AudioFlinger.h revision fe3156ec6fd9fa57dde913fd8567530d095a6550
1/* 2** 3** Copyright 2007, 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 ANDROID_AUDIO_FLINGER_H 19#define ANDROID_AUDIO_FLINGER_H 20 21#include <stdint.h> 22#include <sys/types.h> 23#include <limits.h> 24 25#include <common_time/cc_helper.h> 26 27#include <media/IAudioFlinger.h> 28#include <media/IAudioFlingerClient.h> 29#include <media/IAudioTrack.h> 30#include <media/IAudioRecord.h> 31#include <media/AudioSystem.h> 32#include <media/AudioTrack.h> 33 34#include <utils/Atomic.h> 35#include <utils/Errors.h> 36#include <utils/threads.h> 37#include <utils/SortedVector.h> 38#include <utils/TypeHelpers.h> 39#include <utils/Vector.h> 40 41#include <binder/BinderService.h> 42#include <binder/MemoryDealer.h> 43 44#include <system/audio.h> 45#include <hardware/audio.h> 46#include <hardware/audio_policy.h> 47 48#include <media/AudioBufferProvider.h> 49#include <media/ExtendedAudioBufferProvider.h> 50#include "FastMixer.h" 51#include <media/nbaio/NBAIO.h> 52#include "AudioWatchdog.h" 53 54#include <powermanager/IPowerManager.h> 55 56namespace android { 57 58class audio_track_cblk_t; 59class effect_param_cblk_t; 60class AudioMixer; 61class AudioBuffer; 62class AudioResampler; 63class FastMixer; 64 65// ---------------------------------------------------------------------------- 66 67// AudioFlinger has a hard-coded upper limit of 2 channels for capture and playback. 68// There is support for > 2 channel tracks down-mixed to 2 channel output via a down-mix effect. 69// Adding full support for > 2 channel capture or playback would require more than simply changing 70// this #define. There is an independent hard-coded upper limit in AudioMixer; 71// removing that AudioMixer limit would be necessary but insufficient to support > 2 channels. 72// The macro FCC_2 highlights some (but not all) places where there is are 2-channel assumptions. 73// Search also for "2", "left", "right", "[0]", "[1]", ">> 16", "<< 16", etc. 74#define FCC_2 2 // FCC_2 = Fixed Channel Count 2 75 76static const nsecs_t kDefaultStandbyTimeInNsecs = seconds(3); 77 78class AudioFlinger : 79 public BinderService<AudioFlinger>, 80 public BnAudioFlinger 81{ 82 friend class BinderService<AudioFlinger>; // for AudioFlinger() 83public: 84 static const char* getServiceName() { return "media.audio_flinger"; } 85 86 virtual status_t dump(int fd, const Vector<String16>& args); 87 88 // IAudioFlinger interface, in binder opcode order 89 virtual sp<IAudioTrack> createTrack( 90 pid_t pid, 91 audio_stream_type_t streamType, 92 uint32_t sampleRate, 93 audio_format_t format, 94 audio_channel_mask_t channelMask, 95 int frameCount, 96 IAudioFlinger::track_flags_t flags, 97 const sp<IMemory>& sharedBuffer, 98 audio_io_handle_t output, 99 pid_t tid, 100 int *sessionId, 101 status_t *status); 102 103 virtual sp<IAudioRecord> openRecord( 104 pid_t pid, 105 audio_io_handle_t input, 106 uint32_t sampleRate, 107 audio_format_t format, 108 audio_channel_mask_t channelMask, 109 int frameCount, 110 IAudioFlinger::track_flags_t flags, 111 pid_t tid, 112 int *sessionId, 113 status_t *status); 114 115 virtual uint32_t sampleRate(audio_io_handle_t output) const; 116 virtual int channelCount(audio_io_handle_t output) const; 117 virtual audio_format_t format(audio_io_handle_t output) const; 118 virtual size_t frameCount(audio_io_handle_t output) const; 119 virtual uint32_t latency(audio_io_handle_t output) const; 120 121 virtual status_t setMasterVolume(float value); 122 virtual status_t setMasterMute(bool muted); 123 124 virtual float masterVolume() const; 125 virtual bool masterMute() const; 126 127 virtual status_t setStreamVolume(audio_stream_type_t stream, float value, 128 audio_io_handle_t output); 129 virtual status_t setStreamMute(audio_stream_type_t stream, bool muted); 130 131 virtual float streamVolume(audio_stream_type_t stream, 132 audio_io_handle_t output) const; 133 virtual bool streamMute(audio_stream_type_t stream) const; 134 135 virtual status_t setMode(audio_mode_t mode); 136 137 virtual status_t setMicMute(bool state); 138 virtual bool getMicMute() const; 139 140 virtual status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs); 141 virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) const; 142 143 virtual void registerClient(const sp<IAudioFlingerClient>& client); 144 145 virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, 146 audio_channel_mask_t channelMask) const; 147 148 virtual audio_io_handle_t openOutput(audio_module_handle_t module, 149 audio_devices_t *pDevices, 150 uint32_t *pSamplingRate, 151 audio_format_t *pFormat, 152 audio_channel_mask_t *pChannelMask, 153 uint32_t *pLatencyMs, 154 audio_output_flags_t flags); 155 156 virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1, 157 audio_io_handle_t output2); 158 159 virtual status_t closeOutput(audio_io_handle_t output); 160 161 virtual status_t suspendOutput(audio_io_handle_t output); 162 163 virtual status_t restoreOutput(audio_io_handle_t output); 164 165 virtual audio_io_handle_t openInput(audio_module_handle_t module, 166 audio_devices_t *pDevices, 167 uint32_t *pSamplingRate, 168 audio_format_t *pFormat, 169 audio_channel_mask_t *pChannelMask); 170 171 virtual status_t closeInput(audio_io_handle_t input); 172 173 virtual status_t setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output); 174 175 virtual status_t setVoiceVolume(float volume); 176 177 virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 178 audio_io_handle_t output) const; 179 180 virtual unsigned int getInputFramesLost(audio_io_handle_t ioHandle) const; 181 182 virtual int newAudioSessionId(); 183 184 virtual void acquireAudioSessionId(int audioSession); 185 186 virtual void releaseAudioSessionId(int audioSession); 187 188 virtual status_t queryNumberEffects(uint32_t *numEffects) const; 189 190 virtual status_t queryEffect(uint32_t index, effect_descriptor_t *descriptor) const; 191 192 virtual status_t getEffectDescriptor(const effect_uuid_t *pUuid, 193 effect_descriptor_t *descriptor) const; 194 195 virtual sp<IEffect> createEffect(pid_t pid, 196 effect_descriptor_t *pDesc, 197 const sp<IEffectClient>& effectClient, 198 int32_t priority, 199 audio_io_handle_t io, 200 int sessionId, 201 status_t *status, 202 int *id, 203 int *enabled); 204 205 virtual status_t moveEffects(int sessionId, audio_io_handle_t srcOutput, 206 audio_io_handle_t dstOutput); 207 208 virtual audio_module_handle_t loadHwModule(const char *name); 209 210 virtual status_t onTransact( 211 uint32_t code, 212 const Parcel& data, 213 Parcel* reply, 214 uint32_t flags); 215 216 // end of IAudioFlinger interface 217 218 class SyncEvent; 219 220 typedef void (*sync_event_callback_t)(const wp<SyncEvent>& event) ; 221 222 class SyncEvent : public RefBase { 223 public: 224 SyncEvent(AudioSystem::sync_event_t type, 225 int triggerSession, 226 int listenerSession, 227 sync_event_callback_t callBack, 228 void *cookie) 229 : mType(type), mTriggerSession(triggerSession), mListenerSession(listenerSession), 230 mCallback(callBack), mCookie(cookie) 231 {} 232 233 virtual ~SyncEvent() {} 234 235 void trigger() { Mutex::Autolock _l(mLock); if (mCallback) mCallback(this); } 236 bool isCancelled() const { Mutex::Autolock _l(mLock); return (mCallback == NULL); } 237 void cancel() { Mutex::Autolock _l(mLock); mCallback = NULL; } 238 AudioSystem::sync_event_t type() const { return mType; } 239 int triggerSession() const { return mTriggerSession; } 240 int listenerSession() const { return mListenerSession; } 241 void *cookie() const { return mCookie; } 242 243 private: 244 const AudioSystem::sync_event_t mType; 245 const int mTriggerSession; 246 const int mListenerSession; 247 sync_event_callback_t mCallback; 248 void * const mCookie; 249 mutable Mutex mLock; 250 }; 251 252 sp<SyncEvent> createSyncEvent(AudioSystem::sync_event_t type, 253 int triggerSession, 254 int listenerSession, 255 sync_event_callback_t callBack, 256 void *cookie); 257 258private: 259 class AudioHwDevice; // fwd declaration for findSuitableHwDev_l 260 261 audio_mode_t getMode() const { return mMode; } 262 263 bool btNrecIsOff() const { return mBtNrecIsOff; } 264 265 AudioFlinger(); 266 virtual ~AudioFlinger(); 267 268 // call in any IAudioFlinger method that accesses mPrimaryHardwareDev 269 status_t initCheck() const { return mPrimaryHardwareDev == NULL ? NO_INIT : NO_ERROR; } 270 271 // RefBase 272 virtual void onFirstRef(); 273 274 AudioHwDevice* findSuitableHwDev_l(audio_module_handle_t module, audio_devices_t devices); 275 void purgeStaleEffects_l(); 276 277 // standby delay for MIXER and DUPLICATING playback threads is read from property 278 // ro.audio.flinger_standbytime_ms or defaults to kDefaultStandbyTimeInNsecs 279 static nsecs_t mStandbyTimeInNsecs; 280 281 // Internal dump utilities. 282 void dumpPermissionDenial(int fd, const Vector<String16>& args); 283 void dumpClients(int fd, const Vector<String16>& args); 284 void dumpInternals(int fd, const Vector<String16>& args); 285 286 // --- Client --- 287 class Client : public RefBase { 288 public: 289 Client(const sp<AudioFlinger>& audioFlinger, pid_t pid); 290 virtual ~Client(); 291 sp<MemoryDealer> heap() const; 292 pid_t pid() const { return mPid; } 293 sp<AudioFlinger> audioFlinger() const { return mAudioFlinger; } 294 295 bool reserveTimedTrack(); 296 void releaseTimedTrack(); 297 298 private: 299 Client(const Client&); 300 Client& operator = (const Client&); 301 const sp<AudioFlinger> mAudioFlinger; 302 const sp<MemoryDealer> mMemoryDealer; 303 const pid_t mPid; 304 305 Mutex mTimedTrackLock; 306 int mTimedTrackCount; 307 }; 308 309 // --- Notification Client --- 310 class NotificationClient : public IBinder::DeathRecipient { 311 public: 312 NotificationClient(const sp<AudioFlinger>& audioFlinger, 313 const sp<IAudioFlingerClient>& client, 314 pid_t pid); 315 virtual ~NotificationClient(); 316 317 sp<IAudioFlingerClient> audioFlingerClient() const { return mAudioFlingerClient; } 318 319 // IBinder::DeathRecipient 320 virtual void binderDied(const wp<IBinder>& who); 321 322 private: 323 NotificationClient(const NotificationClient&); 324 NotificationClient& operator = (const NotificationClient&); 325 326 const sp<AudioFlinger> mAudioFlinger; 327 const pid_t mPid; 328 const sp<IAudioFlingerClient> mAudioFlingerClient; 329 }; 330 331 class TrackHandle; 332 class RecordHandle; 333 class RecordThread; 334 class PlaybackThread; 335 class MixerThread; 336 class DirectOutputThread; 337 class DuplicatingThread; 338 class Track; 339 class RecordTrack; 340 class EffectModule; 341 class EffectHandle; 342 class EffectChain; 343 struct AudioStreamOut; 344 struct AudioStreamIn; 345 346 class ThreadBase : public Thread { 347 public: 348 349 enum type_t { 350 MIXER, // Thread class is MixerThread 351 DIRECT, // Thread class is DirectOutputThread 352 DUPLICATING, // Thread class is DuplicatingThread 353 RECORD // Thread class is RecordThread 354 }; 355 356 ThreadBase (const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 357 audio_devices_t outDevice, audio_devices_t inDevice, type_t type); 358 virtual ~ThreadBase(); 359 360 void dumpBase(int fd, const Vector<String16>& args); 361 void dumpEffectChains(int fd, const Vector<String16>& args); 362 363 void clearPowerManager(); 364 365 // base for record and playback 366 class TrackBase : public ExtendedAudioBufferProvider, public RefBase { 367 368 public: 369 enum track_state { 370 IDLE, 371 TERMINATED, 372 FLUSHED, 373 STOPPED, 374 // next 2 states are currently used for fast tracks only 375 STOPPING_1, // waiting for first underrun 376 STOPPING_2, // waiting for presentation complete 377 RESUMING, 378 ACTIVE, 379 PAUSING, 380 PAUSED 381 }; 382 383 TrackBase(ThreadBase *thread, 384 const sp<Client>& client, 385 uint32_t sampleRate, 386 audio_format_t format, 387 audio_channel_mask_t channelMask, 388 int frameCount, 389 const sp<IMemory>& sharedBuffer, 390 int sessionId); 391 virtual ~TrackBase(); 392 393 virtual status_t start(AudioSystem::sync_event_t event, 394 int triggerSession) = 0; 395 virtual void stop() = 0; 396 sp<IMemory> getCblk() const { return mCblkMemory; } 397 audio_track_cblk_t* cblk() const { return mCblk; } 398 int sessionId() const { return mSessionId; } 399 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 400 401 protected: 402 TrackBase(const TrackBase&); 403 TrackBase& operator = (const TrackBase&); 404 405 // AudioBufferProvider interface 406 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) = 0; 407 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 408 409 // ExtendedAudioBufferProvider interface is only needed for Track, 410 // but putting it in TrackBase avoids the complexity of virtual inheritance 411 virtual size_t framesReady() const { return SIZE_MAX; } 412 413 audio_format_t format() const { 414 return mFormat; 415 } 416 417 int channelCount() const { return mChannelCount; } 418 419 audio_channel_mask_t channelMask() const { return mChannelMask; } 420 421 int sampleRate() const; // FIXME inline after cblk sr moved 422 423 // Return a pointer to the start of a contiguous slice of the track buffer. 424 // Parameter 'offset' is the requested start position, expressed in 425 // monotonically increasing frame units relative to the track epoch. 426 // Parameter 'frames' is the requested length, also in frame units. 427 // Always returns non-NULL. It is the caller's responsibility to 428 // verify that this will be successful; the result of calling this 429 // function with invalid 'offset' or 'frames' is undefined. 430 void* getBuffer(uint32_t offset, uint32_t frames) const; 431 432 bool isStopped() const { 433 return (mState == STOPPED || mState == FLUSHED); 434 } 435 436 // for fast tracks only 437 bool isStopping() const { 438 return mState == STOPPING_1 || mState == STOPPING_2; 439 } 440 bool isStopping_1() const { 441 return mState == STOPPING_1; 442 } 443 bool isStopping_2() const { 444 return mState == STOPPING_2; 445 } 446 447 bool isTerminated() const { 448 return mState == TERMINATED; 449 } 450 451 bool step(); 452 void reset(); 453 454 const wp<ThreadBase> mThread; 455 /*const*/ sp<Client> mClient; // see explanation at ~TrackBase() why not const 456 sp<IMemory> mCblkMemory; 457 audio_track_cblk_t* mCblk; 458 void* mBuffer; // start of track buffer, typically in shared memory 459 void* mBufferEnd; // &mBuffer[mFrameCount * frameSize], where frameSize 460 // is based on mChannelCount and 16-bit samples 461 uint32_t mFrameCount; 462 // we don't really need a lock for these 463 track_state mState; 464 const uint32_t mSampleRate; // initial sample rate only; for tracks which 465 // support dynamic rates, the current value is in control block 466 const audio_format_t mFormat; 467 bool mStepServerFailed; 468 const int mSessionId; 469 uint8_t mChannelCount; 470 audio_channel_mask_t mChannelMask; 471 Vector < sp<SyncEvent> >mSyncEvents; 472 }; 473 474 class ConfigEvent { 475 public: 476 ConfigEvent() : mEvent(0), mParam(0) {} 477 478 int mEvent; 479 int mParam; 480 }; 481 482 class PMDeathRecipient : public IBinder::DeathRecipient { 483 public: 484 PMDeathRecipient(const wp<ThreadBase>& thread) : mThread(thread) {} 485 virtual ~PMDeathRecipient() {} 486 487 // IBinder::DeathRecipient 488 virtual void binderDied(const wp<IBinder>& who); 489 490 private: 491 PMDeathRecipient(const PMDeathRecipient&); 492 PMDeathRecipient& operator = (const PMDeathRecipient&); 493 494 wp<ThreadBase> mThread; 495 }; 496 497 virtual status_t initCheck() const = 0; 498 499 // static externally-visible 500 type_t type() const { return mType; } 501 audio_io_handle_t id() const { return mId;} 502 503 // dynamic externally-visible 504 uint32_t sampleRate() const { return mSampleRate; } 505 int channelCount() const { return mChannelCount; } 506 audio_channel_mask_t channelMask() const { return mChannelMask; } 507 audio_format_t format() const { return mFormat; } 508 // Called by AudioFlinger::frameCount(audio_io_handle_t output) and effects, 509 // and returns the normal mix buffer's frame count. No API for HAL frame count. 510 size_t frameCount() const { return mNormalFrameCount; } 511 512 // Should be "virtual status_t requestExitAndWait()" and override same 513 // method in Thread, but Thread::requestExitAndWait() is not yet virtual. 514 void exit(); 515 virtual bool checkForNewParameters_l() = 0; 516 virtual status_t setParameters(const String8& keyValuePairs); 517 virtual String8 getParameters(const String8& keys) = 0; 518 virtual void audioConfigChanged_l(int event, int param = 0) = 0; 519 void sendConfigEvent(int event, int param = 0); 520 void sendConfigEvent_l(int event, int param = 0); 521 void processConfigEvents(); 522 523 // see note at declaration of mStandby, mOutDevice and mInDevice 524 bool standby() const { return mStandby; } 525 audio_devices_t outDevice() const { return mOutDevice; } 526 audio_devices_t inDevice() const { return mInDevice; } 527 528 virtual audio_stream_t* stream() const = 0; 529 530 sp<EffectHandle> createEffect_l( 531 const sp<AudioFlinger::Client>& client, 532 const sp<IEffectClient>& effectClient, 533 int32_t priority, 534 int sessionId, 535 effect_descriptor_t *desc, 536 int *enabled, 537 status_t *status); 538 void disconnectEffect(const sp< EffectModule>& effect, 539 EffectHandle *handle, 540 bool unpinIfLast); 541 542 // return values for hasAudioSession (bit field) 543 enum effect_state { 544 EFFECT_SESSION = 0x1, // the audio session corresponds to at least one 545 // effect 546 TRACK_SESSION = 0x2 // the audio session corresponds to at least one 547 // track 548 }; 549 550 // get effect chain corresponding to session Id. 551 sp<EffectChain> getEffectChain(int sessionId); 552 // same as getEffectChain() but must be called with ThreadBase mutex locked 553 sp<EffectChain> getEffectChain_l(int sessionId) const; 554 // add an effect chain to the chain list (mEffectChains) 555 virtual status_t addEffectChain_l(const sp<EffectChain>& chain) = 0; 556 // remove an effect chain from the chain list (mEffectChains) 557 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain) = 0; 558 // lock all effect chains Mutexes. Must be called before releasing the 559 // ThreadBase mutex before processing the mixer and effects. This guarantees the 560 // integrity of the chains during the process. 561 // Also sets the parameter 'effectChains' to current value of mEffectChains. 562 void lockEffectChains_l(Vector< sp<EffectChain> >& effectChains); 563 // unlock effect chains after process 564 void unlockEffectChains(const Vector< sp<EffectChain> >& effectChains); 565 // set audio mode to all effect chains 566 void setMode(audio_mode_t mode); 567 // get effect module with corresponding ID on specified audio session 568 sp<AudioFlinger::EffectModule> getEffect(int sessionId, int effectId); 569 sp<AudioFlinger::EffectModule> getEffect_l(int sessionId, int effectId); 570 // add and effect module. Also creates the effect chain is none exists for 571 // the effects audio session 572 status_t addEffect_l(const sp< EffectModule>& effect); 573 // remove and effect module. Also removes the effect chain is this was the last 574 // effect 575 void removeEffect_l(const sp< EffectModule>& effect); 576 // detach all tracks connected to an auxiliary effect 577 virtual void detachAuxEffect_l(int effectId) {} 578 // returns either EFFECT_SESSION if effects on this audio session exist in one 579 // chain, or TRACK_SESSION if tracks on this audio session exist, or both 580 virtual uint32_t hasAudioSession(int sessionId) const = 0; 581 // the value returned by default implementation is not important as the 582 // strategy is only meaningful for PlaybackThread which implements this method 583 virtual uint32_t getStrategyForSession_l(int sessionId) { return 0; } 584 585 // suspend or restore effect according to the type of effect passed. a NULL 586 // type pointer means suspend all effects in the session 587 void setEffectSuspended(const effect_uuid_t *type, 588 bool suspend, 589 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 590 // check if some effects must be suspended/restored when an effect is enabled 591 // or disabled 592 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 593 bool enabled, 594 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 595 void checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 596 bool enabled, 597 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 598 599 virtual status_t setSyncEvent(const sp<SyncEvent>& event) = 0; 600 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const = 0; 601 602 603 mutable Mutex mLock; 604 605 protected: 606 607 // entry describing an effect being suspended in mSuspendedSessions keyed vector 608 class SuspendedSessionDesc : public RefBase { 609 public: 610 SuspendedSessionDesc() : mRefCount(0) {} 611 612 int mRefCount; // number of active suspend requests 613 effect_uuid_t mType; // effect type UUID 614 }; 615 616 void acquireWakeLock(); 617 void acquireWakeLock_l(); 618 void releaseWakeLock(); 619 void releaseWakeLock_l(); 620 void setEffectSuspended_l(const effect_uuid_t *type, 621 bool suspend, 622 int sessionId); 623 // updated mSuspendedSessions when an effect suspended or restored 624 void updateSuspendedSessions_l(const effect_uuid_t *type, 625 bool suspend, 626 int sessionId); 627 // check if some effects must be suspended when an effect chain is added 628 void checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain); 629 630 friend class AudioFlinger; // for mEffectChains 631 632 const type_t mType; 633 634 // Used by parameters, config events, addTrack_l, exit 635 Condition mWaitWorkCV; 636 637 const sp<AudioFlinger> mAudioFlinger; 638 uint32_t mSampleRate; 639 size_t mFrameCount; // output HAL, direct output, record 640 size_t mNormalFrameCount; // normal mixer and effects 641 audio_channel_mask_t mChannelMask; 642 uint16_t mChannelCount; 643 size_t mFrameSize; 644 audio_format_t mFormat; 645 646 // Parameter sequence by client: binder thread calling setParameters(): 647 // 1. Lock mLock 648 // 2. Append to mNewParameters 649 // 3. mWaitWorkCV.signal 650 // 4. mParamCond.waitRelative with timeout 651 // 5. read mParamStatus 652 // 6. mWaitWorkCV.signal 653 // 7. Unlock 654 // 655 // Parameter sequence by server: threadLoop calling checkForNewParameters_l(): 656 // 1. Lock mLock 657 // 2. If there is an entry in mNewParameters proceed ... 658 // 2. Read first entry in mNewParameters 659 // 3. Process 660 // 4. Remove first entry from mNewParameters 661 // 5. Set mParamStatus 662 // 6. mParamCond.signal 663 // 7. mWaitWorkCV.wait with timeout (this is to avoid overwriting mParamStatus) 664 // 8. Unlock 665 Condition mParamCond; 666 Vector<String8> mNewParameters; 667 status_t mParamStatus; 668 669 Vector<ConfigEvent> mConfigEvents; 670 671 // These fields are written and read by thread itself without lock or barrier, 672 // and read by other threads without lock or barrier via standby() , outDevice() 673 // and inDevice(). 674 // Because of the absence of a lock or barrier, any other thread that reads 675 // these fields must use the information in isolation, or be prepared to deal 676 // with possibility that it might be inconsistent with other information. 677 bool mStandby; // Whether thread is currently in standby. 678 audio_devices_t mOutDevice; // output device 679 audio_devices_t mInDevice; // input device 680 audio_source_t mAudioSource; // (see audio.h, audio_source_t) 681 682 const audio_io_handle_t mId; 683 Vector< sp<EffectChain> > mEffectChains; 684 685 static const int kNameLength = 16; // prctl(PR_SET_NAME) limit 686 char mName[kNameLength]; 687 sp<IPowerManager> mPowerManager; 688 sp<IBinder> mWakeLockToken; 689 const sp<PMDeathRecipient> mDeathRecipient; 690 // list of suspended effects per session and per type. The first vector is 691 // keyed by session ID, the second by type UUID timeLow field 692 KeyedVector< int, KeyedVector< int, sp<SuspendedSessionDesc> > > mSuspendedSessions; 693 }; 694 695 struct stream_type_t { 696 stream_type_t() 697 : volume(1.0f), 698 mute(false) 699 { 700 } 701 float volume; 702 bool mute; 703 }; 704 705 // --- PlaybackThread --- 706 class PlaybackThread : public ThreadBase { 707 public: 708 709 enum mixer_state { 710 MIXER_IDLE, // no active tracks 711 MIXER_TRACKS_ENABLED, // at least one active track, but no track has any data ready 712 MIXER_TRACKS_READY // at least one active track, and at least one track has data 713 // standby mode does not have an enum value 714 // suspend by audio policy manager is orthogonal to mixer state 715 }; 716 717 // playback track 718 class Track : public TrackBase, public VolumeProvider { 719 public: 720 Track( PlaybackThread *thread, 721 const sp<Client>& client, 722 audio_stream_type_t streamType, 723 uint32_t sampleRate, 724 audio_format_t format, 725 audio_channel_mask_t channelMask, 726 int frameCount, 727 const sp<IMemory>& sharedBuffer, 728 int sessionId, 729 IAudioFlinger::track_flags_t flags); 730 virtual ~Track(); 731 732 static void appendDumpHeader(String8& result); 733 void dump(char* buffer, size_t size); 734 virtual status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 735 int triggerSession = 0); 736 virtual void stop(); 737 void pause(); 738 739 void flush(); 740 void destroy(); 741 void mute(bool); 742 int name() const { return mName; } 743 744 audio_stream_type_t streamType() const { 745 return mStreamType; 746 } 747 status_t attachAuxEffect(int EffectId); 748 void setAuxBuffer(int EffectId, int32_t *buffer); 749 int32_t *auxBuffer() const { return mAuxBuffer; } 750 void setMainBuffer(int16_t *buffer) { mMainBuffer = buffer; } 751 int16_t *mainBuffer() const { return mMainBuffer; } 752 int auxEffectId() const { return mAuxEffectId; } 753 754 // implement FastMixerState::VolumeProvider interface 755 virtual uint32_t getVolumeLR(); 756 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 757 758 protected: 759 // for numerous 760 friend class PlaybackThread; 761 friend class MixerThread; 762 friend class DirectOutputThread; 763 764 Track(const Track&); 765 Track& operator = (const Track&); 766 767 // AudioBufferProvider interface 768 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts = kInvalidPTS); 769 // releaseBuffer() not overridden 770 771 virtual size_t framesReady() const; 772 773 bool isMuted() const { return mMute; } 774 bool isPausing() const { 775 return mState == PAUSING; 776 } 777 bool isPaused() const { 778 return mState == PAUSED; 779 } 780 bool isResuming() const { 781 return mState == RESUMING; 782 } 783 bool isReady() const; 784 void setPaused() { mState = PAUSED; } 785 void reset(); 786 787 bool isOutputTrack() const { 788 return (mStreamType == AUDIO_STREAM_CNT); 789 } 790 791 sp<IMemory> sharedBuffer() const { return mSharedBuffer; } 792 793 bool presentationComplete(size_t framesWritten, size_t audioHalFrames); 794 795 public: 796 void triggerEvents(AudioSystem::sync_event_t type); 797 virtual bool isTimedTrack() const { return false; } 798 bool isFastTrack() const { return (mFlags & IAudioFlinger::TRACK_FAST) != 0; } 799 800 protected: 801 802 // written by Track::mute() called by binder thread(s), without a mutex or barrier. 803 // read by Track::isMuted() called by playback thread, also without a mutex or barrier. 804 // The lack of mutex or barrier is safe because the mute status is only used by itself. 805 bool mMute; 806 807 // FILLED state is used for suppressing volume ramp at begin of playing 808 enum {FS_INVALID, FS_FILLING, FS_FILLED, FS_ACTIVE}; 809 mutable uint8_t mFillingUpStatus; 810 int8_t mRetryCount; 811 const sp<IMemory> mSharedBuffer; 812 bool mResetDone; 813 const audio_stream_type_t mStreamType; 814 int mName; // track name on the normal mixer, 815 // allocated statically at track creation time, 816 // and is even allocated (though unused) for fast tracks 817 // FIXME don't allocate track name for fast tracks 818 int16_t *mMainBuffer; 819 int32_t *mAuxBuffer; 820 int mAuxEffectId; 821 bool mHasVolumeController; 822 size_t mPresentationCompleteFrames; // number of frames written to the audio HAL 823 // when this track will be fully rendered 824 private: 825 IAudioFlinger::track_flags_t mFlags; 826 827 // The following fields are only for fast tracks, and should be in a subclass 828 int mFastIndex; // index within FastMixerState::mFastTracks[]; 829 // either mFastIndex == -1 if not isFastTrack() 830 // or 0 < mFastIndex < FastMixerState::kMaxFast because 831 // index 0 is reserved for normal mixer's submix; 832 // index is allocated statically at track creation time 833 // but the slot is only used if track is active 834 FastTrackUnderruns mObservedUnderruns; // Most recently observed value of 835 // mFastMixerDumpState.mTracks[mFastIndex].mUnderruns 836 uint32_t mUnderrunCount; // Counter of total number of underruns, never reset 837 volatile float mCachedVolume; // combined master volume and stream type volume; 838 // 'volatile' means accessed without lock or 839 // barrier, but is read/written atomically 840 }; // end of Track 841 842 class TimedTrack : public Track { 843 public: 844 static sp<TimedTrack> create(PlaybackThread *thread, 845 const sp<Client>& client, 846 audio_stream_type_t streamType, 847 uint32_t sampleRate, 848 audio_format_t format, 849 audio_channel_mask_t channelMask, 850 int frameCount, 851 const sp<IMemory>& sharedBuffer, 852 int sessionId); 853 virtual ~TimedTrack(); 854 855 class TimedBuffer { 856 public: 857 TimedBuffer(); 858 TimedBuffer(const sp<IMemory>& buffer, int64_t pts); 859 const sp<IMemory>& buffer() const { return mBuffer; } 860 int64_t pts() const { return mPTS; } 861 uint32_t position() const { return mPosition; } 862 void setPosition(uint32_t pos) { mPosition = pos; } 863 private: 864 sp<IMemory> mBuffer; 865 int64_t mPTS; 866 uint32_t mPosition; 867 }; 868 869 // Mixer facing methods. 870 virtual bool isTimedTrack() const { return true; } 871 virtual size_t framesReady() const; 872 873 // AudioBufferProvider interface 874 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, 875 int64_t pts); 876 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 877 878 // Client/App facing methods. 879 status_t allocateTimedBuffer(size_t size, 880 sp<IMemory>* buffer); 881 status_t queueTimedBuffer(const sp<IMemory>& buffer, 882 int64_t pts); 883 status_t setMediaTimeTransform(const LinearTransform& xform, 884 TimedAudioTrack::TargetTimeline target); 885 886 private: 887 TimedTrack(PlaybackThread *thread, 888 const sp<Client>& client, 889 audio_stream_type_t streamType, 890 uint32_t sampleRate, 891 audio_format_t format, 892 audio_channel_mask_t channelMask, 893 int frameCount, 894 const sp<IMemory>& sharedBuffer, 895 int sessionId); 896 897 void timedYieldSamples_l(AudioBufferProvider::Buffer* buffer); 898 void timedYieldSilence_l(uint32_t numFrames, 899 AudioBufferProvider::Buffer* buffer); 900 void trimTimedBufferQueue_l(); 901 void trimTimedBufferQueueHead_l(const char* logTag); 902 void updateFramesPendingAfterTrim_l(const TimedBuffer& buf, 903 const char* logTag); 904 905 uint64_t mLocalTimeFreq; 906 LinearTransform mLocalTimeToSampleTransform; 907 LinearTransform mMediaTimeToSampleTransform; 908 sp<MemoryDealer> mTimedMemoryDealer; 909 910 Vector<TimedBuffer> mTimedBufferQueue; 911 bool mQueueHeadInFlight; 912 bool mTrimQueueHeadOnRelease; 913 uint32_t mFramesPendingInQueue; 914 915 uint8_t* mTimedSilenceBuffer; 916 uint32_t mTimedSilenceBufferSize; 917 mutable Mutex mTimedBufferQueueLock; 918 bool mTimedAudioOutputOnTime; 919 CCHelper mCCHelper; 920 921 Mutex mMediaTimeTransformLock; 922 LinearTransform mMediaTimeTransform; 923 bool mMediaTimeTransformValid; 924 TimedAudioTrack::TargetTimeline mMediaTimeTransformTarget; 925 }; 926 927 928 // playback track 929 class OutputTrack : public Track { 930 public: 931 932 class Buffer: public AudioBufferProvider::Buffer { 933 public: 934 int16_t *mBuffer; 935 }; 936 937 OutputTrack(PlaybackThread *thread, 938 DuplicatingThread *sourceThread, 939 uint32_t sampleRate, 940 audio_format_t format, 941 audio_channel_mask_t channelMask, 942 int frameCount); 943 virtual ~OutputTrack(); 944 945 virtual status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 946 int triggerSession = 0); 947 virtual void stop(); 948 bool write(int16_t* data, uint32_t frames); 949 bool bufferQueueEmpty() const { return mBufferQueue.size() == 0; } 950 bool isActive() const { return mActive; } 951 const wp<ThreadBase>& thread() const { return mThread; } 952 953 private: 954 955 enum { 956 NO_MORE_BUFFERS = 0x80000001, // same in AudioTrack.h, ok to be different value 957 }; 958 959 status_t obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs); 960 void clearBufferQueue(); 961 962 // Maximum number of pending buffers allocated by OutputTrack::write() 963 static const uint8_t kMaxOverFlowBuffers = 10; 964 965 Vector < Buffer* > mBufferQueue; 966 AudioBufferProvider::Buffer mOutBuffer; 967 bool mActive; 968 DuplicatingThread* const mSourceThread; // for waitTimeMs() in write() 969 }; // end of OutputTrack 970 971 PlaybackThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 972 audio_io_handle_t id, audio_devices_t device, type_t type); 973 virtual ~PlaybackThread(); 974 975 void dump(int fd, const Vector<String16>& args); 976 977 // Thread virtuals 978 virtual status_t readyToRun(); 979 virtual bool threadLoop(); 980 981 // RefBase 982 virtual void onFirstRef(); 983 984protected: 985 // Code snippets that were lifted up out of threadLoop() 986 virtual void threadLoop_mix() = 0; 987 virtual void threadLoop_sleepTime() = 0; 988 virtual void threadLoop_write(); 989 virtual void threadLoop_standby(); 990 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 991 992 // prepareTracks_l reads and writes mActiveTracks, and returns 993 // the pending set of tracks to remove via Vector 'tracksToRemove'. The caller 994 // is responsible for clearing or destroying this Vector later on, when it 995 // is safe to do so. That will drop the final ref count and destroy the tracks. 996 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove) = 0; 997 998public: 999 1000 virtual status_t initCheck() const { return (mOutput == NULL) ? NO_INIT : NO_ERROR; } 1001 1002 // return estimated latency in milliseconds, as reported by HAL 1003 uint32_t latency() const; 1004 // same, but lock must already be held 1005 uint32_t latency_l() const; 1006 1007 void setMasterVolume(float value); 1008 void setMasterMute(bool muted); 1009 1010 void setStreamVolume(audio_stream_type_t stream, float value); 1011 void setStreamMute(audio_stream_type_t stream, bool muted); 1012 1013 float streamVolume(audio_stream_type_t stream) const; 1014 1015 sp<Track> createTrack_l( 1016 const sp<AudioFlinger::Client>& client, 1017 audio_stream_type_t streamType, 1018 uint32_t sampleRate, 1019 audio_format_t format, 1020 audio_channel_mask_t channelMask, 1021 int frameCount, 1022 const sp<IMemory>& sharedBuffer, 1023 int sessionId, 1024 IAudioFlinger::track_flags_t flags, 1025 pid_t tid, 1026 status_t *status); 1027 1028 AudioStreamOut* getOutput() const; 1029 AudioStreamOut* clearOutput(); 1030 virtual audio_stream_t* stream() const; 1031 1032 // a very large number of suspend() will eventually wraparound, but unlikely 1033 void suspend() { (void) android_atomic_inc(&mSuspended); } 1034 void restore() 1035 { 1036 // if restore() is done without suspend(), get back into 1037 // range so that the next suspend() will operate correctly 1038 if (android_atomic_dec(&mSuspended) <= 0) { 1039 android_atomic_release_store(0, &mSuspended); 1040 } 1041 } 1042 bool isSuspended() const 1043 { return android_atomic_acquire_load(&mSuspended) > 0; } 1044 1045 virtual String8 getParameters(const String8& keys); 1046 virtual void audioConfigChanged_l(int event, int param = 0); 1047 status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames); 1048 int16_t *mixBuffer() const { return mMixBuffer; }; 1049 1050 virtual void detachAuxEffect_l(int effectId); 1051 status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track> track, 1052 int EffectId); 1053 status_t attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track> track, 1054 int EffectId); 1055 1056 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1057 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1058 virtual uint32_t hasAudioSession(int sessionId) const; 1059 virtual uint32_t getStrategyForSession_l(int sessionId); 1060 1061 1062 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 1063 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 1064 void invalidateTracks(audio_stream_type_t streamType); 1065 1066 1067 protected: 1068 int16_t* mMixBuffer; 1069 1070 // suspend count, > 0 means suspended. While suspended, the thread continues to pull from 1071 // tracks and mix, but doesn't write to HAL. A2DP and SCO HAL implementations can't handle 1072 // concurrent use of both of them, so Audio Policy Service suspends one of the threads to 1073 // workaround that restriction. 1074 // 'volatile' means accessed via atomic operations and no lock. 1075 volatile int32_t mSuspended; 1076 1077 int mBytesWritten; 1078 private: 1079 // mMasterMute is in both PlaybackThread and in AudioFlinger. When a 1080 // PlaybackThread needs to find out if master-muted, it checks it's local 1081 // copy rather than the one in AudioFlinger. This optimization saves a lock. 1082 bool mMasterMute; 1083 void setMasterMute_l(bool muted) { mMasterMute = muted; } 1084 protected: 1085 SortedVector< wp<Track> > mActiveTracks; // FIXME check if this could be sp<> 1086 1087 // Allocate a track name for a given channel mask. 1088 // Returns name >= 0 if successful, -1 on failure. 1089 virtual int getTrackName_l(audio_channel_mask_t channelMask, int sessionId) = 0; 1090 virtual void deleteTrackName_l(int name) = 0; 1091 1092 // Time to sleep between cycles when: 1093 virtual uint32_t activeSleepTimeUs() const; // mixer state MIXER_TRACKS_ENABLED 1094 virtual uint32_t idleSleepTimeUs() const = 0; // mixer state MIXER_IDLE 1095 virtual uint32_t suspendSleepTimeUs() const = 0; // audio policy manager suspended us 1096 // No sleep when mixer state == MIXER_TRACKS_READY; relies on audio HAL stream->write() 1097 // No sleep in standby mode; waits on a condition 1098 1099 // Code snippets that are temporarily lifted up out of threadLoop() until the merge 1100 void checkSilentMode_l(); 1101 1102 // Non-trivial for DUPLICATING only 1103 virtual void saveOutputTracks() { } 1104 virtual void clearOutputTracks() { } 1105 1106 // Cache various calculated values, at threadLoop() entry and after a parameter change 1107 virtual void cacheParameters_l(); 1108 1109 virtual uint32_t correctLatency(uint32_t latency) const; 1110 1111 private: 1112 1113 friend class AudioFlinger; // for numerous 1114 1115 PlaybackThread(const Client&); 1116 PlaybackThread& operator = (const PlaybackThread&); 1117 1118 status_t addTrack_l(const sp<Track>& track); 1119 void destroyTrack_l(const sp<Track>& track); 1120 void removeTrack_l(const sp<Track>& track); 1121 1122 void readOutputParameters(); 1123 1124 virtual void dumpInternals(int fd, const Vector<String16>& args); 1125 void dumpTracks(int fd, const Vector<String16>& args); 1126 1127 SortedVector< sp<Track> > mTracks; 1128 // mStreamTypes[] uses 1 additional stream type internally for the OutputTrack used by DuplicatingThread 1129 stream_type_t mStreamTypes[AUDIO_STREAM_CNT + 1]; 1130 AudioStreamOut *mOutput; 1131 1132 float mMasterVolume; 1133 nsecs_t mLastWriteTime; 1134 int mNumWrites; 1135 int mNumDelayedWrites; 1136 bool mInWrite; 1137 1138 // FIXME rename these former local variables of threadLoop to standard "m" names 1139 nsecs_t standbyTime; 1140 size_t mixBufferSize; 1141 1142 // cached copies of activeSleepTimeUs() and idleSleepTimeUs() made by cacheParameters_l() 1143 uint32_t activeSleepTime; 1144 uint32_t idleSleepTime; 1145 1146 uint32_t sleepTime; 1147 1148 // mixer status returned by prepareTracks_l() 1149 mixer_state mMixerStatus; // current cycle 1150 // previous cycle when in prepareTracks_l() 1151 mixer_state mMixerStatusIgnoringFastTracks; 1152 // FIXME or a separate ready state per track 1153 1154 // FIXME move these declarations into the specific sub-class that needs them 1155 // MIXER only 1156 uint32_t sleepTimeShift; 1157 1158 // same as AudioFlinger::mStandbyTimeInNsecs except for DIRECT which uses a shorter value 1159 nsecs_t standbyDelay; 1160 1161 // MIXER only 1162 nsecs_t maxPeriod; 1163 1164 // DUPLICATING only 1165 uint32_t writeFrames; 1166 1167 private: 1168 // The HAL output sink is treated as non-blocking, but current implementation is blocking 1169 sp<NBAIO_Sink> mOutputSink; 1170 // If a fast mixer is present, the blocking pipe sink, otherwise clear 1171 sp<NBAIO_Sink> mPipeSink; 1172 // The current sink for the normal mixer to write it's (sub)mix, mOutputSink or mPipeSink 1173 sp<NBAIO_Sink> mNormalSink; 1174 // For dumpsys 1175 sp<NBAIO_Sink> mTeeSink; 1176 sp<NBAIO_Source> mTeeSource; 1177 uint32_t mScreenState; // cached copy of gScreenState 1178 public: 1179 virtual bool hasFastMixer() const = 0; 1180 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const 1181 { FastTrackUnderruns dummy; return dummy; } 1182 1183 protected: 1184 // accessed by both binder threads and within threadLoop(), lock on mutex needed 1185 unsigned mFastTrackAvailMask; // bit i set if fast track [i] is available 1186 1187 }; 1188 1189 class MixerThread : public PlaybackThread { 1190 public: 1191 MixerThread (const sp<AudioFlinger>& audioFlinger, 1192 AudioStreamOut* output, 1193 audio_io_handle_t id, 1194 audio_devices_t device, 1195 type_t type = MIXER); 1196 virtual ~MixerThread(); 1197 1198 // Thread virtuals 1199 1200 virtual bool checkForNewParameters_l(); 1201 virtual void dumpInternals(int fd, const Vector<String16>& args); 1202 1203 protected: 1204 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1205 virtual int getTrackName_l(audio_channel_mask_t channelMask, int sessionId); 1206 virtual void deleteTrackName_l(int name); 1207 virtual uint32_t idleSleepTimeUs() const; 1208 virtual uint32_t suspendSleepTimeUs() const; 1209 virtual void cacheParameters_l(); 1210 1211 // threadLoop snippets 1212 virtual void threadLoop_write(); 1213 virtual void threadLoop_standby(); 1214 virtual void threadLoop_mix(); 1215 virtual void threadLoop_sleepTime(); 1216 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 1217 virtual uint32_t correctLatency(uint32_t latency) const; 1218 1219 AudioMixer* mAudioMixer; // normal mixer 1220 private: 1221 // one-time initialization, no locks required 1222 FastMixer* mFastMixer; // non-NULL if there is also a fast mixer 1223 sp<AudioWatchdog> mAudioWatchdog; // non-0 if there is an audio watchdog thread 1224 1225 // contents are not guaranteed to be consistent, no locks required 1226 FastMixerDumpState mFastMixerDumpState; 1227#ifdef STATE_QUEUE_DUMP 1228 StateQueueObserverDump mStateQueueObserverDump; 1229 StateQueueMutatorDump mStateQueueMutatorDump; 1230#endif 1231 AudioWatchdogDump mAudioWatchdogDump; 1232 1233 // accessible only within the threadLoop(), no locks required 1234 // mFastMixer->sq() // for mutating and pushing state 1235 int32_t mFastMixerFutex; // for cold idle 1236 1237 public: 1238 virtual bool hasFastMixer() const { return mFastMixer != NULL; } 1239 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const { 1240 ALOG_ASSERT(fastIndex < FastMixerState::kMaxFastTracks); 1241 return mFastMixerDumpState.mTracks[fastIndex].mUnderruns; 1242 } 1243 }; 1244 1245 class DirectOutputThread : public PlaybackThread { 1246 public: 1247 1248 DirectOutputThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1249 audio_io_handle_t id, audio_devices_t device); 1250 virtual ~DirectOutputThread(); 1251 1252 // Thread virtuals 1253 1254 virtual bool checkForNewParameters_l(); 1255 1256 protected: 1257 virtual int getTrackName_l(audio_channel_mask_t channelMask, int sessionId); 1258 virtual void deleteTrackName_l(int name); 1259 virtual uint32_t activeSleepTimeUs() const; 1260 virtual uint32_t idleSleepTimeUs() const; 1261 virtual uint32_t suspendSleepTimeUs() const; 1262 virtual void cacheParameters_l(); 1263 1264 // threadLoop snippets 1265 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1266 virtual void threadLoop_mix(); 1267 virtual void threadLoop_sleepTime(); 1268 1269 // volumes last sent to audio HAL with stream->set_volume() 1270 float mLeftVolFloat; 1271 float mRightVolFloat; 1272 1273private: 1274 // prepareTracks_l() tells threadLoop_mix() the name of the single active track 1275 sp<Track> mActiveTrack; 1276 public: 1277 virtual bool hasFastMixer() const { return false; } 1278 }; 1279 1280 class DuplicatingThread : public MixerThread { 1281 public: 1282 DuplicatingThread (const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread, 1283 audio_io_handle_t id); 1284 virtual ~DuplicatingThread(); 1285 1286 // Thread virtuals 1287 void addOutputTrack(MixerThread* thread); 1288 void removeOutputTrack(MixerThread* thread); 1289 uint32_t waitTimeMs() const { return mWaitTimeMs; } 1290 protected: 1291 virtual uint32_t activeSleepTimeUs() const; 1292 1293 private: 1294 bool outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks); 1295 protected: 1296 // threadLoop snippets 1297 virtual void threadLoop_mix(); 1298 virtual void threadLoop_sleepTime(); 1299 virtual void threadLoop_write(); 1300 virtual void threadLoop_standby(); 1301 virtual void cacheParameters_l(); 1302 1303 private: 1304 // called from threadLoop, addOutputTrack, removeOutputTrack 1305 virtual void updateWaitTime_l(); 1306 protected: 1307 virtual void saveOutputTracks(); 1308 virtual void clearOutputTracks(); 1309 private: 1310 1311 uint32_t mWaitTimeMs; 1312 SortedVector < sp<OutputTrack> > outputTracks; 1313 SortedVector < sp<OutputTrack> > mOutputTracks; 1314 public: 1315 virtual bool hasFastMixer() const { return false; } 1316 }; 1317 1318 PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const; 1319 MixerThread *checkMixerThread_l(audio_io_handle_t output) const; 1320 RecordThread *checkRecordThread_l(audio_io_handle_t input) const; 1321 // no range check, AudioFlinger::mLock held 1322 bool streamMute_l(audio_stream_type_t stream) const 1323 { return mStreamTypes[stream].mute; } 1324 // no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held 1325 float streamVolume_l(audio_stream_type_t stream) const 1326 { return mStreamTypes[stream].volume; } 1327 void audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2); 1328 1329 // allocate an audio_io_handle_t, session ID, or effect ID 1330 uint32_t nextUniqueId(); 1331 1332 status_t moveEffectChain_l(int sessionId, 1333 PlaybackThread *srcThread, 1334 PlaybackThread *dstThread, 1335 bool reRegister); 1336 // return thread associated with primary hardware device, or NULL 1337 PlaybackThread *primaryPlaybackThread_l() const; 1338 audio_devices_t primaryOutputDevice_l() const; 1339 1340 sp<PlaybackThread> getEffectThread_l(int sessionId, int EffectId); 1341 1342 // server side of the client's IAudioTrack 1343 class TrackHandle : public android::BnAudioTrack { 1344 public: 1345 TrackHandle(const sp<PlaybackThread::Track>& track); 1346 virtual ~TrackHandle(); 1347 virtual sp<IMemory> getCblk() const; 1348 virtual status_t start(); 1349 virtual void stop(); 1350 virtual void flush(); 1351 virtual void mute(bool); 1352 virtual void pause(); 1353 virtual status_t attachAuxEffect(int effectId); 1354 virtual status_t allocateTimedBuffer(size_t size, 1355 sp<IMemory>* buffer); 1356 virtual status_t queueTimedBuffer(const sp<IMemory>& buffer, 1357 int64_t pts); 1358 virtual status_t setMediaTimeTransform(const LinearTransform& xform, 1359 int target); 1360 virtual status_t onTransact( 1361 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 1362 private: 1363 const sp<PlaybackThread::Track> mTrack; 1364 }; 1365 1366 void removeClient_l(pid_t pid); 1367 void removeNotificationClient(pid_t pid); 1368 1369 1370 // record thread 1371 class RecordThread : public ThreadBase, public AudioBufferProvider 1372 // derives from AudioBufferProvider interface for use by resampler 1373 { 1374 public: 1375 1376 // record track 1377 class RecordTrack : public TrackBase { 1378 public: 1379 RecordTrack(RecordThread *thread, 1380 const sp<Client>& client, 1381 uint32_t sampleRate, 1382 audio_format_t format, 1383 audio_channel_mask_t channelMask, 1384 int frameCount, 1385 int sessionId); 1386 virtual ~RecordTrack(); 1387 1388 virtual status_t start(AudioSystem::sync_event_t event, int triggerSession); 1389 virtual void stop(); 1390 1391 void destroy(); 1392 1393 // clear the buffer overflow flag 1394 void clearOverflow() { mOverflow = false; } 1395 // set the buffer overflow flag and return previous value 1396 bool setOverflow() { bool tmp = mOverflow; mOverflow = true; return tmp; } 1397 1398 static void appendDumpHeader(String8& result); 1399 void dump(char* buffer, size_t size); 1400 1401 private: 1402 friend class AudioFlinger; // for mState 1403 1404 RecordTrack(const RecordTrack&); 1405 RecordTrack& operator = (const RecordTrack&); 1406 1407 // AudioBufferProvider interface 1408 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts = kInvalidPTS); 1409 // releaseBuffer() not overridden 1410 1411 bool mOverflow; // overflow on most recent attempt to fill client buffer 1412 }; 1413 1414 RecordThread(const sp<AudioFlinger>& audioFlinger, 1415 AudioStreamIn *input, 1416 uint32_t sampleRate, 1417 audio_channel_mask_t channelMask, 1418 audio_io_handle_t id, 1419 audio_devices_t device); 1420 virtual ~RecordThread(); 1421 1422 // no addTrack_l ? 1423 void destroyTrack_l(const sp<RecordTrack>& track); 1424 void removeTrack_l(const sp<RecordTrack>& track); 1425 1426 void dumpInternals(int fd, const Vector<String16>& args); 1427 void dumpTracks(int fd, const Vector<String16>& args); 1428 1429 // Thread virtuals 1430 virtual bool threadLoop(); 1431 virtual status_t readyToRun(); 1432 1433 // RefBase 1434 virtual void onFirstRef(); 1435 1436 virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; } 1437 sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l( 1438 const sp<AudioFlinger::Client>& client, 1439 uint32_t sampleRate, 1440 audio_format_t format, 1441 audio_channel_mask_t channelMask, 1442 int frameCount, 1443 int sessionId, 1444 IAudioFlinger::track_flags_t flags, 1445 pid_t tid, 1446 status_t *status); 1447 1448 status_t start(RecordTrack* recordTrack, 1449 AudioSystem::sync_event_t event, 1450 int triggerSession); 1451 1452 // ask the thread to stop the specified track, and 1453 // return true if the caller should then do it's part of the stopping process 1454 bool stop_l(RecordTrack* recordTrack); 1455 1456 void dump(int fd, const Vector<String16>& args); 1457 AudioStreamIn* clearInput(); 1458 virtual audio_stream_t* stream() const; 1459 1460 // AudioBufferProvider interface 1461 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts); 1462 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 1463 1464 virtual bool checkForNewParameters_l(); 1465 virtual String8 getParameters(const String8& keys); 1466 virtual void audioConfigChanged_l(int event, int param = 0); 1467 void readInputParameters(); 1468 virtual unsigned int getInputFramesLost(); 1469 1470 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1471 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1472 virtual uint32_t hasAudioSession(int sessionId) const; 1473 1474 // Return the set of unique session IDs across all tracks. 1475 // The keys are the session IDs, and the associated values are meaningless. 1476 // FIXME replace by Set [and implement Bag/Multiset for other uses]. 1477 KeyedVector<int, bool> sessionIds() const; 1478 1479 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 1480 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 1481 1482 static void syncStartEventCallback(const wp<SyncEvent>& event); 1483 void handleSyncStartEvent(const sp<SyncEvent>& event); 1484 1485 private: 1486 void clearSyncStartEvent(); 1487 1488 // Enter standby if not already in standby, and set mStandby flag 1489 void standby(); 1490 1491 // Call the HAL standby method unconditionally, and don't change mStandby flag 1492 void inputStandBy(); 1493 1494 AudioStreamIn *mInput; 1495 SortedVector < sp<RecordTrack> > mTracks; 1496 // mActiveTrack has dual roles: it indicates the current active track, and 1497 // is used together with mStartStopCond to indicate start()/stop() progress 1498 sp<RecordTrack> mActiveTrack; 1499 Condition mStartStopCond; 1500 AudioResampler *mResampler; 1501 int32_t *mRsmpOutBuffer; 1502 int16_t *mRsmpInBuffer; 1503 size_t mRsmpInIndex; 1504 size_t mInputBytes; 1505 const int mReqChannelCount; 1506 const uint32_t mReqSampleRate; 1507 ssize_t mBytesRead; 1508 // sync event triggering actual audio capture. Frames read before this event will 1509 // be dropped and therefore not read by the application. 1510 sp<SyncEvent> mSyncStartEvent; 1511 // number of captured frames to drop after the start sync event has been received. 1512 // when < 0, maximum frames to drop before starting capture even if sync event is 1513 // not received 1514 ssize_t mFramestoDrop; 1515 }; 1516 1517 // server side of the client's IAudioRecord 1518 class RecordHandle : public android::BnAudioRecord { 1519 public: 1520 RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack); 1521 virtual ~RecordHandle(); 1522 virtual sp<IMemory> getCblk() const; 1523 virtual status_t start(int /*AudioSystem::sync_event_t*/ event, int triggerSession); 1524 virtual void stop(); 1525 virtual status_t onTransact( 1526 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 1527 private: 1528 const sp<RecordThread::RecordTrack> mRecordTrack; 1529 1530 // for use from destructor 1531 void stop_nonvirtual(); 1532 }; 1533 1534 //--- Audio Effect Management 1535 1536 // EffectModule and EffectChain classes both have their own mutex to protect 1537 // state changes or resource modifications. Always respect the following order 1538 // if multiple mutexes must be acquired to avoid cross deadlock: 1539 // AudioFlinger -> ThreadBase -> EffectChain -> EffectModule 1540 1541 // The EffectModule class is a wrapper object controlling the effect engine implementation 1542 // in the effect library. It prevents concurrent calls to process() and command() functions 1543 // from different client threads. It keeps a list of EffectHandle objects corresponding 1544 // to all client applications using this effect and notifies applications of effect state, 1545 // control or parameter changes. It manages the activation state machine to send appropriate 1546 // reset, enable, disable commands to effect engine and provide volume 1547 // ramping when effects are activated/deactivated. 1548 // When controlling an auxiliary effect, the EffectModule also provides an input buffer used by 1549 // the attached track(s) to accumulate their auxiliary channel. 1550 class EffectModule: public RefBase { 1551 public: 1552 EffectModule(ThreadBase *thread, 1553 const wp<AudioFlinger::EffectChain>& chain, 1554 effect_descriptor_t *desc, 1555 int id, 1556 int sessionId); 1557 virtual ~EffectModule(); 1558 1559 enum effect_state { 1560 IDLE, 1561 RESTART, 1562 STARTING, 1563 ACTIVE, 1564 STOPPING, 1565 STOPPED, 1566 DESTROYED 1567 }; 1568 1569 int id() const { return mId; } 1570 void process(); 1571 void updateState(); 1572 status_t command(uint32_t cmdCode, 1573 uint32_t cmdSize, 1574 void *pCmdData, 1575 uint32_t *replySize, 1576 void *pReplyData); 1577 1578 void reset_l(); 1579 status_t configure(); 1580 status_t init(); 1581 effect_state state() const { 1582 return mState; 1583 } 1584 uint32_t status() { 1585 return mStatus; 1586 } 1587 int sessionId() const { 1588 return mSessionId; 1589 } 1590 status_t setEnabled(bool enabled); 1591 status_t setEnabled_l(bool enabled); 1592 bool isEnabled() const; 1593 bool isProcessEnabled() const; 1594 1595 void setInBuffer(int16_t *buffer) { mConfig.inputCfg.buffer.s16 = buffer; } 1596 int16_t *inBuffer() { return mConfig.inputCfg.buffer.s16; } 1597 void setOutBuffer(int16_t *buffer) { mConfig.outputCfg.buffer.s16 = buffer; } 1598 int16_t *outBuffer() { return mConfig.outputCfg.buffer.s16; } 1599 void setChain(const wp<EffectChain>& chain) { mChain = chain; } 1600 void setThread(const wp<ThreadBase>& thread) { mThread = thread; } 1601 const wp<ThreadBase>& thread() { return mThread; } 1602 1603 status_t addHandle(EffectHandle *handle); 1604 size_t disconnect(EffectHandle *handle, bool unpinIfLast); 1605 size_t removeHandle(EffectHandle *handle); 1606 1607 const effect_descriptor_t& desc() const { return mDescriptor; } 1608 wp<EffectChain>& chain() { return mChain; } 1609 1610 status_t setDevice(audio_devices_t device); 1611 status_t setVolume(uint32_t *left, uint32_t *right, bool controller); 1612 status_t setMode(audio_mode_t mode); 1613 status_t setAudioSource(audio_source_t source); 1614 status_t start(); 1615 status_t stop(); 1616 void setSuspended(bool suspended); 1617 bool suspended() const; 1618 1619 EffectHandle* controlHandle_l(); 1620 1621 bool isPinned() const { return mPinned; } 1622 void unPin() { mPinned = false; } 1623 bool purgeHandles(); 1624 void lock() { mLock.lock(); } 1625 void unlock() { mLock.unlock(); } 1626 1627 void dump(int fd, const Vector<String16>& args); 1628 1629 protected: 1630 friend class AudioFlinger; // for mHandles 1631 bool mPinned; 1632 1633 // Maximum time allocated to effect engines to complete the turn off sequence 1634 static const uint32_t MAX_DISABLE_TIME_MS = 10000; 1635 1636 EffectModule(const EffectModule&); 1637 EffectModule& operator = (const EffectModule&); 1638 1639 status_t start_l(); 1640 status_t stop_l(); 1641 1642mutable Mutex mLock; // mutex for process, commands and handles list protection 1643 wp<ThreadBase> mThread; // parent thread 1644 wp<EffectChain> mChain; // parent effect chain 1645 const int mId; // this instance unique ID 1646 const int mSessionId; // audio session ID 1647 const effect_descriptor_t mDescriptor;// effect descriptor received from effect engine 1648 effect_config_t mConfig; // input and output audio configuration 1649 effect_handle_t mEffectInterface; // Effect module C API 1650 status_t mStatus; // initialization status 1651 effect_state mState; // current activation state 1652 Vector<EffectHandle *> mHandles; // list of client handles 1653 // First handle in mHandles has highest priority and controls the effect module 1654 uint32_t mMaxDisableWaitCnt; // maximum grace period before forcing an effect off after 1655 // sending disable command. 1656 uint32_t mDisableWaitCnt; // current process() calls count during disable period. 1657 bool mSuspended; // effect is suspended: temporarily disabled by framework 1658 }; 1659 1660 // The EffectHandle class implements the IEffect interface. It provides resources 1661 // to receive parameter updates, keeps track of effect control 1662 // ownership and state and has a pointer to the EffectModule object it is controlling. 1663 // There is one EffectHandle object for each application controlling (or using) 1664 // an effect module. 1665 // The EffectHandle is obtained by calling AudioFlinger::createEffect(). 1666 class EffectHandle: public android::BnEffect { 1667 public: 1668 1669 EffectHandle(const sp<EffectModule>& effect, 1670 const sp<AudioFlinger::Client>& client, 1671 const sp<IEffectClient>& effectClient, 1672 int32_t priority); 1673 virtual ~EffectHandle(); 1674 1675 // IEffect 1676 virtual status_t enable(); 1677 virtual status_t disable(); 1678 virtual status_t command(uint32_t cmdCode, 1679 uint32_t cmdSize, 1680 void *pCmdData, 1681 uint32_t *replySize, 1682 void *pReplyData); 1683 virtual void disconnect(); 1684 private: 1685 void disconnect(bool unpinIfLast); 1686 public: 1687 virtual sp<IMemory> getCblk() const { return mCblkMemory; } 1688 virtual status_t onTransact(uint32_t code, const Parcel& data, 1689 Parcel* reply, uint32_t flags); 1690 1691 1692 // Give or take control of effect module 1693 // - hasControl: true if control is given, false if removed 1694 // - signal: true client app should be signaled of change, false otherwise 1695 // - enabled: state of the effect when control is passed 1696 void setControl(bool hasControl, bool signal, bool enabled); 1697 void commandExecuted(uint32_t cmdCode, 1698 uint32_t cmdSize, 1699 void *pCmdData, 1700 uint32_t replySize, 1701 void *pReplyData); 1702 void setEnabled(bool enabled); 1703 bool enabled() const { return mEnabled; } 1704 1705 // Getters 1706 int id() const { return mEffect->id(); } 1707 int priority() const { return mPriority; } 1708 bool hasControl() const { return mHasControl; } 1709 sp<EffectModule> effect() const { return mEffect; } 1710 // destroyed_l() must be called with the associated EffectModule mLock held 1711 bool destroyed_l() const { return mDestroyed; } 1712 1713 void dump(char* buffer, size_t size); 1714 1715 protected: 1716 friend class AudioFlinger; // for mEffect, mHasControl, mEnabled 1717 EffectHandle(const EffectHandle&); 1718 EffectHandle& operator =(const EffectHandle&); 1719 1720 sp<EffectModule> mEffect; // pointer to controlled EffectModule 1721 sp<IEffectClient> mEffectClient; // callback interface for client notifications 1722 /*const*/ sp<Client> mClient; // client for shared memory allocation, see disconnect() 1723 sp<IMemory> mCblkMemory; // shared memory for control block 1724 effect_param_cblk_t* mCblk; // control block for deferred parameter setting via shared memory 1725 uint8_t* mBuffer; // pointer to parameter area in shared memory 1726 int mPriority; // client application priority to control the effect 1727 bool mHasControl; // true if this handle is controlling the effect 1728 bool mEnabled; // cached enable state: needed when the effect is 1729 // restored after being suspended 1730 bool mDestroyed; // Set to true by destructor. Access with EffectModule 1731 // mLock held 1732 }; 1733 1734 // the EffectChain class represents a group of effects associated to one audio session. 1735 // There can be any number of EffectChain objects per output mixer thread (PlaybackThread). 1736 // The EffecChain with session ID 0 contains global effects applied to the output mix. 1737 // Effects in this chain can be insert or auxiliary. Effects in other chains (attached to tracks) 1738 // are insert only. The EffectChain maintains an ordered list of effect module, the order corresponding 1739 // in the effect process order. When attached to a track (session ID != 0), it also provide it's own 1740 // input buffer used by the track as accumulation buffer. 1741 class EffectChain: public RefBase { 1742 public: 1743 EffectChain(const wp<ThreadBase>& wThread, int sessionId); 1744 EffectChain(ThreadBase *thread, int sessionId); 1745 virtual ~EffectChain(); 1746 1747 // special key used for an entry in mSuspendedEffects keyed vector 1748 // corresponding to a suspend all request. 1749 static const int kKeyForSuspendAll = 0; 1750 1751 // minimum duration during which we force calling effect process when last track on 1752 // a session is stopped or removed to allow effect tail to be rendered 1753 static const int kProcessTailDurationMs = 1000; 1754 1755 void process_l(); 1756 1757 void lock() { 1758 mLock.lock(); 1759 } 1760 void unlock() { 1761 mLock.unlock(); 1762 } 1763 1764 status_t addEffect_l(const sp<EffectModule>& handle); 1765 size_t removeEffect_l(const sp<EffectModule>& handle); 1766 1767 int sessionId() const { return mSessionId; } 1768 void setSessionId(int sessionId) { mSessionId = sessionId; } 1769 1770 sp<EffectModule> getEffectFromDesc_l(effect_descriptor_t *descriptor); 1771 sp<EffectModule> getEffectFromId_l(int id); 1772 sp<EffectModule> getEffectFromType_l(const effect_uuid_t *type); 1773 bool setVolume_l(uint32_t *left, uint32_t *right); 1774 void setDevice_l(audio_devices_t device); 1775 void setMode_l(audio_mode_t mode); 1776 void setAudioSource_l(audio_source_t source); 1777 1778 void setInBuffer(int16_t *buffer, bool ownsBuffer = false) { 1779 mInBuffer = buffer; 1780 mOwnInBuffer = ownsBuffer; 1781 } 1782 int16_t *inBuffer() const { 1783 return mInBuffer; 1784 } 1785 void setOutBuffer(int16_t *buffer) { 1786 mOutBuffer = buffer; 1787 } 1788 int16_t *outBuffer() const { 1789 return mOutBuffer; 1790 } 1791 1792 void incTrackCnt() { android_atomic_inc(&mTrackCnt); } 1793 void decTrackCnt() { android_atomic_dec(&mTrackCnt); } 1794 int32_t trackCnt() const { return android_atomic_acquire_load(&mTrackCnt); } 1795 1796 void incActiveTrackCnt() { android_atomic_inc(&mActiveTrackCnt); 1797 mTailBufferCount = mMaxTailBuffers; } 1798 void decActiveTrackCnt() { android_atomic_dec(&mActiveTrackCnt); } 1799 int32_t activeTrackCnt() const { return android_atomic_acquire_load(&mActiveTrackCnt); } 1800 1801 uint32_t strategy() const { return mStrategy; } 1802 void setStrategy(uint32_t strategy) 1803 { mStrategy = strategy; } 1804 1805 // suspend effect of the given type 1806 void setEffectSuspended_l(const effect_uuid_t *type, 1807 bool suspend); 1808 // suspend all eligible effects 1809 void setEffectSuspendedAll_l(bool suspend); 1810 // check if effects should be suspend or restored when a given effect is enable or disabled 1811 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1812 bool enabled); 1813 1814 void clearInputBuffer(); 1815 1816 void dump(int fd, const Vector<String16>& args); 1817 1818 protected: 1819 friend class AudioFlinger; // for mThread, mEffects 1820 EffectChain(const EffectChain&); 1821 EffectChain& operator =(const EffectChain&); 1822 1823 class SuspendedEffectDesc : public RefBase { 1824 public: 1825 SuspendedEffectDesc() : mRefCount(0) {} 1826 1827 int mRefCount; 1828 effect_uuid_t mType; 1829 wp<EffectModule> mEffect; 1830 }; 1831 1832 // get a list of effect modules to suspend when an effect of the type 1833 // passed is enabled. 1834 void getSuspendEligibleEffects(Vector< sp<EffectModule> > &effects); 1835 1836 // get an effect module if it is currently enable 1837 sp<EffectModule> getEffectIfEnabled(const effect_uuid_t *type); 1838 // true if the effect whose descriptor is passed can be suspended 1839 // OEMs can modify the rules implemented in this method to exclude specific effect 1840 // types or implementations from the suspend/restore mechanism. 1841 bool isEffectEligibleForSuspend(const effect_descriptor_t& desc); 1842 1843 void clearInputBuffer_l(sp<ThreadBase> thread); 1844 1845 wp<ThreadBase> mThread; // parent mixer thread 1846 Mutex mLock; // mutex protecting effect list 1847 Vector< sp<EffectModule> > mEffects; // list of effect modules 1848 int mSessionId; // audio session ID 1849 int16_t *mInBuffer; // chain input buffer 1850 int16_t *mOutBuffer; // chain output buffer 1851 1852 // 'volatile' here means these are accessed with atomic operations instead of mutex 1853 volatile int32_t mActiveTrackCnt; // number of active tracks connected 1854 volatile int32_t mTrackCnt; // number of tracks connected 1855 1856 int32_t mTailBufferCount; // current effect tail buffer count 1857 int32_t mMaxTailBuffers; // maximum effect tail buffers 1858 bool mOwnInBuffer; // true if the chain owns its input buffer 1859 int mVolumeCtrlIdx; // index of insert effect having control over volume 1860 uint32_t mLeftVolume; // previous volume on left channel 1861 uint32_t mRightVolume; // previous volume on right channel 1862 uint32_t mNewLeftVolume; // new volume on left channel 1863 uint32_t mNewRightVolume; // new volume on right channel 1864 uint32_t mStrategy; // strategy for this effect chain 1865 // mSuspendedEffects lists all effects currently suspended in the chain. 1866 // Use effect type UUID timelow field as key. There is no real risk of identical 1867 // timeLow fields among effect type UUIDs. 1868 // Updated by updateSuspendedSessions_l() only. 1869 KeyedVector< int, sp<SuspendedEffectDesc> > mSuspendedEffects; 1870 }; 1871 1872 class AudioHwDevice { 1873 public: 1874 enum Flags { 1875 AHWD_CAN_SET_MASTER_VOLUME = 0x1, 1876 AHWD_CAN_SET_MASTER_MUTE = 0x2, 1877 }; 1878 1879 AudioHwDevice(const char *moduleName, 1880 audio_hw_device_t *hwDevice, 1881 Flags flags) 1882 : mModuleName(strdup(moduleName)) 1883 , mHwDevice(hwDevice) 1884 , mFlags(flags) { } 1885 /*virtual*/ ~AudioHwDevice() { free((void *)mModuleName); } 1886 1887 bool canSetMasterVolume() const { 1888 return (0 != (mFlags & AHWD_CAN_SET_MASTER_VOLUME)); 1889 } 1890 1891 bool canSetMasterMute() const { 1892 return (0 != (mFlags & AHWD_CAN_SET_MASTER_MUTE)); 1893 } 1894 1895 const char *moduleName() const { return mModuleName; } 1896 audio_hw_device_t *hwDevice() const { return mHwDevice; } 1897 private: 1898 const char * const mModuleName; 1899 audio_hw_device_t * const mHwDevice; 1900 Flags mFlags; 1901 }; 1902 1903 // AudioStreamOut and AudioStreamIn are immutable, so their fields are const. 1904 // For emphasis, we could also make all pointers to them be "const *", 1905 // but that would clutter the code unnecessarily. 1906 1907 struct AudioStreamOut { 1908 AudioHwDevice* const audioHwDev; 1909 audio_stream_out_t* const stream; 1910 1911 audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); } 1912 1913 AudioStreamOut(AudioHwDevice *dev, audio_stream_out_t *out) : 1914 audioHwDev(dev), stream(out) {} 1915 }; 1916 1917 struct AudioStreamIn { 1918 AudioHwDevice* const audioHwDev; 1919 audio_stream_in_t* const stream; 1920 1921 audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); } 1922 1923 AudioStreamIn(AudioHwDevice *dev, audio_stream_in_t *in) : 1924 audioHwDev(dev), stream(in) {} 1925 }; 1926 1927 // for mAudioSessionRefs only 1928 struct AudioSessionRef { 1929 AudioSessionRef(int sessionid, pid_t pid) : 1930 mSessionid(sessionid), mPid(pid), mCnt(1) {} 1931 const int mSessionid; 1932 const pid_t mPid; 1933 int mCnt; 1934 }; 1935 1936 mutable Mutex mLock; 1937 1938 DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client() 1939 1940 mutable Mutex mHardwareLock; 1941 // NOTE: If both mLock and mHardwareLock mutexes must be held, 1942 // always take mLock before mHardwareLock 1943 1944 // These two fields are immutable after onFirstRef(), so no lock needed to access 1945 AudioHwDevice* mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL 1946 DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*> mAudioHwDevs; 1947 1948 // for dump, indicates which hardware operation is currently in progress (but not stream ops) 1949 enum hardware_call_state { 1950 AUDIO_HW_IDLE = 0, // no operation in progress 1951 AUDIO_HW_INIT, // init_check 1952 AUDIO_HW_OUTPUT_OPEN, // open_output_stream 1953 AUDIO_HW_OUTPUT_CLOSE, // unused 1954 AUDIO_HW_INPUT_OPEN, // unused 1955 AUDIO_HW_INPUT_CLOSE, // unused 1956 AUDIO_HW_STANDBY, // unused 1957 AUDIO_HW_SET_MASTER_VOLUME, // set_master_volume 1958 AUDIO_HW_GET_ROUTING, // unused 1959 AUDIO_HW_SET_ROUTING, // unused 1960 AUDIO_HW_GET_MODE, // unused 1961 AUDIO_HW_SET_MODE, // set_mode 1962 AUDIO_HW_GET_MIC_MUTE, // get_mic_mute 1963 AUDIO_HW_SET_MIC_MUTE, // set_mic_mute 1964 AUDIO_HW_SET_VOICE_VOLUME, // set_voice_volume 1965 AUDIO_HW_SET_PARAMETER, // set_parameters 1966 AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size 1967 AUDIO_HW_GET_MASTER_VOLUME, // get_master_volume 1968 AUDIO_HW_GET_PARAMETER, // get_parameters 1969 AUDIO_HW_SET_MASTER_MUTE, // set_master_mute 1970 AUDIO_HW_GET_MASTER_MUTE, // get_master_mute 1971 }; 1972 1973 mutable hardware_call_state mHardwareStatus; // for dump only 1974 1975 1976 DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads; 1977 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 1978 1979 // member variables below are protected by mLock 1980 float mMasterVolume; 1981 bool mMasterMute; 1982 // end of variables protected by mLock 1983 1984 DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads; 1985 1986 DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients; 1987 volatile int32_t mNextUniqueId; // updated by android_atomic_inc 1988 audio_mode_t mMode; 1989 bool mBtNrecIsOff; 1990 1991 // protected by mLock 1992 Vector<AudioSessionRef*> mAudioSessionRefs; 1993 1994 float masterVolume_l() const; 1995 bool masterMute_l() const; 1996 audio_module_handle_t loadHwModule_l(const char *name); 1997 1998 Vector < sp<SyncEvent> > mPendingSyncEvents; // sync events awaiting for a session 1999 // to be created 2000 2001private: 2002 sp<Client> registerPid_l(pid_t pid); // always returns non-0 2003 2004 // for use from destructor 2005 status_t closeOutput_nonvirtual(audio_io_handle_t output); 2006 status_t closeInput_nonvirtual(audio_io_handle_t input); 2007}; 2008 2009 2010// ---------------------------------------------------------------------------- 2011 2012}; // namespace android 2013 2014#endif // ANDROID_AUDIO_FLINGER_H 2015