Threads.h revision 69aed5f0f4a3be3996d1e78a0473e1a72c1547da
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 ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 36 audio_devices_t outDevice, audio_devices_t inDevice, type_t type); 37 virtual ~ThreadBase(); 38 39 virtual status_t readyToRun(); 40 41 void dumpBase(int fd, const Vector<String16>& args); 42 void dumpEffectChains(int fd, const Vector<String16>& args); 43 44 void clearPowerManager(); 45 46 // base for record and playback 47 enum { 48 CFG_EVENT_IO, 49 CFG_EVENT_PRIO 50 }; 51 52 class ConfigEvent { 53 public: 54 ConfigEvent(int type) : mType(type) {} 55 virtual ~ConfigEvent() {} 56 57 int type() const { return mType; } 58 59 virtual void dump(char *buffer, size_t size) = 0; 60 61 private: 62 const int mType; 63 }; 64 65 class IoConfigEvent : public ConfigEvent { 66 public: 67 IoConfigEvent(int event, int param) : 68 ConfigEvent(CFG_EVENT_IO), mEvent(event), mParam(param) {} 69 virtual ~IoConfigEvent() {} 70 71 int event() const { return mEvent; } 72 int param() const { return mParam; } 73 74 virtual void dump(char *buffer, size_t size) { 75 snprintf(buffer, size, "IO event: event %d, param %d\n", mEvent, mParam); 76 } 77 78 private: 79 const int mEvent; 80 const int mParam; 81 }; 82 83 class PrioConfigEvent : public ConfigEvent { 84 public: 85 PrioConfigEvent(pid_t pid, pid_t tid, int32_t prio) : 86 ConfigEvent(CFG_EVENT_PRIO), mPid(pid), mTid(tid), mPrio(prio) {} 87 virtual ~PrioConfigEvent() {} 88 89 pid_t pid() const { return mPid; } 90 pid_t tid() const { return mTid; } 91 int32_t prio() const { return mPrio; } 92 93 virtual void dump(char *buffer, size_t size) { 94 snprintf(buffer, size, "Prio event: pid %d, tid %d, prio %d\n", mPid, mTid, mPrio); 95 } 96 97 private: 98 const pid_t mPid; 99 const pid_t mTid; 100 const int32_t mPrio; 101 }; 102 103 104 class PMDeathRecipient : public IBinder::DeathRecipient { 105 public: 106 PMDeathRecipient(const wp<ThreadBase>& thread) : mThread(thread) {} 107 virtual ~PMDeathRecipient() {} 108 109 // IBinder::DeathRecipient 110 virtual void binderDied(const wp<IBinder>& who); 111 112 private: 113 PMDeathRecipient(const PMDeathRecipient&); 114 PMDeathRecipient& operator = (const PMDeathRecipient&); 115 116 wp<ThreadBase> mThread; 117 }; 118 119 virtual status_t initCheck() const = 0; 120 121 // static externally-visible 122 type_t type() const { return mType; } 123 audio_io_handle_t id() const { return mId;} 124 125 // dynamic externally-visible 126 uint32_t sampleRate() const { return mSampleRate; } 127 uint32_t channelCount() const { return mChannelCount; } 128 audio_channel_mask_t channelMask() const { return mChannelMask; } 129 audio_format_t format() const { return mFormat; } 130 // Called by AudioFlinger::frameCount(audio_io_handle_t output) and effects, 131 // and returns the [normal mix] buffer's frame count. 132 virtual size_t frameCount() const = 0; 133 size_t frameSize() const { return mFrameSize; } 134 135 // Should be "virtual status_t requestExitAndWait()" and override same 136 // method in Thread, but Thread::requestExitAndWait() is not yet virtual. 137 void exit(); 138 virtual bool checkForNewParameters_l() = 0; 139 virtual status_t setParameters(const String8& keyValuePairs); 140 virtual String8 getParameters(const String8& keys) = 0; 141 virtual void audioConfigChanged_l(int event, int param = 0) = 0; 142 void sendIoConfigEvent(int event, int param = 0); 143 void sendIoConfigEvent_l(int event, int param = 0); 144 void sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio); 145 void processConfigEvents(); 146 void processConfigEvents_l(); 147 148 // see note at declaration of mStandby, mOutDevice and mInDevice 149 bool standby() const { return mStandby; } 150 audio_devices_t outDevice() const { return mOutDevice; } 151 audio_devices_t inDevice() const { return mInDevice; } 152 153 virtual audio_stream_t* stream() const = 0; 154 155 sp<EffectHandle> createEffect_l( 156 const sp<AudioFlinger::Client>& client, 157 const sp<IEffectClient>& effectClient, 158 int32_t priority, 159 int sessionId, 160 effect_descriptor_t *desc, 161 int *enabled, 162 status_t *status /*non-NULL*/); 163 void disconnectEffect(const sp< EffectModule>& effect, 164 EffectHandle *handle, 165 bool unpinIfLast); 166 167 // return values for hasAudioSession (bit field) 168 enum effect_state { 169 EFFECT_SESSION = 0x1, // the audio session corresponds to at least one 170 // effect 171 TRACK_SESSION = 0x2 // the audio session corresponds to at least one 172 // track 173 }; 174 175 // get effect chain corresponding to session Id. 176 sp<EffectChain> getEffectChain(int sessionId); 177 // same as getEffectChain() but must be called with ThreadBase mutex locked 178 sp<EffectChain> getEffectChain_l(int sessionId) const; 179 // add an effect chain to the chain list (mEffectChains) 180 virtual status_t addEffectChain_l(const sp<EffectChain>& chain) = 0; 181 // remove an effect chain from the chain list (mEffectChains) 182 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain) = 0; 183 // lock all effect chains Mutexes. Must be called before releasing the 184 // ThreadBase mutex before processing the mixer and effects. This guarantees the 185 // integrity of the chains during the process. 186 // Also sets the parameter 'effectChains' to current value of mEffectChains. 187 void lockEffectChains_l(Vector< sp<EffectChain> >& effectChains); 188 // unlock effect chains after process 189 void unlockEffectChains(const Vector< sp<EffectChain> >& effectChains); 190 // get a copy of mEffectChains vector 191 Vector< sp<EffectChain> > getEffectChains_l() const { return mEffectChains; }; 192 // set audio mode to all effect chains 193 void setMode(audio_mode_t mode); 194 // get effect module with corresponding ID on specified audio session 195 sp<AudioFlinger::EffectModule> getEffect(int sessionId, int effectId); 196 sp<AudioFlinger::EffectModule> getEffect_l(int sessionId, int effectId); 197 // add and effect module. Also creates the effect chain is none exists for 198 // the effects audio session 199 status_t addEffect_l(const sp< EffectModule>& effect); 200 // remove and effect module. Also removes the effect chain is this was the last 201 // effect 202 void removeEffect_l(const sp< EffectModule>& effect); 203 // detach all tracks connected to an auxiliary effect 204 virtual void detachAuxEffect_l(int effectId __unused) {} 205 // returns either EFFECT_SESSION if effects on this audio session exist in one 206 // chain, or TRACK_SESSION if tracks on this audio session exist, or both 207 virtual uint32_t hasAudioSession(int sessionId) const = 0; 208 // the value returned by default implementation is not important as the 209 // strategy is only meaningful for PlaybackThread which implements this method 210 virtual uint32_t getStrategyForSession_l(int sessionId __unused) { return 0; } 211 212 // suspend or restore effect according to the type of effect passed. a NULL 213 // type pointer means suspend all effects in the session 214 void setEffectSuspended(const effect_uuid_t *type, 215 bool suspend, 216 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 217 // check if some effects must be suspended/restored when an effect is enabled 218 // or disabled 219 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 220 bool enabled, 221 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 222 void checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 223 bool enabled, 224 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 225 226 virtual status_t setSyncEvent(const sp<SyncEvent>& event) = 0; 227 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const = 0; 228 229 230 mutable Mutex mLock; 231 232protected: 233 234 // entry describing an effect being suspended in mSuspendedSessions keyed vector 235 class SuspendedSessionDesc : public RefBase { 236 public: 237 SuspendedSessionDesc() : mRefCount(0) {} 238 239 int mRefCount; // number of active suspend requests 240 effect_uuid_t mType; // effect type UUID 241 }; 242 243 void acquireWakeLock(int uid = -1); 244 void acquireWakeLock_l(int uid = -1); 245 void releaseWakeLock(); 246 void releaseWakeLock_l(); 247 void updateWakeLockUids(const SortedVector<int> &uids); 248 void updateWakeLockUids_l(const SortedVector<int> &uids); 249 void getPowerManager_l(); 250 void setEffectSuspended_l(const effect_uuid_t *type, 251 bool suspend, 252 int sessionId); 253 // updated mSuspendedSessions when an effect suspended or restored 254 void updateSuspendedSessions_l(const effect_uuid_t *type, 255 bool suspend, 256 int sessionId); 257 // check if some effects must be suspended when an effect chain is added 258 void checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain); 259 260 String16 getWakeLockTag(); 261 262 virtual void preExit() { } 263 264 friend class AudioFlinger; // for mEffectChains 265 266 const type_t mType; 267 268 // Used by parameters, config events, addTrack_l, exit 269 Condition mWaitWorkCV; 270 271 const sp<AudioFlinger> mAudioFlinger; 272 273 // updated by PlaybackThread::readOutputParameters_l() or 274 // RecordThread::readInputParameters_l() 275 uint32_t mSampleRate; 276 size_t mFrameCount; // output HAL, direct output, record 277 audio_channel_mask_t mChannelMask; 278 uint32_t mChannelCount; 279 size_t mFrameSize; 280 audio_format_t mFormat; 281 size_t mBufferSize; // HAL buffer size for read() or write() 282 283 // Parameter sequence by client: binder thread calling setParameters(): 284 // 1. Lock mLock 285 // 2. Append to mNewParameters 286 // 3. mWaitWorkCV.signal 287 // 4. mParamCond.waitRelative with timeout 288 // 5. read mParamStatus 289 // 6. mWaitWorkCV.signal 290 // 7. Unlock 291 // 292 // Parameter sequence by server: threadLoop calling checkForNewParameters_l(): 293 // 1. Lock mLock 294 // 2. If there is an entry in mNewParameters proceed ... 295 // 2. Read first entry in mNewParameters 296 // 3. Process 297 // 4. Remove first entry from mNewParameters 298 // 5. Set mParamStatus 299 // 6. mParamCond.signal 300 // 7. mWaitWorkCV.wait with timeout (this is to avoid overwriting mParamStatus) 301 // 8. Unlock 302 Condition mParamCond; 303 Vector<String8> mNewParameters; 304 status_t mParamStatus; 305 306 // vector owns each ConfigEvent *, so must delete after removing 307 Vector<ConfigEvent *> mConfigEvents; 308 309 // These fields are written and read by thread itself without lock or barrier, 310 // and read by other threads without lock or barrier via standby(), outDevice() 311 // and inDevice(). 312 // Because of the absence of a lock or barrier, any other thread that reads 313 // these fields must use the information in isolation, or be prepared to deal 314 // with possibility that it might be inconsistent with other information. 315 bool mStandby; // Whether thread is currently in standby. 316 audio_devices_t mOutDevice; // output device 317 audio_devices_t mInDevice; // input device 318 audio_source_t mAudioSource; // (see audio.h, audio_source_t) 319 320 const audio_io_handle_t mId; 321 Vector< sp<EffectChain> > mEffectChains; 322 323 static const int kNameLength = 16; // prctl(PR_SET_NAME) limit 324 char mName[kNameLength]; 325 sp<IPowerManager> mPowerManager; 326 sp<IBinder> mWakeLockToken; 327 const sp<PMDeathRecipient> mDeathRecipient; 328 // list of suspended effects per session and per type. The first vector is 329 // keyed by session ID, the second by type UUID timeLow field 330 KeyedVector< int, KeyedVector< int, sp<SuspendedSessionDesc> > > 331 mSuspendedSessions; 332 static const size_t kLogSize = 4 * 1024; 333 sp<NBLog::Writer> mNBLogWriter; 334}; 335 336// --- PlaybackThread --- 337class PlaybackThread : public ThreadBase { 338public: 339 340#include "PlaybackTracks.h" 341 342 enum mixer_state { 343 MIXER_IDLE, // no active tracks 344 MIXER_TRACKS_ENABLED, // at least one active track, but no track has any data ready 345 MIXER_TRACKS_READY, // at least one active track, and at least one track has data 346 MIXER_DRAIN_TRACK, // drain currently playing track 347 MIXER_DRAIN_ALL, // fully drain the hardware 348 // standby mode does not have an enum value 349 // suspend by audio policy manager is orthogonal to mixer state 350 }; 351 352 // retry count before removing active track in case of underrun on offloaded thread: 353 // we need to make sure that AudioTrack client has enough time to send large buffers 354//FIXME may be more appropriate if expressed in time units. Need to revise how underrun is handled 355 // for offloaded tracks 356 static const int8_t kMaxTrackRetriesOffload = 20; 357 358 PlaybackThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 359 audio_io_handle_t id, audio_devices_t device, type_t type); 360 virtual ~PlaybackThread(); 361 362 void dump(int fd, const Vector<String16>& args); 363 364 // Thread virtuals 365 virtual bool threadLoop(); 366 367 // RefBase 368 virtual void onFirstRef(); 369 370protected: 371 // Code snippets that were lifted up out of threadLoop() 372 virtual void threadLoop_mix() = 0; 373 virtual void threadLoop_sleepTime() = 0; 374 virtual ssize_t threadLoop_write(); 375 virtual void threadLoop_drain(); 376 virtual void threadLoop_standby(); 377 virtual void threadLoop_exit(); 378 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 379 380 // prepareTracks_l reads and writes mActiveTracks, and returns 381 // the pending set of tracks to remove via Vector 'tracksToRemove'. The caller 382 // is responsible for clearing or destroying this Vector later on, when it 383 // is safe to do so. That will drop the final ref count and destroy the tracks. 384 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove) = 0; 385 void removeTracks_l(const Vector< sp<Track> >& tracksToRemove); 386 387 void writeCallback(); 388 void resetWriteBlocked(uint32_t sequence); 389 void drainCallback(); 390 void resetDraining(uint32_t sequence); 391 392 static int asyncCallback(stream_callback_event_t event, void *param, void *cookie); 393 394 virtual bool waitingAsyncCallback(); 395 virtual bool waitingAsyncCallback_l(); 396 virtual bool shouldStandby_l(); 397 virtual void onAddNewTrack_l(); 398 399 // ThreadBase virtuals 400 virtual void preExit(); 401 402public: 403 404 virtual status_t initCheck() const { return (mOutput == NULL) ? NO_INIT : NO_ERROR; } 405 406 // return estimated latency in milliseconds, as reported by HAL 407 uint32_t latency() const; 408 // same, but lock must already be held 409 uint32_t latency_l() const; 410 411 void setMasterVolume(float value); 412 void setMasterMute(bool muted); 413 414 void setStreamVolume(audio_stream_type_t stream, float value); 415 void setStreamMute(audio_stream_type_t stream, bool muted); 416 417 float streamVolume(audio_stream_type_t stream) const; 418 419 sp<Track> createTrack_l( 420 const sp<AudioFlinger::Client>& client, 421 audio_stream_type_t streamType, 422 uint32_t sampleRate, 423 audio_format_t format, 424 audio_channel_mask_t channelMask, 425 size_t *pFrameCount, 426 const sp<IMemory>& sharedBuffer, 427 int sessionId, 428 IAudioFlinger::track_flags_t *flags, 429 pid_t tid, 430 int uid, 431 status_t *status /*non-NULL*/); 432 433 AudioStreamOut* getOutput() const; 434 AudioStreamOut* clearOutput(); 435 virtual audio_stream_t* stream() const; 436 437 // a very large number of suspend() will eventually wraparound, but unlikely 438 void suspend() { (void) android_atomic_inc(&mSuspended); } 439 void restore() 440 { 441 // if restore() is done without suspend(), get back into 442 // range so that the next suspend() will operate correctly 443 if (android_atomic_dec(&mSuspended) <= 0) { 444 android_atomic_release_store(0, &mSuspended); 445 } 446 } 447 bool isSuspended() const 448 { return android_atomic_acquire_load(&mSuspended) > 0; } 449 450 virtual String8 getParameters(const String8& keys); 451 virtual void audioConfigChanged_l(int event, int param = 0); 452 status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames); 453 // TODO: rename mixBuffer() to sinkBuffer() or try to remove external use. 454 int16_t *mixBuffer() const { return mSinkBuffer; }; 455 456 virtual void detachAuxEffect_l(int effectId); 457 status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track> track, 458 int EffectId); 459 status_t attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track> track, 460 int EffectId); 461 462 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 463 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 464 virtual uint32_t hasAudioSession(int sessionId) const; 465 virtual uint32_t getStrategyForSession_l(int sessionId); 466 467 468 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 469 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 470 471 // called with AudioFlinger lock held 472 void invalidateTracks(audio_stream_type_t streamType); 473 474 virtual size_t frameCount() const { return mNormalFrameCount; } 475 476 // Return's the HAL's frame count i.e. fast mixer buffer size. 477 size_t frameCountHAL() const { return mFrameCount; } 478 479 status_t getTimestamp_l(AudioTimestamp& timestamp); 480 481protected: 482 // updated by readOutputParameters_l() 483 size_t mNormalFrameCount; // normal mixer and effects 484 485 int16_t* mSinkBuffer; // frame size aligned sink buffer 486 487 // Mixer Buffer (mMixerBuffer*) 488 // 489 // In the case of floating point or multichannel data, which is not in the 490 // sink format, it is required to accumulate in a higher precision or greater channel count 491 // buffer before downmixing or data conversion to the sink buffer. 492 493 // Set to "true" to enable the Mixer Buffer otherwise mixer output goes to sink buffer. 494 bool mMixerBufferEnabled; 495 496 // Storage, 32 byte aligned (may make this alignment a requirement later). 497 // Due to constraints on mNormalFrameCount, the buffer size is a multiple of 16 frames. 498 void* mMixerBuffer; 499 500 // Size of mMixerBuffer in bytes: mNormalFrameCount * #channels * sampsize. 501 size_t mMixerBufferSize; 502 503 // The audio format of mMixerBuffer. Set to AUDIO_FORMAT_PCM_(FLOAT|16_BIT) only. 504 audio_format_t mMixerBufferFormat; 505 506 // An internal flag set to true by MixerThread::prepareTracks_l() 507 // when mMixerBuffer contains valid data after mixing. 508 bool mMixerBufferValid; 509 510 // suspend count, > 0 means suspended. While suspended, the thread continues to pull from 511 // tracks and mix, but doesn't write to HAL. A2DP and SCO HAL implementations can't handle 512 // concurrent use of both of them, so Audio Policy Service suspends one of the threads to 513 // workaround that restriction. 514 // 'volatile' means accessed via atomic operations and no lock. 515 volatile int32_t mSuspended; 516 517 // FIXME overflows every 6+ hours at 44.1 kHz stereo 16-bit samples 518 // mFramesWritten would be better, or 64-bit even better 519 size_t mBytesWritten; 520private: 521 // mMasterMute is in both PlaybackThread and in AudioFlinger. When a 522 // PlaybackThread needs to find out if master-muted, it checks it's local 523 // copy rather than the one in AudioFlinger. This optimization saves a lock. 524 bool mMasterMute; 525 void setMasterMute_l(bool muted) { mMasterMute = muted; } 526protected: 527 SortedVector< wp<Track> > mActiveTracks; // FIXME check if this could be sp<> 528 SortedVector<int> mWakeLockUids; 529 int mActiveTracksGeneration; 530 wp<Track> mLatestActiveTrack; // latest track added to mActiveTracks 531 532 // Allocate a track name for a given channel mask. 533 // Returns name >= 0 if successful, -1 on failure. 534 virtual int getTrackName_l(audio_channel_mask_t channelMask, int sessionId) = 0; 535 virtual void deleteTrackName_l(int name) = 0; 536 537 // Time to sleep between cycles when: 538 virtual uint32_t activeSleepTimeUs() const; // mixer state MIXER_TRACKS_ENABLED 539 virtual uint32_t idleSleepTimeUs() const = 0; // mixer state MIXER_IDLE 540 virtual uint32_t suspendSleepTimeUs() const = 0; // audio policy manager suspended us 541 // No sleep when mixer state == MIXER_TRACKS_READY; relies on audio HAL stream->write() 542 // No sleep in standby mode; waits on a condition 543 544 // Code snippets that are temporarily lifted up out of threadLoop() until the merge 545 void checkSilentMode_l(); 546 547 // Non-trivial for DUPLICATING only 548 virtual void saveOutputTracks() { } 549 virtual void clearOutputTracks() { } 550 551 // Cache various calculated values, at threadLoop() entry and after a parameter change 552 virtual void cacheParameters_l(); 553 554 virtual uint32_t correctLatency_l(uint32_t latency) const; 555 556private: 557 558 friend class AudioFlinger; // for numerous 559 560 PlaybackThread(const Client&); 561 PlaybackThread& operator = (const PlaybackThread&); 562 563 status_t addTrack_l(const sp<Track>& track); 564 bool destroyTrack_l(const sp<Track>& track); 565 void removeTrack_l(const sp<Track>& track); 566 void broadcast_l(); 567 568 void readOutputParameters_l(); 569 570 virtual void dumpInternals(int fd, const Vector<String16>& args); 571 void dumpTracks(int fd, const Vector<String16>& args); 572 573 SortedVector< sp<Track> > mTracks; 574 // mStreamTypes[] uses 1 additional stream type internally for the OutputTrack used by 575 // DuplicatingThread 576 stream_type_t mStreamTypes[AUDIO_STREAM_CNT + 1]; 577 AudioStreamOut *mOutput; 578 579 float mMasterVolume; 580 nsecs_t mLastWriteTime; 581 int mNumWrites; 582 int mNumDelayedWrites; 583 bool mInWrite; 584 585 // FIXME rename these former local variables of threadLoop to standard "m" names 586 nsecs_t standbyTime; 587 size_t mSinkBufferSize; 588 589 // cached copies of activeSleepTimeUs() and idleSleepTimeUs() made by cacheParameters_l() 590 uint32_t activeSleepTime; 591 uint32_t idleSleepTime; 592 593 uint32_t sleepTime; 594 595 // mixer status returned by prepareTracks_l() 596 mixer_state mMixerStatus; // current cycle 597 // previous cycle when in prepareTracks_l() 598 mixer_state mMixerStatusIgnoringFastTracks; 599 // FIXME or a separate ready state per track 600 601 // FIXME move these declarations into the specific sub-class that needs them 602 // MIXER only 603 uint32_t sleepTimeShift; 604 605 // same as AudioFlinger::mStandbyTimeInNsecs except for DIRECT which uses a shorter value 606 nsecs_t standbyDelay; 607 608 // MIXER only 609 nsecs_t maxPeriod; 610 611 // DUPLICATING only 612 uint32_t writeFrames; 613 614 size_t mBytesRemaining; 615 size_t mCurrentWriteLength; 616 bool mUseAsyncWrite; 617 // mWriteAckSequence contains current write sequence on bits 31-1. The write sequence is 618 // incremented each time a write(), a flush() or a standby() occurs. 619 // Bit 0 is set when a write blocks and indicates a callback is expected. 620 // Bit 0 is reset by the async callback thread calling resetWriteBlocked(). Out of sequence 621 // callbacks are ignored. 622 uint32_t mWriteAckSequence; 623 // mDrainSequence contains current drain sequence on bits 31-1. The drain sequence is 624 // incremented each time a drain is requested or a flush() or standby() occurs. 625 // Bit 0 is set when the drain() command is called at the HAL and indicates a callback is 626 // expected. 627 // Bit 0 is reset by the async callback thread calling resetDraining(). Out of sequence 628 // callbacks are ignored. 629 uint32_t mDrainSequence; 630 // A condition that must be evaluated by prepareTrack_l() has changed and we must not wait 631 // for async write callback in the thread loop before evaluating it 632 bool mSignalPending; 633 sp<AsyncCallbackThread> mCallbackThread; 634 635private: 636 // The HAL output sink is treated as non-blocking, but current implementation is blocking 637 sp<NBAIO_Sink> mOutputSink; 638 // If a fast mixer is present, the blocking pipe sink, otherwise clear 639 sp<NBAIO_Sink> mPipeSink; 640 // The current sink for the normal mixer to write it's (sub)mix, mOutputSink or mPipeSink 641 sp<NBAIO_Sink> mNormalSink; 642#ifdef TEE_SINK 643 // For dumpsys 644 sp<NBAIO_Sink> mTeeSink; 645 sp<NBAIO_Source> mTeeSource; 646#endif 647 uint32_t mScreenState; // cached copy of gScreenState 648 static const size_t kFastMixerLogSize = 4 * 1024; 649 sp<NBLog::Writer> mFastMixerNBLogWriter; 650public: 651 virtual bool hasFastMixer() const = 0; 652 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex __unused) const 653 { FastTrackUnderruns dummy; return dummy; } 654 655protected: 656 // accessed by both binder threads and within threadLoop(), lock on mutex needed 657 unsigned mFastTrackAvailMask; // bit i set if fast track [i] is available 658 659private: 660 // timestamp latch: 661 // D input is written by threadLoop_write while mutex is unlocked, and read while locked 662 // Q output is written while locked, and read while locked 663 struct { 664 AudioTimestamp mTimestamp; 665 uint32_t mUnpresentedFrames; 666 } mLatchD, mLatchQ; 667 bool mLatchDValid; // true means mLatchD is valid, and clock it into latch at next opportunity 668 bool mLatchQValid; // true means mLatchQ is valid 669}; 670 671class MixerThread : public PlaybackThread { 672public: 673 MixerThread(const sp<AudioFlinger>& audioFlinger, 674 AudioStreamOut* output, 675 audio_io_handle_t id, 676 audio_devices_t device, 677 type_t type = MIXER); 678 virtual ~MixerThread(); 679 680 // Thread virtuals 681 682 virtual bool checkForNewParameters_l(); 683 virtual void dumpInternals(int fd, const Vector<String16>& args); 684 685protected: 686 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 687 virtual int getTrackName_l(audio_channel_mask_t channelMask, int sessionId); 688 virtual void deleteTrackName_l(int name); 689 virtual uint32_t idleSleepTimeUs() const; 690 virtual uint32_t suspendSleepTimeUs() const; 691 virtual void cacheParameters_l(); 692 693 // threadLoop snippets 694 virtual ssize_t threadLoop_write(); 695 virtual void threadLoop_standby(); 696 virtual void threadLoop_mix(); 697 virtual void threadLoop_sleepTime(); 698 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 699 virtual uint32_t correctLatency_l(uint32_t latency) const; 700 701 AudioMixer* mAudioMixer; // normal mixer 702private: 703 // one-time initialization, no locks required 704 FastMixer* mFastMixer; // non-NULL if there is also a fast mixer 705 sp<AudioWatchdog> mAudioWatchdog; // non-0 if there is an audio watchdog thread 706 707 // contents are not guaranteed to be consistent, no locks required 708 FastMixerDumpState mFastMixerDumpState; 709#ifdef STATE_QUEUE_DUMP 710 StateQueueObserverDump mStateQueueObserverDump; 711 StateQueueMutatorDump mStateQueueMutatorDump; 712#endif 713 AudioWatchdogDump mAudioWatchdogDump; 714 715 // accessible only within the threadLoop(), no locks required 716 // mFastMixer->sq() // for mutating and pushing state 717 int32_t mFastMixerFutex; // for cold idle 718 719public: 720 virtual bool hasFastMixer() const { return mFastMixer != NULL; } 721 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const { 722 ALOG_ASSERT(fastIndex < FastMixerState::kMaxFastTracks); 723 return mFastMixerDumpState.mTracks[fastIndex].mUnderruns; 724 } 725}; 726 727class DirectOutputThread : public PlaybackThread { 728public: 729 730 DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 731 audio_io_handle_t id, audio_devices_t device); 732 virtual ~DirectOutputThread(); 733 734 // Thread virtuals 735 736 virtual bool checkForNewParameters_l(); 737 738protected: 739 virtual int getTrackName_l(audio_channel_mask_t channelMask, int sessionId); 740 virtual void deleteTrackName_l(int name); 741 virtual uint32_t activeSleepTimeUs() const; 742 virtual uint32_t idleSleepTimeUs() const; 743 virtual uint32_t suspendSleepTimeUs() const; 744 virtual void cacheParameters_l(); 745 746 // threadLoop snippets 747 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 748 virtual void threadLoop_mix(); 749 virtual void threadLoop_sleepTime(); 750 751 // volumes last sent to audio HAL with stream->set_volume() 752 float mLeftVolFloat; 753 float mRightVolFloat; 754 755 DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 756 audio_io_handle_t id, uint32_t device, ThreadBase::type_t type); 757 void processVolume_l(Track *track, bool lastTrack); 758 759 // prepareTracks_l() tells threadLoop_mix() the name of the single active track 760 sp<Track> mActiveTrack; 761public: 762 virtual bool hasFastMixer() const { return false; } 763}; 764 765class OffloadThread : public DirectOutputThread { 766public: 767 768 OffloadThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 769 audio_io_handle_t id, uint32_t device); 770 virtual ~OffloadThread() {}; 771 772protected: 773 // threadLoop snippets 774 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 775 virtual void threadLoop_exit(); 776 777 virtual bool waitingAsyncCallback(); 778 virtual bool waitingAsyncCallback_l(); 779 virtual bool shouldStandby_l(); 780 virtual void onAddNewTrack_l(); 781 782private: 783 void flushHw_l(); 784 785private: 786 bool mHwPaused; 787 bool mFlushPending; 788 size_t mPausedWriteLength; // length in bytes of write interrupted by pause 789 size_t mPausedBytesRemaining; // bytes still waiting in mixbuffer after resume 790 wp<Track> mPreviousTrack; // used to detect track switch 791}; 792 793class AsyncCallbackThread : public Thread { 794public: 795 796 AsyncCallbackThread(const wp<PlaybackThread>& playbackThread); 797 798 virtual ~AsyncCallbackThread(); 799 800 // Thread virtuals 801 virtual bool threadLoop(); 802 803 // RefBase 804 virtual void onFirstRef(); 805 806 void exit(); 807 void setWriteBlocked(uint32_t sequence); 808 void resetWriteBlocked(); 809 void setDraining(uint32_t sequence); 810 void resetDraining(); 811 812private: 813 const wp<PlaybackThread> mPlaybackThread; 814 // mWriteAckSequence corresponds to the last write sequence passed by the offload thread via 815 // setWriteBlocked(). The sequence is shifted one bit to the left and the lsb is used 816 // to indicate that the callback has been received via resetWriteBlocked() 817 uint32_t mWriteAckSequence; 818 // mDrainSequence corresponds to the last drain sequence passed by the offload thread via 819 // setDraining(). The sequence is shifted one bit to the left and the lsb is used 820 // to indicate that the callback has been received via resetDraining() 821 uint32_t mDrainSequence; 822 Condition mWaitWorkCV; 823 Mutex mLock; 824}; 825 826class DuplicatingThread : public MixerThread { 827public: 828 DuplicatingThread(const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread, 829 audio_io_handle_t id); 830 virtual ~DuplicatingThread(); 831 832 // Thread virtuals 833 void addOutputTrack(MixerThread* thread); 834 void removeOutputTrack(MixerThread* thread); 835 uint32_t waitTimeMs() const { return mWaitTimeMs; } 836protected: 837 virtual uint32_t activeSleepTimeUs() const; 838 839private: 840 bool outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks); 841protected: 842 // threadLoop snippets 843 virtual void threadLoop_mix(); 844 virtual void threadLoop_sleepTime(); 845 virtual ssize_t threadLoop_write(); 846 virtual void threadLoop_standby(); 847 virtual void cacheParameters_l(); 848 849private: 850 // called from threadLoop, addOutputTrack, removeOutputTrack 851 virtual void updateWaitTime_l(); 852protected: 853 virtual void saveOutputTracks(); 854 virtual void clearOutputTracks(); 855private: 856 857 uint32_t mWaitTimeMs; 858 SortedVector < sp<OutputTrack> > outputTracks; 859 SortedVector < sp<OutputTrack> > mOutputTracks; 860public: 861 virtual bool hasFastMixer() const { return false; } 862}; 863 864 865// record thread 866class RecordThread : public ThreadBase 867{ 868public: 869 870 class RecordTrack; 871 class ResamplerBufferProvider : public AudioBufferProvider 872 // derives from AudioBufferProvider interface for use by resampler 873 { 874 public: 875 ResamplerBufferProvider(RecordTrack* recordTrack) : mRecordTrack(recordTrack) { } 876 virtual ~ResamplerBufferProvider() { } 877 // AudioBufferProvider interface 878 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts); 879 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 880 private: 881 RecordTrack * const mRecordTrack; 882 }; 883 884#include "RecordTracks.h" 885 886 RecordThread(const sp<AudioFlinger>& audioFlinger, 887 AudioStreamIn *input, 888 audio_io_handle_t id, 889 audio_devices_t outDevice, 890 audio_devices_t inDevice 891#ifdef TEE_SINK 892 , const sp<NBAIO_Sink>& teeSink 893#endif 894 ); 895 virtual ~RecordThread(); 896 897 // no addTrack_l ? 898 void destroyTrack_l(const sp<RecordTrack>& track); 899 void removeTrack_l(const sp<RecordTrack>& track); 900 901 void dumpInternals(int fd, const Vector<String16>& args); 902 void dumpTracks(int fd, const Vector<String16>& args); 903 904 // Thread virtuals 905 virtual bool threadLoop(); 906 907 // RefBase 908 virtual void onFirstRef(); 909 910 virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; } 911 912 sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l( 913 const sp<AudioFlinger::Client>& client, 914 uint32_t sampleRate, 915 audio_format_t format, 916 audio_channel_mask_t channelMask, 917 size_t *pFrameCount, 918 int sessionId, 919 int uid, 920 IAudioFlinger::track_flags_t *flags, 921 pid_t tid, 922 status_t *status /*non-NULL*/); 923 924 status_t start(RecordTrack* recordTrack, 925 AudioSystem::sync_event_t event, 926 int triggerSession); 927 928 // ask the thread to stop the specified track, and 929 // return true if the caller should then do it's part of the stopping process 930 bool stop(RecordTrack* recordTrack); 931 932 void dump(int fd, const Vector<String16>& args); 933 AudioStreamIn* clearInput(); 934 virtual audio_stream_t* stream() const; 935 936 937 virtual bool checkForNewParameters_l(); 938 virtual String8 getParameters(const String8& keys); 939 virtual void audioConfigChanged_l(int event, int param = 0); 940 void readInputParameters_l(); 941 virtual uint32_t getInputFramesLost(); 942 943 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 944 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 945 virtual uint32_t hasAudioSession(int sessionId) const; 946 947 // Return the set of unique session IDs across all tracks. 948 // The keys are the session IDs, and the associated values are meaningless. 949 // FIXME replace by Set [and implement Bag/Multiset for other uses]. 950 KeyedVector<int, bool> sessionIds() const; 951 952 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 953 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 954 955 static void syncStartEventCallback(const wp<SyncEvent>& event); 956 957 virtual size_t frameCount() const { return mFrameCount; } 958 bool hasFastRecorder() const { return false; } 959 960private: 961 // Enter standby if not already in standby, and set mStandby flag 962 void standbyIfNotAlreadyInStandby(); 963 964 // Call the HAL standby method unconditionally, and don't change mStandby flag 965 void inputStandBy(); 966 967 AudioStreamIn *mInput; 968 SortedVector < sp<RecordTrack> > mTracks; 969 // mActiveTracks has dual roles: it indicates the current active track(s), and 970 // is used together with mStartStopCond to indicate start()/stop() progress 971 SortedVector< sp<RecordTrack> > mActiveTracks; 972 // generation counter for mActiveTracks 973 int mActiveTracksGen; 974 Condition mStartStopCond; 975 976 // resampler converts input at HAL Hz to output at AudioRecord client Hz 977 int16_t *mRsmpInBuffer; // see new[] for details on the size 978 size_t mRsmpInFrames; // size of resampler input in frames 979 size_t mRsmpInFramesP2;// size rounded up to a power-of-2 980 981 // rolling index that is never cleared 982 int32_t mRsmpInRear; // last filled frame + 1 983 984 // For dumpsys 985 const sp<NBAIO_Sink> mTeeSink; 986}; 987