void deleteTrackName_l(int name); 892 virtual uint32_t idleSleepTimeUs() const; 893 virtual uint32_t suspendSleepTimeUs() const; 894 virtual void cacheParameters_l(); 895 896 virtual void acquireWakeLock_l(int uid = -1) { 897 PlaybackThread::acquireWakeLock_l(uid); 898 if (hasFastMixer()) { 899 mFastMixer->setBoottimeOffset( 900 mTimestamp.mTimebaseOffset[ExtendedTimestamp::TIMEBASE_BOOTTIME]); 901 } 902 } 903 904 // threadLoop snippets 905 virtual ssize_t threadLoop_write(); 906 virtual void threadLoop_standby(); 907 virtual void threadLoop_mix(); 908 virtual void threadLoop_sleepTime(); 909 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 910 virtual uint32_t correctLatency_l(uint32_t latency) const; 911 912 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 913 audio_patch_handle_t *handle); 914 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 915 916 AudioMixer* mAudioMixer; // normal mixer 917private: 918 // one-time initialization, no locks required 919 sp<FastMixer> mFastMixer; // non-0 if there is also a fast mixer 920 sp<AudioWatchdog> mAudioWatchdog; // non-0 if there is an audio watchdog thread 921 922 // contents are not guaranteed to be consistent, no locks required 923 FastMixerDumpState mFastMixerDumpState; 924#ifdef STATE_QUEUE_DUMP 925 StateQueueObserverDump mStateQueueObserverDump; 926 StateQueueMutatorDump mStateQueueMutatorDump; 927#endif 928 AudioWatchdogDump mAudioWatchdogDump; 929 930 // accessible only within the threadLoop(), no locks required 931 // mFastMixer->sq() // for mutating and pushing state 932 int32_t mFastMixerFutex; // for cold idle 933 934 std::atomic_bool mMasterMono; 935public: 936 virtual bool hasFastMixer() const { return mFastMixer != 0; } 937 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const { 938 ALOG_ASSERT(fastIndex < FastMixerState::sMaxFastTracks); 939 return mFastMixerDumpState.mTracks[fastIndex].mUnderruns; 940 } 941 942protected: 943 virtual void setMasterMono_l(bool mono) { 944 mMasterMono.store(mono); 945 if (mFastMixer != nullptr) { /* hasFastMixer() */ 946 mFastMixer->setMasterMono(mMasterMono); 947 } 948 } 949 // the FastMixer performs mono blend if it exists. 950 // Blending with limiter is not idempotent, 951 // and blending without limiter is idempotent but inefficient to do twice. 952 virtual bool requireMonoBlend() { return mMasterMono.load() && !hasFastMixer(); } 953}; 954 955class DirectOutputThread : public PlaybackThread { 956public: 957 958 DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 959 audio_io_handle_t id, audio_devices_t device, bool systemReady); 960 virtual ~DirectOutputThread(); 961 962 // Thread virtuals 963 964 virtual bool checkForNewParameter_l(const String8& keyValuePair, 965 status_t& status); 966 virtual void flushHw_l(); 967 968protected: 969 virtual int getTrackName_l(audio_channel_mask_t channelMask, 970 audio_format_t format, audio_session_t sessionId); 971 virtual void deleteTrackName_l(int name); 972 virtual uint32_t activeSleepTimeUs() const; 973 virtual uint32_t idleSleepTimeUs() const; 974 virtual uint32_t suspendSleepTimeUs() const; 975 virtual void cacheParameters_l(); 976 977 // threadLoop snippets 978 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 979 virtual void threadLoop_mix(); 980 virtual void threadLoop_sleepTime(); 981 virtual void threadLoop_exit(); 982 virtual bool shouldStandby_l(); 983 984 virtual void onAddNewTrack_l(); 985 986 // volumes last sent to audio HAL with stream->set_volume() 987 float mLeftVolFloat; 988 float mRightVolFloat; 989 990 DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 991 audio_io_handle_t id, uint32_t device, ThreadBase::type_t type, 992 bool systemReady); 993 void processVolume_l(Track *track, bool lastTrack); 994 995 // prepareTracks_l() tells threadLoop_mix() the name of the single active track 996 sp<Track> mActiveTrack; 997 998 wp<Track> mPreviousTrack; // used to detect track switch 999 1000public: 1001 virtual bool hasFastMixer() const { return false; } 1002}; 1003 1004class OffloadThread : public DirectOutputThread { 1005public: 1006 1007 OffloadThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1008 audio_io_handle_t id, uint32_t device, bool systemReady); 1009 virtual ~OffloadThread() {}; 1010 virtual void flushHw_l(); 1011 1012protected: 1013 // threadLoop snippets 1014 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1015 virtual void threadLoop_exit(); 1016 1017 virtual bool waitingAsyncCallback(); 1018 virtual bool waitingAsyncCallback_l(); 1019 virtual void invalidateTracks(audio_stream_type_t streamType); 1020 1021 virtual bool keepWakeLock() const { return mKeepWakeLock; } 1022 1023private: 1024 size_t mPausedWriteLength; // length in bytes of write interrupted by pause 1025 size_t mPausedBytesRemaining; // bytes still waiting in mixbuffer after resume 1026 bool mKeepWakeLock; // keep wake lock while waiting for write callback 1027}; 1028 1029class AsyncCallbackThread : public Thread { 1030public: 1031 1032 AsyncCallbackThread(const wp<PlaybackThread>& playbackThread); 1033 1034 virtual ~AsyncCallbackThread(); 1035 1036 // Thread virtuals 1037 virtual bool threadLoop(); 1038 1039 // RefBase 1040 virtual void onFirstRef(); 1041 1042 void exit(); 1043 void setWriteBlocked(uint32_t sequence); 1044 void resetWriteBlocked(); 1045 void setDraining(uint32_t sequence); 1046 void resetDraining(); 1047 1048private: 1049 const wp<PlaybackThread> mPlaybackThread; 1050 // mWriteAckSequence corresponds to the last write sequence passed by the offload thread via 1051 // setWriteBlocked(). The sequence is shifted one bit to the left and the lsb is used 1052 // to indicate that the callback has been received via resetWriteBlocked() 1053 uint32_t mWriteAckSequence; 1054 // mDrainSequence corresponds to the last drain sequence passed by the offload thread via 1055 // setDraining(). The sequence is shifted one bit to the left and the lsb is used 1056 // to indicate that the callback has been received via resetDraining() 1057 uint32_t mDrainSequence; 1058 Condition mWaitWorkCV; 1059 Mutex mLock; 1060}; 1061 1062class DuplicatingThread : public MixerThread { 1063public: 1064 DuplicatingThread(const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread, 1065 audio_io_handle_t id, bool systemReady); 1066 virtual ~DuplicatingThread(); 1067 1068 // Thread virtuals 1069 void addOutputTrack(MixerThread* thread); 1070 void removeOutputTrack(MixerThread* thread); 1071 uint32_t waitTimeMs() const { return mWaitTimeMs; } 1072protected: 1073 virtual uint32_t activeSleepTimeUs() const; 1074 1075private: 1076 bool outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks); 1077protected: 1078 // threadLoop snippets 1079 virtual void threadLoop_mix(); 1080 virtual void threadLoop_sleepTime(); 1081 virtual ssize_t threadLoop_write(); 1082 virtual void threadLoop_standby(); 1083 virtual void cacheParameters_l(); 1084 1085private: 1086 // called from threadLoop, addOutputTrack, removeOutputTrack 1087 virtual void updateWaitTime_l(); 1088protected: 1089 virtual void saveOutputTracks(); 1090 virtual void clearOutputTracks(); 1091private: 1092 1093 uint32_t mWaitTimeMs; 1094 SortedVector < sp<OutputTrack> > outputTracks; 1095 SortedVector < sp<OutputTrack> > mOutputTracks; 1096public: 1097 virtual bool hasFastMixer() const { return false; } 1098}; 1099 1100 1101// record thread 1102class RecordThread : public ThreadBase 1103{ 1104public: 1105 1106 class RecordTrack; 1107 1108 /* The ResamplerBufferProvider is used to retrieve recorded input data from the 1109 * RecordThread. It maintains local state on the relative position of the read 1110 * position of the RecordTrack compared with the RecordThread. 1111 */ 1112 class ResamplerBufferProvider : public AudioBufferProvider 1113 { 1114 public: 1115 ResamplerBufferProvider(RecordTrack* recordTrack) : 1116 mRecordTrack(recordTrack), 1117 mRsmpInUnrel(0), mRsmpInFront(0) { } 1118 virtual ~ResamplerBufferProvider() { } 1119 1120 // called to set the ResamplerBufferProvider to head of the RecordThread data buffer, 1121 // skipping any previous data read from the hal. 1122 virtual void reset(); 1123 1124 /* Synchronizes RecordTrack position with the RecordThread. 1125 * Calculates available frames and handle overruns if the RecordThread 1126 * has advanced faster than the ResamplerBufferProvider has retrieved data. 1127 * TODO: why not do this for every getNextBuffer? 1128 * 1129 * Parameters 1130 * framesAvailable: pointer to optional output size_t to store record track 1131 * frames available. 1132 * hasOverrun: pointer to optional boolean, returns true if track has overrun. 1133 */ 1134 1135 virtual void sync(size_t *framesAvailable = NULL, bool *hasOverrun = NULL); 1136 1137 // AudioBufferProvider interface 1138 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer); 1139 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 1140 private: 1141 RecordTrack * const mRecordTrack; 1142 size_t mRsmpInUnrel; // unreleased frames remaining from 1143 // most recent getNextBuffer 1144 // for debug only 1145 int32_t mRsmpInFront; // next available frame 1146 // rolling counter that is never cleared 1147 }; 1148 1149 /* The RecordBufferConverter is used for format, channel, and sample rate 1150 * conversion for a RecordTrack. 1151 * 1152 * TODO: Self contained, so move to a separate file later. 1153 * 1154 * RecordBufferConverter uses the convert() method rather than exposing a 1155 * buffer provider interface; this is to save a memory copy. 1156 */ 1157 class RecordBufferConverter 1158 { 1159 public: 1160 RecordBufferConverter( 1161 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 1162 uint32_t srcSampleRate, 1163 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 1164 uint32_t dstSampleRate); 1165 1166 ~RecordBufferConverter(); 1167 1168 /* Converts input data from an AudioBufferProvider by format, channelMask, 1169 * and sampleRate to a destination buffer. 1170 * 1171 * Parameters 1172 * dst: buffer to place the converted data. 1173 * provider: buffer provider to obtain source data. 1174 * frames: number of frames to convert 1175 * 1176 * Returns the number of frames converted. 1177 */ 1178 size_t convert(void *dst, AudioBufferProvider *provider, size_t frames); 1179 1180 // returns NO_ERROR if constructor was successful 1181 status_t initCheck() const { 1182 // mSrcChannelMask set on successful updateParameters 1183 return mSrcChannelMask != AUDIO_CHANNEL_INVALID ? NO_ERROR : NO_INIT; 1184 } 1185 1186 // allows dynamic reconfigure of all parameters 1187 status_t updateParameters( 1188 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 1189 uint32_t srcSampleRate, 1190 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 1191 uint32_t dstSampleRate); 1192 1193 // called to reset resampler buffers on record track discontinuity 1194 void reset() { 1195 if (mResampler != NULL) { 1196 mResampler->reset(); 1197 } 1198 } 1199 1200 private: 1201 // format conversion when not using resampler 1202 void convertNoResampler(void *dst, const void *src, size_t frames); 1203 1204 // format conversion when using resampler; modifies src in-place 1205 void convertResampler(void *dst, /*not-a-const*/ void *src, size_t frames); 1206 1207 // user provided information 1208 audio_channel_mask_t mSrcChannelMask; 1209 audio_format_t mSrcFormat; 1210 uint32_t mSrcSampleRate; 1211 audio_channel_mask_t mDstChannelMask; 1212 audio_format_t mDstFormat; 1213 uint32_t mDstSampleRate; 1214 1215 // derived information 1216 uint32_t mSrcChannelCount; 1217 uint32_t mDstChannelCount; 1218 size_t mDstFrameSize; 1219 1220 // format conversion buffer 1221 void *mBuf; 1222 size_t mBufFrames; 1223 size_t mBufFrameSize; 1224 1225 // resampler info 1226 AudioResampler *mResampler; 1227 1228 bool mIsLegacyDownmix; // legacy stereo to mono conversion needed 1229 bool mIsLegacyUpmix; // legacy mono to stereo conversion needed 1230 bool mRequiresFloat; // data processing requires float (e.g. resampler) 1231 PassthruBufferProvider *mInputConverterProvider; // converts input to float 1232 int8_t mIdxAry[sizeof(uint32_t) * 8]; // used for channel mask conversion 1233 }; 1234 1235#include "RecordTracks.h" 1236 1237 RecordThread(const sp<AudioFlinger>& audioFlinger, 1238 AudioStreamIn *input, 1239 audio_io_handle_t id, 1240 audio_devices_t outDevice, 1241 audio_devices_t inDevice, 1242 bool systemReady 1243#ifdef TEE_SINK 1244 , const sp<NBAIO_Sink>& teeSink 1245#endif 1246 ); 1247 virtual ~RecordThread(); 1248 1249 // no addTrack_l ? 1250 void destroyTrack_l(const sp<RecordTrack>& track); 1251 void removeTrack_l(const sp<RecordTrack>& track); 1252 1253 void dumpInternals(int fd, const Vector<String16>& args); 1254 void dumpTracks(int fd, const Vector<String16>& args); 1255 1256 // Thread virtuals 1257 virtual bool threadLoop(); 1258 1259 // RefBase 1260 virtual void onFirstRef(); 1261 1262 virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; } 1263 1264 virtual sp<MemoryDealer> readOnlyHeap() const { return mReadOnlyHeap; } 1265 1266 virtual sp<IMemory> pipeMemory() const { return mPipeMemory; } 1267 1268 sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l( 1269 const sp<AudioFlinger::Client>& client, 1270 uint32_t sampleRate, 1271 audio_format_t format, 1272 audio_channel_mask_t channelMask, 1273 size_t *pFrameCount, 1274 audio_session_t sessionId, 1275 size_t *notificationFrames, 1276 int uid, 1277 audio_input_flags_t *flags, 1278 pid_t tid, 1279 status_t *status /*non-NULL*/); 1280 1281 status_t start(RecordTrack* recordTrack, 1282 AudioSystem::sync_event_t event, 1283 audio_session_t triggerSession); 1284 1285 // ask the thread to stop the specified track, and 1286 // return true if the caller should then do it's part of the stopping process 1287 bool stop(RecordTrack* recordTrack); 1288 1289 void dump(int fd, const Vector<String16>& args); 1290 AudioStreamIn* clearInput(); 1291 virtual audio_stream_t* stream() const; 1292 1293 1294 virtual bool checkForNewParameter_l(const String8& keyValuePair, 1295 status_t& status); 1296 virtual void cacheParameters_l() {} 1297 virtual String8 getParameters(const String8& keys); 1298 virtual void ioConfigChanged(audio_io_config_event event, pid_t pid = 0); 1299 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 1300 audio_patch_handle_t *handle); 1301 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 1302 1303 void addPatchRecord(const sp<PatchRecord>& record); 1304 void deletePatchRecord(const sp<PatchRecord>& record); 1305 1306 void readInputParameters_l(); 1307 virtual uint32_t getInputFramesLost(); 1308 1309 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1310 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1311 virtual uint32_t hasAudioSession_l(audio_session_t sessionId) const; 1312 1313 // Return the set of unique session IDs across all tracks. 1314 // The keys are the session IDs, and the associated values are meaningless. 1315 // FIXME replace by Set [and implement Bag/Multiset for other uses]. 1316 KeyedVector<audio_session_t, bool> sessionIds() const; 1317 1318 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 1319 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 1320 1321 static void syncStartEventCallback(const wp<SyncEvent>& event); 1322 1323 virtual size_t frameCount() const { return mFrameCount; } 1324 bool hasFastCapture() const { return mFastCapture != 0; } 1325 virtual void getAudioPortConfig(struct audio_port_config *config); 1326 1327 virtual status_t checkEffectCompatibility_l(const effect_descriptor_t *desc, 1328 audio_session_t sessionId); 1329 1330private: 1331 // Enter standby if not already in standby, and set mStandby flag 1332 void standbyIfNotAlreadyInStandby(); 1333 1334 // Call the HAL standby method unconditionally, and don't change mStandby flag 1335 void inputStandBy(); 1336 1337 AudioStreamIn *mInput; 1338 SortedVector < sp<RecordTrack> > mTracks; 1339 // mActiveTracks has dual roles: it indicates the current active track(s), and 1340 // is used together with mStartStopCond to indicate start()/stop() progress 1341 SortedVector< sp<RecordTrack> > mActiveTracks; 1342 // generation counter for mActiveTracks 1343 int mActiveTracksGen; 1344 Condition mStartStopCond; 1345 1346 // resampler converts input at HAL Hz to output at AudioRecord client Hz 1347 void *mRsmpInBuffer; // 1348 size_t mRsmpInFrames; // size of resampler input in frames 1349 size_t mRsmpInFramesP2;// size rounded up to a power-of-2 1350 1351 // rolling index that is never cleared 1352 int32_t mRsmpInRear; // last filled frame + 1 1353 1354 // For dumpsys 1355 const sp<NBAIO_Sink> mTeeSink; 1356 1357 const sp<MemoryDealer> mReadOnlyHeap; 1358 1359 // one-time initialization, no locks required 1360 sp<FastCapture> mFastCapture; // non-0 if there is also 1361 // a fast capture 1362 1363 // FIXME audio watchdog thread 1364 1365 // contents are not guaranteed to be consistent, no locks required 1366 FastCaptureDumpState mFastCaptureDumpState; 1367#ifdef STATE_QUEUE_DUMP 1368 // FIXME StateQueue observer and mutator dump fields 1369#endif 1370 // FIXME audio watchdog dump 1371 1372 // accessible only within the threadLoop(), no locks required 1373 // mFastCapture->sq() // for mutating and pushing state 1374 int32_t mFastCaptureFutex; // for cold idle 1375 1376 // The HAL input source is treated as non-blocking, 1377 // but current implementation is blocking 1378 sp<NBAIO_Source> mInputSource; 1379 // The source for the normal capture thread to read from: mInputSource or mPipeSource 1380 sp<NBAIO_Source> mNormalSource; 1381 // If a fast capture is present, the non-blocking pipe sink written to by fast capture, 1382 // otherwise clear 1383 sp<NBAIO_Sink> mPipeSink; 1384 // If a fast capture is present, the non-blocking pipe source read by normal thread, 1385 // otherwise clear 1386 sp<NBAIO_Source> mPipeSource; 1387 // Depth of pipe from fast capture to normal thread and fast clients, always power of 2 1388 size_t mPipeFramesP2; 1389 // If a fast capture is present, the Pipe as IMemory, otherwise clear 1390 sp<IMemory> mPipeMemory; 1391 1392 static const size_t kFastCaptureLogSize = 4 * 1024; 1393 sp<NBLog::Writer> mFastCaptureNBLogWriter; 1394 1395 bool mFastTrackAvail; // true if fast track available 1396}; 1397