AudioFlinger.h revision 91de9b56282d126ffb36344266af5fee3cefcfdd
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 "AudioBufferProvider.h" 49#include "ExtendedAudioBufferProvider.h" 50#include "FastMixer.h" 51#include "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 float masterVolumeSW() const; 126 virtual bool masterMute() const; 127 virtual bool masterMuteSW() const; 128 129 virtual status_t setStreamVolume(audio_stream_type_t stream, float value, 130 audio_io_handle_t output); 131 virtual status_t setStreamMute(audio_stream_type_t stream, bool muted); 132 133 virtual float streamVolume(audio_stream_type_t stream, 134 audio_io_handle_t output) const; 135 virtual bool streamMute(audio_stream_type_t stream) const; 136 137 virtual status_t setMode(audio_mode_t mode); 138 139 virtual status_t setMicMute(bool state); 140 virtual bool getMicMute() const; 141 142 virtual status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs); 143 virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) const; 144 145 virtual void registerClient(const sp<IAudioFlingerClient>& client); 146 147 virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, 148 audio_channel_mask_t channelMask) const; 149 150 virtual audio_io_handle_t openOutput(audio_module_handle_t module, 151 audio_devices_t *pDevices, 152 uint32_t *pSamplingRate, 153 audio_format_t *pFormat, 154 audio_channel_mask_t *pChannelMask, 155 uint32_t *pLatencyMs, 156 audio_output_flags_t flags); 157 158 virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1, 159 audio_io_handle_t output2); 160 161 virtual status_t closeOutput(audio_io_handle_t output); 162 163 virtual status_t suspendOutput(audio_io_handle_t output); 164 165 virtual status_t restoreOutput(audio_io_handle_t output); 166 167 virtual audio_io_handle_t openInput(audio_module_handle_t module, 168 audio_devices_t *pDevices, 169 uint32_t *pSamplingRate, 170 audio_format_t *pFormat, 171 audio_channel_mask_t *pChannelMask); 172 173 virtual status_t closeInput(audio_io_handle_t input); 174 175 virtual status_t setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output); 176 177 virtual status_t setVoiceVolume(float volume); 178 179 virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 180 audio_io_handle_t output) const; 181 182 virtual unsigned int getInputFramesLost(audio_io_handle_t ioHandle) const; 183 184 virtual int newAudioSessionId(); 185 186 virtual void acquireAudioSessionId(int audioSession); 187 188 virtual void releaseAudioSessionId(int audioSession); 189 190 virtual status_t queryNumberEffects(uint32_t *numEffects) const; 191 192 virtual status_t queryEffect(uint32_t index, effect_descriptor_t *descriptor) const; 193 194 virtual status_t getEffectDescriptor(const effect_uuid_t *pUuid, 195 effect_descriptor_t *descriptor) const; 196 197 virtual sp<IEffect> createEffect(pid_t pid, 198 effect_descriptor_t *pDesc, 199 const sp<IEffectClient>& effectClient, 200 int32_t priority, 201 audio_io_handle_t io, 202 int sessionId, 203 status_t *status, 204 int *id, 205 int *enabled); 206 207 virtual status_t moveEffects(int sessionId, audio_io_handle_t srcOutput, 208 audio_io_handle_t dstOutput); 209 210 virtual audio_module_handle_t loadHwModule(const char *name); 211 212 virtual status_t onTransact( 213 uint32_t code, 214 const Parcel& data, 215 Parcel* reply, 216 uint32_t flags); 217 218 // end of IAudioFlinger interface 219 220 class SyncEvent; 221 222 typedef void (*sync_event_callback_t)(const wp<SyncEvent>& event) ; 223 224 class SyncEvent : public RefBase { 225 public: 226 SyncEvent(AudioSystem::sync_event_t type, 227 int triggerSession, 228 int listenerSession, 229 sync_event_callback_t callBack, 230 void *cookie) 231 : mType(type), mTriggerSession(triggerSession), mListenerSession(listenerSession), 232 mCallback(callBack), mCookie(cookie) 233 {} 234 235 virtual ~SyncEvent() {} 236 237 void trigger() { Mutex::Autolock _l(mLock); if (mCallback) mCallback(this); } 238 bool isCancelled() { Mutex::Autolock _l(mLock); return (mCallback == NULL); } 239 void cancel() {Mutex::Autolock _l(mLock); mCallback = NULL; } 240 AudioSystem::sync_event_t type() const { return mType; } 241 int triggerSession() const { return mTriggerSession; } 242 int listenerSession() const { return mListenerSession; } 243 void *cookie() const { return mCookie; } 244 245 private: 246 const AudioSystem::sync_event_t mType; 247 const int mTriggerSession; 248 const int mListenerSession; 249 sync_event_callback_t mCallback; 250 void * const mCookie; 251 Mutex mLock; 252 }; 253 254 sp<SyncEvent> createSyncEvent(AudioSystem::sync_event_t type, 255 int triggerSession, 256 int listenerSession, 257 sync_event_callback_t callBack, 258 void *cookie); 259 260private: 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 audio_hw_device_t* 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; 458 void* mBufferEnd; 459 uint32_t mFrameCount; 460 // we don't really need a lock for these 461 track_state mState; 462 const uint32_t mSampleRate; // initial sample rate only; for tracks which 463 // support dynamic rates, the current value is in control block 464 const audio_format_t mFormat; 465 bool mStepServerFailed; 466 const int mSessionId; 467 uint8_t mChannelCount; 468 audio_channel_mask_t mChannelMask; 469 Vector < sp<SyncEvent> >mSyncEvents; 470 }; 471 472 class ConfigEvent { 473 public: 474 ConfigEvent() : mEvent(0), mParam(0) {} 475 476 int mEvent; 477 int mParam; 478 }; 479 480 class PMDeathRecipient : public IBinder::DeathRecipient { 481 public: 482 PMDeathRecipient(const wp<ThreadBase>& thread) : mThread(thread) {} 483 virtual ~PMDeathRecipient() {} 484 485 // IBinder::DeathRecipient 486 virtual void binderDied(const wp<IBinder>& who); 487 488 private: 489 PMDeathRecipient(const PMDeathRecipient&); 490 PMDeathRecipient& operator = (const PMDeathRecipient&); 491 492 wp<ThreadBase> mThread; 493 }; 494 495 virtual status_t initCheck() const = 0; 496 497 // static externally-visible 498 type_t type() const { return mType; } 499 audio_io_handle_t id() const { return mId;} 500 501 // dynamic externally-visible 502 uint32_t sampleRate() const { return mSampleRate; } 503 int channelCount() const { return mChannelCount; } 504 audio_channel_mask_t channelMask() const { return mChannelMask; } 505 audio_format_t format() const { return mFormat; } 506 // Called by AudioFlinger::frameCount(audio_io_handle_t output) and effects, 507 // and returns the normal mix buffer's frame count. No API for HAL frame count. 508 size_t frameCount() const { return mNormalFrameCount; } 509 510 // Should be "virtual status_t requestExitAndWait()" and override same 511 // method in Thread, but Thread::requestExitAndWait() is not yet virtual. 512 void exit(); 513 virtual bool checkForNewParameters_l() = 0; 514 virtual status_t setParameters(const String8& keyValuePairs); 515 virtual String8 getParameters(const String8& keys) = 0; 516 virtual void audioConfigChanged_l(int event, int param = 0) = 0; 517 void sendConfigEvent(int event, int param = 0); 518 void sendConfigEvent_l(int event, int param = 0); 519 void processConfigEvents(); 520 521 // see note at declaration of mStandby and mDevice 522 bool standby() const { return mStandby; } 523 audio_devices_t device() const { return mDevice; } 524 525 virtual audio_stream_t* stream() const = 0; 526 527 sp<EffectHandle> createEffect_l( 528 const sp<AudioFlinger::Client>& client, 529 const sp<IEffectClient>& effectClient, 530 int32_t priority, 531 int sessionId, 532 effect_descriptor_t *desc, 533 int *enabled, 534 status_t *status); 535 void disconnectEffect(const sp< EffectModule>& effect, 536 EffectHandle *handle, 537 bool unpinIfLast); 538 539 // return values for hasAudioSession (bit field) 540 enum effect_state { 541 EFFECT_SESSION = 0x1, // the audio session corresponds to at least one 542 // effect 543 TRACK_SESSION = 0x2 // the audio session corresponds to at least one 544 // track 545 }; 546 547 // get effect chain corresponding to session Id. 548 sp<EffectChain> getEffectChain(int sessionId); 549 // same as getEffectChain() but must be called with ThreadBase mutex locked 550 sp<EffectChain> getEffectChain_l(int sessionId); 551 // add an effect chain to the chain list (mEffectChains) 552 virtual status_t addEffectChain_l(const sp<EffectChain>& chain) = 0; 553 // remove an effect chain from the chain list (mEffectChains) 554 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain) = 0; 555 // lock all effect chains Mutexes. Must be called before releasing the 556 // ThreadBase mutex before processing the mixer and effects. This guarantees the 557 // integrity of the chains during the process. 558 // Also sets the parameter 'effectChains' to current value of mEffectChains. 559 void lockEffectChains_l(Vector< sp<EffectChain> >& effectChains); 560 // unlock effect chains after process 561 void unlockEffectChains(const Vector< sp<EffectChain> >& effectChains); 562 // set audio mode to all effect chains 563 void setMode(audio_mode_t mode); 564 // get effect module with corresponding ID on specified audio session 565 sp<AudioFlinger::EffectModule> getEffect(int sessionId, int effectId); 566 sp<AudioFlinger::EffectModule> getEffect_l(int sessionId, int effectId); 567 // add and effect module. Also creates the effect chain is none exists for 568 // the effects audio session 569 status_t addEffect_l(const sp< EffectModule>& effect); 570 // remove and effect module. Also removes the effect chain is this was the last 571 // effect 572 void removeEffect_l(const sp< EffectModule>& effect); 573 // detach all tracks connected to an auxiliary effect 574 virtual void detachAuxEffect_l(int effectId) {} 575 // returns either EFFECT_SESSION if effects on this audio session exist in one 576 // chain, or TRACK_SESSION if tracks on this audio session exist, or both 577 virtual uint32_t hasAudioSession(int sessionId) = 0; 578 // the value returned by default implementation is not important as the 579 // strategy is only meaningful for PlaybackThread which implements this method 580 virtual uint32_t getStrategyForSession_l(int sessionId) { return 0; } 581 582 // suspend or restore effect according to the type of effect passed. a NULL 583 // type pointer means suspend all effects in the session 584 void setEffectSuspended(const effect_uuid_t *type, 585 bool suspend, 586 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 587 // check if some effects must be suspended/restored when an effect is enabled 588 // or disabled 589 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 590 bool enabled, 591 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 592 void checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 593 bool enabled, 594 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 595 596 virtual status_t setSyncEvent(const sp<SyncEvent>& event) = 0; 597 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) = 0; 598 599 600 mutable Mutex mLock; 601 602 protected: 603 604 // entry describing an effect being suspended in mSuspendedSessions keyed vector 605 class SuspendedSessionDesc : public RefBase { 606 public: 607 SuspendedSessionDesc() : mRefCount(0) {} 608 609 int mRefCount; // number of active suspend requests 610 effect_uuid_t mType; // effect type UUID 611 }; 612 613 void acquireWakeLock(); 614 void acquireWakeLock_l(); 615 void releaseWakeLock(); 616 void releaseWakeLock_l(); 617 void setEffectSuspended_l(const effect_uuid_t *type, 618 bool suspend, 619 int sessionId); 620 // updated mSuspendedSessions when an effect suspended or restored 621 void updateSuspendedSessions_l(const effect_uuid_t *type, 622 bool suspend, 623 int sessionId); 624 // check if some effects must be suspended when an effect chain is added 625 void checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain); 626 627 friend class AudioFlinger; // for mEffectChains 628 629 const type_t mType; 630 631 // Used by parameters, config events, addTrack_l, exit 632 Condition mWaitWorkCV; 633 634 const sp<AudioFlinger> mAudioFlinger; 635 uint32_t mSampleRate; 636 size_t mFrameCount; // output HAL, direct output, record 637 size_t mNormalFrameCount; // normal mixer and effects 638 audio_channel_mask_t mChannelMask; 639 uint16_t mChannelCount; 640 size_t mFrameSize; 641 audio_format_t mFormat; 642 643 // Parameter sequence by client: binder thread calling setParameters(): 644 // 1. Lock mLock 645 // 2. Append to mNewParameters 646 // 3. mWaitWorkCV.signal 647 // 4. mParamCond.waitRelative with timeout 648 // 5. read mParamStatus 649 // 6. mWaitWorkCV.signal 650 // 7. Unlock 651 // 652 // Parameter sequence by server: threadLoop calling checkForNewParameters_l(): 653 // 1. Lock mLock 654 // 2. If there is an entry in mNewParameters proceed ... 655 // 2. Read first entry in mNewParameters 656 // 3. Process 657 // 4. Remove first entry from mNewParameters 658 // 5. Set mParamStatus 659 // 6. mParamCond.signal 660 // 7. mWaitWorkCV.wait with timeout (this is to avoid overwriting mParamStatus) 661 // 8. Unlock 662 Condition mParamCond; 663 Vector<String8> mNewParameters; 664 status_t mParamStatus; 665 666 Vector<ConfigEvent> mConfigEvents; 667 668 // These fields are written and read by thread itself without lock or barrier, 669 // and read by other threads without lock or barrier via standby() and device(). 670 // Because of the absence of a lock or barrier, any other thread that reads 671 // these fields must use the information in isolation, or be prepared to deal 672 // with possibility that it might be inconsistent with other information. 673 bool mStandby; // Whether thread is currently in standby. 674 audio_devices_t mDevice; // output device for PlaybackThread 675 // input + output devices for RecordThread 676 677 const audio_io_handle_t mId; 678 Vector< sp<EffectChain> > mEffectChains; 679 680 static const int kNameLength = 16; // prctl(PR_SET_NAME) limit 681 char mName[kNameLength]; 682 sp<IPowerManager> mPowerManager; 683 sp<IBinder> mWakeLockToken; 684 const sp<PMDeathRecipient> mDeathRecipient; 685 // list of suspended effects per session and per type. The first vector is 686 // keyed by session ID, the second by type UUID timeLow field 687 KeyedVector< int, KeyedVector< int, sp<SuspendedSessionDesc> > > mSuspendedSessions; 688 }; 689 690 struct stream_type_t { 691 stream_type_t() 692 : volume(1.0f), 693 mute(false) 694 { 695 } 696 float volume; 697 bool mute; 698 }; 699 700 // --- PlaybackThread --- 701 class PlaybackThread : public ThreadBase { 702 public: 703 704 enum mixer_state { 705 MIXER_IDLE, // no active tracks 706 MIXER_TRACKS_ENABLED, // at least one active track, but no track has any data ready 707 MIXER_TRACKS_READY // at least one active track, and at least one track has data 708 // standby mode does not have an enum value 709 // suspend by audio policy manager is orthogonal to mixer state 710 }; 711 712 // playback track 713 class Track : public TrackBase, public VolumeProvider { 714 public: 715 Track( PlaybackThread *thread, 716 const sp<Client>& client, 717 audio_stream_type_t streamType, 718 uint32_t sampleRate, 719 audio_format_t format, 720 audio_channel_mask_t channelMask, 721 int frameCount, 722 const sp<IMemory>& sharedBuffer, 723 int sessionId, 724 IAudioFlinger::track_flags_t flags); 725 virtual ~Track(); 726 727 static void appendDumpHeader(String8& result); 728 void dump(char* buffer, size_t size); 729 virtual status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 730 int triggerSession = 0); 731 virtual void stop(); 732 void pause(); 733 734 void flush(); 735 void destroy(); 736 void mute(bool); 737 int name() const { return mName; } 738 739 audio_stream_type_t streamType() const { 740 return mStreamType; 741 } 742 status_t attachAuxEffect(int EffectId); 743 void setAuxBuffer(int EffectId, int32_t *buffer); 744 int32_t *auxBuffer() const { return mAuxBuffer; } 745 void setMainBuffer(int16_t *buffer) { mMainBuffer = buffer; } 746 int16_t *mainBuffer() const { return mMainBuffer; } 747 int auxEffectId() const { return mAuxEffectId; } 748 749 // implement FastMixerState::VolumeProvider interface 750 virtual uint32_t getVolumeLR(); 751 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 752 753 protected: 754 // for numerous 755 friend class PlaybackThread; 756 friend class MixerThread; 757 friend class DirectOutputThread; 758 759 Track(const Track&); 760 Track& operator = (const Track&); 761 762 // AudioBufferProvider interface 763 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts = kInvalidPTS); 764 // releaseBuffer() not overridden 765 766 virtual size_t framesReady() const; 767 768 bool isMuted() const { return mMute; } 769 bool isPausing() const { 770 return mState == PAUSING; 771 } 772 bool isPaused() const { 773 return mState == PAUSED; 774 } 775 bool isResuming() const { 776 return mState == RESUMING; 777 } 778 bool isReady() const; 779 void setPaused() { mState = PAUSED; } 780 void reset(); 781 782 bool isOutputTrack() const { 783 return (mStreamType == AUDIO_STREAM_CNT); 784 } 785 786 sp<IMemory> sharedBuffer() const { return mSharedBuffer; } 787 788 bool presentationComplete(size_t framesWritten, size_t audioHalFrames); 789 790 public: 791 void triggerEvents(AudioSystem::sync_event_t type); 792 virtual bool isTimedTrack() const { return false; } 793 bool isFastTrack() const { return (mFlags & IAudioFlinger::TRACK_FAST) != 0; } 794 795 protected: 796 797 // written by Track::mute() called by binder thread(s), without a mutex or barrier. 798 // read by Track::isMuted() called by playback thread, also without a mutex or barrier. 799 // The lack of mutex or barrier is safe because the mute status is only used by itself. 800 bool mMute; 801 802 // FILLED state is used for suppressing volume ramp at begin of playing 803 enum {FS_INVALID, FS_FILLING, FS_FILLED, FS_ACTIVE}; 804 mutable uint8_t mFillingUpStatus; 805 int8_t mRetryCount; 806 const sp<IMemory> mSharedBuffer; 807 bool mResetDone; 808 const audio_stream_type_t mStreamType; 809 int mName; // track name on the normal mixer, 810 // allocated statically at track creation time, 811 // and is even allocated (though unused) for fast tracks 812 // FIXME don't allocate track name for fast tracks 813 int16_t *mMainBuffer; 814 int32_t *mAuxBuffer; 815 int mAuxEffectId; 816 bool mHasVolumeController; 817 size_t mPresentationCompleteFrames; // number of frames written to the audio HAL 818 // when this track will be fully rendered 819 private: 820 IAudioFlinger::track_flags_t mFlags; 821 822 // The following fields are only for fast tracks, and should be in a subclass 823 int mFastIndex; // index within FastMixerState::mFastTracks[]; 824 // either mFastIndex == -1 if not isFastTrack() 825 // or 0 < mFastIndex < FastMixerState::kMaxFast because 826 // index 0 is reserved for normal mixer's submix; 827 // index is allocated statically at track creation time 828 // but the slot is only used if track is active 829 FastTrackUnderruns mObservedUnderruns; // Most recently observed value of 830 // mFastMixerDumpState.mTracks[mFastIndex].mUnderruns 831 uint32_t mUnderrunCount; // Counter of total number of underruns, never reset 832 volatile float mCachedVolume; // combined master volume and stream type volume; 833 // 'volatile' means accessed without lock or 834 // barrier, but is read/written atomically 835 }; // end of Track 836 837 class TimedTrack : public Track { 838 public: 839 static sp<TimedTrack> create(PlaybackThread *thread, 840 const sp<Client>& client, 841 audio_stream_type_t streamType, 842 uint32_t sampleRate, 843 audio_format_t format, 844 audio_channel_mask_t channelMask, 845 int frameCount, 846 const sp<IMemory>& sharedBuffer, 847 int sessionId); 848 virtual ~TimedTrack(); 849 850 class TimedBuffer { 851 public: 852 TimedBuffer(); 853 TimedBuffer(const sp<IMemory>& buffer, int64_t pts); 854 const sp<IMemory>& buffer() const { return mBuffer; } 855 int64_t pts() const { return mPTS; } 856 uint32_t position() const { return mPosition; } 857 void setPosition(uint32_t pos) { mPosition = pos; } 858 private: 859 sp<IMemory> mBuffer; 860 int64_t mPTS; 861 uint32_t mPosition; 862 }; 863 864 // Mixer facing methods. 865 virtual bool isTimedTrack() const { return true; } 866 virtual size_t framesReady() const; 867 868 // AudioBufferProvider interface 869 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, 870 int64_t pts); 871 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 872 873 // Client/App facing methods. 874 status_t allocateTimedBuffer(size_t size, 875 sp<IMemory>* buffer); 876 status_t queueTimedBuffer(const sp<IMemory>& buffer, 877 int64_t pts); 878 status_t setMediaTimeTransform(const LinearTransform& xform, 879 TimedAudioTrack::TargetTimeline target); 880 881 private: 882 TimedTrack(PlaybackThread *thread, 883 const sp<Client>& client, 884 audio_stream_type_t streamType, 885 uint32_t sampleRate, 886 audio_format_t format, 887 audio_channel_mask_t channelMask, 888 int frameCount, 889 const sp<IMemory>& sharedBuffer, 890 int sessionId); 891 892 void timedYieldSamples_l(AudioBufferProvider::Buffer* buffer); 893 void timedYieldSilence_l(uint32_t numFrames, 894 AudioBufferProvider::Buffer* buffer); 895 void trimTimedBufferQueue_l(); 896 void trimTimedBufferQueueHead_l(const char* logTag); 897 void updateFramesPendingAfterTrim_l(const TimedBuffer& buf, 898 const char* logTag); 899 900 uint64_t mLocalTimeFreq; 901 LinearTransform mLocalTimeToSampleTransform; 902 LinearTransform mMediaTimeToSampleTransform; 903 sp<MemoryDealer> mTimedMemoryDealer; 904 905 Vector<TimedBuffer> mTimedBufferQueue; 906 bool mQueueHeadInFlight; 907 bool mTrimQueueHeadOnRelease; 908 uint32_t mFramesPendingInQueue; 909 910 uint8_t* mTimedSilenceBuffer; 911 uint32_t mTimedSilenceBufferSize; 912 mutable Mutex mTimedBufferQueueLock; 913 bool mTimedAudioOutputOnTime; 914 CCHelper mCCHelper; 915 916 Mutex mMediaTimeTransformLock; 917 LinearTransform mMediaTimeTransform; 918 bool mMediaTimeTransformValid; 919 TimedAudioTrack::TargetTimeline mMediaTimeTransformTarget; 920 }; 921 922 923 // playback track 924 class OutputTrack : public Track { 925 public: 926 927 class Buffer: public AudioBufferProvider::Buffer { 928 public: 929 int16_t *mBuffer; 930 }; 931 932 OutputTrack(PlaybackThread *thread, 933 DuplicatingThread *sourceThread, 934 uint32_t sampleRate, 935 audio_format_t format, 936 audio_channel_mask_t channelMask, 937 int frameCount); 938 virtual ~OutputTrack(); 939 940 virtual status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 941 int triggerSession = 0); 942 virtual void stop(); 943 bool write(int16_t* data, uint32_t frames); 944 bool bufferQueueEmpty() const { return mBufferQueue.size() == 0; } 945 bool isActive() const { return mActive; } 946 const wp<ThreadBase>& thread() const { return mThread; } 947 948 private: 949 950 enum { 951 NO_MORE_BUFFERS = 0x80000001, // same in AudioTrack.h, ok to be different value 952 }; 953 954 status_t obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs); 955 void clearBufferQueue(); 956 957 // Maximum number of pending buffers allocated by OutputTrack::write() 958 static const uint8_t kMaxOverFlowBuffers = 10; 959 960 Vector < Buffer* > mBufferQueue; 961 AudioBufferProvider::Buffer mOutBuffer; 962 bool mActive; 963 DuplicatingThread* const mSourceThread; // for waitTimeMs() in write() 964 }; // end of OutputTrack 965 966 PlaybackThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 967 audio_io_handle_t id, audio_devices_t device, type_t type); 968 virtual ~PlaybackThread(); 969 970 void dump(int fd, const Vector<String16>& args); 971 972 // Thread virtuals 973 virtual status_t readyToRun(); 974 virtual bool threadLoop(); 975 976 // RefBase 977 virtual void onFirstRef(); 978 979protected: 980 // Code snippets that were lifted up out of threadLoop() 981 virtual void threadLoop_mix() = 0; 982 virtual void threadLoop_sleepTime() = 0; 983 virtual void threadLoop_write(); 984 virtual void threadLoop_standby(); 985 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 986 987 // prepareTracks_l reads and writes mActiveTracks, and returns 988 // the pending set of tracks to remove via Vector 'tracksToRemove'. The caller 989 // is responsible for clearing or destroying this Vector later on, when it 990 // is safe to do so. That will drop the final ref count and destroy the tracks. 991 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove) = 0; 992 993public: 994 995 virtual status_t initCheck() const { return (mOutput == NULL) ? NO_INIT : NO_ERROR; } 996 997 // return estimated latency in milliseconds, as reported by HAL 998 uint32_t latency() const; 999 // same, but lock must already be held 1000 uint32_t latency_l() const; 1001 1002 void setMasterVolume(float value); 1003 void setMasterMute(bool muted); 1004 1005 void setStreamVolume(audio_stream_type_t stream, float value); 1006 void setStreamMute(audio_stream_type_t stream, bool muted); 1007 1008 float streamVolume(audio_stream_type_t stream) const; 1009 1010 sp<Track> createTrack_l( 1011 const sp<AudioFlinger::Client>& client, 1012 audio_stream_type_t streamType, 1013 uint32_t sampleRate, 1014 audio_format_t format, 1015 audio_channel_mask_t channelMask, 1016 int frameCount, 1017 const sp<IMemory>& sharedBuffer, 1018 int sessionId, 1019 IAudioFlinger::track_flags_t flags, 1020 pid_t tid, 1021 status_t *status); 1022 1023 AudioStreamOut* getOutput() const; 1024 AudioStreamOut* clearOutput(); 1025 virtual audio_stream_t* stream() const; 1026 1027 // a very large number of suspend() will eventually wraparound, but unlikely 1028 void suspend() { (void) android_atomic_inc(&mSuspended); } 1029 void restore() 1030 { 1031 // if restore() is done without suspend(), get back into 1032 // range so that the next suspend() will operate correctly 1033 if (android_atomic_dec(&mSuspended) <= 0) { 1034 android_atomic_release_store(0, &mSuspended); 1035 } 1036 } 1037 bool isSuspended() const 1038 { return android_atomic_acquire_load(&mSuspended) > 0; } 1039 1040 virtual String8 getParameters(const String8& keys); 1041 virtual void audioConfigChanged_l(int event, int param = 0); 1042 status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames); 1043 int16_t *mixBuffer() const { return mMixBuffer; }; 1044 1045 virtual void detachAuxEffect_l(int effectId); 1046 status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track> track, 1047 int EffectId); 1048 status_t attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track> track, 1049 int EffectId); 1050 1051 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1052 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1053 virtual uint32_t hasAudioSession(int sessionId); 1054 virtual uint32_t getStrategyForSession_l(int sessionId); 1055 1056 1057 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 1058 virtual bool isValidSyncEvent(const sp<SyncEvent>& event); 1059 void invalidateTracks(audio_stream_type_t streamType); 1060 1061 1062 protected: 1063 int16_t* mMixBuffer; 1064 1065 // suspend count, > 0 means suspended. While suspended, the thread continues to pull from 1066 // tracks and mix, but doesn't write to HAL. A2DP and SCO HAL implementations can't handle 1067 // concurrent use of both of them, so Audio Policy Service suspends one of the threads to 1068 // workaround that restriction. 1069 // 'volatile' means accessed via atomic operations and no lock. 1070 volatile int32_t mSuspended; 1071 1072 int mBytesWritten; 1073 private: 1074 // mMasterMute is in both PlaybackThread and in AudioFlinger. When a 1075 // PlaybackThread needs to find out if master-muted, it checks it's local 1076 // copy rather than the one in AudioFlinger. This optimization saves a lock. 1077 bool mMasterMute; 1078 void setMasterMute_l(bool muted) { mMasterMute = muted; } 1079 protected: 1080 SortedVector< wp<Track> > mActiveTracks; // FIXME check if this could be sp<> 1081 1082 // Allocate a track name for a given channel mask. 1083 // Returns name >= 0 if successful, -1 on failure. 1084 virtual int getTrackName_l(audio_channel_mask_t channelMask) = 0; 1085 virtual void deleteTrackName_l(int name) = 0; 1086 1087 // Time to sleep between cycles when: 1088 virtual uint32_t activeSleepTimeUs() const; // mixer state MIXER_TRACKS_ENABLED 1089 virtual uint32_t idleSleepTimeUs() const = 0; // mixer state MIXER_IDLE 1090 virtual uint32_t suspendSleepTimeUs() const = 0; // audio policy manager suspended us 1091 // No sleep when mixer state == MIXER_TRACKS_READY; relies on audio HAL stream->write() 1092 // No sleep in standby mode; waits on a condition 1093 1094 // Code snippets that are temporarily lifted up out of threadLoop() until the merge 1095 void checkSilentMode_l(); 1096 1097 // Non-trivial for DUPLICATING only 1098 virtual void saveOutputTracks() { } 1099 virtual void clearOutputTracks() { } 1100 1101 // Cache various calculated values, at threadLoop() entry and after a parameter change 1102 virtual void cacheParameters_l(); 1103 1104 virtual uint32_t correctLatency(uint32_t latency) const; 1105 1106 private: 1107 1108 friend class AudioFlinger; // for numerous 1109 1110 PlaybackThread(const Client&); 1111 PlaybackThread& operator = (const PlaybackThread&); 1112 1113 status_t addTrack_l(const sp<Track>& track); 1114 void destroyTrack_l(const sp<Track>& track); 1115 void removeTrack_l(const sp<Track>& track); 1116 1117 void readOutputParameters(); 1118 1119 virtual void dumpInternals(int fd, const Vector<String16>& args); 1120 void dumpTracks(int fd, const Vector<String16>& args); 1121 1122 SortedVector< sp<Track> > mTracks; 1123 // mStreamTypes[] uses 1 additional stream type internally for the OutputTrack used by DuplicatingThread 1124 stream_type_t mStreamTypes[AUDIO_STREAM_CNT + 1]; 1125 AudioStreamOut *mOutput; 1126 1127 float mMasterVolume; 1128 nsecs_t mLastWriteTime; 1129 int mNumWrites; 1130 int mNumDelayedWrites; 1131 bool mInWrite; 1132 1133 // FIXME rename these former local variables of threadLoop to standard "m" names 1134 nsecs_t standbyTime; 1135 size_t mixBufferSize; 1136 1137 // cached copies of activeSleepTimeUs() and idleSleepTimeUs() made by cacheParameters_l() 1138 uint32_t activeSleepTime; 1139 uint32_t idleSleepTime; 1140 1141 uint32_t sleepTime; 1142 1143 // mixer status returned by prepareTracks_l() 1144 mixer_state mMixerStatus; // current cycle 1145 // previous cycle when in prepareTracks_l() 1146 mixer_state mMixerStatusIgnoringFastTracks; 1147 // FIXME or a separate ready state per track 1148 1149 // FIXME move these declarations into the specific sub-class that needs them 1150 // MIXER only 1151 uint32_t sleepTimeShift; 1152 1153 // same as AudioFlinger::mStandbyTimeInNsecs except for DIRECT which uses a shorter value 1154 nsecs_t standbyDelay; 1155 1156 // MIXER only 1157 nsecs_t maxPeriod; 1158 1159 // DUPLICATING only 1160 uint32_t writeFrames; 1161 1162 private: 1163 // The HAL output sink is treated as non-blocking, but current implementation is blocking 1164 sp<NBAIO_Sink> mOutputSink; 1165 // If a fast mixer is present, the blocking pipe sink, otherwise clear 1166 sp<NBAIO_Sink> mPipeSink; 1167 // The current sink for the normal mixer to write it's (sub)mix, mOutputSink or mPipeSink 1168 sp<NBAIO_Sink> mNormalSink; 1169 // For dumpsys 1170 sp<NBAIO_Sink> mTeeSink; 1171 sp<NBAIO_Source> mTeeSource; 1172 uint32_t mScreenState; // cached copy of gScreenState 1173 public: 1174 virtual bool hasFastMixer() const = 0; 1175 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const 1176 { FastTrackUnderruns dummy; return dummy; } 1177 1178 protected: 1179 // accessed by both binder threads and within threadLoop(), lock on mutex needed 1180 unsigned mFastTrackAvailMask; // bit i set if fast track [i] is available 1181 1182 }; 1183 1184 class MixerThread : public PlaybackThread { 1185 public: 1186 MixerThread (const sp<AudioFlinger>& audioFlinger, 1187 AudioStreamOut* output, 1188 audio_io_handle_t id, 1189 audio_devices_t device, 1190 type_t type = MIXER); 1191 virtual ~MixerThread(); 1192 1193 // Thread virtuals 1194 1195 virtual bool checkForNewParameters_l(); 1196 virtual void dumpInternals(int fd, const Vector<String16>& args); 1197 1198 protected: 1199 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1200 virtual int getTrackName_l(audio_channel_mask_t channelMask); 1201 virtual void deleteTrackName_l(int name); 1202 virtual uint32_t idleSleepTimeUs() const; 1203 virtual uint32_t suspendSleepTimeUs() const; 1204 virtual void cacheParameters_l(); 1205 1206 // threadLoop snippets 1207 virtual void threadLoop_write(); 1208 virtual void threadLoop_standby(); 1209 virtual void threadLoop_mix(); 1210 virtual void threadLoop_sleepTime(); 1211 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 1212 virtual uint32_t correctLatency(uint32_t latency) const; 1213 1214 AudioMixer* mAudioMixer; // normal mixer 1215 private: 1216 // one-time initialization, no locks required 1217 FastMixer* mFastMixer; // non-NULL if there is also a fast mixer 1218 sp<AudioWatchdog> mAudioWatchdog; // non-0 if there is an audio watchdog thread 1219 1220 // contents are not guaranteed to be consistent, no locks required 1221 FastMixerDumpState mFastMixerDumpState; 1222#ifdef STATE_QUEUE_DUMP 1223 StateQueueObserverDump mStateQueueObserverDump; 1224 StateQueueMutatorDump mStateQueueMutatorDump; 1225#endif 1226 AudioWatchdogDump mAudioWatchdogDump; 1227 1228 // accessible only within the threadLoop(), no locks required 1229 // mFastMixer->sq() // for mutating and pushing state 1230 int32_t mFastMixerFutex; // for cold idle 1231 1232 public: 1233 virtual bool hasFastMixer() const { return mFastMixer != NULL; } 1234 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const { 1235 ALOG_ASSERT(fastIndex < FastMixerState::kMaxFastTracks); 1236 return mFastMixerDumpState.mTracks[fastIndex].mUnderruns; 1237 } 1238 }; 1239 1240 class DirectOutputThread : public PlaybackThread { 1241 public: 1242 1243 DirectOutputThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1244 audio_io_handle_t id, audio_devices_t device); 1245 virtual ~DirectOutputThread(); 1246 1247 // Thread virtuals 1248 1249 virtual bool checkForNewParameters_l(); 1250 1251 protected: 1252 virtual int getTrackName_l(audio_channel_mask_t channelMask); 1253 virtual void deleteTrackName_l(int name); 1254 virtual uint32_t activeSleepTimeUs() const; 1255 virtual uint32_t idleSleepTimeUs() const; 1256 virtual uint32_t suspendSleepTimeUs() const; 1257 virtual void cacheParameters_l(); 1258 1259 // threadLoop snippets 1260 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1261 virtual void threadLoop_mix(); 1262 virtual void threadLoop_sleepTime(); 1263 1264 // volumes last sent to audio HAL with stream->set_volume() 1265 float mLeftVolFloat; 1266 float mRightVolFloat; 1267 1268private: 1269 // prepareTracks_l() tells threadLoop_mix() the name of the single active track 1270 sp<Track> mActiveTrack; 1271 public: 1272 virtual bool hasFastMixer() const { return false; } 1273 }; 1274 1275 class DuplicatingThread : public MixerThread { 1276 public: 1277 DuplicatingThread (const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread, 1278 audio_io_handle_t id); 1279 virtual ~DuplicatingThread(); 1280 1281 // Thread virtuals 1282 void addOutputTrack(MixerThread* thread); 1283 void removeOutputTrack(MixerThread* thread); 1284 uint32_t waitTimeMs() const { return mWaitTimeMs; } 1285 protected: 1286 virtual uint32_t activeSleepTimeUs() const; 1287 1288 private: 1289 bool outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks); 1290 protected: 1291 // threadLoop snippets 1292 virtual void threadLoop_mix(); 1293 virtual void threadLoop_sleepTime(); 1294 virtual void threadLoop_write(); 1295 virtual void threadLoop_standby(); 1296 virtual void cacheParameters_l(); 1297 1298 private: 1299 // called from threadLoop, addOutputTrack, removeOutputTrack 1300 virtual void updateWaitTime_l(); 1301 protected: 1302 virtual void saveOutputTracks(); 1303 virtual void clearOutputTracks(); 1304 private: 1305 1306 uint32_t mWaitTimeMs; 1307 SortedVector < sp<OutputTrack> > outputTracks; 1308 SortedVector < sp<OutputTrack> > mOutputTracks; 1309 public: 1310 virtual bool hasFastMixer() const { return false; } 1311 }; 1312 1313 PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const; 1314 MixerThread *checkMixerThread_l(audio_io_handle_t output) const; 1315 RecordThread *checkRecordThread_l(audio_io_handle_t input) const; 1316 // no range check, AudioFlinger::mLock held 1317 bool streamMute_l(audio_stream_type_t stream) const 1318 { return mStreamTypes[stream].mute; } 1319 // no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held 1320 float streamVolume_l(audio_stream_type_t stream) const 1321 { return mStreamTypes[stream].volume; } 1322 void audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2); 1323 1324 // allocate an audio_io_handle_t, session ID, or effect ID 1325 uint32_t nextUniqueId(); 1326 1327 status_t moveEffectChain_l(int sessionId, 1328 PlaybackThread *srcThread, 1329 PlaybackThread *dstThread, 1330 bool reRegister); 1331 // return thread associated with primary hardware device, or NULL 1332 PlaybackThread *primaryPlaybackThread_l() const; 1333 audio_devices_t primaryOutputDevice_l() const; 1334 1335 sp<PlaybackThread> getEffectThread_l(int sessionId, int EffectId); 1336 1337 // server side of the client's IAudioTrack 1338 class TrackHandle : public android::BnAudioTrack { 1339 public: 1340 TrackHandle(const sp<PlaybackThread::Track>& track); 1341 virtual ~TrackHandle(); 1342 virtual sp<IMemory> getCblk() const; 1343 virtual status_t start(); 1344 virtual void stop(); 1345 virtual void flush(); 1346 virtual void mute(bool); 1347 virtual void pause(); 1348 virtual status_t attachAuxEffect(int effectId); 1349 virtual status_t allocateTimedBuffer(size_t size, 1350 sp<IMemory>* buffer); 1351 virtual status_t queueTimedBuffer(const sp<IMemory>& buffer, 1352 int64_t pts); 1353 virtual status_t setMediaTimeTransform(const LinearTransform& xform, 1354 int target); 1355 virtual status_t onTransact( 1356 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 1357 private: 1358 const sp<PlaybackThread::Track> mTrack; 1359 }; 1360 1361 void removeClient_l(pid_t pid); 1362 void removeNotificationClient(pid_t pid); 1363 1364 1365 // record thread 1366 class RecordThread : public ThreadBase, public AudioBufferProvider 1367 { 1368 public: 1369 1370 // record track 1371 class RecordTrack : public TrackBase { 1372 public: 1373 RecordTrack(RecordThread *thread, 1374 const sp<Client>& client, 1375 uint32_t sampleRate, 1376 audio_format_t format, 1377 audio_channel_mask_t channelMask, 1378 int frameCount, 1379 int sessionId); 1380 virtual ~RecordTrack(); 1381 1382 virtual status_t start(AudioSystem::sync_event_t event, int triggerSession); 1383 virtual void stop(); 1384 1385 // clear the buffer overflow flag 1386 void clearOverflow() { mOverflow = false; } 1387 // set the buffer overflow flag and return previous value 1388 bool setOverflow() { bool tmp = mOverflow; mOverflow = true; return tmp; } 1389 1390 void dump(char* buffer, size_t size); 1391 1392 private: 1393 friend class AudioFlinger; // for mState 1394 1395 RecordTrack(const RecordTrack&); 1396 RecordTrack& operator = (const RecordTrack&); 1397 1398 // AudioBufferProvider interface 1399 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts = kInvalidPTS); 1400 // releaseBuffer() not overridden 1401 1402 bool mOverflow; // overflow on most recent attempt to fill client buffer 1403 }; 1404 1405 RecordThread(const sp<AudioFlinger>& audioFlinger, 1406 AudioStreamIn *input, 1407 uint32_t sampleRate, 1408 audio_channel_mask_t channelMask, 1409 audio_io_handle_t id, 1410 audio_devices_t device); 1411 virtual ~RecordThread(); 1412 1413 // Thread 1414 virtual bool threadLoop(); 1415 virtual status_t readyToRun(); 1416 1417 // RefBase 1418 virtual void onFirstRef(); 1419 1420 virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; } 1421 sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l( 1422 const sp<AudioFlinger::Client>& client, 1423 uint32_t sampleRate, 1424 audio_format_t format, 1425 audio_channel_mask_t channelMask, 1426 int frameCount, 1427 int sessionId, 1428 IAudioFlinger::track_flags_t flags, 1429 pid_t tid, 1430 status_t *status); 1431 1432 status_t start(RecordTrack* recordTrack, 1433 AudioSystem::sync_event_t event, 1434 int triggerSession); 1435 1436 // ask the thread to stop the specified track, and 1437 // return true if the caller should then do it's part of the stopping process 1438 bool stop_l(RecordTrack* recordTrack); 1439 1440 void dump(int fd, const Vector<String16>& args); 1441 AudioStreamIn* clearInput(); 1442 virtual audio_stream_t* stream() const; 1443 1444 // AudioBufferProvider interface 1445 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts); 1446 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 1447 1448 virtual bool checkForNewParameters_l(); 1449 virtual String8 getParameters(const String8& keys); 1450 virtual void audioConfigChanged_l(int event, int param = 0); 1451 void readInputParameters(); 1452 virtual unsigned int getInputFramesLost(); 1453 1454 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1455 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1456 virtual uint32_t hasAudioSession(int sessionId); 1457 RecordTrack* track(); 1458 1459 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 1460 virtual bool isValidSyncEvent(const sp<SyncEvent>& event); 1461 1462 static void syncStartEventCallback(const wp<SyncEvent>& event); 1463 void handleSyncStartEvent(const sp<SyncEvent>& event); 1464 1465 private: 1466 void clearSyncStartEvent(); 1467 1468 // Enter standby if not already in standby, and set mStandby flag 1469 void standby(); 1470 1471 // Call the HAL standby method unconditionally, and don't change mStandby flag 1472 void inputStandBy(); 1473 1474 AudioStreamIn *mInput; 1475 RecordTrack* mTrack; 1476 sp<RecordTrack> mActiveTrack; 1477 Condition mStartStopCond; 1478 AudioResampler *mResampler; 1479 int32_t *mRsmpOutBuffer; 1480 int16_t *mRsmpInBuffer; 1481 size_t mRsmpInIndex; 1482 size_t mInputBytes; 1483 const int mReqChannelCount; 1484 const uint32_t mReqSampleRate; 1485 ssize_t mBytesRead; 1486 // sync event triggering actual audio capture. Frames read before this event will 1487 // be dropped and therefore not read by the application. 1488 sp<SyncEvent> mSyncStartEvent; 1489 // number of captured frames to drop after the start sync event has been received. 1490 // when < 0, maximum frames to drop before starting capture even if sync event is 1491 // not received 1492 ssize_t mFramestoDrop; 1493 }; 1494 1495 // server side of the client's IAudioRecord 1496 class RecordHandle : public android::BnAudioRecord { 1497 public: 1498 RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack); 1499 virtual ~RecordHandle(); 1500 virtual sp<IMemory> getCblk() const; 1501 virtual status_t start(int /*AudioSystem::sync_event_t*/ event, int triggerSession); 1502 virtual void stop(); 1503 virtual status_t onTransact( 1504 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 1505 private: 1506 const sp<RecordThread::RecordTrack> mRecordTrack; 1507 1508 // for use from destructor 1509 void stop_nonvirtual(); 1510 }; 1511 1512 //--- Audio Effect Management 1513 1514 // EffectModule and EffectChain classes both have their own mutex to protect 1515 // state changes or resource modifications. Always respect the following order 1516 // if multiple mutexes must be acquired to avoid cross deadlock: 1517 // AudioFlinger -> ThreadBase -> EffectChain -> EffectModule 1518 1519 // The EffectModule class is a wrapper object controlling the effect engine implementation 1520 // in the effect library. It prevents concurrent calls to process() and command() functions 1521 // from different client threads. It keeps a list of EffectHandle objects corresponding 1522 // to all client applications using this effect and notifies applications of effect state, 1523 // control or parameter changes. It manages the activation state machine to send appropriate 1524 // reset, enable, disable commands to effect engine and provide volume 1525 // ramping when effects are activated/deactivated. 1526 // When controlling an auxiliary effect, the EffectModule also provides an input buffer used by 1527 // the attached track(s) to accumulate their auxiliary channel. 1528 class EffectModule: public RefBase { 1529 public: 1530 EffectModule(ThreadBase *thread, 1531 const wp<AudioFlinger::EffectChain>& chain, 1532 effect_descriptor_t *desc, 1533 int id, 1534 int sessionId); 1535 virtual ~EffectModule(); 1536 1537 enum effect_state { 1538 IDLE, 1539 RESTART, 1540 STARTING, 1541 ACTIVE, 1542 STOPPING, 1543 STOPPED, 1544 DESTROYED 1545 }; 1546 1547 int id() const { return mId; } 1548 void process(); 1549 void updateState(); 1550 status_t command(uint32_t cmdCode, 1551 uint32_t cmdSize, 1552 void *pCmdData, 1553 uint32_t *replySize, 1554 void *pReplyData); 1555 1556 void reset_l(); 1557 status_t configure(); 1558 status_t init(); 1559 effect_state state() const { 1560 return mState; 1561 } 1562 uint32_t status() { 1563 return mStatus; 1564 } 1565 int sessionId() const { 1566 return mSessionId; 1567 } 1568 status_t setEnabled(bool enabled); 1569 status_t setEnabled_l(bool enabled); 1570 bool isEnabled() const; 1571 bool isProcessEnabled() const; 1572 1573 void setInBuffer(int16_t *buffer) { mConfig.inputCfg.buffer.s16 = buffer; } 1574 int16_t *inBuffer() { return mConfig.inputCfg.buffer.s16; } 1575 void setOutBuffer(int16_t *buffer) { mConfig.outputCfg.buffer.s16 = buffer; } 1576 int16_t *outBuffer() { return mConfig.outputCfg.buffer.s16; } 1577 void setChain(const wp<EffectChain>& chain) { mChain = chain; } 1578 void setThread(const wp<ThreadBase>& thread) { mThread = thread; } 1579 const wp<ThreadBase>& thread() { return mThread; } 1580 1581 status_t addHandle(EffectHandle *handle); 1582 size_t disconnect(EffectHandle *handle, bool unpinIfLast); 1583 size_t removeHandle(EffectHandle *handle); 1584 1585 const effect_descriptor_t& desc() const { return mDescriptor; } 1586 wp<EffectChain>& chain() { return mChain; } 1587 1588 status_t setDevice(audio_devices_t device); 1589 status_t setVolume(uint32_t *left, uint32_t *right, bool controller); 1590 status_t setMode(audio_mode_t mode); 1591 status_t start(); 1592 status_t stop(); 1593 void setSuspended(bool suspended); 1594 bool suspended() const; 1595 1596 EffectHandle* controlHandle_l(); 1597 1598 bool isPinned() const { return mPinned; } 1599 void unPin() { mPinned = false; } 1600 bool purgeHandles(); 1601 void lock() { mLock.lock(); } 1602 void unlock() { mLock.unlock(); } 1603 1604 void dump(int fd, const Vector<String16>& args); 1605 1606 protected: 1607 friend class AudioFlinger; // for mHandles 1608 bool mPinned; 1609 1610 // Maximum time allocated to effect engines to complete the turn off sequence 1611 static const uint32_t MAX_DISABLE_TIME_MS = 10000; 1612 1613 EffectModule(const EffectModule&); 1614 EffectModule& operator = (const EffectModule&); 1615 1616 status_t start_l(); 1617 status_t stop_l(); 1618 1619mutable Mutex mLock; // mutex for process, commands and handles list protection 1620 wp<ThreadBase> mThread; // parent thread 1621 wp<EffectChain> mChain; // parent effect chain 1622 const int mId; // this instance unique ID 1623 const int mSessionId; // audio session ID 1624 const effect_descriptor_t mDescriptor;// effect descriptor received from effect engine 1625 effect_config_t mConfig; // input and output audio configuration 1626 effect_handle_t mEffectInterface; // Effect module C API 1627 status_t mStatus; // initialization status 1628 effect_state mState; // current activation state 1629 Vector<EffectHandle *> mHandles; // list of client handles 1630 // First handle in mHandles has highest priority and controls the effect module 1631 uint32_t mMaxDisableWaitCnt; // maximum grace period before forcing an effect off after 1632 // sending disable command. 1633 uint32_t mDisableWaitCnt; // current process() calls count during disable period. 1634 bool mSuspended; // effect is suspended: temporarily disabled by framework 1635 }; 1636 1637 // The EffectHandle class implements the IEffect interface. It provides resources 1638 // to receive parameter updates, keeps track of effect control 1639 // ownership and state and has a pointer to the EffectModule object it is controlling. 1640 // There is one EffectHandle object for each application controlling (or using) 1641 // an effect module. 1642 // The EffectHandle is obtained by calling AudioFlinger::createEffect(). 1643 class EffectHandle: public android::BnEffect { 1644 public: 1645 1646 EffectHandle(const sp<EffectModule>& effect, 1647 const sp<AudioFlinger::Client>& client, 1648 const sp<IEffectClient>& effectClient, 1649 int32_t priority); 1650 virtual ~EffectHandle(); 1651 1652 // IEffect 1653 virtual status_t enable(); 1654 virtual status_t disable(); 1655 virtual status_t command(uint32_t cmdCode, 1656 uint32_t cmdSize, 1657 void *pCmdData, 1658 uint32_t *replySize, 1659 void *pReplyData); 1660 virtual void disconnect(); 1661 private: 1662 void disconnect(bool unpinIfLast); 1663 public: 1664 virtual sp<IMemory> getCblk() const { return mCblkMemory; } 1665 virtual status_t onTransact(uint32_t code, const Parcel& data, 1666 Parcel* reply, uint32_t flags); 1667 1668 1669 // Give or take control of effect module 1670 // - hasControl: true if control is given, false if removed 1671 // - signal: true client app should be signaled of change, false otherwise 1672 // - enabled: state of the effect when control is passed 1673 void setControl(bool hasControl, bool signal, bool enabled); 1674 void commandExecuted(uint32_t cmdCode, 1675 uint32_t cmdSize, 1676 void *pCmdData, 1677 uint32_t replySize, 1678 void *pReplyData); 1679 void setEnabled(bool enabled); 1680 bool enabled() const { return mEnabled; } 1681 1682 // Getters 1683 int id() const { return mEffect->id(); } 1684 int priority() const { return mPriority; } 1685 bool hasControl() const { return mHasControl; } 1686 sp<EffectModule> effect() const { return mEffect; } 1687 // destroyed_l() must be called with the associated EffectModule mLock held 1688 bool destroyed_l() const { return mDestroyed; } 1689 1690 void dump(char* buffer, size_t size); 1691 1692 protected: 1693 friend class AudioFlinger; // for mEffect, mHasControl, mEnabled 1694 EffectHandle(const EffectHandle&); 1695 EffectHandle& operator =(const EffectHandle&); 1696 1697 sp<EffectModule> mEffect; // pointer to controlled EffectModule 1698 sp<IEffectClient> mEffectClient; // callback interface for client notifications 1699 /*const*/ sp<Client> mClient; // client for shared memory allocation, see disconnect() 1700 sp<IMemory> mCblkMemory; // shared memory for control block 1701 effect_param_cblk_t* mCblk; // control block for deferred parameter setting via shared memory 1702 uint8_t* mBuffer; // pointer to parameter area in shared memory 1703 int mPriority; // client application priority to control the effect 1704 bool mHasControl; // true if this handle is controlling the effect 1705 bool mEnabled; // cached enable state: needed when the effect is 1706 // restored after being suspended 1707 bool mDestroyed; // Set to true by destructor. Access with EffectModule 1708 // mLock held 1709 }; 1710 1711 // the EffectChain class represents a group of effects associated to one audio session. 1712 // There can be any number of EffectChain objects per output mixer thread (PlaybackThread). 1713 // The EffecChain with session ID 0 contains global effects applied to the output mix. 1714 // Effects in this chain can be insert or auxiliary. Effects in other chains (attached to tracks) 1715 // are insert only. The EffectChain maintains an ordered list of effect module, the order corresponding 1716 // in the effect process order. When attached to a track (session ID != 0), it also provide it's own 1717 // input buffer used by the track as accumulation buffer. 1718 class EffectChain: public RefBase { 1719 public: 1720 EffectChain(const wp<ThreadBase>& wThread, int sessionId); 1721 EffectChain(ThreadBase *thread, int sessionId); 1722 virtual ~EffectChain(); 1723 1724 // special key used for an entry in mSuspendedEffects keyed vector 1725 // corresponding to a suspend all request. 1726 static const int kKeyForSuspendAll = 0; 1727 1728 // minimum duration during which we force calling effect process when last track on 1729 // a session is stopped or removed to allow effect tail to be rendered 1730 static const int kProcessTailDurationMs = 1000; 1731 1732 void process_l(); 1733 1734 void lock() { 1735 mLock.lock(); 1736 } 1737 void unlock() { 1738 mLock.unlock(); 1739 } 1740 1741 status_t addEffect_l(const sp<EffectModule>& handle); 1742 size_t removeEffect_l(const sp<EffectModule>& handle); 1743 1744 int sessionId() const { return mSessionId; } 1745 void setSessionId(int sessionId) { mSessionId = sessionId; } 1746 1747 sp<EffectModule> getEffectFromDesc_l(effect_descriptor_t *descriptor); 1748 sp<EffectModule> getEffectFromId_l(int id); 1749 sp<EffectModule> getEffectFromType_l(const effect_uuid_t *type); 1750 bool setVolume_l(uint32_t *left, uint32_t *right); 1751 void setDevice_l(audio_devices_t device); 1752 void setMode_l(audio_mode_t mode); 1753 1754 void setInBuffer(int16_t *buffer, bool ownsBuffer = false) { 1755 mInBuffer = buffer; 1756 mOwnInBuffer = ownsBuffer; 1757 } 1758 int16_t *inBuffer() const { 1759 return mInBuffer; 1760 } 1761 void setOutBuffer(int16_t *buffer) { 1762 mOutBuffer = buffer; 1763 } 1764 int16_t *outBuffer() const { 1765 return mOutBuffer; 1766 } 1767 1768 void incTrackCnt() { android_atomic_inc(&mTrackCnt); } 1769 void decTrackCnt() { android_atomic_dec(&mTrackCnt); } 1770 int32_t trackCnt() const { return android_atomic_acquire_load(&mTrackCnt); } 1771 1772 void incActiveTrackCnt() { android_atomic_inc(&mActiveTrackCnt); 1773 mTailBufferCount = mMaxTailBuffers; } 1774 void decActiveTrackCnt() { android_atomic_dec(&mActiveTrackCnt); } 1775 int32_t activeTrackCnt() const { return android_atomic_acquire_load(&mActiveTrackCnt); } 1776 1777 uint32_t strategy() const { return mStrategy; } 1778 void setStrategy(uint32_t strategy) 1779 { mStrategy = strategy; } 1780 1781 // suspend effect of the given type 1782 void setEffectSuspended_l(const effect_uuid_t *type, 1783 bool suspend); 1784 // suspend all eligible effects 1785 void setEffectSuspendedAll_l(bool suspend); 1786 // check if effects should be suspend or restored when a given effect is enable or disabled 1787 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1788 bool enabled); 1789 1790 void clearInputBuffer(); 1791 1792 void dump(int fd, const Vector<String16>& args); 1793 1794 protected: 1795 friend class AudioFlinger; // for mThread, mEffects 1796 EffectChain(const EffectChain&); 1797 EffectChain& operator =(const EffectChain&); 1798 1799 class SuspendedEffectDesc : public RefBase { 1800 public: 1801 SuspendedEffectDesc() : mRefCount(0) {} 1802 1803 int mRefCount; 1804 effect_uuid_t mType; 1805 wp<EffectModule> mEffect; 1806 }; 1807 1808 // get a list of effect modules to suspend when an effect of the type 1809 // passed is enabled. 1810 void getSuspendEligibleEffects(Vector< sp<EffectModule> > &effects); 1811 1812 // get an effect module if it is currently enable 1813 sp<EffectModule> getEffectIfEnabled(const effect_uuid_t *type); 1814 // true if the effect whose descriptor is passed can be suspended 1815 // OEMs can modify the rules implemented in this method to exclude specific effect 1816 // types or implementations from the suspend/restore mechanism. 1817 bool isEffectEligibleForSuspend(const effect_descriptor_t& desc); 1818 1819 void clearInputBuffer_l(sp<ThreadBase> thread); 1820 1821 wp<ThreadBase> mThread; // parent mixer thread 1822 Mutex mLock; // mutex protecting effect list 1823 Vector< sp<EffectModule> > mEffects; // list of effect modules 1824 int mSessionId; // audio session ID 1825 int16_t *mInBuffer; // chain input buffer 1826 int16_t *mOutBuffer; // chain output buffer 1827 1828 // 'volatile' here means these are accessed with atomic operations instead of mutex 1829 volatile int32_t mActiveTrackCnt; // number of active tracks connected 1830 volatile int32_t mTrackCnt; // number of tracks connected 1831 1832 int32_t mTailBufferCount; // current effect tail buffer count 1833 int32_t mMaxTailBuffers; // maximum effect tail buffers 1834 bool mOwnInBuffer; // true if the chain owns its input buffer 1835 int mVolumeCtrlIdx; // index of insert effect having control over volume 1836 uint32_t mLeftVolume; // previous volume on left channel 1837 uint32_t mRightVolume; // previous volume on right channel 1838 uint32_t mNewLeftVolume; // new volume on left channel 1839 uint32_t mNewRightVolume; // new volume on right channel 1840 uint32_t mStrategy; // strategy for this effect chain 1841 // mSuspendedEffects lists all effects currently suspended in the chain. 1842 // Use effect type UUID timelow field as key. There is no real risk of identical 1843 // timeLow fields among effect type UUIDs. 1844 // Updated by updateSuspendedSessions_l() only. 1845 KeyedVector< int, sp<SuspendedEffectDesc> > mSuspendedEffects; 1846 }; 1847 1848 // AudioStreamOut and AudioStreamIn are immutable, so their fields are const. 1849 // For emphasis, we could also make all pointers to them be "const *", 1850 // but that would clutter the code unnecessarily. 1851 1852 struct AudioStreamOut { 1853 audio_hw_device_t* const hwDev; 1854 audio_stream_out_t* const stream; 1855 1856 AudioStreamOut(audio_hw_device_t *dev, audio_stream_out_t *out) : 1857 hwDev(dev), stream(out) {} 1858 }; 1859 1860 struct AudioStreamIn { 1861 audio_hw_device_t* const hwDev; 1862 audio_stream_in_t* const stream; 1863 1864 AudioStreamIn(audio_hw_device_t *dev, audio_stream_in_t *in) : 1865 hwDev(dev), stream(in) {} 1866 }; 1867 1868 // for mAudioSessionRefs only 1869 struct AudioSessionRef { 1870 AudioSessionRef(int sessionid, pid_t pid) : 1871 mSessionid(sessionid), mPid(pid), mCnt(1) {} 1872 const int mSessionid; 1873 const pid_t mPid; 1874 int mCnt; 1875 }; 1876 1877 enum master_volume_support { 1878 // MVS_NONE: 1879 // Audio HAL has no support for master volume, either setting or 1880 // getting. All master volume control must be implemented in SW by the 1881 // AudioFlinger mixing core. 1882 MVS_NONE, 1883 1884 // MVS_SETONLY: 1885 // Audio HAL has support for setting master volume, but not for getting 1886 // master volume (original HAL design did not include a getter). 1887 // AudioFlinger needs to keep track of the last set master volume in 1888 // addition to needing to set an initial, default, master volume at HAL 1889 // load time. 1890 MVS_SETONLY, 1891 1892 // MVS_FULL: 1893 // Audio HAL has support both for setting and getting master volume. 1894 // AudioFlinger should send all set and get master volume requests 1895 // directly to the HAL. 1896 MVS_FULL, 1897 }; 1898 1899 enum master_mute_support { 1900 // MMS_NONE: 1901 // Audio HAL has no support for master mute, either setting or getting. 1902 // All master mute control must be implemented in SW by the 1903 // AudioFlinger mixing core. 1904 MMS_NONE, 1905 1906 // MMS_SETONLY: 1907 // Audio HAL has support for setting master mute, but not for getting 1908 // master mute. AudioFlinger needs to keep track of the last set 1909 // master mute in addition to needing to set an initial, default, 1910 // master mute at HAL load time. 1911 MMS_SETONLY, 1912 1913 // MMS_FULL: 1914 // Audio HAL has support both for setting and getting master mute. 1915 // AudioFlinger should send all set and get master mute requests 1916 // directly to the HAL. 1917 MMS_FULL, 1918 }; 1919 1920 class AudioHwDevice { 1921 public: 1922 AudioHwDevice(const char *moduleName, audio_hw_device_t *hwDevice) : 1923 mModuleName(strdup(moduleName)), mHwDevice(hwDevice){} 1924 /*virtual*/ ~AudioHwDevice() { free((void *)mModuleName); } 1925 1926 const char *moduleName() const { return mModuleName; } 1927 audio_hw_device_t *hwDevice() const { return mHwDevice; } 1928 private: 1929 const char * const mModuleName; 1930 audio_hw_device_t * const mHwDevice; 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 audio_hw_device_t* 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 // both are protected by mLock 1977 float mMasterVolume; 1978 float mMasterVolumeSW; 1979 master_volume_support mMasterVolumeSupportLvl; 1980 1981 bool mMasterMute; 1982 bool mMasterMuteSW; 1983 master_mute_support mMasterMuteSupportLvl; 1984 1985 DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads; 1986 1987 DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients; 1988 volatile int32_t mNextUniqueId; // updated by android_atomic_inc 1989 audio_mode_t mMode; 1990 bool mBtNrecIsOff; 1991 1992 // protected by mLock 1993 Vector<AudioSessionRef*> mAudioSessionRefs; 1994 1995 float masterVolume_l() const; 1996 float masterVolumeSW_l() const { return mMasterVolumeSW; } 1997 bool masterMute_l() const; 1998 bool masterMuteSW_l() const { return mMasterMuteSW; } 1999 audio_module_handle_t loadHwModule_l(const char *name); 2000 2001 Vector < sp<SyncEvent> > mPendingSyncEvents; // sync events awaiting for a session 2002 // to be created 2003 2004private: 2005 sp<Client> registerPid_l(pid_t pid); // always returns non-0 2006 2007 // for use from destructor 2008 status_t closeOutput_nonvirtual(audio_io_handle_t output); 2009 status_t closeInput_nonvirtual(audio_io_handle_t input); 2010}; 2011 2012 2013// ---------------------------------------------------------------------------- 2014 2015}; // namespace android 2016 2017#endif // ANDROID_AUDIO_FLINGER_H 2018