AudioFlinger.h revision 57b2dd1e78af53115985f18d31ec5421c9da947e
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, audio_devices_t device, type_t type); 357 virtual ~ThreadBase(); 358 359 void dumpBase(int fd, const Vector<String16>& args); 360 void dumpEffectChains(int fd, const Vector<String16>& args); 361 362 void clearPowerManager(); 363 364 // base for record and playback 365 class TrackBase : public ExtendedAudioBufferProvider, public RefBase { 366 367 public: 368 enum track_state { 369 IDLE, 370 TERMINATED, 371 FLUSHED, 372 STOPPED, 373 // next 2 states are currently used for fast tracks only 374 STOPPING_1, // waiting for first underrun 375 STOPPING_2, // waiting for presentation complete 376 RESUMING, 377 ACTIVE, 378 PAUSING, 379 PAUSED 380 }; 381 382 TrackBase(ThreadBase *thread, 383 const sp<Client>& client, 384 uint32_t sampleRate, 385 audio_format_t format, 386 audio_channel_mask_t channelMask, 387 int frameCount, 388 const sp<IMemory>& sharedBuffer, 389 int sessionId); 390 virtual ~TrackBase(); 391 392 virtual status_t start(AudioSystem::sync_event_t event, 393 int triggerSession) = 0; 394 virtual void stop() = 0; 395 sp<IMemory> getCblk() const { return mCblkMemory; } 396 audio_track_cblk_t* cblk() const { return mCblk; } 397 int sessionId() const { return mSessionId; } 398 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 399 400 protected: 401 TrackBase(const TrackBase&); 402 TrackBase& operator = (const TrackBase&); 403 404 // AudioBufferProvider interface 405 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) = 0; 406 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 407 408 // ExtendedAudioBufferProvider interface is only needed for Track, 409 // but putting it in TrackBase avoids the complexity of virtual inheritance 410 virtual size_t framesReady() const { return SIZE_MAX; } 411 412 audio_format_t format() const { 413 return mFormat; 414 } 415 416 int channelCount() const { return mChannelCount; } 417 418 audio_channel_mask_t channelMask() const { return mChannelMask; } 419 420 int sampleRate() const; // FIXME inline after cblk sr moved 421 422 // Return a pointer to the start of a contiguous slice of the track buffer. 423 // Parameter 'offset' is the requested start position, expressed in 424 // monotonically increasing frame units relative to the track epoch. 425 // Parameter 'frames' is the requested length, also in frame units. 426 // Always returns non-NULL. It is the caller's responsibility to 427 // verify that this will be successful; the result of calling this 428 // function with invalid 'offset' or 'frames' is undefined. 429 void* getBuffer(uint32_t offset, uint32_t frames) const; 430 431 bool isStopped() const { 432 return (mState == STOPPED || mState == FLUSHED); 433 } 434 435 // for fast tracks only 436 bool isStopping() const { 437 return mState == STOPPING_1 || mState == STOPPING_2; 438 } 439 bool isStopping_1() const { 440 return mState == STOPPING_1; 441 } 442 bool isStopping_2() const { 443 return mState == STOPPING_2; 444 } 445 446 bool isTerminated() const { 447 return mState == TERMINATED; 448 } 449 450 bool step(); 451 void reset(); 452 453 const wp<ThreadBase> mThread; 454 /*const*/ sp<Client> mClient; // see explanation at ~TrackBase() why not const 455 sp<IMemory> mCblkMemory; 456 audio_track_cblk_t* mCblk; 457 void* mBuffer; // start of track buffer, typically in shared memory 458 void* mBufferEnd; // &mBuffer[mFrameCount * frameSize], where frameSize 459 // is based on mChannelCount and 16-bit samples 460 uint32_t mFrameCount; 461 // we don't really need a lock for these 462 track_state mState; 463 const uint32_t mSampleRate; // initial sample rate only; for tracks which 464 // support dynamic rates, the current value is in control block 465 const audio_format_t mFormat; 466 bool mStepServerFailed; 467 const int mSessionId; 468 uint8_t mChannelCount; 469 audio_channel_mask_t mChannelMask; 470 Vector < sp<SyncEvent> >mSyncEvents; 471 }; 472 473 class ConfigEvent { 474 public: 475 ConfigEvent() : mEvent(0), mParam(0) {} 476 477 int mEvent; 478 int mParam; 479 }; 480 481 class PMDeathRecipient : public IBinder::DeathRecipient { 482 public: 483 PMDeathRecipient(const wp<ThreadBase>& thread) : mThread(thread) {} 484 virtual ~PMDeathRecipient() {} 485 486 // IBinder::DeathRecipient 487 virtual void binderDied(const wp<IBinder>& who); 488 489 private: 490 PMDeathRecipient(const PMDeathRecipient&); 491 PMDeathRecipient& operator = (const PMDeathRecipient&); 492 493 wp<ThreadBase> mThread; 494 }; 495 496 virtual status_t initCheck() const = 0; 497 498 // static externally-visible 499 type_t type() const { return mType; } 500 audio_io_handle_t id() const { return mId;} 501 502 // dynamic externally-visible 503 uint32_t sampleRate() const { return mSampleRate; } 504 int channelCount() const { return mChannelCount; } 505 audio_channel_mask_t channelMask() const { return mChannelMask; } 506 audio_format_t format() const { return mFormat; } 507 // Called by AudioFlinger::frameCount(audio_io_handle_t output) and effects, 508 // and returns the normal mix buffer's frame count. No API for HAL frame count. 509 size_t frameCount() const { return mNormalFrameCount; } 510 511 // Should be "virtual status_t requestExitAndWait()" and override same 512 // method in Thread, but Thread::requestExitAndWait() is not yet virtual. 513 void exit(); 514 virtual bool checkForNewParameters_l() = 0; 515 virtual status_t setParameters(const String8& keyValuePairs); 516 virtual String8 getParameters(const String8& keys) = 0; 517 virtual void audioConfigChanged_l(int event, int param = 0) = 0; 518 void sendConfigEvent(int event, int param = 0); 519 void sendConfigEvent_l(int event, int param = 0); 520 void processConfigEvents(); 521 522 // see note at declaration of mStandby and mDevice 523 bool standby() const { return mStandby; } 524 audio_devices_t device() const { return mDevice; } 525 526 virtual audio_stream_t* stream() const = 0; 527 528 sp<EffectHandle> createEffect_l( 529 const sp<AudioFlinger::Client>& client, 530 const sp<IEffectClient>& effectClient, 531 int32_t priority, 532 int sessionId, 533 effect_descriptor_t *desc, 534 int *enabled, 535 status_t *status); 536 void disconnectEffect(const sp< EffectModule>& effect, 537 EffectHandle *handle, 538 bool unpinIfLast); 539 540 // return values for hasAudioSession (bit field) 541 enum effect_state { 542 EFFECT_SESSION = 0x1, // the audio session corresponds to at least one 543 // effect 544 TRACK_SESSION = 0x2 // the audio session corresponds to at least one 545 // track 546 }; 547 548 // get effect chain corresponding to session Id. 549 sp<EffectChain> getEffectChain(int sessionId); 550 // same as getEffectChain() but must be called with ThreadBase mutex locked 551 sp<EffectChain> getEffectChain_l(int sessionId) const; 552 // add an effect chain to the chain list (mEffectChains) 553 virtual status_t addEffectChain_l(const sp<EffectChain>& chain) = 0; 554 // remove an effect chain from the chain list (mEffectChains) 555 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain) = 0; 556 // lock all effect chains Mutexes. Must be called before releasing the 557 // ThreadBase mutex before processing the mixer and effects. This guarantees the 558 // integrity of the chains during the process. 559 // Also sets the parameter 'effectChains' to current value of mEffectChains. 560 void lockEffectChains_l(Vector< sp<EffectChain> >& effectChains); 561 // unlock effect chains after process 562 void unlockEffectChains(const Vector< sp<EffectChain> >& effectChains); 563 // set audio mode to all effect chains 564 void setMode(audio_mode_t mode); 565 // get effect module with corresponding ID on specified audio session 566 sp<AudioFlinger::EffectModule> getEffect(int sessionId, int effectId); 567 sp<AudioFlinger::EffectModule> getEffect_l(int sessionId, int effectId); 568 // add and effect module. Also creates the effect chain is none exists for 569 // the effects audio session 570 status_t addEffect_l(const sp< EffectModule>& effect); 571 // remove and effect module. Also removes the effect chain is this was the last 572 // effect 573 void removeEffect_l(const sp< EffectModule>& effect); 574 // detach all tracks connected to an auxiliary effect 575 virtual void detachAuxEffect_l(int effectId) {} 576 // returns either EFFECT_SESSION if effects on this audio session exist in one 577 // chain, or TRACK_SESSION if tracks on this audio session exist, or both 578 virtual uint32_t hasAudioSession(int sessionId) const = 0; 579 // the value returned by default implementation is not important as the 580 // strategy is only meaningful for PlaybackThread which implements this method 581 virtual uint32_t getStrategyForSession_l(int sessionId) { return 0; } 582 583 // suspend or restore effect according to the type of effect passed. a NULL 584 // type pointer means suspend all effects in the session 585 void setEffectSuspended(const effect_uuid_t *type, 586 bool suspend, 587 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 588 // check if some effects must be suspended/restored when an effect is enabled 589 // or disabled 590 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 591 bool enabled, 592 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 593 void checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 594 bool enabled, 595 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 596 597 virtual status_t setSyncEvent(const sp<SyncEvent>& event) = 0; 598 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const = 0; 599 600 601 mutable Mutex mLock; 602 603 protected: 604 605 // entry describing an effect being suspended in mSuspendedSessions keyed vector 606 class SuspendedSessionDesc : public RefBase { 607 public: 608 SuspendedSessionDesc() : mRefCount(0) {} 609 610 int mRefCount; // number of active suspend requests 611 effect_uuid_t mType; // effect type UUID 612 }; 613 614 void acquireWakeLock(); 615 void acquireWakeLock_l(); 616 void releaseWakeLock(); 617 void releaseWakeLock_l(); 618 void setEffectSuspended_l(const effect_uuid_t *type, 619 bool suspend, 620 int sessionId); 621 // updated mSuspendedSessions when an effect suspended or restored 622 void updateSuspendedSessions_l(const effect_uuid_t *type, 623 bool suspend, 624 int sessionId); 625 // check if some effects must be suspended when an effect chain is added 626 void checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain); 627 628 friend class AudioFlinger; // for mEffectChains 629 630 const type_t mType; 631 632 // Used by parameters, config events, addTrack_l, exit 633 Condition mWaitWorkCV; 634 635 const sp<AudioFlinger> mAudioFlinger; 636 uint32_t mSampleRate; 637 size_t mFrameCount; // output HAL, direct output, record 638 size_t mNormalFrameCount; // normal mixer and effects 639 audio_channel_mask_t mChannelMask; 640 uint16_t mChannelCount; 641 size_t mFrameSize; 642 audio_format_t mFormat; 643 644 // Parameter sequence by client: binder thread calling setParameters(): 645 // 1. Lock mLock 646 // 2. Append to mNewParameters 647 // 3. mWaitWorkCV.signal 648 // 4. mParamCond.waitRelative with timeout 649 // 5. read mParamStatus 650 // 6. mWaitWorkCV.signal 651 // 7. Unlock 652 // 653 // Parameter sequence by server: threadLoop calling checkForNewParameters_l(): 654 // 1. Lock mLock 655 // 2. If there is an entry in mNewParameters proceed ... 656 // 2. Read first entry in mNewParameters 657 // 3. Process 658 // 4. Remove first entry from mNewParameters 659 // 5. Set mParamStatus 660 // 6. mParamCond.signal 661 // 7. mWaitWorkCV.wait with timeout (this is to avoid overwriting mParamStatus) 662 // 8. Unlock 663 Condition mParamCond; 664 Vector<String8> mNewParameters; 665 status_t mParamStatus; 666 667 Vector<ConfigEvent> mConfigEvents; 668 669 // These fields are written and read by thread itself without lock or barrier, 670 // and read by other threads without lock or barrier via standby() and device(). 671 // Because of the absence of a lock or barrier, any other thread that reads 672 // these fields must use the information in isolation, or be prepared to deal 673 // with possibility that it might be inconsistent with other information. 674 bool mStandby; // Whether thread is currently in standby. 675 audio_devices_t mDevice; // output device for PlaybackThread 676 // input + output devices for RecordThread 677 audio_source_t mAudioSource; // (see audio.h, audio_source_t) 678 679 const audio_io_handle_t mId; 680 Vector< sp<EffectChain> > mEffectChains; 681 682 static const int kNameLength = 16; // prctl(PR_SET_NAME) limit 683 char mName[kNameLength]; 684 sp<IPowerManager> mPowerManager; 685 sp<IBinder> mWakeLockToken; 686 const sp<PMDeathRecipient> mDeathRecipient; 687 // list of suspended effects per session and per type. The first vector is 688 // keyed by session ID, the second by type UUID timeLow field 689 KeyedVector< int, KeyedVector< int, sp<SuspendedSessionDesc> > > mSuspendedSessions; 690 }; 691 692 struct stream_type_t { 693 stream_type_t() 694 : volume(1.0f), 695 mute(false) 696 { 697 } 698 float volume; 699 bool mute; 700 }; 701 702 // --- PlaybackThread --- 703 class PlaybackThread : public ThreadBase { 704 public: 705 706 enum mixer_state { 707 MIXER_IDLE, // no active tracks 708 MIXER_TRACKS_ENABLED, // at least one active track, but no track has any data ready 709 MIXER_TRACKS_READY // at least one active track, and at least one track has data 710 // standby mode does not have an enum value 711 // suspend by audio policy manager is orthogonal to mixer state 712 }; 713 714 // playback track 715 class Track : public TrackBase, public VolumeProvider { 716 public: 717 Track( PlaybackThread *thread, 718 const sp<Client>& client, 719 audio_stream_type_t streamType, 720 uint32_t sampleRate, 721 audio_format_t format, 722 audio_channel_mask_t channelMask, 723 int frameCount, 724 const sp<IMemory>& sharedBuffer, 725 int sessionId, 726 IAudioFlinger::track_flags_t flags); 727 virtual ~Track(); 728 729 static void appendDumpHeader(String8& result); 730 void dump(char* buffer, size_t size); 731 virtual status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 732 int triggerSession = 0); 733 virtual void stop(); 734 void pause(); 735 736 void flush(); 737 void destroy(); 738 void mute(bool); 739 int name() const { return mName; } 740 741 audio_stream_type_t streamType() const { 742 return mStreamType; 743 } 744 status_t attachAuxEffect(int EffectId); 745 void setAuxBuffer(int EffectId, int32_t *buffer); 746 int32_t *auxBuffer() const { return mAuxBuffer; } 747 void setMainBuffer(int16_t *buffer) { mMainBuffer = buffer; } 748 int16_t *mainBuffer() const { return mMainBuffer; } 749 int auxEffectId() const { return mAuxEffectId; } 750 751 // implement FastMixerState::VolumeProvider interface 752 virtual uint32_t getVolumeLR(); 753 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 754 755 protected: 756 // for numerous 757 friend class PlaybackThread; 758 friend class MixerThread; 759 friend class DirectOutputThread; 760 761 Track(const Track&); 762 Track& operator = (const Track&); 763 764 // AudioBufferProvider interface 765 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts = kInvalidPTS); 766 // releaseBuffer() not overridden 767 768 virtual size_t framesReady() const; 769 770 bool isMuted() const { return mMute; } 771 bool isPausing() const { 772 return mState == PAUSING; 773 } 774 bool isPaused() const { 775 return mState == PAUSED; 776 } 777 bool isResuming() const { 778 return mState == RESUMING; 779 } 780 bool isReady() const; 781 void setPaused() { mState = PAUSED; } 782 void reset(); 783 784 bool isOutputTrack() const { 785 return (mStreamType == AUDIO_STREAM_CNT); 786 } 787 788 sp<IMemory> sharedBuffer() const { return mSharedBuffer; } 789 790 bool presentationComplete(size_t framesWritten, size_t audioHalFrames); 791 792 public: 793 void triggerEvents(AudioSystem::sync_event_t type); 794 virtual bool isTimedTrack() const { return false; } 795 bool isFastTrack() const { return (mFlags & IAudioFlinger::TRACK_FAST) != 0; } 796 797 protected: 798 799 // written by Track::mute() called by binder thread(s), without a mutex or barrier. 800 // read by Track::isMuted() called by playback thread, also without a mutex or barrier. 801 // The lack of mutex or barrier is safe because the mute status is only used by itself. 802 bool mMute; 803 804 // FILLED state is used for suppressing volume ramp at begin of playing 805 enum {FS_INVALID, FS_FILLING, FS_FILLED, FS_ACTIVE}; 806 mutable uint8_t mFillingUpStatus; 807 int8_t mRetryCount; 808 const sp<IMemory> mSharedBuffer; 809 bool mResetDone; 810 const audio_stream_type_t mStreamType; 811 int mName; // track name on the normal mixer, 812 // allocated statically at track creation time, 813 // and is even allocated (though unused) for fast tracks 814 // FIXME don't allocate track name for fast tracks 815 int16_t *mMainBuffer; 816 int32_t *mAuxBuffer; 817 int mAuxEffectId; 818 bool mHasVolumeController; 819 size_t mPresentationCompleteFrames; // number of frames written to the audio HAL 820 // when this track will be fully rendered 821 private: 822 IAudioFlinger::track_flags_t mFlags; 823 824 // The following fields are only for fast tracks, and should be in a subclass 825 int mFastIndex; // index within FastMixerState::mFastTracks[]; 826 // either mFastIndex == -1 if not isFastTrack() 827 // or 0 < mFastIndex < FastMixerState::kMaxFast because 828 // index 0 is reserved for normal mixer's submix; 829 // index is allocated statically at track creation time 830 // but the slot is only used if track is active 831 FastTrackUnderruns mObservedUnderruns; // Most recently observed value of 832 // mFastMixerDumpState.mTracks[mFastIndex].mUnderruns 833 uint32_t mUnderrunCount; // Counter of total number of underruns, never reset 834 volatile float mCachedVolume; // combined master volume and stream type volume; 835 // 'volatile' means accessed without lock or 836 // barrier, but is read/written atomically 837 }; // end of Track 838 839 class TimedTrack : public Track { 840 public: 841 static sp<TimedTrack> create(PlaybackThread *thread, 842 const sp<Client>& client, 843 audio_stream_type_t streamType, 844 uint32_t sampleRate, 845 audio_format_t format, 846 audio_channel_mask_t channelMask, 847 int frameCount, 848 const sp<IMemory>& sharedBuffer, 849 int sessionId); 850 virtual ~TimedTrack(); 851 852 class TimedBuffer { 853 public: 854 TimedBuffer(); 855 TimedBuffer(const sp<IMemory>& buffer, int64_t pts); 856 const sp<IMemory>& buffer() const { return mBuffer; } 857 int64_t pts() const { return mPTS; } 858 uint32_t position() const { return mPosition; } 859 void setPosition(uint32_t pos) { mPosition = pos; } 860 private: 861 sp<IMemory> mBuffer; 862 int64_t mPTS; 863 uint32_t mPosition; 864 }; 865 866 // Mixer facing methods. 867 virtual bool isTimedTrack() const { return true; } 868 virtual size_t framesReady() const; 869 870 // AudioBufferProvider interface 871 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, 872 int64_t pts); 873 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 874 875 // Client/App facing methods. 876 status_t allocateTimedBuffer(size_t size, 877 sp<IMemory>* buffer); 878 status_t queueTimedBuffer(const sp<IMemory>& buffer, 879 int64_t pts); 880 status_t setMediaTimeTransform(const LinearTransform& xform, 881 TimedAudioTrack::TargetTimeline target); 882 883 private: 884 TimedTrack(PlaybackThread *thread, 885 const sp<Client>& client, 886 audio_stream_type_t streamType, 887 uint32_t sampleRate, 888 audio_format_t format, 889 audio_channel_mask_t channelMask, 890 int frameCount, 891 const sp<IMemory>& sharedBuffer, 892 int sessionId); 893 894 void timedYieldSamples_l(AudioBufferProvider::Buffer* buffer); 895 void timedYieldSilence_l(uint32_t numFrames, 896 AudioBufferProvider::Buffer* buffer); 897 void trimTimedBufferQueue_l(); 898 void trimTimedBufferQueueHead_l(const char* logTag); 899 void updateFramesPendingAfterTrim_l(const TimedBuffer& buf, 900 const char* logTag); 901 902 uint64_t mLocalTimeFreq; 903 LinearTransform mLocalTimeToSampleTransform; 904 LinearTransform mMediaTimeToSampleTransform; 905 sp<MemoryDealer> mTimedMemoryDealer; 906 907 Vector<TimedBuffer> mTimedBufferQueue; 908 bool mQueueHeadInFlight; 909 bool mTrimQueueHeadOnRelease; 910 uint32_t mFramesPendingInQueue; 911 912 uint8_t* mTimedSilenceBuffer; 913 uint32_t mTimedSilenceBufferSize; 914 mutable Mutex mTimedBufferQueueLock; 915 bool mTimedAudioOutputOnTime; 916 CCHelper mCCHelper; 917 918 Mutex mMediaTimeTransformLock; 919 LinearTransform mMediaTimeTransform; 920 bool mMediaTimeTransformValid; 921 TimedAudioTrack::TargetTimeline mMediaTimeTransformTarget; 922 }; 923 924 925 // playback track 926 class OutputTrack : public Track { 927 public: 928 929 class Buffer: public AudioBufferProvider::Buffer { 930 public: 931 int16_t *mBuffer; 932 }; 933 934 OutputTrack(PlaybackThread *thread, 935 DuplicatingThread *sourceThread, 936 uint32_t sampleRate, 937 audio_format_t format, 938 audio_channel_mask_t channelMask, 939 int frameCount); 940 virtual ~OutputTrack(); 941 942 virtual status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 943 int triggerSession = 0); 944 virtual void stop(); 945 bool write(int16_t* data, uint32_t frames); 946 bool bufferQueueEmpty() const { return mBufferQueue.size() == 0; } 947 bool isActive() const { return mActive; } 948 const wp<ThreadBase>& thread() const { return mThread; } 949 950 private: 951 952 enum { 953 NO_MORE_BUFFERS = 0x80000001, // same in AudioTrack.h, ok to be different value 954 }; 955 956 status_t obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs); 957 void clearBufferQueue(); 958 959 // Maximum number of pending buffers allocated by OutputTrack::write() 960 static const uint8_t kMaxOverFlowBuffers = 10; 961 962 Vector < Buffer* > mBufferQueue; 963 AudioBufferProvider::Buffer mOutBuffer; 964 bool mActive; 965 DuplicatingThread* const mSourceThread; // for waitTimeMs() in write() 966 }; // end of OutputTrack 967 968 PlaybackThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 969 audio_io_handle_t id, audio_devices_t device, type_t type); 970 virtual ~PlaybackThread(); 971 972 void dump(int fd, const Vector<String16>& args); 973 974 // Thread virtuals 975 virtual status_t readyToRun(); 976 virtual bool threadLoop(); 977 978 // RefBase 979 virtual void onFirstRef(); 980 981protected: 982 // Code snippets that were lifted up out of threadLoop() 983 virtual void threadLoop_mix() = 0; 984 virtual void threadLoop_sleepTime() = 0; 985 virtual void threadLoop_write(); 986 virtual void threadLoop_standby(); 987 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 988 989 // prepareTracks_l reads and writes mActiveTracks, and returns 990 // the pending set of tracks to remove via Vector 'tracksToRemove'. The caller 991 // is responsible for clearing or destroying this Vector later on, when it 992 // is safe to do so. That will drop the final ref count and destroy the tracks. 993 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove) = 0; 994 995public: 996 997 virtual status_t initCheck() const { return (mOutput == NULL) ? NO_INIT : NO_ERROR; } 998 999 // return estimated latency in milliseconds, as reported by HAL 1000 uint32_t latency() const; 1001 // same, but lock must already be held 1002 uint32_t latency_l() const; 1003 1004 void setMasterVolume(float value); 1005 void setMasterMute(bool muted); 1006 1007 void setStreamVolume(audio_stream_type_t stream, float value); 1008 void setStreamMute(audio_stream_type_t stream, bool muted); 1009 1010 float streamVolume(audio_stream_type_t stream) const; 1011 1012 sp<Track> createTrack_l( 1013 const sp<AudioFlinger::Client>& client, 1014 audio_stream_type_t streamType, 1015 uint32_t sampleRate, 1016 audio_format_t format, 1017 audio_channel_mask_t channelMask, 1018 int frameCount, 1019 const sp<IMemory>& sharedBuffer, 1020 int sessionId, 1021 IAudioFlinger::track_flags_t flags, 1022 pid_t tid, 1023 status_t *status); 1024 1025 AudioStreamOut* getOutput() const; 1026 AudioStreamOut* clearOutput(); 1027 virtual audio_stream_t* stream() const; 1028 1029 // a very large number of suspend() will eventually wraparound, but unlikely 1030 void suspend() { (void) android_atomic_inc(&mSuspended); } 1031 void restore() 1032 { 1033 // if restore() is done without suspend(), get back into 1034 // range so that the next suspend() will operate correctly 1035 if (android_atomic_dec(&mSuspended) <= 0) { 1036 android_atomic_release_store(0, &mSuspended); 1037 } 1038 } 1039 bool isSuspended() const 1040 { return android_atomic_acquire_load(&mSuspended) > 0; } 1041 1042 virtual String8 getParameters(const String8& keys); 1043 virtual void audioConfigChanged_l(int event, int param = 0); 1044 status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames); 1045 int16_t *mixBuffer() const { return mMixBuffer; }; 1046 1047 virtual void detachAuxEffect_l(int effectId); 1048 status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track> track, 1049 int EffectId); 1050 status_t attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track> track, 1051 int EffectId); 1052 1053 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1054 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1055 virtual uint32_t hasAudioSession(int sessionId) const; 1056 virtual uint32_t getStrategyForSession_l(int sessionId); 1057 1058 1059 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 1060 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 1061 void invalidateTracks(audio_stream_type_t streamType); 1062 1063 1064 protected: 1065 int16_t* mMixBuffer; 1066 1067 // suspend count, > 0 means suspended. While suspended, the thread continues to pull from 1068 // tracks and mix, but doesn't write to HAL. A2DP and SCO HAL implementations can't handle 1069 // concurrent use of both of them, so Audio Policy Service suspends one of the threads to 1070 // workaround that restriction. 1071 // 'volatile' means accessed via atomic operations and no lock. 1072 volatile int32_t mSuspended; 1073 1074 int mBytesWritten; 1075 private: 1076 // mMasterMute is in both PlaybackThread and in AudioFlinger. When a 1077 // PlaybackThread needs to find out if master-muted, it checks it's local 1078 // copy rather than the one in AudioFlinger. This optimization saves a lock. 1079 bool mMasterMute; 1080 void setMasterMute_l(bool muted) { mMasterMute = muted; } 1081 protected: 1082 SortedVector< wp<Track> > mActiveTracks; // FIXME check if this could be sp<> 1083 1084 // Allocate a track name for a given channel mask. 1085 // Returns name >= 0 if successful, -1 on failure. 1086 virtual int getTrackName_l(audio_channel_mask_t channelMask) = 0; 1087 virtual void deleteTrackName_l(int name) = 0; 1088 1089 // Time to sleep between cycles when: 1090 virtual uint32_t activeSleepTimeUs() const; // mixer state MIXER_TRACKS_ENABLED 1091 virtual uint32_t idleSleepTimeUs() const = 0; // mixer state MIXER_IDLE 1092 virtual uint32_t suspendSleepTimeUs() const = 0; // audio policy manager suspended us 1093 // No sleep when mixer state == MIXER_TRACKS_READY; relies on audio HAL stream->write() 1094 // No sleep in standby mode; waits on a condition 1095 1096 // Code snippets that are temporarily lifted up out of threadLoop() until the merge 1097 void checkSilentMode_l(); 1098 1099 // Non-trivial for DUPLICATING only 1100 virtual void saveOutputTracks() { } 1101 virtual void clearOutputTracks() { } 1102 1103 // Cache various calculated values, at threadLoop() entry and after a parameter change 1104 virtual void cacheParameters_l(); 1105 1106 virtual uint32_t correctLatency(uint32_t latency) const; 1107 1108 private: 1109 1110 friend class AudioFlinger; // for numerous 1111 1112 PlaybackThread(const Client&); 1113 PlaybackThread& operator = (const PlaybackThread&); 1114 1115 status_t addTrack_l(const sp<Track>& track); 1116 void destroyTrack_l(const sp<Track>& track); 1117 void removeTrack_l(const sp<Track>& track); 1118 1119 void readOutputParameters(); 1120 1121 virtual void dumpInternals(int fd, const Vector<String16>& args); 1122 void dumpTracks(int fd, const Vector<String16>& args); 1123 1124 SortedVector< sp<Track> > mTracks; 1125 // mStreamTypes[] uses 1 additional stream type internally for the OutputTrack used by DuplicatingThread 1126 stream_type_t mStreamTypes[AUDIO_STREAM_CNT + 1]; 1127 AudioStreamOut *mOutput; 1128 1129 float mMasterVolume; 1130 nsecs_t mLastWriteTime; 1131 int mNumWrites; 1132 int mNumDelayedWrites; 1133 bool mInWrite; 1134 1135 // FIXME rename these former local variables of threadLoop to standard "m" names 1136 nsecs_t standbyTime; 1137 size_t mixBufferSize; 1138 1139 // cached copies of activeSleepTimeUs() and idleSleepTimeUs() made by cacheParameters_l() 1140 uint32_t activeSleepTime; 1141 uint32_t idleSleepTime; 1142 1143 uint32_t sleepTime; 1144 1145 // mixer status returned by prepareTracks_l() 1146 mixer_state mMixerStatus; // current cycle 1147 // previous cycle when in prepareTracks_l() 1148 mixer_state mMixerStatusIgnoringFastTracks; 1149 // FIXME or a separate ready state per track 1150 1151 // FIXME move these declarations into the specific sub-class that needs them 1152 // MIXER only 1153 uint32_t sleepTimeShift; 1154 1155 // same as AudioFlinger::mStandbyTimeInNsecs except for DIRECT which uses a shorter value 1156 nsecs_t standbyDelay; 1157 1158 // MIXER only 1159 nsecs_t maxPeriod; 1160 1161 // DUPLICATING only 1162 uint32_t writeFrames; 1163 1164 private: 1165 // The HAL output sink is treated as non-blocking, but current implementation is blocking 1166 sp<NBAIO_Sink> mOutputSink; 1167 // If a fast mixer is present, the blocking pipe sink, otherwise clear 1168 sp<NBAIO_Sink> mPipeSink; 1169 // The current sink for the normal mixer to write it's (sub)mix, mOutputSink or mPipeSink 1170 sp<NBAIO_Sink> mNormalSink; 1171 // For dumpsys 1172 sp<NBAIO_Sink> mTeeSink; 1173 sp<NBAIO_Source> mTeeSource; 1174 uint32_t mScreenState; // cached copy of gScreenState 1175 public: 1176 virtual bool hasFastMixer() const = 0; 1177 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const 1178 { FastTrackUnderruns dummy; return dummy; } 1179 1180 protected: 1181 // accessed by both binder threads and within threadLoop(), lock on mutex needed 1182 unsigned mFastTrackAvailMask; // bit i set if fast track [i] is available 1183 1184 }; 1185 1186 class MixerThread : public PlaybackThread { 1187 public: 1188 MixerThread (const sp<AudioFlinger>& audioFlinger, 1189 AudioStreamOut* output, 1190 audio_io_handle_t id, 1191 audio_devices_t device, 1192 type_t type = MIXER); 1193 virtual ~MixerThread(); 1194 1195 // Thread virtuals 1196 1197 virtual bool checkForNewParameters_l(); 1198 virtual void dumpInternals(int fd, const Vector<String16>& args); 1199 1200 protected: 1201 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1202 virtual int getTrackName_l(audio_channel_mask_t channelMask); 1203 virtual void deleteTrackName_l(int name); 1204 virtual uint32_t idleSleepTimeUs() const; 1205 virtual uint32_t suspendSleepTimeUs() const; 1206 virtual void cacheParameters_l(); 1207 1208 // threadLoop snippets 1209 virtual void threadLoop_write(); 1210 virtual void threadLoop_standby(); 1211 virtual void threadLoop_mix(); 1212 virtual void threadLoop_sleepTime(); 1213 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 1214 virtual uint32_t correctLatency(uint32_t latency) const; 1215 1216 AudioMixer* mAudioMixer; // normal mixer 1217 private: 1218 // one-time initialization, no locks required 1219 FastMixer* mFastMixer; // non-NULL if there is also a fast mixer 1220 sp<AudioWatchdog> mAudioWatchdog; // non-0 if there is an audio watchdog thread 1221 1222 // contents are not guaranteed to be consistent, no locks required 1223 FastMixerDumpState mFastMixerDumpState; 1224#ifdef STATE_QUEUE_DUMP 1225 StateQueueObserverDump mStateQueueObserverDump; 1226 StateQueueMutatorDump mStateQueueMutatorDump; 1227#endif 1228 AudioWatchdogDump mAudioWatchdogDump; 1229 1230 // accessible only within the threadLoop(), no locks required 1231 // mFastMixer->sq() // for mutating and pushing state 1232 int32_t mFastMixerFutex; // for cold idle 1233 1234 public: 1235 virtual bool hasFastMixer() const { return mFastMixer != NULL; } 1236 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const { 1237 ALOG_ASSERT(fastIndex < FastMixerState::kMaxFastTracks); 1238 return mFastMixerDumpState.mTracks[fastIndex].mUnderruns; 1239 } 1240 }; 1241 1242 class DirectOutputThread : public PlaybackThread { 1243 public: 1244 1245 DirectOutputThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1246 audio_io_handle_t id, audio_devices_t device); 1247 virtual ~DirectOutputThread(); 1248 1249 // Thread virtuals 1250 1251 virtual bool checkForNewParameters_l(); 1252 1253 protected: 1254 virtual int getTrackName_l(audio_channel_mask_t channelMask); 1255 virtual void deleteTrackName_l(int name); 1256 virtual uint32_t activeSleepTimeUs() const; 1257 virtual uint32_t idleSleepTimeUs() const; 1258 virtual uint32_t suspendSleepTimeUs() const; 1259 virtual void cacheParameters_l(); 1260 1261 // threadLoop snippets 1262 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1263 virtual void threadLoop_mix(); 1264 virtual void threadLoop_sleepTime(); 1265 1266 // volumes last sent to audio HAL with stream->set_volume() 1267 float mLeftVolFloat; 1268 float mRightVolFloat; 1269 1270private: 1271 // prepareTracks_l() tells threadLoop_mix() the name of the single active track 1272 sp<Track> mActiveTrack; 1273 public: 1274 virtual bool hasFastMixer() const { return false; } 1275 }; 1276 1277 class DuplicatingThread : public MixerThread { 1278 public: 1279 DuplicatingThread (const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread, 1280 audio_io_handle_t id); 1281 virtual ~DuplicatingThread(); 1282 1283 // Thread virtuals 1284 void addOutputTrack(MixerThread* thread); 1285 void removeOutputTrack(MixerThread* thread); 1286 uint32_t waitTimeMs() const { return mWaitTimeMs; } 1287 protected: 1288 virtual uint32_t activeSleepTimeUs() const; 1289 1290 private: 1291 bool outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks); 1292 protected: 1293 // threadLoop snippets 1294 virtual void threadLoop_mix(); 1295 virtual void threadLoop_sleepTime(); 1296 virtual void threadLoop_write(); 1297 virtual void threadLoop_standby(); 1298 virtual void cacheParameters_l(); 1299 1300 private: 1301 // called from threadLoop, addOutputTrack, removeOutputTrack 1302 virtual void updateWaitTime_l(); 1303 protected: 1304 virtual void saveOutputTracks(); 1305 virtual void clearOutputTracks(); 1306 private: 1307 1308 uint32_t mWaitTimeMs; 1309 SortedVector < sp<OutputTrack> > outputTracks; 1310 SortedVector < sp<OutputTrack> > mOutputTracks; 1311 public: 1312 virtual bool hasFastMixer() const { return false; } 1313 }; 1314 1315 PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const; 1316 MixerThread *checkMixerThread_l(audio_io_handle_t output) const; 1317 RecordThread *checkRecordThread_l(audio_io_handle_t input) const; 1318 // no range check, AudioFlinger::mLock held 1319 bool streamMute_l(audio_stream_type_t stream) const 1320 { return mStreamTypes[stream].mute; } 1321 // no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held 1322 float streamVolume_l(audio_stream_type_t stream) const 1323 { return mStreamTypes[stream].volume; } 1324 void audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2); 1325 1326 // allocate an audio_io_handle_t, session ID, or effect ID 1327 uint32_t nextUniqueId(); 1328 1329 status_t moveEffectChain_l(int sessionId, 1330 PlaybackThread *srcThread, 1331 PlaybackThread *dstThread, 1332 bool reRegister); 1333 // return thread associated with primary hardware device, or NULL 1334 PlaybackThread *primaryPlaybackThread_l() const; 1335 audio_devices_t primaryOutputDevice_l() const; 1336 1337 sp<PlaybackThread> getEffectThread_l(int sessionId, int EffectId); 1338 1339 // server side of the client's IAudioTrack 1340 class TrackHandle : public android::BnAudioTrack { 1341 public: 1342 TrackHandle(const sp<PlaybackThread::Track>& track); 1343 virtual ~TrackHandle(); 1344 virtual sp<IMemory> getCblk() const; 1345 virtual status_t start(); 1346 virtual void stop(); 1347 virtual void flush(); 1348 virtual void mute(bool); 1349 virtual void pause(); 1350 virtual status_t attachAuxEffect(int effectId); 1351 virtual status_t allocateTimedBuffer(size_t size, 1352 sp<IMemory>* buffer); 1353 virtual status_t queueTimedBuffer(const sp<IMemory>& buffer, 1354 int64_t pts); 1355 virtual status_t setMediaTimeTransform(const LinearTransform& xform, 1356 int target); 1357 virtual status_t onTransact( 1358 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 1359 private: 1360 const sp<PlaybackThread::Track> mTrack; 1361 }; 1362 1363 void removeClient_l(pid_t pid); 1364 void removeNotificationClient(pid_t pid); 1365 1366 1367 // record thread 1368 class RecordThread : public ThreadBase, public AudioBufferProvider 1369 // derives from AudioBufferProvider interface for use by resampler 1370 { 1371 public: 1372 1373 // record track 1374 class RecordTrack : public TrackBase { 1375 public: 1376 RecordTrack(RecordThread *thread, 1377 const sp<Client>& client, 1378 uint32_t sampleRate, 1379 audio_format_t format, 1380 audio_channel_mask_t channelMask, 1381 int frameCount, 1382 int sessionId); 1383 virtual ~RecordTrack(); 1384 1385 virtual status_t start(AudioSystem::sync_event_t event, int triggerSession); 1386 virtual void stop(); 1387 1388 void destroy(); 1389 1390 // clear the buffer overflow flag 1391 void clearOverflow() { mOverflow = false; } 1392 // set the buffer overflow flag and return previous value 1393 bool setOverflow() { bool tmp = mOverflow; mOverflow = true; return tmp; } 1394 1395 static void appendDumpHeader(String8& result); 1396 void dump(char* buffer, size_t size); 1397 1398 private: 1399 friend class AudioFlinger; // for mState 1400 1401 RecordTrack(const RecordTrack&); 1402 RecordTrack& operator = (const RecordTrack&); 1403 1404 // AudioBufferProvider interface 1405 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts = kInvalidPTS); 1406 // releaseBuffer() not overridden 1407 1408 bool mOverflow; // overflow on most recent attempt to fill client buffer 1409 }; 1410 1411 RecordThread(const sp<AudioFlinger>& audioFlinger, 1412 AudioStreamIn *input, 1413 uint32_t sampleRate, 1414 audio_channel_mask_t channelMask, 1415 audio_io_handle_t id, 1416 audio_devices_t device); 1417 virtual ~RecordThread(); 1418 1419 // no addTrack_l ? 1420 void destroyTrack_l(const sp<RecordTrack>& track); 1421 void removeTrack_l(const sp<RecordTrack>& track); 1422 1423 void dumpInternals(int fd, const Vector<String16>& args); 1424 void dumpTracks(int fd, const Vector<String16>& args); 1425 1426 // Thread virtuals 1427 virtual bool threadLoop(); 1428 virtual status_t readyToRun(); 1429 1430 // RefBase 1431 virtual void onFirstRef(); 1432 1433 virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; } 1434 sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l( 1435 const sp<AudioFlinger::Client>& client, 1436 uint32_t sampleRate, 1437 audio_format_t format, 1438 audio_channel_mask_t channelMask, 1439 int frameCount, 1440 int sessionId, 1441 IAudioFlinger::track_flags_t flags, 1442 pid_t tid, 1443 status_t *status); 1444 1445 status_t start(RecordTrack* recordTrack, 1446 AudioSystem::sync_event_t event, 1447 int triggerSession); 1448 1449 // ask the thread to stop the specified track, and 1450 // return true if the caller should then do it's part of the stopping process 1451 bool stop_l(RecordTrack* recordTrack); 1452 1453 void dump(int fd, const Vector<String16>& args); 1454 AudioStreamIn* clearInput(); 1455 virtual audio_stream_t* stream() const; 1456 1457 // AudioBufferProvider interface 1458 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts); 1459 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 1460 1461 virtual bool checkForNewParameters_l(); 1462 virtual String8 getParameters(const String8& keys); 1463 virtual void audioConfigChanged_l(int event, int param = 0); 1464 void readInputParameters(); 1465 virtual unsigned int getInputFramesLost(); 1466 1467 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1468 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1469 virtual uint32_t hasAudioSession(int sessionId) const; 1470 1471 // Return the set of unique session IDs across all tracks. 1472 // The keys are the session IDs, and the associated values are meaningless. 1473 // FIXME replace by Set [and implement Bag/Multiset for other uses]. 1474 KeyedVector<int, bool> sessionIds() const; 1475 1476 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 1477 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 1478 1479 static void syncStartEventCallback(const wp<SyncEvent>& event); 1480 void handleSyncStartEvent(const sp<SyncEvent>& event); 1481 1482 private: 1483 void clearSyncStartEvent(); 1484 1485 // Enter standby if not already in standby, and set mStandby flag 1486 void standby(); 1487 1488 // Call the HAL standby method unconditionally, and don't change mStandby flag 1489 void inputStandBy(); 1490 1491 AudioStreamIn *mInput; 1492 SortedVector < sp<RecordTrack> > mTracks; 1493 // mActiveTrack has dual roles: it indicates the current active track, and 1494 // is used together with mStartStopCond to indicate start()/stop() progress 1495 sp<RecordTrack> mActiveTrack; 1496 Condition mStartStopCond; 1497 AudioResampler *mResampler; 1498 int32_t *mRsmpOutBuffer; 1499 int16_t *mRsmpInBuffer; 1500 size_t mRsmpInIndex; 1501 size_t mInputBytes; 1502 const int mReqChannelCount; 1503 const uint32_t mReqSampleRate; 1504 ssize_t mBytesRead; 1505 // sync event triggering actual audio capture. Frames read before this event will 1506 // be dropped and therefore not read by the application. 1507 sp<SyncEvent> mSyncStartEvent; 1508 // number of captured frames to drop after the start sync event has been received. 1509 // when < 0, maximum frames to drop before starting capture even if sync event is 1510 // not received 1511 ssize_t mFramestoDrop; 1512 }; 1513 1514 // server side of the client's IAudioRecord 1515 class RecordHandle : public android::BnAudioRecord { 1516 public: 1517 RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack); 1518 virtual ~RecordHandle(); 1519 virtual sp<IMemory> getCblk() const; 1520 virtual status_t start(int /*AudioSystem::sync_event_t*/ event, int triggerSession); 1521 virtual void stop(); 1522 virtual status_t onTransact( 1523 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 1524 private: 1525 const sp<RecordThread::RecordTrack> mRecordTrack; 1526 1527 // for use from destructor 1528 void stop_nonvirtual(); 1529 }; 1530 1531 //--- Audio Effect Management 1532 1533 // EffectModule and EffectChain classes both have their own mutex to protect 1534 // state changes or resource modifications. Always respect the following order 1535 // if multiple mutexes must be acquired to avoid cross deadlock: 1536 // AudioFlinger -> ThreadBase -> EffectChain -> EffectModule 1537 1538 // The EffectModule class is a wrapper object controlling the effect engine implementation 1539 // in the effect library. It prevents concurrent calls to process() and command() functions 1540 // from different client threads. It keeps a list of EffectHandle objects corresponding 1541 // to all client applications using this effect and notifies applications of effect state, 1542 // control or parameter changes. It manages the activation state machine to send appropriate 1543 // reset, enable, disable commands to effect engine and provide volume 1544 // ramping when effects are activated/deactivated. 1545 // When controlling an auxiliary effect, the EffectModule also provides an input buffer used by 1546 // the attached track(s) to accumulate their auxiliary channel. 1547 class EffectModule: public RefBase { 1548 public: 1549 EffectModule(ThreadBase *thread, 1550 const wp<AudioFlinger::EffectChain>& chain, 1551 effect_descriptor_t *desc, 1552 int id, 1553 int sessionId); 1554 virtual ~EffectModule(); 1555 1556 enum effect_state { 1557 IDLE, 1558 RESTART, 1559 STARTING, 1560 ACTIVE, 1561 STOPPING, 1562 STOPPED, 1563 DESTROYED 1564 }; 1565 1566 int id() const { return mId; } 1567 void process(); 1568 void updateState(); 1569 status_t command(uint32_t cmdCode, 1570 uint32_t cmdSize, 1571 void *pCmdData, 1572 uint32_t *replySize, 1573 void *pReplyData); 1574 1575 void reset_l(); 1576 status_t configure(); 1577 status_t init(); 1578 effect_state state() const { 1579 return mState; 1580 } 1581 uint32_t status() { 1582 return mStatus; 1583 } 1584 int sessionId() const { 1585 return mSessionId; 1586 } 1587 status_t setEnabled(bool enabled); 1588 status_t setEnabled_l(bool enabled); 1589 bool isEnabled() const; 1590 bool isProcessEnabled() const; 1591 1592 void setInBuffer(int16_t *buffer) { mConfig.inputCfg.buffer.s16 = buffer; } 1593 int16_t *inBuffer() { return mConfig.inputCfg.buffer.s16; } 1594 void setOutBuffer(int16_t *buffer) { mConfig.outputCfg.buffer.s16 = buffer; } 1595 int16_t *outBuffer() { return mConfig.outputCfg.buffer.s16; } 1596 void setChain(const wp<EffectChain>& chain) { mChain = chain; } 1597 void setThread(const wp<ThreadBase>& thread) { mThread = thread; } 1598 const wp<ThreadBase>& thread() { return mThread; } 1599 1600 status_t addHandle(EffectHandle *handle); 1601 size_t disconnect(EffectHandle *handle, bool unpinIfLast); 1602 size_t removeHandle(EffectHandle *handle); 1603 1604 const effect_descriptor_t& desc() const { return mDescriptor; } 1605 wp<EffectChain>& chain() { return mChain; } 1606 1607 status_t setDevice(audio_devices_t device); 1608 status_t setVolume(uint32_t *left, uint32_t *right, bool controller); 1609 status_t setMode(audio_mode_t mode); 1610 status_t setAudioSource(audio_source_t source); 1611 status_t start(); 1612 status_t stop(); 1613 void setSuspended(bool suspended); 1614 bool suspended() const; 1615 1616 EffectHandle* controlHandle_l(); 1617 1618 bool isPinned() const { return mPinned; } 1619 void unPin() { mPinned = false; } 1620 bool purgeHandles(); 1621 void lock() { mLock.lock(); } 1622 void unlock() { mLock.unlock(); } 1623 1624 void dump(int fd, const Vector<String16>& args); 1625 1626 protected: 1627 friend class AudioFlinger; // for mHandles 1628 bool mPinned; 1629 1630 // Maximum time allocated to effect engines to complete the turn off sequence 1631 static const uint32_t MAX_DISABLE_TIME_MS = 10000; 1632 1633 EffectModule(const EffectModule&); 1634 EffectModule& operator = (const EffectModule&); 1635 1636 status_t start_l(); 1637 status_t stop_l(); 1638 1639mutable Mutex mLock; // mutex for process, commands and handles list protection 1640 wp<ThreadBase> mThread; // parent thread 1641 wp<EffectChain> mChain; // parent effect chain 1642 const int mId; // this instance unique ID 1643 const int mSessionId; // audio session ID 1644 const effect_descriptor_t mDescriptor;// effect descriptor received from effect engine 1645 effect_config_t mConfig; // input and output audio configuration 1646 effect_handle_t mEffectInterface; // Effect module C API 1647 status_t mStatus; // initialization status 1648 effect_state mState; // current activation state 1649 Vector<EffectHandle *> mHandles; // list of client handles 1650 // First handle in mHandles has highest priority and controls the effect module 1651 uint32_t mMaxDisableWaitCnt; // maximum grace period before forcing an effect off after 1652 // sending disable command. 1653 uint32_t mDisableWaitCnt; // current process() calls count during disable period. 1654 bool mSuspended; // effect is suspended: temporarily disabled by framework 1655 }; 1656 1657 // The EffectHandle class implements the IEffect interface. It provides resources 1658 // to receive parameter updates, keeps track of effect control 1659 // ownership and state and has a pointer to the EffectModule object it is controlling. 1660 // There is one EffectHandle object for each application controlling (or using) 1661 // an effect module. 1662 // The EffectHandle is obtained by calling AudioFlinger::createEffect(). 1663 class EffectHandle: public android::BnEffect { 1664 public: 1665 1666 EffectHandle(const sp<EffectModule>& effect, 1667 const sp<AudioFlinger::Client>& client, 1668 const sp<IEffectClient>& effectClient, 1669 int32_t priority); 1670 virtual ~EffectHandle(); 1671 1672 // IEffect 1673 virtual status_t enable(); 1674 virtual status_t disable(); 1675 virtual status_t command(uint32_t cmdCode, 1676 uint32_t cmdSize, 1677 void *pCmdData, 1678 uint32_t *replySize, 1679 void *pReplyData); 1680 virtual void disconnect(); 1681 private: 1682 void disconnect(bool unpinIfLast); 1683 public: 1684 virtual sp<IMemory> getCblk() const { return mCblkMemory; } 1685 virtual status_t onTransact(uint32_t code, const Parcel& data, 1686 Parcel* reply, uint32_t flags); 1687 1688 1689 // Give or take control of effect module 1690 // - hasControl: true if control is given, false if removed 1691 // - signal: true client app should be signaled of change, false otherwise 1692 // - enabled: state of the effect when control is passed 1693 void setControl(bool hasControl, bool signal, bool enabled); 1694 void commandExecuted(uint32_t cmdCode, 1695 uint32_t cmdSize, 1696 void *pCmdData, 1697 uint32_t replySize, 1698 void *pReplyData); 1699 void setEnabled(bool enabled); 1700 bool enabled() const { return mEnabled; } 1701 1702 // Getters 1703 int id() const { return mEffect->id(); } 1704 int priority() const { return mPriority; } 1705 bool hasControl() const { return mHasControl; } 1706 sp<EffectModule> effect() const { return mEffect; } 1707 // destroyed_l() must be called with the associated EffectModule mLock held 1708 bool destroyed_l() const { return mDestroyed; } 1709 1710 void dump(char* buffer, size_t size); 1711 1712 protected: 1713 friend class AudioFlinger; // for mEffect, mHasControl, mEnabled 1714 EffectHandle(const EffectHandle&); 1715 EffectHandle& operator =(const EffectHandle&); 1716 1717 sp<EffectModule> mEffect; // pointer to controlled EffectModule 1718 sp<IEffectClient> mEffectClient; // callback interface for client notifications 1719 /*const*/ sp<Client> mClient; // client for shared memory allocation, see disconnect() 1720 sp<IMemory> mCblkMemory; // shared memory for control block 1721 effect_param_cblk_t* mCblk; // control block for deferred parameter setting via shared memory 1722 uint8_t* mBuffer; // pointer to parameter area in shared memory 1723 int mPriority; // client application priority to control the effect 1724 bool mHasControl; // true if this handle is controlling the effect 1725 bool mEnabled; // cached enable state: needed when the effect is 1726 // restored after being suspended 1727 bool mDestroyed; // Set to true by destructor. Access with EffectModule 1728 // mLock held 1729 }; 1730 1731 // the EffectChain class represents a group of effects associated to one audio session. 1732 // There can be any number of EffectChain objects per output mixer thread (PlaybackThread). 1733 // The EffecChain with session ID 0 contains global effects applied to the output mix. 1734 // Effects in this chain can be insert or auxiliary. Effects in other chains (attached to tracks) 1735 // are insert only. The EffectChain maintains an ordered list of effect module, the order corresponding 1736 // in the effect process order. When attached to a track (session ID != 0), it also provide it's own 1737 // input buffer used by the track as accumulation buffer. 1738 class EffectChain: public RefBase { 1739 public: 1740 EffectChain(const wp<ThreadBase>& wThread, int sessionId); 1741 EffectChain(ThreadBase *thread, int sessionId); 1742 virtual ~EffectChain(); 1743 1744 // special key used for an entry in mSuspendedEffects keyed vector 1745 // corresponding to a suspend all request. 1746 static const int kKeyForSuspendAll = 0; 1747 1748 // minimum duration during which we force calling effect process when last track on 1749 // a session is stopped or removed to allow effect tail to be rendered 1750 static const int kProcessTailDurationMs = 1000; 1751 1752 void process_l(); 1753 1754 void lock() { 1755 mLock.lock(); 1756 } 1757 void unlock() { 1758 mLock.unlock(); 1759 } 1760 1761 status_t addEffect_l(const sp<EffectModule>& handle); 1762 size_t removeEffect_l(const sp<EffectModule>& handle); 1763 1764 int sessionId() const { return mSessionId; } 1765 void setSessionId(int sessionId) { mSessionId = sessionId; } 1766 1767 sp<EffectModule> getEffectFromDesc_l(effect_descriptor_t *descriptor); 1768 sp<EffectModule> getEffectFromId_l(int id); 1769 sp<EffectModule> getEffectFromType_l(const effect_uuid_t *type); 1770 bool setVolume_l(uint32_t *left, uint32_t *right); 1771 void setDevice_l(audio_devices_t device); 1772 void setMode_l(audio_mode_t mode); 1773 void setAudioSource_l(audio_source_t source); 1774 1775 void setInBuffer(int16_t *buffer, bool ownsBuffer = false) { 1776 mInBuffer = buffer; 1777 mOwnInBuffer = ownsBuffer; 1778 } 1779 int16_t *inBuffer() const { 1780 return mInBuffer; 1781 } 1782 void setOutBuffer(int16_t *buffer) { 1783 mOutBuffer = buffer; 1784 } 1785 int16_t *outBuffer() const { 1786 return mOutBuffer; 1787 } 1788 1789 void incTrackCnt() { android_atomic_inc(&mTrackCnt); } 1790 void decTrackCnt() { android_atomic_dec(&mTrackCnt); } 1791 int32_t trackCnt() const { return android_atomic_acquire_load(&mTrackCnt); } 1792 1793 void incActiveTrackCnt() { android_atomic_inc(&mActiveTrackCnt); 1794 mTailBufferCount = mMaxTailBuffers; } 1795 void decActiveTrackCnt() { android_atomic_dec(&mActiveTrackCnt); } 1796 int32_t activeTrackCnt() const { return android_atomic_acquire_load(&mActiveTrackCnt); } 1797 1798 uint32_t strategy() const { return mStrategy; } 1799 void setStrategy(uint32_t strategy) 1800 { mStrategy = strategy; } 1801 1802 // suspend effect of the given type 1803 void setEffectSuspended_l(const effect_uuid_t *type, 1804 bool suspend); 1805 // suspend all eligible effects 1806 void setEffectSuspendedAll_l(bool suspend); 1807 // check if effects should be suspend or restored when a given effect is enable or disabled 1808 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1809 bool enabled); 1810 1811 void clearInputBuffer(); 1812 1813 void dump(int fd, const Vector<String16>& args); 1814 1815 protected: 1816 friend class AudioFlinger; // for mThread, mEffects 1817 EffectChain(const EffectChain&); 1818 EffectChain& operator =(const EffectChain&); 1819 1820 class SuspendedEffectDesc : public RefBase { 1821 public: 1822 SuspendedEffectDesc() : mRefCount(0) {} 1823 1824 int mRefCount; 1825 effect_uuid_t mType; 1826 wp<EffectModule> mEffect; 1827 }; 1828 1829 // get a list of effect modules to suspend when an effect of the type 1830 // passed is enabled. 1831 void getSuspendEligibleEffects(Vector< sp<EffectModule> > &effects); 1832 1833 // get an effect module if it is currently enable 1834 sp<EffectModule> getEffectIfEnabled(const effect_uuid_t *type); 1835 // true if the effect whose descriptor is passed can be suspended 1836 // OEMs can modify the rules implemented in this method to exclude specific effect 1837 // types or implementations from the suspend/restore mechanism. 1838 bool isEffectEligibleForSuspend(const effect_descriptor_t& desc); 1839 1840 void clearInputBuffer_l(sp<ThreadBase> thread); 1841 1842 wp<ThreadBase> mThread; // parent mixer thread 1843 Mutex mLock; // mutex protecting effect list 1844 Vector< sp<EffectModule> > mEffects; // list of effect modules 1845 int mSessionId; // audio session ID 1846 int16_t *mInBuffer; // chain input buffer 1847 int16_t *mOutBuffer; // chain output buffer 1848 1849 // 'volatile' here means these are accessed with atomic operations instead of mutex 1850 volatile int32_t mActiveTrackCnt; // number of active tracks connected 1851 volatile int32_t mTrackCnt; // number of tracks connected 1852 1853 int32_t mTailBufferCount; // current effect tail buffer count 1854 int32_t mMaxTailBuffers; // maximum effect tail buffers 1855 bool mOwnInBuffer; // true if the chain owns its input buffer 1856 int mVolumeCtrlIdx; // index of insert effect having control over volume 1857 uint32_t mLeftVolume; // previous volume on left channel 1858 uint32_t mRightVolume; // previous volume on right channel 1859 uint32_t mNewLeftVolume; // new volume on left channel 1860 uint32_t mNewRightVolume; // new volume on right channel 1861 uint32_t mStrategy; // strategy for this effect chain 1862 // mSuspendedEffects lists all effects currently suspended in the chain. 1863 // Use effect type UUID timelow field as key. There is no real risk of identical 1864 // timeLow fields among effect type UUIDs. 1865 // Updated by updateSuspendedSessions_l() only. 1866 KeyedVector< int, sp<SuspendedEffectDesc> > mSuspendedEffects; 1867 }; 1868 1869 class AudioHwDevice { 1870 public: 1871 enum Flags { 1872 AHWD_CAN_SET_MASTER_VOLUME = 0x1, 1873 AHWD_CAN_SET_MASTER_MUTE = 0x2, 1874 }; 1875 1876 AudioHwDevice(const char *moduleName, 1877 audio_hw_device_t *hwDevice, 1878 Flags flags) 1879 : mModuleName(strdup(moduleName)) 1880 , mHwDevice(hwDevice) 1881 , mFlags(flags) { } 1882 /*virtual*/ ~AudioHwDevice() { free((void *)mModuleName); } 1883 1884 bool canSetMasterVolume() const { 1885 return (0 != (mFlags & AHWD_CAN_SET_MASTER_VOLUME)); 1886 } 1887 1888 bool canSetMasterMute() const { 1889 return (0 != (mFlags & AHWD_CAN_SET_MASTER_MUTE)); 1890 } 1891 1892 const char *moduleName() const { return mModuleName; } 1893 audio_hw_device_t *hwDevice() const { return mHwDevice; } 1894 private: 1895 const char * const mModuleName; 1896 audio_hw_device_t * const mHwDevice; 1897 Flags mFlags; 1898 }; 1899 1900 // AudioStreamOut and AudioStreamIn are immutable, so their fields are const. 1901 // For emphasis, we could also make all pointers to them be "const *", 1902 // but that would clutter the code unnecessarily. 1903 1904 struct AudioStreamOut { 1905 AudioHwDevice* const audioHwDev; 1906 audio_stream_out_t* const stream; 1907 1908 audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); } 1909 1910 AudioStreamOut(AudioHwDevice *dev, audio_stream_out_t *out) : 1911 audioHwDev(dev), stream(out) {} 1912 }; 1913 1914 struct AudioStreamIn { 1915 AudioHwDevice* const audioHwDev; 1916 audio_stream_in_t* const stream; 1917 1918 audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); } 1919 1920 AudioStreamIn(AudioHwDevice *dev, audio_stream_in_t *in) : 1921 audioHwDev(dev), stream(in) {} 1922 }; 1923 1924 // for mAudioSessionRefs only 1925 struct AudioSessionRef { 1926 AudioSessionRef(int sessionid, pid_t pid) : 1927 mSessionid(sessionid), mPid(pid), mCnt(1) {} 1928 const int mSessionid; 1929 const pid_t mPid; 1930 int mCnt; 1931 }; 1932 1933 mutable Mutex mLock; 1934 1935 DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client() 1936 1937 mutable Mutex mHardwareLock; 1938 // NOTE: If both mLock and mHardwareLock mutexes must be held, 1939 // always take mLock before mHardwareLock 1940 1941 // These two fields are immutable after onFirstRef(), so no lock needed to access 1942 AudioHwDevice* mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL 1943 DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*> mAudioHwDevs; 1944 1945 // for dump, indicates which hardware operation is currently in progress (but not stream ops) 1946 enum hardware_call_state { 1947 AUDIO_HW_IDLE = 0, // no operation in progress 1948 AUDIO_HW_INIT, // init_check 1949 AUDIO_HW_OUTPUT_OPEN, // open_output_stream 1950 AUDIO_HW_OUTPUT_CLOSE, // unused 1951 AUDIO_HW_INPUT_OPEN, // unused 1952 AUDIO_HW_INPUT_CLOSE, // unused 1953 AUDIO_HW_STANDBY, // unused 1954 AUDIO_HW_SET_MASTER_VOLUME, // set_master_volume 1955 AUDIO_HW_GET_ROUTING, // unused 1956 AUDIO_HW_SET_ROUTING, // unused 1957 AUDIO_HW_GET_MODE, // unused 1958 AUDIO_HW_SET_MODE, // set_mode 1959 AUDIO_HW_GET_MIC_MUTE, // get_mic_mute 1960 AUDIO_HW_SET_MIC_MUTE, // set_mic_mute 1961 AUDIO_HW_SET_VOICE_VOLUME, // set_voice_volume 1962 AUDIO_HW_SET_PARAMETER, // set_parameters 1963 AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size 1964 AUDIO_HW_GET_MASTER_VOLUME, // get_master_volume 1965 AUDIO_HW_GET_PARAMETER, // get_parameters 1966 AUDIO_HW_SET_MASTER_MUTE, // set_master_mute 1967 AUDIO_HW_GET_MASTER_MUTE, // get_master_mute 1968 }; 1969 1970 mutable hardware_call_state mHardwareStatus; // for dump only 1971 1972 1973 DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads; 1974 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 1975 1976 // member variables below are protected by mLock 1977 float mMasterVolume; 1978 bool mMasterMute; 1979 // end of variables protected by mLock 1980 1981 DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads; 1982 1983 DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients; 1984 volatile int32_t mNextUniqueId; // updated by android_atomic_inc 1985 audio_mode_t mMode; 1986 bool mBtNrecIsOff; 1987 1988 // protected by mLock 1989 Vector<AudioSessionRef*> mAudioSessionRefs; 1990 1991 float masterVolume_l() const; 1992 bool masterMute_l() const; 1993 audio_module_handle_t loadHwModule_l(const char *name); 1994 1995 Vector < sp<SyncEvent> > mPendingSyncEvents; // sync events awaiting for a session 1996 // to be created 1997 1998private: 1999 sp<Client> registerPid_l(pid_t pid); // always returns non-0 2000 2001 // for use from destructor 2002 status_t closeOutput_nonvirtual(audio_io_handle_t output); 2003 status_t closeInput_nonvirtual(audio_io_handle_t input); 2004}; 2005 2006 2007// ---------------------------------------------------------------------------- 2008 2009}; // namespace android 2010 2011#endif // ANDROID_AUDIO_FLINGER_H 2012