Threads.h revision 054d9d3dea1390294650ac704acb4aa0a0731217
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 virtual ~ThreadBase(); 40 41 virtual status_t readyToRun(); 42 43 void dumpBase(int fd, const Vector<String16>& args); 44 void dumpEffectChains(int fd, const Vector<String16>& args); 45 46 void clearPowerManager(); 47 48 // base for record and playback 49 enum { 50 CFG_EVENT_IO, 51 CFG_EVENT_PRIO, 52 CFG_EVENT_SET_PARAMETER, 53 CFG_EVENT_CREATE_AUDIO_PATCH, 54 CFG_EVENT_RELEASE_AUDIO_PATCH, 55 }; 56 57 class ConfigEventData: public RefBase { 58 public: 59 virtual ~ConfigEventData() {} 60 61 virtual void dump(char *buffer, size_t size) = 0; 62 protected: 63 ConfigEventData() {} 64 }; 65 66 // Config event sequence by client if status needed (e.g binder thread calling setParameters()): 67 // 1. create SetParameterConfigEvent. This sets mWaitStatus in config event 68 // 2. Lock mLock 69 // 3. Call sendConfigEvent_l(): Append to mConfigEvents and mWaitWorkCV.signal 70 // 4. sendConfigEvent_l() reads status from event->mStatus; 71 // 5. sendConfigEvent_l() returns status 72 // 6. Unlock 73 // 74 // Parameter sequence by server: threadLoop calling processConfigEvents_l(): 75 // 1. Lock mLock 76 // 2. If there is an entry in mConfigEvents proceed ... 77 // 3. Read first entry in mConfigEvents 78 // 4. Remove first entry from mConfigEvents 79 // 5. Process 80 // 6. Set event->mStatus 81 // 7. event->mCond.signal 82 // 8. Unlock 83 84 class ConfigEvent: public RefBase { 85 public: 86 virtual ~ConfigEvent() {} 87 88 void dump(char *buffer, size_t size) { mData->dump(buffer, size); } 89 90 const int mType; // event type e.g. CFG_EVENT_IO 91 Mutex mLock; // mutex associated with mCond 92 Condition mCond; // condition for status return 93 status_t mStatus; // status communicated to sender 94 bool mWaitStatus; // true if sender is waiting for status 95 sp<ConfigEventData> mData; // event specific parameter data 96 97 protected: 98 ConfigEvent(int type) : mType(type), mStatus(NO_ERROR), mWaitStatus(false), mData(NULL) {} 99 }; 100 101 class IoConfigEventData : public ConfigEventData { 102 public: 103 IoConfigEventData(int event, int param) : 104 mEvent(event), mParam(param) {} 105 106 virtual void dump(char *buffer, size_t size) { 107 snprintf(buffer, size, "IO event: event %d, param %d\n", mEvent, mParam); 108 } 109 110 const int mEvent; 111 const int mParam; 112 }; 113 114 class IoConfigEvent : public ConfigEvent { 115 public: 116 IoConfigEvent(int event, int param) : 117 ConfigEvent(CFG_EVENT_IO) { 118 mData = new IoConfigEventData(event, param); 119 } 120 virtual ~IoConfigEvent() {} 121 }; 122 123 class PrioConfigEventData : public ConfigEventData { 124 public: 125 PrioConfigEventData(pid_t pid, pid_t tid, int32_t prio) : 126 mPid(pid), mTid(tid), mPrio(prio) {} 127 128 virtual void dump(char *buffer, size_t size) { 129 snprintf(buffer, size, "Prio event: pid %d, tid %d, prio %d\n", mPid, mTid, mPrio); 130 } 131 132 const pid_t mPid; 133 const pid_t mTid; 134 const int32_t mPrio; 135 }; 136 137 class PrioConfigEvent : public ConfigEvent { 138 public: 139 PrioConfigEvent(pid_t pid, pid_t tid, int32_t prio) : 140 ConfigEvent(CFG_EVENT_PRIO) { 141 mData = new PrioConfigEventData(pid, tid, prio); 142 } 143 virtual ~PrioConfigEvent() {} 144 }; 145 146 class SetParameterConfigEventData : public ConfigEventData { 147 public: 148 SetParameterConfigEventData(String8 keyValuePairs) : 149 mKeyValuePairs(keyValuePairs) {} 150 151 virtual void dump(char *buffer, size_t size) { 152 snprintf(buffer, size, "KeyValue: %s\n", mKeyValuePairs.string()); 153 } 154 155 const String8 mKeyValuePairs; 156 }; 157 158 class SetParameterConfigEvent : public ConfigEvent { 159 public: 160 SetParameterConfigEvent(String8 keyValuePairs) : 161 ConfigEvent(CFG_EVENT_SET_PARAMETER) { 162 mData = new SetParameterConfigEventData(keyValuePairs); 163 mWaitStatus = true; 164 } 165 virtual ~SetParameterConfigEvent() {} 166 }; 167 168 class CreateAudioPatchConfigEventData : public ConfigEventData { 169 public: 170 CreateAudioPatchConfigEventData(const struct audio_patch patch, 171 audio_patch_handle_t handle) : 172 mPatch(patch), mHandle(handle) {} 173 174 virtual void dump(char *buffer, size_t size) { 175 snprintf(buffer, size, "Patch handle: %u\n", mHandle); 176 } 177 178 const struct audio_patch mPatch; 179 audio_patch_handle_t mHandle; 180 }; 181 182 class CreateAudioPatchConfigEvent : public ConfigEvent { 183 public: 184 CreateAudioPatchConfigEvent(const struct audio_patch patch, 185 audio_patch_handle_t handle) : 186 ConfigEvent(CFG_EVENT_CREATE_AUDIO_PATCH) { 187 mData = new CreateAudioPatchConfigEventData(patch, handle); 188 mWaitStatus = true; 189 } 190 virtual ~CreateAudioPatchConfigEvent() {} 191 }; 192 193 class ReleaseAudioPatchConfigEventData : public ConfigEventData { 194 public: 195 ReleaseAudioPatchConfigEventData(const audio_patch_handle_t handle) : 196 mHandle(handle) {} 197 198 virtual void dump(char *buffer, size_t size) { 199 snprintf(buffer, size, "Patch handle: %u\n", mHandle); 200 } 201 202 audio_patch_handle_t mHandle; 203 }; 204 205 class ReleaseAudioPatchConfigEvent : public ConfigEvent { 206 public: 207 ReleaseAudioPatchConfigEvent(const audio_patch_handle_t handle) : 208 ConfigEvent(CFG_EVENT_RELEASE_AUDIO_PATCH) { 209 mData = new ReleaseAudioPatchConfigEventData(handle); 210 mWaitStatus = true; 211 } 212 virtual ~ReleaseAudioPatchConfigEvent() {} 213 }; 214 215 class PMDeathRecipient : public IBinder::DeathRecipient { 216 public: 217 PMDeathRecipient(const wp<ThreadBase>& thread) : mThread(thread) {} 218 virtual ~PMDeathRecipient() {} 219 220 // IBinder::DeathRecipient 221 virtual void binderDied(const wp<IBinder>& who); 222 223 private: 224 PMDeathRecipient(const PMDeathRecipient&); 225 PMDeathRecipient& operator = (const PMDeathRecipient&); 226 227 wp<ThreadBase> mThread; 228 }; 229 230 virtual status_t initCheck() const = 0; 231 232 // static externally-visible 233 type_t type() const { return mType; } 234 audio_io_handle_t id() const { return mId;} 235 236 // dynamic externally-visible 237 uint32_t sampleRate() const { return mSampleRate; } 238 audio_channel_mask_t channelMask() const { return mChannelMask; } 239 audio_format_t format() const { return mHALFormat; } 240 uint32_t channelCount() const { return mChannelCount; } 241 // Called by AudioFlinger::frameCount(audio_io_handle_t output) and effects, 242 // and returns the [normal mix] buffer's frame count. 243 virtual size_t frameCount() const = 0; 244 size_t frameSize() const { return mFrameSize; } 245 246 // Should be "virtual status_t requestExitAndWait()" and override same 247 // method in Thread, but Thread::requestExitAndWait() is not yet virtual. 248 void exit(); 249 virtual bool checkForNewParameter_l(const String8& keyValuePair, 250 status_t& status) = 0; 251 virtual status_t setParameters(const String8& keyValuePairs); 252 virtual String8 getParameters(const String8& keys) = 0; 253 virtual void audioConfigChanged(int event, int param = 0) = 0; 254 // sendConfigEvent_l() must be called with ThreadBase::mLock held 255 // Can temporarily release the lock if waiting for a reply from 256 // processConfigEvents_l(). 257 status_t sendConfigEvent_l(sp<ConfigEvent>& event); 258 void sendIoConfigEvent(int event, int param = 0); 259 void sendIoConfigEvent_l(int event, int param = 0); 260 void sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio); 261 status_t sendSetParameterConfigEvent_l(const String8& keyValuePair); 262 status_t sendCreateAudioPatchConfigEvent(const struct audio_patch *patch, 263 audio_patch_handle_t *handle); 264 status_t sendReleaseAudioPatchConfigEvent(audio_patch_handle_t handle); 265 void processConfigEvents_l(); 266 virtual void cacheParameters_l() = 0; 267 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 268 audio_patch_handle_t *handle) = 0; 269 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle) = 0; 270 virtual void getAudioPortConfig(struct audio_port_config *config) = 0; 271 272 273 // see note at declaration of mStandby, mOutDevice and mInDevice 274 bool standby() const { return mStandby; } 275 audio_devices_t outDevice() const { return mOutDevice; } 276 audio_devices_t inDevice() const { return mInDevice; } 277 278 virtual audio_stream_t* stream() const = 0; 279 280 sp<EffectHandle> createEffect_l( 281 const sp<AudioFlinger::Client>& client, 282 const sp<IEffectClient>& effectClient, 283 int32_t priority, 284 int sessionId, 285 effect_descriptor_t *desc, 286 int *enabled, 287 status_t *status /*non-NULL*/); 288 289 // return values for hasAudioSession (bit field) 290 enum effect_state { 291 EFFECT_SESSION = 0x1, // the audio session corresponds to at least one 292 // effect 293 TRACK_SESSION = 0x2 // the audio session corresponds to at least one 294 // track 295 }; 296 297 // get effect chain corresponding to session Id. 298 sp<EffectChain> getEffectChain(int sessionId); 299 // same as getEffectChain() but must be called with ThreadBase mutex locked 300 sp<EffectChain> getEffectChain_l(int sessionId) const; 301 // add an effect chain to the chain list (mEffectChains) 302 virtual status_t addEffectChain_l(const sp<EffectChain>& chain) = 0; 303 // remove an effect chain from the chain list (mEffectChains) 304 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain) = 0; 305 // lock all effect chains Mutexes. Must be called before releasing the 306 // ThreadBase mutex before processing the mixer and effects. This guarantees the 307 // integrity of the chains during the process. 308 // Also sets the parameter 'effectChains' to current value of mEffectChains. 309 void lockEffectChains_l(Vector< sp<EffectChain> >& effectChains); 310 // unlock effect chains after process 311 void unlockEffectChains(const Vector< sp<EffectChain> >& effectChains); 312 // get a copy of mEffectChains vector 313 Vector< sp<EffectChain> > getEffectChains_l() const { return mEffectChains; }; 314 // set audio mode to all effect chains 315 void setMode(audio_mode_t mode); 316 // get effect module with corresponding ID on specified audio session 317 sp<AudioFlinger::EffectModule> getEffect(int sessionId, int effectId); 318 sp<AudioFlinger::EffectModule> getEffect_l(int sessionId, int effectId); 319 // add and effect module. Also creates the effect chain is none exists for 320 // the effects audio session 321 status_t addEffect_l(const sp< EffectModule>& effect); 322 // remove and effect module. Also removes the effect chain is this was the last 323 // effect 324 void removeEffect_l(const sp< EffectModule>& effect); 325 // detach all tracks connected to an auxiliary effect 326 virtual void detachAuxEffect_l(int effectId __unused) {} 327 // returns either EFFECT_SESSION if effects on this audio session exist in one 328 // chain, or TRACK_SESSION if tracks on this audio session exist, or both 329 virtual uint32_t hasAudioSession(int sessionId) const = 0; 330 // the value returned by default implementation is not important as the 331 // strategy is only meaningful for PlaybackThread which implements this method 332 virtual uint32_t getStrategyForSession_l(int sessionId __unused) { return 0; } 333 334 // suspend or restore effect according to the type of effect passed. a NULL 335 // type pointer means suspend all effects in the session 336 void setEffectSuspended(const effect_uuid_t *type, 337 bool suspend, 338 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 339 // check if some effects must be suspended/restored when an effect is enabled 340 // or disabled 341 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 342 bool enabled, 343 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 344 void checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 345 bool enabled, 346 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 347 348 virtual status_t setSyncEvent(const sp<SyncEvent>& event) = 0; 349 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const = 0; 350 351 // Return a reference to a per-thread heap which can be used to allocate IMemory 352 // objects that will be read-only to client processes, read/write to mediaserver, 353 // and shared by all client processes of the thread. 354 // The heap is per-thread rather than common across all threads, because 355 // clients can't be trusted not to modify the offset of the IMemory they receive. 356 // If a thread does not have such a heap, this method returns 0. 357 virtual sp<MemoryDealer> readOnlyHeap() const { return 0; } 358 359 virtual sp<IMemory> pipeMemory() const { return 0; } 360 361 mutable Mutex mLock; 362 363protected: 364 365 // entry describing an effect being suspended in mSuspendedSessions keyed vector 366 class SuspendedSessionDesc : public RefBase { 367 public: 368 SuspendedSessionDesc() : mRefCount(0) {} 369 370 int mRefCount; // number of active suspend requests 371 effect_uuid_t mType; // effect type UUID 372 }; 373 374 void acquireWakeLock(int uid = -1); 375 void acquireWakeLock_l(int uid = -1); 376 void releaseWakeLock(); 377 void releaseWakeLock_l(); 378 void updateWakeLockUids(const SortedVector<int> &uids); 379 void updateWakeLockUids_l(const SortedVector<int> &uids); 380 void getPowerManager_l(); 381 void setEffectSuspended_l(const effect_uuid_t *type, 382 bool suspend, 383 int sessionId); 384 // updated mSuspendedSessions when an effect suspended or restored 385 void updateSuspendedSessions_l(const effect_uuid_t *type, 386 bool suspend, 387 int sessionId); 388 // check if some effects must be suspended when an effect chain is added 389 void checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain); 390 391 String16 getWakeLockTag(); 392 393 virtual void preExit() { } 394 395 friend class AudioFlinger; // for mEffectChains 396 397 const type_t mType; 398 399 // Used by parameters, config events, addTrack_l, exit 400 Condition mWaitWorkCV; 401 402 const sp<AudioFlinger> mAudioFlinger; 403 404 // updated by PlaybackThread::readOutputParameters_l() or 405 // RecordThread::readInputParameters_l() 406 uint32_t mSampleRate; 407 size_t mFrameCount; // output HAL, direct output, record 408 audio_channel_mask_t mChannelMask; 409 uint32_t mChannelCount; 410 size_t mFrameSize; 411 // not HAL frame size, this is for output sink (to pipe to fast mixer) 412 audio_format_t mFormat; // Source format for Recording and 413 // Sink format for Playback. 414 // Sink format may be different than 415 // HAL format if Fastmixer is used. 416 audio_format_t mHALFormat; 417 size_t mBufferSize; // HAL buffer size for read() or write() 418 419 Vector< sp<ConfigEvent> > mConfigEvents; 420 421 // These fields are written and read by thread itself without lock or barrier, 422 // and read by other threads without lock or barrier via standby(), outDevice() 423 // and inDevice(). 424 // Because of the absence of a lock or barrier, any other thread that reads 425 // these fields must use the information in isolation, or be prepared to deal 426 // with possibility that it might be inconsistent with other information. 427 bool mStandby; // Whether thread is currently in standby. 428 audio_devices_t mOutDevice; // output device 429 audio_devices_t mInDevice; // input device 430 audio_source_t mAudioSource; 431 432 const audio_io_handle_t mId; 433 Vector< sp<EffectChain> > mEffectChains; 434 435 static const int kThreadNameLength = 16; // prctl(PR_SET_NAME) limit 436 char mThreadName[kThreadNameLength]; // guaranteed NUL-terminated 437 sp<IPowerManager> mPowerManager; 438 sp<IBinder> mWakeLockToken; 439 const sp<PMDeathRecipient> mDeathRecipient; 440 // list of suspended effects per session and per type. The first vector is 441 // keyed by session ID, the second by type UUID timeLow field 442 KeyedVector< int, KeyedVector< int, sp<SuspendedSessionDesc> > > 443 mSuspendedSessions; 444 static const size_t kLogSize = 4 * 1024; 445 sp<NBLog::Writer> mNBLogWriter; 446}; 447 448// --- PlaybackThread --- 449class PlaybackThread : public ThreadBase { 450public: 451 452#include "PlaybackTracks.h" 453 454 enum mixer_state { 455 MIXER_IDLE, // no active tracks 456 MIXER_TRACKS_ENABLED, // at least one active track, but no track has any data ready 457 MIXER_TRACKS_READY, // at least one active track, and at least one track has data 458 MIXER_DRAIN_TRACK, // drain currently playing track 459 MIXER_DRAIN_ALL, // fully drain the hardware 460 // standby mode does not have an enum value 461 // suspend by audio policy manager is orthogonal to mixer state 462 }; 463 464 // retry count before removing active track in case of underrun on offloaded thread: 465 // we need to make sure that AudioTrack client has enough time to send large buffers 466//FIXME may be more appropriate if expressed in time units. Need to revise how underrun is handled 467 // for offloaded tracks 468 static const int8_t kMaxTrackRetriesOffload = 20; 469 470 PlaybackThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 471 audio_io_handle_t id, audio_devices_t device, type_t type); 472 virtual ~PlaybackThread(); 473 474 void dump(int fd, const Vector<String16>& args); 475 476 // Thread virtuals 477 virtual bool threadLoop(); 478 479 // RefBase 480 virtual void onFirstRef(); 481 482protected: 483 // Code snippets that were lifted up out of threadLoop() 484 virtual void threadLoop_mix() = 0; 485 virtual void threadLoop_sleepTime() = 0; 486 virtual ssize_t threadLoop_write(); 487 virtual void threadLoop_drain(); 488 virtual void threadLoop_standby(); 489 virtual void threadLoop_exit(); 490 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 491 492 // prepareTracks_l reads and writes mActiveTracks, and returns 493 // the pending set of tracks to remove via Vector 'tracksToRemove'. The caller 494 // is responsible for clearing or destroying this Vector later on, when it 495 // is safe to do so. That will drop the final ref count and destroy the tracks. 496 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove) = 0; 497 void removeTracks_l(const Vector< sp<Track> >& tracksToRemove); 498 499 void writeCallback(); 500 void resetWriteBlocked(uint32_t sequence); 501 void drainCallback(); 502 void resetDraining(uint32_t sequence); 503 504 static int asyncCallback(stream_callback_event_t event, void *param, void *cookie); 505 506 virtual bool waitingAsyncCallback(); 507 virtual bool waitingAsyncCallback_l(); 508 virtual bool shouldStandby_l(); 509 virtual void onAddNewTrack_l(); 510 511 // ThreadBase virtuals 512 virtual void preExit(); 513 514public: 515 516 virtual status_t initCheck() const { return (mOutput == NULL) ? NO_INIT : NO_ERROR; } 517 518 // return estimated latency in milliseconds, as reported by HAL 519 uint32_t latency() const; 520 // same, but lock must already be held 521 uint32_t latency_l() const; 522 523 void setMasterVolume(float value); 524 void setMasterMute(bool muted); 525 526 void setStreamVolume(audio_stream_type_t stream, float value); 527 void setStreamMute(audio_stream_type_t stream, bool muted); 528 529 float streamVolume(audio_stream_type_t stream) const; 530 531 sp<Track> createTrack_l( 532 const sp<AudioFlinger::Client>& client, 533 audio_stream_type_t streamType, 534 uint32_t sampleRate, 535 audio_format_t format, 536 audio_channel_mask_t channelMask, 537 size_t *pFrameCount, 538 const sp<IMemory>& sharedBuffer, 539 int sessionId, 540 IAudioFlinger::track_flags_t *flags, 541 pid_t tid, 542 int uid, 543 status_t *status /*non-NULL*/); 544 545 AudioStreamOut* getOutput() const; 546 AudioStreamOut* clearOutput(); 547 virtual audio_stream_t* stream() const; 548 549 // a very large number of suspend() will eventually wraparound, but unlikely 550 void suspend() { (void) android_atomic_inc(&mSuspended); } 551 void restore() 552 { 553 // if restore() is done without suspend(), get back into 554 // range so that the next suspend() will operate correctly 555 if (android_atomic_dec(&mSuspended) <= 0) { 556 android_atomic_release_store(0, &mSuspended); 557 } 558 } 559 bool isSuspended() const 560 { return android_atomic_acquire_load(&mSuspended) > 0; } 561 562 virtual String8 getParameters(const String8& keys); 563 virtual void audioConfigChanged(int event, int param = 0); 564 status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames); 565 // FIXME rename mixBuffer() to sinkBuffer() and remove int16_t* dependency. 566 // Consider also removing and passing an explicit mMainBuffer initialization 567 // parameter to AF::PlaybackThread::Track::Track(). 568 int16_t *mixBuffer() const { 569 return reinterpret_cast<int16_t *>(mSinkBuffer); }; 570 571 virtual void detachAuxEffect_l(int effectId); 572 status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track> track, 573 int EffectId); 574 status_t attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track> track, 575 int EffectId); 576 577 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 578 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 579 virtual uint32_t hasAudioSession(int sessionId) const; 580 virtual uint32_t getStrategyForSession_l(int sessionId); 581 582 583 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 584 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 585 586 // called with AudioFlinger lock held 587 void invalidateTracks(audio_stream_type_t streamType); 588 589 virtual size_t frameCount() const { return mNormalFrameCount; } 590 591 // Return's the HAL's frame count i.e. fast mixer buffer size. 592 size_t frameCountHAL() const { return mFrameCount; } 593 594 status_t getTimestamp_l(AudioTimestamp& timestamp); 595 596 void addPatchTrack(const sp<PatchTrack>& track); 597 void deletePatchTrack(const sp<PatchTrack>& track); 598 599 virtual void getAudioPortConfig(struct audio_port_config *config); 600 601protected: 602 // updated by readOutputParameters_l() 603 size_t mNormalFrameCount; // normal mixer and effects 604 605 void* mSinkBuffer; // frame size aligned sink buffer 606 607 // TODO: 608 // Rearrange the buffer info into a struct/class with 609 // clear, copy, construction, destruction methods. 610 // 611 // mSinkBuffer also has associated with it: 612 // 613 // mSinkBufferSize: Sink Buffer Size 614 // mFormat: Sink Buffer Format 615 616 // Mixer Buffer (mMixerBuffer*) 617 // 618 // In the case of floating point or multichannel data, which is not in the 619 // sink format, it is required to accumulate in a higher precision or greater channel count 620 // buffer before downmixing or data conversion to the sink buffer. 621 622 // Set to "true" to enable the Mixer Buffer otherwise mixer output goes to sink buffer. 623 bool mMixerBufferEnabled; 624 625 // Storage, 32 byte aligned (may make this alignment a requirement later). 626 // Due to constraints on mNormalFrameCount, the buffer size is a multiple of 16 frames. 627 void* mMixerBuffer; 628 629 // Size of mMixerBuffer in bytes: mNormalFrameCount * #channels * sampsize. 630 size_t mMixerBufferSize; 631 632 // The audio format of mMixerBuffer. Set to AUDIO_FORMAT_PCM_(FLOAT|16_BIT) only. 633 audio_format_t mMixerBufferFormat; 634 635 // An internal flag set to true by MixerThread::prepareTracks_l() 636 // when mMixerBuffer contains valid data after mixing. 637 bool mMixerBufferValid; 638 639 // Effects Buffer (mEffectsBuffer*) 640 // 641 // In the case of effects data, which is not in the sink format, 642 // it is required to accumulate in a different buffer before data conversion 643 // to the sink buffer. 644 645 // Set to "true" to enable the Effects Buffer otherwise effects output goes to sink buffer. 646 bool mEffectBufferEnabled; 647 648 // Storage, 32 byte aligned (may make this alignment a requirement later). 649 // Due to constraints on mNormalFrameCount, the buffer size is a multiple of 16 frames. 650 void* mEffectBuffer; 651 652 // Size of mEffectsBuffer in bytes: mNormalFrameCount * #channels * sampsize. 653 size_t mEffectBufferSize; 654 655 // The audio format of mEffectsBuffer. Set to AUDIO_FORMAT_PCM_16_BIT only. 656 audio_format_t mEffectBufferFormat; 657 658 // An internal flag set to true by MixerThread::prepareTracks_l() 659 // when mEffectsBuffer contains valid data after mixing. 660 // 661 // When this is set, all mixer data is routed into the effects buffer 662 // for any processing (including output processing). 663 bool mEffectBufferValid; 664 665 // suspend count, > 0 means suspended. While suspended, the thread continues to pull from 666 // tracks and mix, but doesn't write to HAL. A2DP and SCO HAL implementations can't handle 667 // concurrent use of both of them, so Audio Policy Service suspends one of the threads to 668 // workaround that restriction. 669 // 'volatile' means accessed via atomic operations and no lock. 670 volatile int32_t mSuspended; 671 672 // FIXME overflows every 6+ hours at 44.1 kHz stereo 16-bit samples 673 // mFramesWritten would be better, or 64-bit even better 674 size_t mBytesWritten; 675private: 676 // mMasterMute is in both PlaybackThread and in AudioFlinger. When a 677 // PlaybackThread needs to find out if master-muted, it checks it's local 678 // copy rather than the one in AudioFlinger. This optimization saves a lock. 679 bool mMasterMute; 680 void setMasterMute_l(bool muted) { mMasterMute = muted; } 681protected: 682 SortedVector< wp<Track> > mActiveTracks; // FIXME check if this could be sp<> 683 SortedVector<int> mWakeLockUids; 684 int mActiveTracksGeneration; 685 wp<Track> mLatestActiveTrack; // latest track added to mActiveTracks 686 687 // Allocate a track name for a given channel mask. 688 // Returns name >= 0 if successful, -1 on failure. 689 virtual int getTrackName_l(audio_channel_mask_t channelMask, 690 audio_format_t format, int sessionId) = 0; 691 virtual void deleteTrackName_l(int name) = 0; 692 693 // Time to sleep between cycles when: 694 virtual uint32_t activeSleepTimeUs() const; // mixer state MIXER_TRACKS_ENABLED 695 virtual uint32_t idleSleepTimeUs() const = 0; // mixer state MIXER_IDLE 696 virtual uint32_t suspendSleepTimeUs() const = 0; // audio policy manager suspended us 697 // No sleep when mixer state == MIXER_TRACKS_READY; relies on audio HAL stream->write() 698 // No sleep in standby mode; waits on a condition 699 700 // Code snippets that are temporarily lifted up out of threadLoop() until the merge 701 void checkSilentMode_l(); 702 703 // Non-trivial for DUPLICATING only 704 virtual void saveOutputTracks() { } 705 virtual void clearOutputTracks() { } 706 707 // Cache various calculated values, at threadLoop() entry and after a parameter change 708 virtual void cacheParameters_l(); 709 710 virtual uint32_t correctLatency_l(uint32_t latency) const; 711 712 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 713 audio_patch_handle_t *handle); 714 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 715 716 bool usesHwAvSync() const { return (mType == DIRECT) && (mOutput != NULL) && 717 (mOutput->flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC); } 718 719private: 720 721 friend class AudioFlinger; // for numerous 722 723 PlaybackThread& operator = (const PlaybackThread&); 724 725 status_t addTrack_l(const sp<Track>& track); 726 bool destroyTrack_l(const sp<Track>& track); 727 void removeTrack_l(const sp<Track>& track); 728 void broadcast_l(); 729 730 void readOutputParameters_l(); 731 732 virtual void dumpInternals(int fd, const Vector<String16>& args); 733 void dumpTracks(int fd, const Vector<String16>& args); 734 735 SortedVector< sp<Track> > mTracks; 736 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 737 AudioStreamOut *mOutput; 738 739 float mMasterVolume; 740 nsecs_t mLastWriteTime; 741 int mNumWrites; 742 int mNumDelayedWrites; 743 bool mInWrite; 744 745 // FIXME rename these former local variables of threadLoop to standard "m" names 746 nsecs_t standbyTime; 747 size_t mSinkBufferSize; 748 749 // cached copies of activeSleepTimeUs() and idleSleepTimeUs() made by cacheParameters_l() 750 uint32_t activeSleepTime; 751 uint32_t idleSleepTime; 752 753 uint32_t sleepTime; 754 755 // mixer status returned by prepareTracks_l() 756 mixer_state mMixerStatus; // current cycle 757 // previous cycle when in prepareTracks_l() 758 mixer_state mMixerStatusIgnoringFastTracks; 759 // FIXME or a separate ready state per track 760 761 // FIXME move these declarations into the specific sub-class that needs them 762 // MIXER only 763 uint32_t sleepTimeShift; 764 765 // same as AudioFlinger::mStandbyTimeInNsecs except for DIRECT which uses a shorter value 766 nsecs_t standbyDelay; 767 768 // MIXER only 769 nsecs_t maxPeriod; 770 771 // DUPLICATING only 772 uint32_t writeFrames; 773 774 size_t mBytesRemaining; 775 size_t mCurrentWriteLength; 776 bool mUseAsyncWrite; 777 // mWriteAckSequence contains current write sequence on bits 31-1. The write sequence is 778 // incremented each time a write(), a flush() or a standby() occurs. 779 // Bit 0 is set when a write blocks and indicates a callback is expected. 780 // Bit 0 is reset by the async callback thread calling resetWriteBlocked(). Out of sequence 781 // callbacks are ignored. 782 uint32_t mWriteAckSequence; 783 // mDrainSequence contains current drain sequence on bits 31-1. The drain sequence is 784 // incremented each time a drain is requested or a flush() or standby() occurs. 785 // Bit 0 is set when the drain() command is called at the HAL and indicates a callback is 786 // expected. 787 // Bit 0 is reset by the async callback thread calling resetDraining(). Out of sequence 788 // callbacks are ignored. 789 uint32_t mDrainSequence; 790 // A condition that must be evaluated by prepareTrack_l() has changed and we must not wait 791 // for async write callback in the thread loop before evaluating it 792 bool mSignalPending; 793 sp<AsyncCallbackThread> mCallbackThread; 794 795private: 796 // The HAL output sink is treated as non-blocking, but current implementation is blocking 797 sp<NBAIO_Sink> mOutputSink; 798 // If a fast mixer is present, the blocking pipe sink, otherwise clear 799 sp<NBAIO_Sink> mPipeSink; 800 // The current sink for the normal mixer to write it's (sub)mix, mOutputSink or mPipeSink 801 sp<NBAIO_Sink> mNormalSink; 802#ifdef TEE_SINK 803 // For dumpsys 804 sp<NBAIO_Sink> mTeeSink; 805 sp<NBAIO_Source> mTeeSource; 806#endif 807 uint32_t mScreenState; // cached copy of gScreenState 808 static const size_t kFastMixerLogSize = 4 * 1024; 809 sp<NBLog::Writer> mFastMixerNBLogWriter; 810public: 811 virtual bool hasFastMixer() const = 0; 812 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex __unused) const 813 { FastTrackUnderruns dummy; return dummy; } 814 815protected: 816 // accessed by both binder threads and within threadLoop(), lock on mutex needed 817 unsigned mFastTrackAvailMask; // bit i set if fast track [i] is available 818 bool mHwSupportsPause; 819 bool mHwPaused; 820 bool mFlushPending; 821private: 822 // timestamp latch: 823 // D input is written by threadLoop_write while mutex is unlocked, and read while locked 824 // Q output is written while locked, and read while locked 825 struct { 826 AudioTimestamp mTimestamp; 827 uint32_t mUnpresentedFrames; 828 KeyedVector<Track *, uint32_t> mFramesReleased; 829 } mLatchD, mLatchQ; 830 bool mLatchDValid; // true means mLatchD is valid 831 // (except for mFramesReleased which is filled in later), 832 // and clock it into latch at next opportunity 833 bool mLatchQValid; // true means mLatchQ is valid 834}; 835 836class MixerThread : public PlaybackThread { 837public: 838 MixerThread(const sp<AudioFlinger>& audioFlinger, 839 AudioStreamOut* output, 840 audio_io_handle_t id, 841 audio_devices_t device, 842 type_t type = MIXER); 843 virtual ~MixerThread(); 844 845 // Thread virtuals 846 847 virtual bool checkForNewParameter_l(const String8& keyValuePair, 848 status_t& status); 849 virtual void dumpInternals(int fd, const Vector<String16>& args); 850 851protected: 852 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 853 virtual int getTrackName_l(audio_channel_mask_t channelMask, 854 audio_format_t format, int sessionId); 855 virtual void deleteTrackName_l(int name); 856 virtual uint32_t idleSleepTimeUs() const; 857 virtual uint32_t suspendSleepTimeUs() const; 858 virtual void cacheParameters_l(); 859 860 // threadLoop snippets 861 virtual ssize_t threadLoop_write(); 862 virtual void threadLoop_standby(); 863 virtual void threadLoop_mix(); 864 virtual void threadLoop_sleepTime(); 865 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 866 virtual uint32_t correctLatency_l(uint32_t latency) const; 867 868 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 869 audio_patch_handle_t *handle); 870 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 871 872 AudioMixer* mAudioMixer; // normal mixer 873private: 874 // one-time initialization, no locks required 875 sp<FastMixer> mFastMixer; // non-0 if there is also a fast mixer 876 sp<AudioWatchdog> mAudioWatchdog; // non-0 if there is an audio watchdog thread 877 878 // contents are not guaranteed to be consistent, no locks required 879 FastMixerDumpState mFastMixerDumpState; 880#ifdef STATE_QUEUE_DUMP 881 StateQueueObserverDump mStateQueueObserverDump; 882 StateQueueMutatorDump mStateQueueMutatorDump; 883#endif 884 AudioWatchdogDump mAudioWatchdogDump; 885 886 // accessible only within the threadLoop(), no locks required 887 // mFastMixer->sq() // for mutating and pushing state 888 int32_t mFastMixerFutex; // for cold idle 889 890public: 891 virtual bool hasFastMixer() const { return mFastMixer != 0; } 892 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const { 893 ALOG_ASSERT(fastIndex < FastMixerState::kMaxFastTracks); 894 return mFastMixerDumpState.mTracks[fastIndex].mUnderruns; 895 } 896 897}; 898 899class DirectOutputThread : public PlaybackThread { 900public: 901 902 DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 903 audio_io_handle_t id, audio_devices_t device); 904 virtual ~DirectOutputThread(); 905 906 // Thread virtuals 907 908 virtual bool checkForNewParameter_l(const String8& keyValuePair, 909 status_t& status); 910 virtual void flushHw_l(); 911 912protected: 913 virtual int getTrackName_l(audio_channel_mask_t channelMask, 914 audio_format_t format, int sessionId); 915 virtual void deleteTrackName_l(int name); 916 virtual uint32_t activeSleepTimeUs() const; 917 virtual uint32_t idleSleepTimeUs() const; 918 virtual uint32_t suspendSleepTimeUs() const; 919 virtual void cacheParameters_l(); 920 921 // threadLoop snippets 922 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 923 virtual void threadLoop_mix(); 924 virtual void threadLoop_sleepTime(); 925 virtual void threadLoop_exit(); 926 virtual bool shouldStandby_l(); 927 928 // volumes last sent to audio HAL with stream->set_volume() 929 float mLeftVolFloat; 930 float mRightVolFloat; 931 932 DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 933 audio_io_handle_t id, uint32_t device, ThreadBase::type_t type); 934 void processVolume_l(Track *track, bool lastTrack); 935 936 // prepareTracks_l() tells threadLoop_mix() the name of the single active track 937 sp<Track> mActiveTrack; 938public: 939 virtual bool hasFastMixer() const { return false; } 940}; 941 942class OffloadThread : public DirectOutputThread { 943public: 944 945 OffloadThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 946 audio_io_handle_t id, uint32_t device); 947 virtual ~OffloadThread() {}; 948 virtual void flushHw_l(); 949 950protected: 951 // threadLoop snippets 952 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 953 virtual void threadLoop_exit(); 954 955 virtual bool waitingAsyncCallback(); 956 virtual bool waitingAsyncCallback_l(); 957 virtual void onAddNewTrack_l(); 958 959private: 960 size_t mPausedWriteLength; // length in bytes of write interrupted by pause 961 size_t mPausedBytesRemaining; // bytes still waiting in mixbuffer after resume 962 wp<Track> mPreviousTrack; // used to detect track switch 963}; 964 965class AsyncCallbackThread : public Thread { 966public: 967 968 AsyncCallbackThread(const wp<PlaybackThread>& playbackThread); 969 970 virtual ~AsyncCallbackThread(); 971 972 // Thread virtuals 973 virtual bool threadLoop(); 974 975 // RefBase 976 virtual void onFirstRef(); 977 978 void exit(); 979 void setWriteBlocked(uint32_t sequence); 980 void resetWriteBlocked(); 981 void setDraining(uint32_t sequence); 982 void resetDraining(); 983 984private: 985 const wp<PlaybackThread> mPlaybackThread; 986 // mWriteAckSequence corresponds to the last write sequence passed by the offload thread via 987 // setWriteBlocked(). The sequence is shifted one bit to the left and the lsb is used 988 // to indicate that the callback has been received via resetWriteBlocked() 989 uint32_t mWriteAckSequence; 990 // mDrainSequence corresponds to the last drain sequence passed by the offload thread via 991 // setDraining(). The sequence is shifted one bit to the left and the lsb is used 992 // to indicate that the callback has been received via resetDraining() 993 uint32_t mDrainSequence; 994 Condition mWaitWorkCV; 995 Mutex mLock; 996}; 997 998class DuplicatingThread : public MixerThread { 999public: 1000 DuplicatingThread(const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread, 1001 audio_io_handle_t id); 1002 virtual ~DuplicatingThread(); 1003 1004 // Thread virtuals 1005 void addOutputTrack(MixerThread* thread); 1006 void removeOutputTrack(MixerThread* thread); 1007 uint32_t waitTimeMs() const { return mWaitTimeMs; } 1008protected: 1009 virtual uint32_t activeSleepTimeUs() const; 1010 1011private: 1012 bool outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks); 1013protected: 1014 // threadLoop snippets 1015 virtual void threadLoop_mix(); 1016 virtual void threadLoop_sleepTime(); 1017 virtual ssize_t threadLoop_write(); 1018 virtual void threadLoop_standby(); 1019 virtual void cacheParameters_l(); 1020 1021private: 1022 // called from threadLoop, addOutputTrack, removeOutputTrack 1023 virtual void updateWaitTime_l(); 1024protected: 1025 virtual void saveOutputTracks(); 1026 virtual void clearOutputTracks(); 1027private: 1028 1029 uint32_t mWaitTimeMs; 1030 SortedVector < sp<OutputTrack> > outputTracks; 1031 SortedVector < sp<OutputTrack> > mOutputTracks; 1032public: 1033 virtual bool hasFastMixer() const { return false; } 1034}; 1035 1036 1037// record thread 1038class RecordThread : public ThreadBase 1039{ 1040public: 1041 1042 class RecordTrack; 1043 1044 /* The ResamplerBufferProvider is used to retrieve recorded input data from the 1045 * RecordThread. It maintains local state on the relative position of the read 1046 * position of the RecordTrack compared with the RecordThread. 1047 */ 1048 class ResamplerBufferProvider : public AudioBufferProvider 1049 { 1050 public: 1051 ResamplerBufferProvider(RecordTrack* recordTrack) : 1052 mRecordTrack(recordTrack), 1053 mRsmpInUnrel(0), mRsmpInFront(0) { } 1054 virtual ~ResamplerBufferProvider() { } 1055 1056 // called to set the ResamplerBufferProvider to head of the RecordThread data buffer, 1057 // skipping any previous data read from the hal. 1058 virtual void reset(); 1059 1060 /* Synchronizes RecordTrack position with the RecordThread. 1061 * Calculates available frames and handle overruns if the RecordThread 1062 * has advanced faster than the ResamplerBufferProvider has retrieved data. 1063 * TODO: why not do this for every getNextBuffer? 1064 * 1065 * Parameters 1066 * framesAvailable: pointer to optional output size_t to store record track 1067 * frames available. 1068 * hasOverrun: pointer to optional boolean, returns true if track has overrun. 1069 */ 1070 1071 virtual void sync(size_t *framesAvailable = NULL, bool *hasOverrun = NULL); 1072 1073 // AudioBufferProvider interface 1074 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts); 1075 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 1076 private: 1077 RecordTrack * const mRecordTrack; 1078 size_t mRsmpInUnrel; // unreleased frames remaining from 1079 // most recent getNextBuffer 1080 // for debug only 1081 int32_t mRsmpInFront; // next available frame 1082 // rolling counter that is never cleared 1083 }; 1084 1085 /* The RecordBufferConverter is used for format, channel, and sample rate 1086 * conversion for a RecordTrack. 1087 * 1088 * TODO: Self contained, so move to a separate file later. 1089 * 1090 * RecordBufferConverter uses the convert() method rather than exposing a 1091 * buffer provider interface; this is to save a memory copy. 1092 */ 1093 class RecordBufferConverter 1094 { 1095 public: 1096 RecordBufferConverter( 1097 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 1098 uint32_t srcSampleRate, 1099 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 1100 uint32_t dstSampleRate); 1101 1102 ~RecordBufferConverter(); 1103 1104 /* Converts input data from an AudioBufferProvider by format, channelMask, 1105 * and sampleRate to a destination buffer. 1106 * 1107 * Parameters 1108 * dst: buffer to place the converted data. 1109 * provider: buffer provider to obtain source data. 1110 * frames: number of frames to convert 1111 * 1112 * Returns the number of frames converted. 1113 */ 1114 size_t convert(void *dst, AudioBufferProvider *provider, size_t frames); 1115 1116 // returns NO_ERROR if constructor was successful 1117 status_t initCheck() const { 1118 // mSrcChannelMask set on successful updateParameters 1119 return mSrcChannelMask != AUDIO_CHANNEL_INVALID ? NO_ERROR : NO_INIT; 1120 } 1121 1122 // allows dynamic reconfigure of all parameters 1123 status_t updateParameters( 1124 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 1125 uint32_t srcSampleRate, 1126 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 1127 uint32_t dstSampleRate); 1128 1129 // called to reset resampler buffers on record track discontinuity 1130 void reset() { 1131 if (mResampler != NULL) { 1132 mResampler->reset(); 1133 } 1134 } 1135 1136 private: 1137 // format conversion when not using resampler 1138 void convertNoResampler(void *dst, const void *src, size_t frames); 1139 1140 // format conversion when using resampler; modifies src in-place 1141 void convertResampler(void *dst, /*not-a-const*/ void *src, size_t frames); 1142 1143 // user provided information 1144 audio_channel_mask_t mSrcChannelMask; 1145 audio_format_t mSrcFormat; 1146 uint32_t mSrcSampleRate; 1147 audio_channel_mask_t mDstChannelMask; 1148 audio_format_t mDstFormat; 1149 uint32_t mDstSampleRate; 1150 1151 // derived information 1152 uint32_t mSrcChannelCount; 1153 uint32_t mDstChannelCount; 1154 size_t mDstFrameSize; 1155 1156 // format conversion buffer 1157 void *mBuf; 1158 size_t mBufFrames; 1159 size_t mBufFrameSize; 1160 1161 // resampler info 1162 AudioResampler *mResampler; 1163 1164 bool mIsLegacyDownmix; // legacy stereo to mono conversion needed 1165 bool mIsLegacyUpmix; // legacy mono to stereo conversion needed 1166 bool mRequiresFloat; // data processing requires float (e.g. resampler) 1167 PassthruBufferProvider *mInputConverterProvider; // converts input to float 1168 int8_t mIdxAry[sizeof(uint32_t) * 8]; // used for channel mask conversion 1169 }; 1170 1171#include "RecordTracks.h" 1172 1173 RecordThread(const sp<AudioFlinger>& audioFlinger, 1174 AudioStreamIn *input, 1175 audio_io_handle_t id, 1176 audio_devices_t outDevice, 1177 audio_devices_t inDevice 1178#ifdef TEE_SINK 1179 , const sp<NBAIO_Sink>& teeSink 1180#endif 1181 ); 1182 virtual ~RecordThread(); 1183 1184 // no addTrack_l ? 1185 void destroyTrack_l(const sp<RecordTrack>& track); 1186 void removeTrack_l(const sp<RecordTrack>& track); 1187 1188 void dumpInternals(int fd, const Vector<String16>& args); 1189 void dumpTracks(int fd, const Vector<String16>& args); 1190 1191 // Thread virtuals 1192 virtual bool threadLoop(); 1193 1194 // RefBase 1195 virtual void onFirstRef(); 1196 1197 virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; } 1198 1199 virtual sp<MemoryDealer> readOnlyHeap() const { return mReadOnlyHeap; } 1200 1201 virtual sp<IMemory> pipeMemory() const { return mPipeMemory; } 1202 1203 sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l( 1204 const sp<AudioFlinger::Client>& client, 1205 uint32_t sampleRate, 1206 audio_format_t format, 1207 audio_channel_mask_t channelMask, 1208 size_t *pFrameCount, 1209 int sessionId, 1210 size_t *notificationFrames, 1211 int uid, 1212 IAudioFlinger::track_flags_t *flags, 1213 pid_t tid, 1214 status_t *status /*non-NULL*/); 1215 1216 status_t start(RecordTrack* recordTrack, 1217 AudioSystem::sync_event_t event, 1218 int triggerSession); 1219 1220 // ask the thread to stop the specified track, and 1221 // return true if the caller should then do it's part of the stopping process 1222 bool stop(RecordTrack* recordTrack); 1223 1224 void dump(int fd, const Vector<String16>& args); 1225 AudioStreamIn* clearInput(); 1226 virtual audio_stream_t* stream() const; 1227 1228 1229 virtual bool checkForNewParameter_l(const String8& keyValuePair, 1230 status_t& status); 1231 virtual void cacheParameters_l() {} 1232 virtual String8 getParameters(const String8& keys); 1233 virtual void audioConfigChanged(int event, int param = 0); 1234 virtual status_t createAudioPatch_l(const struct audio_patch *patch, 1235 audio_patch_handle_t *handle); 1236 virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); 1237 1238 void addPatchRecord(const sp<PatchRecord>& record); 1239 void deletePatchRecord(const sp<PatchRecord>& record); 1240 1241 void readInputParameters_l(); 1242 virtual uint32_t getInputFramesLost(); 1243 1244 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1245 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1246 virtual uint32_t hasAudioSession(int sessionId) const; 1247 1248 // Return the set of unique session IDs across all tracks. 1249 // The keys are the session IDs, and the associated values are meaningless. 1250 // FIXME replace by Set [and implement Bag/Multiset for other uses]. 1251 KeyedVector<int, bool> sessionIds() const; 1252 1253 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 1254 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 1255 1256 static void syncStartEventCallback(const wp<SyncEvent>& event); 1257 1258 virtual size_t frameCount() const { return mFrameCount; } 1259 bool hasFastCapture() const { return mFastCapture != 0; } 1260 virtual void getAudioPortConfig(struct audio_port_config *config); 1261 1262private: 1263 // Enter standby if not already in standby, and set mStandby flag 1264 void standbyIfNotAlreadyInStandby(); 1265 1266 // Call the HAL standby method unconditionally, and don't change mStandby flag 1267 void inputStandBy(); 1268 1269 AudioStreamIn *mInput; 1270 SortedVector < sp<RecordTrack> > mTracks; 1271 // mActiveTracks has dual roles: it indicates the current active track(s), and 1272 // is used together with mStartStopCond to indicate start()/stop() progress 1273 SortedVector< sp<RecordTrack> > mActiveTracks; 1274 // generation counter for mActiveTracks 1275 int mActiveTracksGen; 1276 Condition mStartStopCond; 1277 1278 // resampler converts input at HAL Hz to output at AudioRecord client Hz 1279 void *mRsmpInBuffer; // 1280 size_t mRsmpInFrames; // size of resampler input in frames 1281 size_t mRsmpInFramesP2;// size rounded up to a power-of-2 1282 1283 // rolling index that is never cleared 1284 int32_t mRsmpInRear; // last filled frame + 1 1285 1286 // For dumpsys 1287 const sp<NBAIO_Sink> mTeeSink; 1288 1289 const sp<MemoryDealer> mReadOnlyHeap; 1290 1291 // one-time initialization, no locks required 1292 sp<FastCapture> mFastCapture; // non-0 if there is also 1293 // a fast capture 1294 // FIXME audio watchdog thread 1295 1296 // contents are not guaranteed to be consistent, no locks required 1297 FastCaptureDumpState mFastCaptureDumpState; 1298#ifdef STATE_QUEUE_DUMP 1299 // FIXME StateQueue observer and mutator dump fields 1300#endif 1301 // FIXME audio watchdog dump 1302 1303 // accessible only within the threadLoop(), no locks required 1304 // mFastCapture->sq() // for mutating and pushing state 1305 int32_t mFastCaptureFutex; // for cold idle 1306 1307 // The HAL input source is treated as non-blocking, 1308 // but current implementation is blocking 1309 sp<NBAIO_Source> mInputSource; 1310 // The source for the normal capture thread to read from: mInputSource or mPipeSource 1311 sp<NBAIO_Source> mNormalSource; 1312 // If a fast capture is present, the non-blocking pipe sink written to by fast capture, 1313 // otherwise clear 1314 sp<NBAIO_Sink> mPipeSink; 1315 // If a fast capture is present, the non-blocking pipe source read by normal thread, 1316 // otherwise clear 1317 sp<NBAIO_Source> mPipeSource; 1318 // Depth of pipe from fast capture to normal thread and fast clients, always power of 2 1319 size_t mPipeFramesP2; 1320 // If a fast capture is present, the Pipe as IMemory, otherwise clear 1321 sp<IMemory> mPipeMemory; 1322 1323 static const size_t kFastCaptureLogSize = 4 * 1024; 1324 sp<NBLog::Writer> mFastCaptureNBLogWriter; 1325 1326 bool mFastTrackAvail; // true if fast track available 1327}; 1328