Threads.h revision e7e676fd2866fa4898712c4effa9e624e969c182
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 // FIXME rename mixBuffer() to sinkBuffer() and remove int16_t* dependency. 454 // Consider also removing and passing an explicit mMainBuffer initialization 455 // parameter to AF::PlaybackThread::Track::Track(). 456 int16_t *mixBuffer() const { 457 return reinterpret_cast<int16_t *>(mSinkBuffer); }; 458 459 virtual void detachAuxEffect_l(int effectId); 460 status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track> track, 461 int EffectId); 462 status_t attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track> track, 463 int EffectId); 464 465 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 466 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 467 virtual uint32_t hasAudioSession(int sessionId) const; 468 virtual uint32_t getStrategyForSession_l(int sessionId); 469 470 471 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 472 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 473 474 // called with AudioFlinger lock held 475 void invalidateTracks(audio_stream_type_t streamType); 476 477 virtual size_t frameCount() const { return mNormalFrameCount; } 478 479 // Return's the HAL's frame count i.e. fast mixer buffer size. 480 size_t frameCountHAL() const { return mFrameCount; } 481 482 status_t getTimestamp_l(AudioTimestamp& timestamp); 483 484protected: 485 // updated by readOutputParameters_l() 486 size_t mNormalFrameCount; // normal mixer and effects 487 488 void* mSinkBuffer; // frame size aligned sink buffer 489 490 // TODO: 491 // Rearrange the buffer info into a struct/class with 492 // clear, copy, construction, destruction methods. 493 // 494 // mSinkBuffer also has associated with it: 495 // 496 // mSinkBufferSize: Sink Buffer Size 497 // mFormat: Sink Buffer Format 498 499 // Mixer Buffer (mMixerBuffer*) 500 // 501 // In the case of floating point or multichannel data, which is not in the 502 // sink format, it is required to accumulate in a higher precision or greater channel count 503 // buffer before downmixing or data conversion to the sink buffer. 504 505 // Set to "true" to enable the Mixer Buffer otherwise mixer output goes to sink buffer. 506 bool mMixerBufferEnabled; 507 508 // Storage, 32 byte aligned (may make this alignment a requirement later). 509 // Due to constraints on mNormalFrameCount, the buffer size is a multiple of 16 frames. 510 void* mMixerBuffer; 511 512 // Size of mMixerBuffer in bytes: mNormalFrameCount * #channels * sampsize. 513 size_t mMixerBufferSize; 514 515 // The audio format of mMixerBuffer. Set to AUDIO_FORMAT_PCM_(FLOAT|16_BIT) only. 516 audio_format_t mMixerBufferFormat; 517 518 // An internal flag set to true by MixerThread::prepareTracks_l() 519 // when mMixerBuffer contains valid data after mixing. 520 bool mMixerBufferValid; 521 522 // Effects Buffer (mEffectsBuffer*) 523 // 524 // In the case of effects data, which is not in the sink format, 525 // it is required to accumulate in a different buffer before data conversion 526 // to the sink buffer. 527 528 // Set to "true" to enable the Effects Buffer otherwise effects output goes to sink buffer. 529 bool mEffectBufferEnabled; 530 531 // Storage, 32 byte aligned (may make this alignment a requirement later). 532 // Due to constraints on mNormalFrameCount, the buffer size is a multiple of 16 frames. 533 void* mEffectBuffer; 534 535 // Size of mEffectsBuffer in bytes: mNormalFrameCount * #channels * sampsize. 536 size_t mEffectBufferSize; 537 538 // The audio format of mEffectsBuffer. Set to AUDIO_FORMAT_PCM_16_BIT only. 539 audio_format_t mEffectBufferFormat; 540 541 // An internal flag set to true by MixerThread::prepareTracks_l() 542 // when mEffectsBuffer contains valid data after mixing. 543 // 544 // When this is set, all mixer data is routed into the effects buffer 545 // for any processing (including output processing). 546 bool mEffectBufferValid; 547 548 // suspend count, > 0 means suspended. While suspended, the thread continues to pull from 549 // tracks and mix, but doesn't write to HAL. A2DP and SCO HAL implementations can't handle 550 // concurrent use of both of them, so Audio Policy Service suspends one of the threads to 551 // workaround that restriction. 552 // 'volatile' means accessed via atomic operations and no lock. 553 volatile int32_t mSuspended; 554 555 // FIXME overflows every 6+ hours at 44.1 kHz stereo 16-bit samples 556 // mFramesWritten would be better, or 64-bit even better 557 size_t mBytesWritten; 558private: 559 // mMasterMute is in both PlaybackThread and in AudioFlinger. When a 560 // PlaybackThread needs to find out if master-muted, it checks it's local 561 // copy rather than the one in AudioFlinger. This optimization saves a lock. 562 bool mMasterMute; 563 void setMasterMute_l(bool muted) { mMasterMute = muted; } 564protected: 565 SortedVector< wp<Track> > mActiveTracks; // FIXME check if this could be sp<> 566 SortedVector<int> mWakeLockUids; 567 int mActiveTracksGeneration; 568 wp<Track> mLatestActiveTrack; // latest track added to mActiveTracks 569 570 // Allocate a track name for a given channel mask. 571 // Returns name >= 0 if successful, -1 on failure. 572 virtual int getTrackName_l(audio_channel_mask_t channelMask, int sessionId) = 0; 573 virtual void deleteTrackName_l(int name) = 0; 574 575 // Time to sleep between cycles when: 576 virtual uint32_t activeSleepTimeUs() const; // mixer state MIXER_TRACKS_ENABLED 577 virtual uint32_t idleSleepTimeUs() const = 0; // mixer state MIXER_IDLE 578 virtual uint32_t suspendSleepTimeUs() const = 0; // audio policy manager suspended us 579 // No sleep when mixer state == MIXER_TRACKS_READY; relies on audio HAL stream->write() 580 // No sleep in standby mode; waits on a condition 581 582 // Code snippets that are temporarily lifted up out of threadLoop() until the merge 583 void checkSilentMode_l(); 584 585 // Non-trivial for DUPLICATING only 586 virtual void saveOutputTracks() { } 587 virtual void clearOutputTracks() { } 588 589 // Cache various calculated values, at threadLoop() entry and after a parameter change 590 virtual void cacheParameters_l(); 591 592 virtual uint32_t correctLatency_l(uint32_t latency) const; 593 594private: 595 596 friend class AudioFlinger; // for numerous 597 598 PlaybackThread(const Client&); 599 PlaybackThread& operator = (const PlaybackThread&); 600 601 status_t addTrack_l(const sp<Track>& track); 602 bool destroyTrack_l(const sp<Track>& track); 603 void removeTrack_l(const sp<Track>& track); 604 void broadcast_l(); 605 606 void readOutputParameters_l(); 607 608 virtual void dumpInternals(int fd, const Vector<String16>& args); 609 void dumpTracks(int fd, const Vector<String16>& args); 610 611 SortedVector< sp<Track> > mTracks; 612 // mStreamTypes[] uses 1 additional stream type internally for the OutputTrack used by 613 // DuplicatingThread 614 stream_type_t mStreamTypes[AUDIO_STREAM_CNT + 1]; 615 AudioStreamOut *mOutput; 616 617 float mMasterVolume; 618 nsecs_t mLastWriteTime; 619 int mNumWrites; 620 int mNumDelayedWrites; 621 bool mInWrite; 622 623 // FIXME rename these former local variables of threadLoop to standard "m" names 624 nsecs_t standbyTime; 625 size_t mSinkBufferSize; 626 627 // cached copies of activeSleepTimeUs() and idleSleepTimeUs() made by cacheParameters_l() 628 uint32_t activeSleepTime; 629 uint32_t idleSleepTime; 630 631 uint32_t sleepTime; 632 633 // mixer status returned by prepareTracks_l() 634 mixer_state mMixerStatus; // current cycle 635 // previous cycle when in prepareTracks_l() 636 mixer_state mMixerStatusIgnoringFastTracks; 637 // FIXME or a separate ready state per track 638 639 // FIXME move these declarations into the specific sub-class that needs them 640 // MIXER only 641 uint32_t sleepTimeShift; 642 643 // same as AudioFlinger::mStandbyTimeInNsecs except for DIRECT which uses a shorter value 644 nsecs_t standbyDelay; 645 646 // MIXER only 647 nsecs_t maxPeriod; 648 649 // DUPLICATING only 650 uint32_t writeFrames; 651 652 size_t mBytesRemaining; 653 size_t mCurrentWriteLength; 654 bool mUseAsyncWrite; 655 // mWriteAckSequence contains current write sequence on bits 31-1. The write sequence is 656 // incremented each time a write(), a flush() or a standby() occurs. 657 // Bit 0 is set when a write blocks and indicates a callback is expected. 658 // Bit 0 is reset by the async callback thread calling resetWriteBlocked(). Out of sequence 659 // callbacks are ignored. 660 uint32_t mWriteAckSequence; 661 // mDrainSequence contains current drain sequence on bits 31-1. The drain sequence is 662 // incremented each time a drain is requested or a flush() or standby() occurs. 663 // Bit 0 is set when the drain() command is called at the HAL and indicates a callback is 664 // expected. 665 // Bit 0 is reset by the async callback thread calling resetDraining(). Out of sequence 666 // callbacks are ignored. 667 uint32_t mDrainSequence; 668 // A condition that must be evaluated by prepareTrack_l() has changed and we must not wait 669 // for async write callback in the thread loop before evaluating it 670 bool mSignalPending; 671 sp<AsyncCallbackThread> mCallbackThread; 672 673private: 674 // The HAL output sink is treated as non-blocking, but current implementation is blocking 675 sp<NBAIO_Sink> mOutputSink; 676 // If a fast mixer is present, the blocking pipe sink, otherwise clear 677 sp<NBAIO_Sink> mPipeSink; 678 // The current sink for the normal mixer to write it's (sub)mix, mOutputSink or mPipeSink 679 sp<NBAIO_Sink> mNormalSink; 680#ifdef TEE_SINK 681 // For dumpsys 682 sp<NBAIO_Sink> mTeeSink; 683 sp<NBAIO_Source> mTeeSource; 684#endif 685 uint32_t mScreenState; // cached copy of gScreenState 686 static const size_t kFastMixerLogSize = 4 * 1024; 687 sp<NBLog::Writer> mFastMixerNBLogWriter; 688public: 689 virtual bool hasFastMixer() const = 0; 690 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex __unused) const 691 { FastTrackUnderruns dummy; return dummy; } 692 693protected: 694 // accessed by both binder threads and within threadLoop(), lock on mutex needed 695 unsigned mFastTrackAvailMask; // bit i set if fast track [i] is available 696 697private: 698 // timestamp latch: 699 // D input is written by threadLoop_write while mutex is unlocked, and read while locked 700 // Q output is written while locked, and read while locked 701 struct { 702 AudioTimestamp mTimestamp; 703 uint32_t mUnpresentedFrames; 704 } mLatchD, mLatchQ; 705 bool mLatchDValid; // true means mLatchD is valid, and clock it into latch at next opportunity 706 bool mLatchQValid; // true means mLatchQ is valid 707}; 708 709class MixerThread : public PlaybackThread { 710public: 711 MixerThread(const sp<AudioFlinger>& audioFlinger, 712 AudioStreamOut* output, 713 audio_io_handle_t id, 714 audio_devices_t device, 715 type_t type = MIXER); 716 virtual ~MixerThread(); 717 718 // Thread virtuals 719 720 virtual bool checkForNewParameters_l(); 721 virtual void dumpInternals(int fd, const Vector<String16>& args); 722 723protected: 724 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 725 virtual int getTrackName_l(audio_channel_mask_t channelMask, int sessionId); 726 virtual void deleteTrackName_l(int name); 727 virtual uint32_t idleSleepTimeUs() const; 728 virtual uint32_t suspendSleepTimeUs() const; 729 virtual void cacheParameters_l(); 730 731 // threadLoop snippets 732 virtual ssize_t threadLoop_write(); 733 virtual void threadLoop_standby(); 734 virtual void threadLoop_mix(); 735 virtual void threadLoop_sleepTime(); 736 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 737 virtual uint32_t correctLatency_l(uint32_t latency) const; 738 739 AudioMixer* mAudioMixer; // normal mixer 740private: 741 // one-time initialization, no locks required 742 FastMixer* mFastMixer; // non-NULL if there is also a fast mixer 743 sp<AudioWatchdog> mAudioWatchdog; // non-0 if there is an audio watchdog thread 744 745 // contents are not guaranteed to be consistent, no locks required 746 FastMixerDumpState mFastMixerDumpState; 747#ifdef STATE_QUEUE_DUMP 748 StateQueueObserverDump mStateQueueObserverDump; 749 StateQueueMutatorDump mStateQueueMutatorDump; 750#endif 751 AudioWatchdogDump mAudioWatchdogDump; 752 753 // accessible only within the threadLoop(), no locks required 754 // mFastMixer->sq() // for mutating and pushing state 755 int32_t mFastMixerFutex; // for cold idle 756 757public: 758 virtual bool hasFastMixer() const { return mFastMixer != NULL; } 759 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const { 760 ALOG_ASSERT(fastIndex < FastMixerState::kMaxFastTracks); 761 return mFastMixerDumpState.mTracks[fastIndex].mUnderruns; 762 } 763}; 764 765class DirectOutputThread : public PlaybackThread { 766public: 767 768 DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 769 audio_io_handle_t id, audio_devices_t device); 770 virtual ~DirectOutputThread(); 771 772 // Thread virtuals 773 774 virtual bool checkForNewParameters_l(); 775 776protected: 777 virtual int getTrackName_l(audio_channel_mask_t channelMask, int sessionId); 778 virtual void deleteTrackName_l(int name); 779 virtual uint32_t activeSleepTimeUs() const; 780 virtual uint32_t idleSleepTimeUs() const; 781 virtual uint32_t suspendSleepTimeUs() const; 782 virtual void cacheParameters_l(); 783 784 // threadLoop snippets 785 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 786 virtual void threadLoop_mix(); 787 virtual void threadLoop_sleepTime(); 788 789 // volumes last sent to audio HAL with stream->set_volume() 790 float mLeftVolFloat; 791 float mRightVolFloat; 792 793 DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 794 audio_io_handle_t id, uint32_t device, ThreadBase::type_t type); 795 void processVolume_l(Track *track, bool lastTrack); 796 797 // prepareTracks_l() tells threadLoop_mix() the name of the single active track 798 sp<Track> mActiveTrack; 799public: 800 virtual bool hasFastMixer() const { return false; } 801}; 802 803class OffloadThread : public DirectOutputThread { 804public: 805 806 OffloadThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 807 audio_io_handle_t id, uint32_t device); 808 virtual ~OffloadThread() {}; 809 810protected: 811 // threadLoop snippets 812 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 813 virtual void threadLoop_exit(); 814 815 virtual bool waitingAsyncCallback(); 816 virtual bool waitingAsyncCallback_l(); 817 virtual bool shouldStandby_l(); 818 virtual void onAddNewTrack_l(); 819 820private: 821 void flushHw_l(); 822 823private: 824 bool mHwPaused; 825 bool mFlushPending; 826 size_t mPausedWriteLength; // length in bytes of write interrupted by pause 827 size_t mPausedBytesRemaining; // bytes still waiting in mixbuffer after resume 828 wp<Track> mPreviousTrack; // used to detect track switch 829}; 830 831class AsyncCallbackThread : public Thread { 832public: 833 834 AsyncCallbackThread(const wp<PlaybackThread>& playbackThread); 835 836 virtual ~AsyncCallbackThread(); 837 838 // Thread virtuals 839 virtual bool threadLoop(); 840 841 // RefBase 842 virtual void onFirstRef(); 843 844 void exit(); 845 void setWriteBlocked(uint32_t sequence); 846 void resetWriteBlocked(); 847 void setDraining(uint32_t sequence); 848 void resetDraining(); 849 850private: 851 const wp<PlaybackThread> mPlaybackThread; 852 // mWriteAckSequence corresponds to the last write sequence passed by the offload thread via 853 // setWriteBlocked(). The sequence is shifted one bit to the left and the lsb is used 854 // to indicate that the callback has been received via resetWriteBlocked() 855 uint32_t mWriteAckSequence; 856 // mDrainSequence corresponds to the last drain sequence passed by the offload thread via 857 // setDraining(). The sequence is shifted one bit to the left and the lsb is used 858 // to indicate that the callback has been received via resetDraining() 859 uint32_t mDrainSequence; 860 Condition mWaitWorkCV; 861 Mutex mLock; 862}; 863 864class DuplicatingThread : public MixerThread { 865public: 866 DuplicatingThread(const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread, 867 audio_io_handle_t id); 868 virtual ~DuplicatingThread(); 869 870 // Thread virtuals 871 void addOutputTrack(MixerThread* thread); 872 void removeOutputTrack(MixerThread* thread); 873 uint32_t waitTimeMs() const { return mWaitTimeMs; } 874protected: 875 virtual uint32_t activeSleepTimeUs() const; 876 877private: 878 bool outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks); 879protected: 880 // threadLoop snippets 881 virtual void threadLoop_mix(); 882 virtual void threadLoop_sleepTime(); 883 virtual ssize_t threadLoop_write(); 884 virtual void threadLoop_standby(); 885 virtual void cacheParameters_l(); 886 887private: 888 // called from threadLoop, addOutputTrack, removeOutputTrack 889 virtual void updateWaitTime_l(); 890protected: 891 virtual void saveOutputTracks(); 892 virtual void clearOutputTracks(); 893private: 894 895 uint32_t mWaitTimeMs; 896 SortedVector < sp<OutputTrack> > outputTracks; 897 SortedVector < sp<OutputTrack> > mOutputTracks; 898public: 899 virtual bool hasFastMixer() const { return false; } 900}; 901 902 903// record thread 904class RecordThread : public ThreadBase 905{ 906public: 907 908 class RecordTrack; 909 class ResamplerBufferProvider : public AudioBufferProvider 910 // derives from AudioBufferProvider interface for use by resampler 911 { 912 public: 913 ResamplerBufferProvider(RecordTrack* recordTrack) : mRecordTrack(recordTrack) { } 914 virtual ~ResamplerBufferProvider() { } 915 // AudioBufferProvider interface 916 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts); 917 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 918 private: 919 RecordTrack * const mRecordTrack; 920 }; 921 922#include "RecordTracks.h" 923 924 RecordThread(const sp<AudioFlinger>& audioFlinger, 925 AudioStreamIn *input, 926 audio_io_handle_t id, 927 audio_devices_t outDevice, 928 audio_devices_t inDevice 929#ifdef TEE_SINK 930 , const sp<NBAIO_Sink>& teeSink 931#endif 932 ); 933 virtual ~RecordThread(); 934 935 // no addTrack_l ? 936 void destroyTrack_l(const sp<RecordTrack>& track); 937 void removeTrack_l(const sp<RecordTrack>& track); 938 939 void dumpInternals(int fd, const Vector<String16>& args); 940 void dumpTracks(int fd, const Vector<String16>& args); 941 942 // Thread virtuals 943 virtual bool threadLoop(); 944 945 // RefBase 946 virtual void onFirstRef(); 947 948 virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; } 949 950 sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l( 951 const sp<AudioFlinger::Client>& client, 952 uint32_t sampleRate, 953 audio_format_t format, 954 audio_channel_mask_t channelMask, 955 size_t *pFrameCount, 956 int sessionId, 957 int uid, 958 IAudioFlinger::track_flags_t *flags, 959 pid_t tid, 960 status_t *status /*non-NULL*/); 961 962 status_t start(RecordTrack* recordTrack, 963 AudioSystem::sync_event_t event, 964 int triggerSession); 965 966 // ask the thread to stop the specified track, and 967 // return true if the caller should then do it's part of the stopping process 968 bool stop(RecordTrack* recordTrack); 969 970 void dump(int fd, const Vector<String16>& args); 971 AudioStreamIn* clearInput(); 972 virtual audio_stream_t* stream() const; 973 974 975 virtual bool checkForNewParameters_l(); 976 virtual String8 getParameters(const String8& keys); 977 virtual void audioConfigChanged_l(int event, int param = 0); 978 void readInputParameters_l(); 979 virtual uint32_t getInputFramesLost(); 980 981 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 982 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 983 virtual uint32_t hasAudioSession(int sessionId) const; 984 985 // Return the set of unique session IDs across all tracks. 986 // The keys are the session IDs, and the associated values are meaningless. 987 // FIXME replace by Set [and implement Bag/Multiset for other uses]. 988 KeyedVector<int, bool> sessionIds() const; 989 990 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 991 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 992 993 static void syncStartEventCallback(const wp<SyncEvent>& event); 994 995 virtual size_t frameCount() const { return mFrameCount; } 996 bool hasFastRecorder() const { return false; } 997 998private: 999 // Enter standby if not already in standby, and set mStandby flag 1000 void standbyIfNotAlreadyInStandby(); 1001 1002 // Call the HAL standby method unconditionally, and don't change mStandby flag 1003 void inputStandBy(); 1004 1005 AudioStreamIn *mInput; 1006 SortedVector < sp<RecordTrack> > mTracks; 1007 // mActiveTracks has dual roles: it indicates the current active track(s), and 1008 // is used together with mStartStopCond to indicate start()/stop() progress 1009 SortedVector< sp<RecordTrack> > mActiveTracks; 1010 // generation counter for mActiveTracks 1011 int mActiveTracksGen; 1012 Condition mStartStopCond; 1013 1014 // resampler converts input at HAL Hz to output at AudioRecord client Hz 1015 int16_t *mRsmpInBuffer; // see new[] for details on the size 1016 size_t mRsmpInFrames; // size of resampler input in frames 1017 size_t mRsmpInFramesP2;// size rounded up to a power-of-2 1018 1019 // rolling index that is never cleared 1020 int32_t mRsmpInRear; // last filled frame + 1 1021 1022 // For dumpsys 1023 const sp<NBAIO_Sink> mTeeSink; 1024}; 1025