AudioFlinger.h revision 2df8f55055fe431a508148cf525df1ba40f03113
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 int *sessionId, 112 status_t *status); 113 114 virtual uint32_t sampleRate(audio_io_handle_t output) const; 115 virtual int channelCount(audio_io_handle_t output) const; 116 virtual audio_format_t format(audio_io_handle_t output) const; 117 virtual size_t frameCount(audio_io_handle_t output) const; 118 virtual uint32_t latency(audio_io_handle_t output) const; 119 120 virtual status_t setMasterVolume(float value); 121 virtual status_t setMasterMute(bool muted); 122 123 virtual float masterVolume() const; 124 virtual float masterVolumeSW() 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() { 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 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 audio_mode_t getMode() const { return mMode; } 260 261 bool btNrecIsOff() const { return mBtNrecIsOff; } 262 263 AudioFlinger(); 264 virtual ~AudioFlinger(); 265 266 // call in any IAudioFlinger method that accesses mPrimaryHardwareDev 267 status_t initCheck() const { return mPrimaryHardwareDev == NULL ? NO_INIT : NO_ERROR; } 268 269 // RefBase 270 virtual void onFirstRef(); 271 272 audio_hw_device_t* findSuitableHwDev_l(audio_module_handle_t module, uint32_t devices); 273 void purgeStaleEffects_l(); 274 275 // standby delay for MIXER and DUPLICATING playback threads is read from property 276 // ro.audio.flinger_standbytime_ms or defaults to kDefaultStandbyTimeInNsecs 277 static nsecs_t mStandbyTimeInNsecs; 278 279 // Internal dump utilites. 280 status_t dumpPermissionDenial(int fd, const Vector<String16>& args); 281 status_t dumpClients(int fd, const Vector<String16>& args); 282 status_t dumpInternals(int fd, const Vector<String16>& args); 283 284 // --- Client --- 285 class Client : public RefBase { 286 public: 287 Client(const sp<AudioFlinger>& audioFlinger, pid_t pid); 288 virtual ~Client(); 289 sp<MemoryDealer> heap() const; 290 pid_t pid() const { return mPid; } 291 sp<AudioFlinger> audioFlinger() const { return mAudioFlinger; } 292 293 bool reserveTimedTrack(); 294 void releaseTimedTrack(); 295 296 private: 297 Client(const Client&); 298 Client& operator = (const Client&); 299 const sp<AudioFlinger> mAudioFlinger; 300 const sp<MemoryDealer> mMemoryDealer; 301 const pid_t mPid; 302 303 Mutex mTimedTrackLock; 304 int mTimedTrackCount; 305 }; 306 307 // --- Notification Client --- 308 class NotificationClient : public IBinder::DeathRecipient { 309 public: 310 NotificationClient(const sp<AudioFlinger>& audioFlinger, 311 const sp<IAudioFlingerClient>& client, 312 pid_t pid); 313 virtual ~NotificationClient(); 314 315 sp<IAudioFlingerClient> audioFlingerClient() const { return mAudioFlingerClient; } 316 317 // IBinder::DeathRecipient 318 virtual void binderDied(const wp<IBinder>& who); 319 320 private: 321 NotificationClient(const NotificationClient&); 322 NotificationClient& operator = (const NotificationClient&); 323 324 const sp<AudioFlinger> mAudioFlinger; 325 const pid_t mPid; 326 const sp<IAudioFlingerClient> mAudioFlingerClient; 327 }; 328 329 class TrackHandle; 330 class RecordHandle; 331 class RecordThread; 332 class PlaybackThread; 333 class MixerThread; 334 class DirectOutputThread; 335 class DuplicatingThread; 336 class Track; 337 class RecordTrack; 338 class EffectModule; 339 class EffectHandle; 340 class EffectChain; 341 struct AudioStreamOut; 342 struct AudioStreamIn; 343 344 class ThreadBase : public Thread { 345 public: 346 347 enum type_t { 348 MIXER, // Thread class is MixerThread 349 DIRECT, // Thread class is DirectOutputThread 350 DUPLICATING, // Thread class is DuplicatingThread 351 RECORD // Thread class is RecordThread 352 }; 353 354 ThreadBase (const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, uint32_t device, type_t type); 355 virtual ~ThreadBase(); 356 357 status_t dumpBase(int fd, const Vector<String16>& args); 358 status_t dumpEffectChains(int fd, const Vector<String16>& args); 359 360 void clearPowerManager(); 361 362 // base for record and playback 363 class TrackBase : public ExtendedAudioBufferProvider, public RefBase { 364 365 public: 366 enum track_state { 367 IDLE, 368 TERMINATED, 369 FLUSHED, 370 STOPPED, 371 // next 2 states are currently used for fast tracks only 372 STOPPING_1, // waiting for first underrun 373 STOPPING_2, // waiting for presentation complete 374 RESUMING, 375 ACTIVE, 376 PAUSING, 377 PAUSED 378 }; 379 380 TrackBase(ThreadBase *thread, 381 const sp<Client>& client, 382 uint32_t sampleRate, 383 audio_format_t format, 384 uint32_t channelMask, 385 int frameCount, 386 const sp<IMemory>& sharedBuffer, 387 int sessionId); 388 virtual ~TrackBase(); 389 390 virtual status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 391 int triggerSession = 0) = 0; 392 virtual void stop() = 0; 393 sp<IMemory> getCblk() const { return mCblkMemory; } 394 audio_track_cblk_t* cblk() const { return mCblk; } 395 int sessionId() const { return mSessionId; } 396 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 397 398 protected: 399 TrackBase(const TrackBase&); 400 TrackBase& operator = (const TrackBase&); 401 402 // AudioBufferProvider interface 403 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) = 0; 404 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 405 406 // ExtendedAudioBufferProvider interface is only needed for Track, 407 // but putting it in TrackBase avoids the complexity of virtual inheritance 408 virtual size_t framesReady() const { return SIZE_MAX; } 409 410 audio_format_t format() const { 411 return mFormat; 412 } 413 414 int channelCount() const { return mChannelCount; } 415 416 uint32_t channelMask() const { return mChannelMask; } 417 418 int sampleRate() const; // FIXME inline after cblk sr moved 419 420 // Return a pointer to the start of a contiguous slice of the track buffer. 421 // Parameter 'offset' is the requested start position, expressed in 422 // monotonically increasing frame units relative to the track epoch. 423 // Parameter 'frames' is the requested length, also in frame units. 424 // Always returns non-NULL. It is the caller's responsibility to 425 // verify that this will be successful; the result of calling this 426 // function with invalid 'offset' or 'frames' is undefined. 427 void* getBuffer(uint32_t offset, uint32_t frames) const; 428 429 bool isStopped() const { 430 return (mState == STOPPED || mState == FLUSHED); 431 } 432 433 // for fast tracks only 434 bool isStopping() const { 435 return mState == STOPPING_1 || mState == STOPPING_2; 436 } 437 bool isStopping_1() const { 438 return mState == STOPPING_1; 439 } 440 bool isStopping_2() const { 441 return mState == STOPPING_2; 442 } 443 444 bool isTerminated() const { 445 return mState == TERMINATED; 446 } 447 448 bool step(); 449 void reset(); 450 451 const wp<ThreadBase> mThread; 452 /*const*/ sp<Client> mClient; // see explanation at ~TrackBase() why not const 453 sp<IMemory> mCblkMemory; 454 audio_track_cblk_t* mCblk; 455 void* mBuffer; 456 void* mBufferEnd; 457 uint32_t mFrameCount; 458 // we don't really need a lock for these 459 track_state mState; 460 const uint32_t mSampleRate; // initial sample rate only; for tracks which 461 // support dynamic rates, the current value is in control block 462 const audio_format_t mFormat; 463 bool mStepServerFailed; 464 const int mSessionId; 465 uint8_t mChannelCount; 466 uint32_t mChannelMask; 467 Vector < sp<SyncEvent> >mSyncEvents; 468 }; 469 470 class ConfigEvent { 471 public: 472 ConfigEvent() : mEvent(0), mParam(0) {} 473 474 int mEvent; 475 int mParam; 476 }; 477 478 class PMDeathRecipient : public IBinder::DeathRecipient { 479 public: 480 PMDeathRecipient(const wp<ThreadBase>& thread) : mThread(thread) {} 481 virtual ~PMDeathRecipient() {} 482 483 // IBinder::DeathRecipient 484 virtual void binderDied(const wp<IBinder>& who); 485 486 private: 487 PMDeathRecipient(const PMDeathRecipient&); 488 PMDeathRecipient& operator = (const PMDeathRecipient&); 489 490 wp<ThreadBase> mThread; 491 }; 492 493 virtual status_t initCheck() const = 0; 494 495 // static externally-visible 496 type_t type() const { return mType; } 497 audio_io_handle_t id() const { return mId;} 498 499 // dynamic externally-visible 500 uint32_t sampleRate() const { return mSampleRate; } 501 int channelCount() const { return mChannelCount; } 502 audio_format_t format() const { return mFormat; } 503 // Called by AudioFlinger::frameCount(audio_io_handle_t output) and effects, 504 // and returns the normal mix buffer's frame count. No API for HAL frame count. 505 size_t frameCount() const { return mNormalFrameCount; } 506 507 void wakeUp() { mWaitWorkCV.broadcast(); } 508 // Should be "virtual status_t requestExitAndWait()" and override same 509 // method in Thread, but Thread::requestExitAndWait() is not yet virtual. 510 void exit(); 511 virtual bool checkForNewParameters_l() = 0; 512 virtual status_t setParameters(const String8& keyValuePairs); 513 virtual String8 getParameters(const String8& keys) = 0; 514 virtual void audioConfigChanged_l(int event, int param = 0) = 0; 515 void sendConfigEvent(int event, int param = 0); 516 void sendConfigEvent_l(int event, int param = 0); 517 void processConfigEvents(); 518 519 // see note at declaration of mStandby and mDevice 520 bool standby() const { return mStandby; } 521 audio_devices_t device() const { return mDevice; } 522 523 virtual audio_stream_t* stream() const = 0; 524 525 sp<EffectHandle> createEffect_l( 526 const sp<AudioFlinger::Client>& client, 527 const sp<IEffectClient>& effectClient, 528 int32_t priority, 529 int sessionId, 530 effect_descriptor_t *desc, 531 int *enabled, 532 status_t *status); 533 void disconnectEffect(const sp< EffectModule>& effect, 534 EffectHandle *handle, 535 bool unpinIfLast); 536 537 // return values for hasAudioSession (bit field) 538 enum effect_state { 539 EFFECT_SESSION = 0x1, // the audio session corresponds to at least one 540 // effect 541 TRACK_SESSION = 0x2 // the audio session corresponds to at least one 542 // track 543 }; 544 545 // get effect chain corresponding to session Id. 546 sp<EffectChain> getEffectChain(int sessionId); 547 // same as getEffectChain() but must be called with ThreadBase mutex locked 548 sp<EffectChain> getEffectChain_l(int sessionId); 549 // add an effect chain to the chain list (mEffectChains) 550 virtual status_t addEffectChain_l(const sp<EffectChain>& chain) = 0; 551 // remove an effect chain from the chain list (mEffectChains) 552 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain) = 0; 553 // lock all effect chains Mutexes. Must be called before releasing the 554 // ThreadBase mutex before processing the mixer and effects. This guarantees the 555 // integrity of the chains during the process. 556 // Also sets the parameter 'effectChains' to current value of mEffectChains. 557 void lockEffectChains_l(Vector< sp<EffectChain> >& effectChains); 558 // unlock effect chains after process 559 void unlockEffectChains(const Vector< sp<EffectChain> >& effectChains); 560 // set audio mode to all effect chains 561 void setMode(audio_mode_t mode); 562 // get effect module with corresponding ID on specified audio session 563 sp<AudioFlinger::EffectModule> getEffect(int sessionId, int effectId); 564 sp<AudioFlinger::EffectModule> getEffect_l(int sessionId, int effectId); 565 // add and effect module. Also creates the effect chain is none exists for 566 // the effects audio session 567 status_t addEffect_l(const sp< EffectModule>& effect); 568 // remove and effect module. Also removes the effect chain is this was the last 569 // effect 570 void removeEffect_l(const sp< EffectModule>& effect); 571 // detach all tracks connected to an auxiliary effect 572 virtual void detachAuxEffect_l(int effectId) {} 573 // returns either EFFECT_SESSION if effects on this audio session exist in one 574 // chain, or TRACK_SESSION if tracks on this audio session exist, or both 575 virtual uint32_t hasAudioSession(int sessionId) = 0; 576 // the value returned by default implementation is not important as the 577 // strategy is only meaningful for PlaybackThread which implements this method 578 virtual uint32_t getStrategyForSession_l(int sessionId) { return 0; } 579 580 // suspend or restore effect according to the type of effect passed. a NULL 581 // type pointer means suspend all effects in the session 582 void setEffectSuspended(const effect_uuid_t *type, 583 bool suspend, 584 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 585 // check if some effects must be suspended/restored when an effect is enabled 586 // or disabled 587 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 588 bool enabled, 589 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 590 void checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 591 bool enabled, 592 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 593 594 virtual status_t setSyncEvent(const sp<SyncEvent>& event) = 0; 595 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) = 0; 596 597 598 mutable Mutex mLock; 599 600 protected: 601 602 // entry describing an effect being suspended in mSuspendedSessions keyed vector 603 class SuspendedSessionDesc : public RefBase { 604 public: 605 SuspendedSessionDesc() : mRefCount(0) {} 606 607 int mRefCount; // number of active suspend requests 608 effect_uuid_t mType; // effect type UUID 609 }; 610 611 void acquireWakeLock(); 612 void acquireWakeLock_l(); 613 void releaseWakeLock(); 614 void releaseWakeLock_l(); 615 void setEffectSuspended_l(const effect_uuid_t *type, 616 bool suspend, 617 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 618 // updated mSuspendedSessions when an effect suspended or restored 619 void updateSuspendedSessions_l(const effect_uuid_t *type, 620 bool suspend, 621 int sessionId); 622 // check if some effects must be suspended when an effect chain is added 623 void checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain); 624 625 friend class AudioFlinger; // for mEffectChains 626 627 const type_t mType; 628 629 // Used by parameters, config events, addTrack_l, exit 630 Condition mWaitWorkCV; 631 632 const sp<AudioFlinger> mAudioFlinger; 633 uint32_t mSampleRate; 634 size_t mFrameCount; // output HAL, direct output, record 635 size_t mNormalFrameCount; // normal mixer and effects 636 uint32_t mChannelMask; 637 uint16_t mChannelCount; 638 size_t mFrameSize; 639 audio_format_t mFormat; 640 641 // Parameter sequence by client: binder thread calling setParameters(): 642 // 1. Lock mLock 643 // 2. Append to mNewParameters 644 // 3. mWaitWorkCV.signal 645 // 4. mParamCond.waitRelative with timeout 646 // 5. read mParamStatus 647 // 6. mWaitWorkCV.signal 648 // 7. Unlock 649 // 650 // Parameter sequence by server: threadLoop calling checkForNewParameters_l(): 651 // 1. Lock mLock 652 // 2. If there is an entry in mNewParameters proceed ... 653 // 2. Read first entry in mNewParameters 654 // 3. Process 655 // 4. Remove first entry from mNewParameters 656 // 5. Set mParamStatus 657 // 6. mParamCond.signal 658 // 7. mWaitWorkCV.wait with timeout (this is to avoid overwriting mParamStatus) 659 // 8. Unlock 660 Condition mParamCond; 661 Vector<String8> mNewParameters; 662 status_t mParamStatus; 663 664 Vector<ConfigEvent> mConfigEvents; 665 666 // These fields are written and read by thread itself without lock or barrier, 667 // and read by other threads without lock or barrier via standby() and device(). 668 // Because of the absence of a lock or barrier, any other thread that reads 669 // these fields must use the information in isolation, or be prepared to deal 670 // with possibility that it might be inconsistent with other information. 671 bool mStandby; // Whether thread is currently in standby. 672 audio_devices_t mDevice; // output device for PlaybackThread 673 // input + output devices for RecordThread 674 675 const audio_io_handle_t mId; 676 Vector< sp<EffectChain> > mEffectChains; 677 678 static const int kNameLength = 16; // prctl(PR_SET_NAME) limit 679 char mName[kNameLength]; 680 sp<IPowerManager> mPowerManager; 681 sp<IBinder> mWakeLockToken; 682 const sp<PMDeathRecipient> mDeathRecipient; 683 // list of suspended effects per session and per type. The first vector is 684 // keyed by session ID, the second by type UUID timeLow field 685 KeyedVector< int, KeyedVector< int, sp<SuspendedSessionDesc> > > mSuspendedSessions; 686 }; 687 688 struct stream_type_t { 689 stream_type_t() 690 : volume(1.0f), 691 mute(false) 692 { 693 } 694 float volume; 695 bool mute; 696 }; 697 698 // --- PlaybackThread --- 699 class PlaybackThread : public ThreadBase { 700 public: 701 702 enum mixer_state { 703 MIXER_IDLE, // no active tracks 704 MIXER_TRACKS_ENABLED, // at least one active track, but no track has any data ready 705 MIXER_TRACKS_READY // at least one active track, and at least one track has data 706 // standby mode does not have an enum value 707 // suspend by audio policy manager is orthogonal to mixer state 708 }; 709 710 // playback track 711 class Track : public TrackBase, public VolumeProvider { 712 public: 713 Track( PlaybackThread *thread, 714 const sp<Client>& client, 715 audio_stream_type_t streamType, 716 uint32_t sampleRate, 717 audio_format_t format, 718 uint32_t channelMask, 719 int frameCount, 720 const sp<IMemory>& sharedBuffer, 721 int sessionId, 722 IAudioFlinger::track_flags_t flags); 723 virtual ~Track(); 724 725 static void appendDumpHeader(String8& result); 726 void dump(char* buffer, size_t size); 727 virtual status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 728 int triggerSession = 0); 729 virtual void stop(); 730 void pause(); 731 732 void flush(); 733 void destroy(); 734 void mute(bool); 735 int name() const { return mName; } 736 737 audio_stream_type_t streamType() const { 738 return mStreamType; 739 } 740 status_t attachAuxEffect(int EffectId); 741 void setAuxBuffer(int EffectId, int32_t *buffer); 742 int32_t *auxBuffer() const { return mAuxBuffer; } 743 void setMainBuffer(int16_t *buffer) { mMainBuffer = buffer; } 744 int16_t *mainBuffer() const { return mMainBuffer; } 745 int auxEffectId() const { return mAuxEffectId; } 746 747 // implement FastMixerState::VolumeProvider interface 748 virtual uint32_t getVolumeLR(); 749 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 750 751 protected: 752 // for numerous 753 friend class PlaybackThread; 754 friend class MixerThread; 755 friend class DirectOutputThread; 756 757 Track(const Track&); 758 Track& operator = (const Track&); 759 760 // AudioBufferProvider interface 761 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts = kInvalidPTS); 762 // releaseBuffer() not overridden 763 764 virtual size_t framesReady() const; 765 766 bool isMuted() const { return mMute; } 767 bool isPausing() const { 768 return mState == PAUSING; 769 } 770 bool isPaused() const { 771 return mState == PAUSED; 772 } 773 bool isResuming() const { 774 return mState == RESUMING; 775 } 776 bool isReady() const; 777 void setPaused() { mState = PAUSED; } 778 void reset(); 779 780 bool isOutputTrack() const { 781 return (mStreamType == AUDIO_STREAM_CNT); 782 } 783 784 sp<IMemory> sharedBuffer() const { return mSharedBuffer; } 785 786 bool presentationComplete(size_t framesWritten, size_t audioHalFrames); 787 788 public: 789 void triggerEvents(AudioSystem::sync_event_t type); 790 virtual bool isTimedTrack() const { return false; } 791 bool isFastTrack() const { return (mFlags & IAudioFlinger::TRACK_FAST) != 0; } 792 793 protected: 794 795 // written by Track::mute() called by binder thread(s), without a mutex or barrier. 796 // read by Track::isMuted() called by playback thread, also without a mutex or barrier. 797 // The lack of mutex or barrier is safe because the mute status is only used by itself. 798 bool mMute; 799 800 // FILLED state is used for suppressing volume ramp at begin of playing 801 enum {FS_INVALID, FS_FILLING, FS_FILLED, FS_ACTIVE}; 802 mutable uint8_t mFillingUpStatus; 803 int8_t mRetryCount; 804 const sp<IMemory> mSharedBuffer; 805 bool mResetDone; 806 const audio_stream_type_t mStreamType; 807 int mName; // track name on the normal mixer, 808 // allocated statically at track creation time, 809 // and is even allocated (though unused) for fast tracks 810 // FIXME don't allocate track name for fast tracks 811 int16_t *mMainBuffer; 812 int32_t *mAuxBuffer; 813 int mAuxEffectId; 814 bool mHasVolumeController; 815 size_t mPresentationCompleteFrames; // number of frames written to the audio HAL 816 // when this track will be fully rendered 817 private: 818 IAudioFlinger::track_flags_t mFlags; 819 820 // The following fields are only for fast tracks, and should be in a subclass 821 int mFastIndex; // index within FastMixerState::mFastTracks[]; 822 // either mFastIndex == -1 if not isFastTrack() 823 // or 0 < mFastIndex < FastMixerState::kMaxFast because 824 // index 0 is reserved for normal mixer's submix; 825 // index is allocated statically at track creation time 826 // but the slot is only used if track is active 827 FastTrackUnderruns mObservedUnderruns; // Most recently observed value of 828 // mFastMixerDumpState.mTracks[mFastIndex].mUnderruns 829 uint32_t mUnderrunCount; // Counter of total number of underruns, never reset 830 volatile float mCachedVolume; // combined master volume and stream type volume; 831 // 'volatile' means accessed without lock or 832 // barrier, but is read/written atomically 833 }; // end of Track 834 835 class TimedTrack : public Track { 836 public: 837 static sp<TimedTrack> create(PlaybackThread *thread, 838 const sp<Client>& client, 839 audio_stream_type_t streamType, 840 uint32_t sampleRate, 841 audio_format_t format, 842 uint32_t channelMask, 843 int frameCount, 844 const sp<IMemory>& sharedBuffer, 845 int sessionId); 846 ~TimedTrack(); 847 848 class TimedBuffer { 849 public: 850 TimedBuffer(); 851 TimedBuffer(const sp<IMemory>& buffer, int64_t pts); 852 const sp<IMemory>& buffer() const { return mBuffer; } 853 int64_t pts() const { return mPTS; } 854 uint32_t position() const { return mPosition; } 855 void setPosition(uint32_t pos) { mPosition = pos; } 856 private: 857 sp<IMemory> mBuffer; 858 int64_t mPTS; 859 uint32_t mPosition; 860 }; 861 862 // Mixer facing methods. 863 virtual bool isTimedTrack() const { return true; } 864 virtual size_t framesReady() const; 865 866 // AudioBufferProvider interface 867 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, 868 int64_t pts); 869 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 870 871 // Client/App facing methods. 872 status_t allocateTimedBuffer(size_t size, 873 sp<IMemory>* buffer); 874 status_t queueTimedBuffer(const sp<IMemory>& buffer, 875 int64_t pts); 876 status_t setMediaTimeTransform(const LinearTransform& xform, 877 TimedAudioTrack::TargetTimeline target); 878 879 private: 880 TimedTrack(PlaybackThread *thread, 881 const sp<Client>& client, 882 audio_stream_type_t streamType, 883 uint32_t sampleRate, 884 audio_format_t format, 885 uint32_t channelMask, 886 int frameCount, 887 const sp<IMemory>& sharedBuffer, 888 int sessionId); 889 890 void timedYieldSamples_l(AudioBufferProvider::Buffer* buffer); 891 void timedYieldSilence_l(uint32_t numFrames, 892 AudioBufferProvider::Buffer* buffer); 893 void trimTimedBufferQueue_l(); 894 void trimTimedBufferQueueHead_l(const char* logTag); 895 void updateFramesPendingAfterTrim_l(const TimedBuffer& buf, 896 const char* logTag); 897 898 uint64_t mLocalTimeFreq; 899 LinearTransform mLocalTimeToSampleTransform; 900 LinearTransform mMediaTimeToSampleTransform; 901 sp<MemoryDealer> mTimedMemoryDealer; 902 903 Vector<TimedBuffer> mTimedBufferQueue; 904 bool mQueueHeadInFlight; 905 bool mTrimQueueHeadOnRelease; 906 uint32_t mFramesPendingInQueue; 907 908 uint8_t* mTimedSilenceBuffer; 909 uint32_t mTimedSilenceBufferSize; 910 mutable Mutex mTimedBufferQueueLock; 911 bool mTimedAudioOutputOnTime; 912 CCHelper mCCHelper; 913 914 Mutex mMediaTimeTransformLock; 915 LinearTransform mMediaTimeTransform; 916 bool mMediaTimeTransformValid; 917 TimedAudioTrack::TargetTimeline mMediaTimeTransformTarget; 918 }; 919 920 921 // playback track 922 class OutputTrack : public Track { 923 public: 924 925 class Buffer: public AudioBufferProvider::Buffer { 926 public: 927 int16_t *mBuffer; 928 }; 929 930 OutputTrack(PlaybackThread *thread, 931 DuplicatingThread *sourceThread, 932 uint32_t sampleRate, 933 audio_format_t format, 934 uint32_t channelMask, 935 int frameCount); 936 virtual ~OutputTrack(); 937 938 virtual status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 939 int triggerSession = 0); 940 virtual void stop(); 941 bool write(int16_t* data, uint32_t frames); 942 bool bufferQueueEmpty() const { return mBufferQueue.size() == 0; } 943 bool isActive() const { return mActive; } 944 const wp<ThreadBase>& thread() const { return mThread; } 945 946 private: 947 948 enum { 949 NO_MORE_BUFFERS = 0x80000001, // same in AudioTrack.h, ok to be different value 950 }; 951 952 status_t obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs); 953 void clearBufferQueue(); 954 955 // Maximum number of pending buffers allocated by OutputTrack::write() 956 static const uint8_t kMaxOverFlowBuffers = 10; 957 958 Vector < Buffer* > mBufferQueue; 959 AudioBufferProvider::Buffer mOutBuffer; 960 bool mActive; 961 DuplicatingThread* const mSourceThread; // for waitTimeMs() in write() 962 }; // end of OutputTrack 963 964 PlaybackThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 965 audio_io_handle_t id, uint32_t device, type_t type); 966 virtual ~PlaybackThread(); 967 968 status_t dump(int fd, const Vector<String16>& args); 969 970 // Thread virtuals 971 virtual status_t readyToRun(); 972 virtual bool threadLoop(); 973 974 // RefBase 975 virtual void onFirstRef(); 976 977protected: 978 // Code snippets that were lifted up out of threadLoop() 979 virtual void threadLoop_mix() = 0; 980 virtual void threadLoop_sleepTime() = 0; 981 virtual void threadLoop_write(); 982 virtual void threadLoop_standby(); 983 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 984 985 // prepareTracks_l reads and writes mActiveTracks, and returns 986 // the pending set of tracks to remove via Vector 'tracksToRemove'. The caller 987 // is responsible for clearing or destroying this Vector later on, when it 988 // is safe to do so. That will drop the final ref count and destroy the tracks. 989 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove) = 0; 990 991public: 992 993 virtual status_t initCheck() const { return (mOutput == NULL) ? NO_INIT : NO_ERROR; } 994 995 // return estimated latency in milliseconds, as reported by HAL 996 uint32_t latency() const; 997 // same, but lock must already be held 998 uint32_t latency_l() const; 999 1000 void setMasterVolume(float value); 1001 void setMasterMute(bool muted); 1002 1003 void setStreamVolume(audio_stream_type_t stream, float value); 1004 void setStreamMute(audio_stream_type_t stream, bool muted); 1005 1006 float streamVolume(audio_stream_type_t stream) const; 1007 1008 sp<Track> createTrack_l( 1009 const sp<AudioFlinger::Client>& client, 1010 audio_stream_type_t streamType, 1011 uint32_t sampleRate, 1012 audio_format_t format, 1013 uint32_t channelMask, 1014 int frameCount, 1015 const sp<IMemory>& sharedBuffer, 1016 int sessionId, 1017 IAudioFlinger::track_flags_t flags, 1018 pid_t tid, 1019 status_t *status); 1020 1021 AudioStreamOut* getOutput() const; 1022 AudioStreamOut* clearOutput(); 1023 virtual audio_stream_t* stream() const; 1024 1025 void suspend() { mSuspended++; } 1026 void restore() { if (mSuspended > 0) mSuspended--; } 1027 bool isSuspended() const { return (mSuspended > 0); } 1028 virtual String8 getParameters(const String8& keys); 1029 virtual void audioConfigChanged_l(int event, int param = 0); 1030 status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames); 1031 int16_t *mixBuffer() const { return mMixBuffer; }; 1032 1033 virtual void detachAuxEffect_l(int effectId); 1034 status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track> track, 1035 int EffectId); 1036 status_t attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track> track, 1037 int EffectId); 1038 1039 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1040 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1041 virtual uint32_t hasAudioSession(int sessionId); 1042 virtual uint32_t getStrategyForSession_l(int sessionId); 1043 1044 1045 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 1046 virtual bool isValidSyncEvent(const sp<SyncEvent>& event); 1047 void invalidateTracks(audio_stream_type_t streamType); 1048 1049 1050 protected: 1051 int16_t* mMixBuffer; 1052 uint32_t mSuspended; // suspend count, > 0 means suspended 1053 int mBytesWritten; 1054 private: 1055 // mMasterMute is in both PlaybackThread and in AudioFlinger. When a 1056 // PlaybackThread needs to find out if master-muted, it checks it's local 1057 // copy rather than the one in AudioFlinger. This optimization saves a lock. 1058 bool mMasterMute; 1059 void setMasterMute_l(bool muted) { mMasterMute = muted; } 1060 protected: 1061 SortedVector< wp<Track> > mActiveTracks; // FIXME check if this could be sp<> 1062 1063 // Allocate a track name for a given channel mask. 1064 // Returns name >= 0 if successful, -1 on failure. 1065 virtual int getTrackName_l(audio_channel_mask_t channelMask) = 0; 1066 virtual void deleteTrackName_l(int name) = 0; 1067 1068 // Time to sleep between cycles when: 1069 virtual uint32_t activeSleepTimeUs() const; // mixer state MIXER_TRACKS_ENABLED 1070 virtual uint32_t idleSleepTimeUs() const = 0; // mixer state MIXER_IDLE 1071 virtual uint32_t suspendSleepTimeUs() const = 0; // audio policy manager suspended us 1072 // No sleep when mixer state == MIXER_TRACKS_READY; relies on audio HAL stream->write() 1073 // No sleep in standby mode; waits on a condition 1074 1075 // Code snippets that are temporarily lifted up out of threadLoop() until the merge 1076 void checkSilentMode_l(); 1077 1078 // Non-trivial for DUPLICATING only 1079 virtual void saveOutputTracks() { } 1080 virtual void clearOutputTracks() { } 1081 1082 // Cache various calculated values, at threadLoop() entry and after a parameter change 1083 virtual void cacheParameters_l(); 1084 1085 virtual uint32_t correctLatency(uint32_t latency) const; 1086 1087 private: 1088 1089 friend class AudioFlinger; // for numerous 1090 1091 PlaybackThread(const Client&); 1092 PlaybackThread& operator = (const PlaybackThread&); 1093 1094 status_t addTrack_l(const sp<Track>& track); 1095 void destroyTrack_l(const sp<Track>& track); 1096 void removeTrack_l(const sp<Track>& track); 1097 1098 void readOutputParameters(); 1099 1100 virtual status_t dumpInternals(int fd, const Vector<String16>& args); 1101 status_t dumpTracks(int fd, const Vector<String16>& args); 1102 1103 SortedVector< sp<Track> > mTracks; 1104 // mStreamTypes[] uses 1 additional stream type internally for the OutputTrack used by DuplicatingThread 1105 stream_type_t mStreamTypes[AUDIO_STREAM_CNT + 1]; 1106 AudioStreamOut *mOutput; 1107 1108 float mMasterVolume; 1109 nsecs_t mLastWriteTime; 1110 int mNumWrites; 1111 int mNumDelayedWrites; 1112 bool mInWrite; 1113 1114 // FIXME rename these former local variables of threadLoop to standard "m" names 1115 nsecs_t standbyTime; 1116 size_t mixBufferSize; 1117 1118 // cached copies of activeSleepTimeUs() and idleSleepTimeUs() made by cacheParameters_l() 1119 uint32_t activeSleepTime; 1120 uint32_t idleSleepTime; 1121 1122 uint32_t sleepTime; 1123 1124 // mixer status returned by prepareTracks_l() 1125 mixer_state mMixerStatus; // current cycle 1126 // previous cycle when in prepareTracks_l() 1127 mixer_state mMixerStatusIgnoringFastTracks; 1128 // FIXME or a separate ready state per track 1129 1130 // FIXME move these declarations into the specific sub-class that needs them 1131 // MIXER only 1132 uint32_t sleepTimeShift; 1133 1134 // same as AudioFlinger::mStandbyTimeInNsecs except for DIRECT which uses a shorter value 1135 nsecs_t standbyDelay; 1136 1137 // MIXER only 1138 nsecs_t maxPeriod; 1139 1140 // DUPLICATING only 1141 uint32_t writeFrames; 1142 1143 private: 1144 // The HAL output sink is treated as non-blocking, but current implementation is blocking 1145 sp<NBAIO_Sink> mOutputSink; 1146 // If a fast mixer is present, the blocking pipe sink, otherwise clear 1147 sp<NBAIO_Sink> mPipeSink; 1148 // The current sink for the normal mixer to write it's (sub)mix, mOutputSink or mPipeSink 1149 sp<NBAIO_Sink> mNormalSink; 1150 // For dumpsys 1151 sp<NBAIO_Sink> mTeeSink; 1152 sp<NBAIO_Source> mTeeSource; 1153 uint32_t mScreenState; // cached copy of gScreenState 1154 public: 1155 virtual bool hasFastMixer() const = 0; 1156 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const 1157 { FastTrackUnderruns dummy; return dummy; } 1158 1159 protected: 1160 // accessed by both binder threads and within threadLoop(), lock on mutex needed 1161 unsigned mFastTrackAvailMask; // bit i set if fast track [i] is available 1162 1163 }; 1164 1165 class MixerThread : public PlaybackThread { 1166 public: 1167 MixerThread (const sp<AudioFlinger>& audioFlinger, 1168 AudioStreamOut* output, 1169 audio_io_handle_t id, 1170 uint32_t device, 1171 type_t type = MIXER); 1172 virtual ~MixerThread(); 1173 1174 // Thread virtuals 1175 1176 virtual bool checkForNewParameters_l(); 1177 virtual status_t dumpInternals(int fd, const Vector<String16>& args); 1178 1179 protected: 1180 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1181 virtual int getTrackName_l(audio_channel_mask_t channelMask); 1182 virtual void deleteTrackName_l(int name); 1183 virtual uint32_t idleSleepTimeUs() const; 1184 virtual uint32_t suspendSleepTimeUs() const; 1185 virtual void cacheParameters_l(); 1186 1187 // threadLoop snippets 1188 virtual void threadLoop_write(); 1189 virtual void threadLoop_standby(); 1190 virtual void threadLoop_mix(); 1191 virtual void threadLoop_sleepTime(); 1192 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 1193 virtual uint32_t correctLatency(uint32_t latency) const; 1194 1195 AudioMixer* mAudioMixer; // normal mixer 1196 private: 1197 // one-time initialization, no locks required 1198 FastMixer* mFastMixer; // non-NULL if there is also a fast mixer 1199 sp<AudioWatchdog> mAudioWatchdog; // non-0 if there is an audio watchdog thread 1200 1201 // contents are not guaranteed to be consistent, no locks required 1202 FastMixerDumpState mFastMixerDumpState; 1203#ifdef STATE_QUEUE_DUMP 1204 StateQueueObserverDump mStateQueueObserverDump; 1205 StateQueueMutatorDump mStateQueueMutatorDump; 1206#endif 1207 AudioWatchdogDump mAudioWatchdogDump; 1208 1209 // accessible only within the threadLoop(), no locks required 1210 // mFastMixer->sq() // for mutating and pushing state 1211 int32_t mFastMixerFutex; // for cold idle 1212 1213 public: 1214 virtual bool hasFastMixer() const { return mFastMixer != NULL; } 1215 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const { 1216 ALOG_ASSERT(fastIndex < FastMixerState::kMaxFastTracks); 1217 return mFastMixerDumpState.mTracks[fastIndex].mUnderruns; 1218 } 1219 }; 1220 1221 class DirectOutputThread : public PlaybackThread { 1222 public: 1223 1224 DirectOutputThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1225 audio_io_handle_t id, uint32_t device); 1226 virtual ~DirectOutputThread(); 1227 1228 // Thread virtuals 1229 1230 virtual bool checkForNewParameters_l(); 1231 1232 protected: 1233 virtual int getTrackName_l(audio_channel_mask_t channelMask); 1234 virtual void deleteTrackName_l(int name); 1235 virtual uint32_t activeSleepTimeUs() const; 1236 virtual uint32_t idleSleepTimeUs() const; 1237 virtual uint32_t suspendSleepTimeUs() const; 1238 virtual void cacheParameters_l(); 1239 1240 // threadLoop snippets 1241 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1242 virtual void threadLoop_mix(); 1243 virtual void threadLoop_sleepTime(); 1244 1245 // volumes last sent to audio HAL with stream->set_volume() 1246 float mLeftVolFloat; 1247 float mRightVolFloat; 1248 1249private: 1250 // prepareTracks_l() tells threadLoop_mix() the name of the single active track 1251 sp<Track> mActiveTrack; 1252 public: 1253 virtual bool hasFastMixer() const { return false; } 1254 }; 1255 1256 class DuplicatingThread : public MixerThread { 1257 public: 1258 DuplicatingThread (const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread, 1259 audio_io_handle_t id); 1260 virtual ~DuplicatingThread(); 1261 1262 // Thread virtuals 1263 void addOutputTrack(MixerThread* thread); 1264 void removeOutputTrack(MixerThread* thread); 1265 uint32_t waitTimeMs() const { return mWaitTimeMs; } 1266 protected: 1267 virtual uint32_t activeSleepTimeUs() const; 1268 1269 private: 1270 bool outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks); 1271 protected: 1272 // threadLoop snippets 1273 virtual void threadLoop_mix(); 1274 virtual void threadLoop_sleepTime(); 1275 virtual void threadLoop_write(); 1276 virtual void threadLoop_standby(); 1277 virtual void cacheParameters_l(); 1278 1279 private: 1280 // called from threadLoop, addOutputTrack, removeOutputTrack 1281 virtual void updateWaitTime_l(); 1282 protected: 1283 virtual void saveOutputTracks(); 1284 virtual void clearOutputTracks(); 1285 private: 1286 1287 uint32_t mWaitTimeMs; 1288 SortedVector < sp<OutputTrack> > outputTracks; 1289 SortedVector < sp<OutputTrack> > mOutputTracks; 1290 public: 1291 virtual bool hasFastMixer() const { return false; } 1292 }; 1293 1294 PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const; 1295 MixerThread *checkMixerThread_l(audio_io_handle_t output) const; 1296 RecordThread *checkRecordThread_l(audio_io_handle_t input) const; 1297 // no range check, AudioFlinger::mLock held 1298 bool streamMute_l(audio_stream_type_t stream) const 1299 { return mStreamTypes[stream].mute; } 1300 // no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held 1301 float streamVolume_l(audio_stream_type_t stream) const 1302 { return mStreamTypes[stream].volume; } 1303 void audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2); 1304 1305 // allocate an audio_io_handle_t, session ID, or effect ID 1306 uint32_t nextUniqueId(); 1307 1308 status_t moveEffectChain_l(int sessionId, 1309 PlaybackThread *srcThread, 1310 PlaybackThread *dstThread, 1311 bool reRegister); 1312 // return thread associated with primary hardware device, or NULL 1313 PlaybackThread *primaryPlaybackThread_l() const; 1314 uint32_t primaryOutputDevice_l() const; 1315 1316 sp<PlaybackThread> getEffectThread_l(int sessionId, int EffectId); 1317 1318 // server side of the client's IAudioTrack 1319 class TrackHandle : public android::BnAudioTrack { 1320 public: 1321 TrackHandle(const sp<PlaybackThread::Track>& track); 1322 virtual ~TrackHandle(); 1323 virtual sp<IMemory> getCblk() const; 1324 virtual status_t start(); 1325 virtual void stop(); 1326 virtual void flush(); 1327 virtual void mute(bool); 1328 virtual void pause(); 1329 virtual status_t attachAuxEffect(int effectId); 1330 virtual status_t allocateTimedBuffer(size_t size, 1331 sp<IMemory>* buffer); 1332 virtual status_t queueTimedBuffer(const sp<IMemory>& buffer, 1333 int64_t pts); 1334 virtual status_t setMediaTimeTransform(const LinearTransform& xform, 1335 int target); 1336 virtual status_t onTransact( 1337 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 1338 private: 1339 const sp<PlaybackThread::Track> mTrack; 1340 }; 1341 1342 void removeClient_l(pid_t pid); 1343 void removeNotificationClient(pid_t pid); 1344 1345 1346 // record thread 1347 class RecordThread : public ThreadBase, public AudioBufferProvider 1348 { 1349 public: 1350 1351 // record track 1352 class RecordTrack : public TrackBase { 1353 public: 1354 RecordTrack(RecordThread *thread, 1355 const sp<Client>& client, 1356 uint32_t sampleRate, 1357 audio_format_t format, 1358 uint32_t channelMask, 1359 int frameCount, 1360 int sessionId); 1361 virtual ~RecordTrack(); 1362 1363 virtual status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 1364 int triggerSession = 0); 1365 virtual void stop(); 1366 1367 bool overflow() { bool tmp = mOverflow; mOverflow = false; return tmp; } 1368 bool setOverflow() { bool tmp = mOverflow; mOverflow = true; return tmp; } 1369 1370 void dump(char* buffer, size_t size); 1371 1372 private: 1373 friend class AudioFlinger; // for mState 1374 1375 RecordTrack(const RecordTrack&); 1376 RecordTrack& operator = (const RecordTrack&); 1377 1378 // AudioBufferProvider interface 1379 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts = kInvalidPTS); 1380 // releaseBuffer() not overridden 1381 1382 bool mOverflow; 1383 }; 1384 1385 1386 RecordThread(const sp<AudioFlinger>& audioFlinger, 1387 AudioStreamIn *input, 1388 uint32_t sampleRate, 1389 uint32_t channels, 1390 audio_io_handle_t id, 1391 uint32_t device); 1392 virtual ~RecordThread(); 1393 1394 // Thread 1395 virtual bool threadLoop(); 1396 virtual status_t readyToRun(); 1397 1398 // RefBase 1399 virtual void onFirstRef(); 1400 1401 virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; } 1402 sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l( 1403 const sp<AudioFlinger::Client>& client, 1404 uint32_t sampleRate, 1405 audio_format_t format, 1406 int channelMask, 1407 int frameCount, 1408 int sessionId, 1409 status_t *status); 1410 1411 status_t start(RecordTrack* recordTrack, 1412 AudioSystem::sync_event_t event, 1413 int triggerSession); 1414 void stop(RecordTrack* recordTrack); 1415 status_t dump(int fd, const Vector<String16>& args); 1416 AudioStreamIn* getInput() const; 1417 AudioStreamIn* clearInput(); 1418 virtual audio_stream_t* stream() const; 1419 1420 // AudioBufferProvider interface 1421 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts); 1422 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 1423 1424 virtual bool checkForNewParameters_l(); 1425 virtual String8 getParameters(const String8& keys); 1426 virtual void audioConfigChanged_l(int event, int param = 0); 1427 void readInputParameters(); 1428 virtual unsigned int getInputFramesLost(); 1429 1430 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1431 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1432 virtual uint32_t hasAudioSession(int sessionId); 1433 RecordTrack* track(); 1434 1435 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 1436 virtual bool isValidSyncEvent(const sp<SyncEvent>& event); 1437 1438 static void syncStartEventCallback(const wp<SyncEvent>& event); 1439 void handleSyncStartEvent(const sp<SyncEvent>& event); 1440 1441 private: 1442 void clearSyncStartEvent(); 1443 1444 RecordThread(); 1445 AudioStreamIn *mInput; 1446 RecordTrack* mTrack; 1447 sp<RecordTrack> mActiveTrack; 1448 Condition mStartStopCond; 1449 AudioResampler *mResampler; 1450 int32_t *mRsmpOutBuffer; 1451 int16_t *mRsmpInBuffer; 1452 size_t mRsmpInIndex; 1453 size_t mInputBytes; 1454 const int mReqChannelCount; 1455 const uint32_t mReqSampleRate; 1456 ssize_t mBytesRead; 1457 // sync event triggering actual audio capture. Frames read before this event will 1458 // be dropped and therefore not read by the application. 1459 sp<SyncEvent> mSyncStartEvent; 1460 // number of captured frames to drop after the start sync event has been received. 1461 // when < 0, maximum frames to drop before starting capture even if sync event is 1462 // not received 1463 ssize_t mFramestoDrop; 1464 }; 1465 1466 // server side of the client's IAudioRecord 1467 class RecordHandle : public android::BnAudioRecord { 1468 public: 1469 RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack); 1470 virtual ~RecordHandle(); 1471 virtual sp<IMemory> getCblk() const; 1472 virtual status_t start(int event, int triggerSession); 1473 virtual void stop(); 1474 virtual status_t onTransact( 1475 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 1476 private: 1477 const sp<RecordThread::RecordTrack> mRecordTrack; 1478 }; 1479 1480 //--- Audio Effect Management 1481 1482 // EffectModule and EffectChain classes both have their own mutex to protect 1483 // state changes or resource modifications. Always respect the following order 1484 // if multiple mutexes must be acquired to avoid cross deadlock: 1485 // AudioFlinger -> ThreadBase -> EffectChain -> EffectModule 1486 1487 // The EffectModule class is a wrapper object controlling the effect engine implementation 1488 // in the effect library. It prevents concurrent calls to process() and command() functions 1489 // from different client threads. It keeps a list of EffectHandle objects corresponding 1490 // to all client applications using this effect and notifies applications of effect state, 1491 // control or parameter changes. It manages the activation state machine to send appropriate 1492 // reset, enable, disable commands to effect engine and provide volume 1493 // ramping when effects are activated/deactivated. 1494 // When controlling an auxiliary effect, the EffectModule also provides an input buffer used by 1495 // the attached track(s) to accumulate their auxiliary channel. 1496 class EffectModule: public RefBase { 1497 public: 1498 EffectModule(ThreadBase *thread, 1499 const wp<AudioFlinger::EffectChain>& chain, 1500 effect_descriptor_t *desc, 1501 int id, 1502 int sessionId); 1503 virtual ~EffectModule(); 1504 1505 enum effect_state { 1506 IDLE, 1507 RESTART, 1508 STARTING, 1509 ACTIVE, 1510 STOPPING, 1511 STOPPED, 1512 DESTROYED 1513 }; 1514 1515 int id() const { return mId; } 1516 void process(); 1517 void updateState(); 1518 status_t command(uint32_t cmdCode, 1519 uint32_t cmdSize, 1520 void *pCmdData, 1521 uint32_t *replySize, 1522 void *pReplyData); 1523 1524 void reset_l(); 1525 status_t configure(); 1526 status_t init(); 1527 effect_state state() const { 1528 return mState; 1529 } 1530 uint32_t status() { 1531 return mStatus; 1532 } 1533 int sessionId() const { 1534 return mSessionId; 1535 } 1536 status_t setEnabled(bool enabled); 1537 status_t setEnabled_l(bool enabled); 1538 bool isEnabled() const; 1539 bool isProcessEnabled() const; 1540 1541 void setInBuffer(int16_t *buffer) { mConfig.inputCfg.buffer.s16 = buffer; } 1542 int16_t *inBuffer() { return mConfig.inputCfg.buffer.s16; } 1543 void setOutBuffer(int16_t *buffer) { mConfig.outputCfg.buffer.s16 = buffer; } 1544 int16_t *outBuffer() { return mConfig.outputCfg.buffer.s16; } 1545 void setChain(const wp<EffectChain>& chain) { mChain = chain; } 1546 void setThread(const wp<ThreadBase>& thread) { mThread = thread; } 1547 const wp<ThreadBase>& thread() { return mThread; } 1548 1549 status_t addHandle(EffectHandle *handle); 1550 size_t disconnect(EffectHandle *handle, bool unpinIfLast); 1551 size_t removeHandle(EffectHandle *handle); 1552 1553 effect_descriptor_t& desc() { return mDescriptor; } 1554 wp<EffectChain>& chain() { return mChain; } 1555 1556 status_t setDevice(uint32_t device); 1557 status_t setVolume(uint32_t *left, uint32_t *right, bool controller); 1558 status_t setMode(audio_mode_t mode); 1559 status_t start(); 1560 status_t stop(); 1561 void setSuspended(bool suspended); 1562 bool suspended() const; 1563 1564 EffectHandle* controlHandle_l(); 1565 1566 bool isPinned() const { return mPinned; } 1567 void unPin() { mPinned = false; } 1568 bool purgeHandles(); 1569 void lock() { mLock.lock(); } 1570 void unlock() { mLock.unlock(); } 1571 1572 status_t dump(int fd, const Vector<String16>& args); 1573 1574 protected: 1575 friend class AudioFlinger; // for mHandles 1576 bool mPinned; 1577 1578 // Maximum time allocated to effect engines to complete the turn off sequence 1579 static const uint32_t MAX_DISABLE_TIME_MS = 10000; 1580 1581 EffectModule(const EffectModule&); 1582 EffectModule& operator = (const EffectModule&); 1583 1584 status_t start_l(); 1585 status_t stop_l(); 1586 1587mutable Mutex mLock; // mutex for process, commands and handles list protection 1588 wp<ThreadBase> mThread; // parent thread 1589 wp<EffectChain> mChain; // parent effect chain 1590 const int mId; // this instance unique ID 1591 const int mSessionId; // audio session ID 1592 effect_descriptor_t mDescriptor;// effect descriptor received from effect engine 1593 effect_config_t mConfig; // input and output audio configuration 1594 effect_handle_t mEffectInterface; // Effect module C API 1595 status_t mStatus; // initialization status 1596 effect_state mState; // current activation state 1597 Vector<EffectHandle *> mHandles; // list of client handles 1598 // First handle in mHandles has highest priority and controls the effect module 1599 uint32_t mMaxDisableWaitCnt; // maximum grace period before forcing an effect off after 1600 // sending disable command. 1601 uint32_t mDisableWaitCnt; // current process() calls count during disable period. 1602 bool mSuspended; // effect is suspended: temporarily disabled by framework 1603 }; 1604 1605 // The EffectHandle class implements the IEffect interface. It provides resources 1606 // to receive parameter updates, keeps track of effect control 1607 // ownership and state and has a pointer to the EffectModule object it is controlling. 1608 // There is one EffectHandle object for each application controlling (or using) 1609 // an effect module. 1610 // The EffectHandle is obtained by calling AudioFlinger::createEffect(). 1611 class EffectHandle: public android::BnEffect { 1612 public: 1613 1614 EffectHandle(const sp<EffectModule>& effect, 1615 const sp<AudioFlinger::Client>& client, 1616 const sp<IEffectClient>& effectClient, 1617 int32_t priority); 1618 virtual ~EffectHandle(); 1619 1620 // IEffect 1621 virtual status_t enable(); 1622 virtual status_t disable(); 1623 virtual status_t command(uint32_t cmdCode, 1624 uint32_t cmdSize, 1625 void *pCmdData, 1626 uint32_t *replySize, 1627 void *pReplyData); 1628 virtual void disconnect(); 1629 private: 1630 void disconnect(bool unpinIfLast); 1631 public: 1632 virtual sp<IMemory> getCblk() const { return mCblkMemory; } 1633 virtual status_t onTransact(uint32_t code, const Parcel& data, 1634 Parcel* reply, uint32_t flags); 1635 1636 1637 // Give or take control of effect module 1638 // - hasControl: true if control is given, false if removed 1639 // - signal: true client app should be signaled of change, false otherwise 1640 // - enabled: state of the effect when control is passed 1641 void setControl(bool hasControl, bool signal, bool enabled); 1642 void commandExecuted(uint32_t cmdCode, 1643 uint32_t cmdSize, 1644 void *pCmdData, 1645 uint32_t replySize, 1646 void *pReplyData); 1647 void setEnabled(bool enabled); 1648 bool enabled() const { return mEnabled; } 1649 1650 // Getters 1651 int id() const { return mEffect->id(); } 1652 int priority() const { return mPriority; } 1653 bool hasControl() const { return mHasControl; } 1654 sp<EffectModule> effect() const { return mEffect; } 1655 // destroyed_l() must be called with the associated EffectModule mLock held 1656 bool destroyed_l() const { return mDestroyed; } 1657 1658 void dump(char* buffer, size_t size); 1659 1660 protected: 1661 friend class AudioFlinger; // for mEffect, mHasControl, mEnabled 1662 EffectHandle(const EffectHandle&); 1663 EffectHandle& operator =(const EffectHandle&); 1664 1665 sp<EffectModule> mEffect; // pointer to controlled EffectModule 1666 sp<IEffectClient> mEffectClient; // callback interface for client notifications 1667 /*const*/ sp<Client> mClient; // client for shared memory allocation, see disconnect() 1668 sp<IMemory> mCblkMemory; // shared memory for control block 1669 effect_param_cblk_t* mCblk; // control block for deferred parameter setting via shared memory 1670 uint8_t* mBuffer; // pointer to parameter area in shared memory 1671 int mPriority; // client application priority to control the effect 1672 bool mHasControl; // true if this handle is controlling the effect 1673 bool mEnabled; // cached enable state: needed when the effect is 1674 // restored after being suspended 1675 bool mDestroyed; // Set to true by destructor. Access with EffectModule 1676 // mLock held 1677 }; 1678 1679 // the EffectChain class represents a group of effects associated to one audio session. 1680 // There can be any number of EffectChain objects per output mixer thread (PlaybackThread). 1681 // The EffecChain with session ID 0 contains global effects applied to the output mix. 1682 // Effects in this chain can be insert or auxiliary. Effects in other chains (attached to tracks) 1683 // are insert only. The EffectChain maintains an ordered list of effect module, the order corresponding 1684 // in the effect process order. When attached to a track (session ID != 0), it also provide it's own 1685 // input buffer used by the track as accumulation buffer. 1686 class EffectChain: public RefBase { 1687 public: 1688 EffectChain(const wp<ThreadBase>& wThread, int sessionId); 1689 EffectChain(ThreadBase *thread, int sessionId); 1690 virtual ~EffectChain(); 1691 1692 // special key used for an entry in mSuspendedEffects keyed vector 1693 // corresponding to a suspend all request. 1694 static const int kKeyForSuspendAll = 0; 1695 1696 // minimum duration during which we force calling effect process when last track on 1697 // a session is stopped or removed to allow effect tail to be rendered 1698 static const int kProcessTailDurationMs = 1000; 1699 1700 void process_l(); 1701 1702 void lock() { 1703 mLock.lock(); 1704 } 1705 void unlock() { 1706 mLock.unlock(); 1707 } 1708 1709 status_t addEffect_l(const sp<EffectModule>& handle); 1710 size_t removeEffect_l(const sp<EffectModule>& handle); 1711 1712 int sessionId() const { return mSessionId; } 1713 void setSessionId(int sessionId) { mSessionId = sessionId; } 1714 1715 sp<EffectModule> getEffectFromDesc_l(effect_descriptor_t *descriptor); 1716 sp<EffectModule> getEffectFromId_l(int id); 1717 sp<EffectModule> getEffectFromType_l(const effect_uuid_t *type); 1718 bool setVolume_l(uint32_t *left, uint32_t *right); 1719 void setDevice_l(uint32_t device); 1720 void setMode_l(audio_mode_t mode); 1721 1722 void setInBuffer(int16_t *buffer, bool ownsBuffer = false) { 1723 mInBuffer = buffer; 1724 mOwnInBuffer = ownsBuffer; 1725 } 1726 int16_t *inBuffer() const { 1727 return mInBuffer; 1728 } 1729 void setOutBuffer(int16_t *buffer) { 1730 mOutBuffer = buffer; 1731 } 1732 int16_t *outBuffer() const { 1733 return mOutBuffer; 1734 } 1735 1736 void incTrackCnt() { android_atomic_inc(&mTrackCnt); } 1737 void decTrackCnt() { android_atomic_dec(&mTrackCnt); } 1738 int32_t trackCnt() const { return android_atomic_acquire_load(&mTrackCnt); } 1739 1740 void incActiveTrackCnt() { android_atomic_inc(&mActiveTrackCnt); 1741 mTailBufferCount = mMaxTailBuffers; } 1742 void decActiveTrackCnt() { android_atomic_dec(&mActiveTrackCnt); } 1743 int32_t activeTrackCnt() const { return android_atomic_acquire_load(&mActiveTrackCnt); } 1744 1745 uint32_t strategy() const { return mStrategy; } 1746 void setStrategy(uint32_t strategy) 1747 { mStrategy = strategy; } 1748 1749 // suspend effect of the given type 1750 void setEffectSuspended_l(const effect_uuid_t *type, 1751 bool suspend); 1752 // suspend all eligible effects 1753 void setEffectSuspendedAll_l(bool suspend); 1754 // check if effects should be suspend or restored when a given effect is enable or disabled 1755 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1756 bool enabled); 1757 1758 void clearInputBuffer(); 1759 1760 status_t dump(int fd, const Vector<String16>& args); 1761 1762 protected: 1763 friend class AudioFlinger; // for mThread, mEffects 1764 EffectChain(const EffectChain&); 1765 EffectChain& operator =(const EffectChain&); 1766 1767 class SuspendedEffectDesc : public RefBase { 1768 public: 1769 SuspendedEffectDesc() : mRefCount(0) {} 1770 1771 int mRefCount; 1772 effect_uuid_t mType; 1773 wp<EffectModule> mEffect; 1774 }; 1775 1776 // get a list of effect modules to suspend when an effect of the type 1777 // passed is enabled. 1778 void getSuspendEligibleEffects(Vector< sp<EffectModule> > &effects); 1779 1780 // get an effect module if it is currently enable 1781 sp<EffectModule> getEffectIfEnabled(const effect_uuid_t *type); 1782 // true if the effect whose descriptor is passed can be suspended 1783 // OEMs can modify the rules implemented in this method to exclude specific effect 1784 // types or implementations from the suspend/restore mechanism. 1785 bool isEffectEligibleForSuspend(const effect_descriptor_t& desc); 1786 1787 void clearInputBuffer_l(sp<ThreadBase> thread); 1788 1789 wp<ThreadBase> mThread; // parent mixer thread 1790 Mutex mLock; // mutex protecting effect list 1791 Vector< sp<EffectModule> > mEffects; // list of effect modules 1792 int mSessionId; // audio session ID 1793 int16_t *mInBuffer; // chain input buffer 1794 int16_t *mOutBuffer; // chain output buffer 1795 1796 // 'volatile' here means these are accessed with atomic operations instead of mutex 1797 volatile int32_t mActiveTrackCnt; // number of active tracks connected 1798 volatile int32_t mTrackCnt; // number of tracks connected 1799 1800 int32_t mTailBufferCount; // current effect tail buffer count 1801 int32_t mMaxTailBuffers; // maximum effect tail buffers 1802 bool mOwnInBuffer; // true if the chain owns its input buffer 1803 int mVolumeCtrlIdx; // index of insert effect having control over volume 1804 uint32_t mLeftVolume; // previous volume on left channel 1805 uint32_t mRightVolume; // previous volume on right channel 1806 uint32_t mNewLeftVolume; // new volume on left channel 1807 uint32_t mNewRightVolume; // new volume on right channel 1808 uint32_t mStrategy; // strategy for this effect chain 1809 // mSuspendedEffects lists all effects currently suspended in the chain. 1810 // Use effect type UUID timelow field as key. There is no real risk of identical 1811 // timeLow fields among effect type UUIDs. 1812 // Updated by updateSuspendedSessions_l() only. 1813 KeyedVector< int, sp<SuspendedEffectDesc> > mSuspendedEffects; 1814 }; 1815 1816 // AudioStreamOut and AudioStreamIn are immutable, so their fields are const. 1817 // For emphasis, we could also make all pointers to them be "const *", 1818 // but that would clutter the code unnecessarily. 1819 1820 struct AudioStreamOut { 1821 audio_hw_device_t* const hwDev; 1822 audio_stream_out_t* const stream; 1823 1824 AudioStreamOut(audio_hw_device_t *dev, audio_stream_out_t *out) : 1825 hwDev(dev), stream(out) {} 1826 }; 1827 1828 struct AudioStreamIn { 1829 audio_hw_device_t* const hwDev; 1830 audio_stream_in_t* const stream; 1831 1832 AudioStreamIn(audio_hw_device_t *dev, audio_stream_in_t *in) : 1833 hwDev(dev), stream(in) {} 1834 }; 1835 1836 // for mAudioSessionRefs only 1837 struct AudioSessionRef { 1838 AudioSessionRef(int sessionid, pid_t pid) : 1839 mSessionid(sessionid), mPid(pid), mCnt(1) {} 1840 const int mSessionid; 1841 const pid_t mPid; 1842 int mCnt; 1843 }; 1844 1845 enum master_volume_support { 1846 // MVS_NONE: 1847 // Audio HAL has no support for master volume, either setting or 1848 // getting. All master volume control must be implemented in SW by the 1849 // AudioFlinger mixing core. 1850 MVS_NONE, 1851 1852 // MVS_SETONLY: 1853 // Audio HAL has support for setting master volume, but not for getting 1854 // master volume (original HAL design did not include a getter). 1855 // AudioFlinger needs to keep track of the last set master volume in 1856 // addition to needing to set an initial, default, master volume at HAL 1857 // load time. 1858 MVS_SETONLY, 1859 1860 // MVS_FULL: 1861 // Audio HAL has support both for setting and getting master volume. 1862 // AudioFlinger should send all set and get master volume requests 1863 // directly to the HAL. 1864 MVS_FULL, 1865 }; 1866 1867 class AudioHwDevice { 1868 public: 1869 AudioHwDevice(const char *moduleName, audio_hw_device_t *hwDevice) : 1870 mModuleName(strdup(moduleName)), mHwDevice(hwDevice){} 1871 ~AudioHwDevice() { free((void *)mModuleName); } 1872 1873 const char *moduleName() const { return mModuleName; } 1874 audio_hw_device_t *hwDevice() const { return mHwDevice; } 1875 private: 1876 const char * const mModuleName; 1877 audio_hw_device_t * const mHwDevice; 1878 }; 1879 1880 mutable Mutex mLock; 1881 1882 DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client() 1883 1884 mutable Mutex mHardwareLock; 1885 // NOTE: If both mLock and mHardwareLock mutexes must be held, 1886 // always take mLock before mHardwareLock 1887 1888 // These two fields are immutable after onFirstRef(), so no lock needed to access 1889 audio_hw_device_t* mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL 1890 DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*> mAudioHwDevs; 1891 1892 // for dump, indicates which hardware operation is currently in progress (but not stream ops) 1893 enum hardware_call_state { 1894 AUDIO_HW_IDLE = 0, // no operation in progress 1895 AUDIO_HW_INIT, // init_check 1896 AUDIO_HW_OUTPUT_OPEN, // open_output_stream 1897 AUDIO_HW_OUTPUT_CLOSE, // unused 1898 AUDIO_HW_INPUT_OPEN, // unused 1899 AUDIO_HW_INPUT_CLOSE, // unused 1900 AUDIO_HW_STANDBY, // unused 1901 AUDIO_HW_SET_MASTER_VOLUME, // set_master_volume 1902 AUDIO_HW_GET_ROUTING, // unused 1903 AUDIO_HW_SET_ROUTING, // unused 1904 AUDIO_HW_GET_MODE, // unused 1905 AUDIO_HW_SET_MODE, // set_mode 1906 AUDIO_HW_GET_MIC_MUTE, // get_mic_mute 1907 AUDIO_HW_SET_MIC_MUTE, // set_mic_mute 1908 AUDIO_HW_SET_VOICE_VOLUME, // set_voice_volume 1909 AUDIO_HW_SET_PARAMETER, // set_parameters 1910 AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size 1911 AUDIO_HW_GET_MASTER_VOLUME, // get_master_volume 1912 AUDIO_HW_GET_PARAMETER, // get_parameters 1913 }; 1914 1915 mutable hardware_call_state mHardwareStatus; // for dump only 1916 1917 1918 DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads; 1919 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 1920 1921 // both are protected by mLock 1922 float mMasterVolume; 1923 float mMasterVolumeSW; 1924 master_volume_support mMasterVolumeSupportLvl; 1925 bool mMasterMute; 1926 1927 DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads; 1928 1929 DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients; 1930 volatile int32_t mNextUniqueId; // updated by android_atomic_inc 1931 audio_mode_t mMode; 1932 bool mBtNrecIsOff; 1933 1934 // protected by mLock 1935 Vector<AudioSessionRef*> mAudioSessionRefs; 1936 1937 float masterVolume_l() const; 1938 float masterVolumeSW_l() const { return mMasterVolumeSW; } 1939 bool masterMute_l() const { return mMasterMute; } 1940 audio_module_handle_t loadHwModule_l(const char *name); 1941 1942 Vector < sp<SyncEvent> > mPendingSyncEvents; // sync events awaiting for a session 1943 // to be created 1944 1945private: 1946 sp<Client> registerPid_l(pid_t pid); // always returns non-0 1947 1948}; 1949 1950 1951// ---------------------------------------------------------------------------- 1952 1953}; // namespace android 1954 1955#endif // ANDROID_AUDIO_FLINGER_H 1956