AudioFlinger.h revision 9f34a36d9cdb9595c288e50ffe00da038bc8abb9
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 protected: 793 794 // we don't really need a lock for these 795 volatile bool mMute; 796 // FILLED state is used for suppressing volume ramp at begin of playing 797 enum {FS_INVALID, FS_FILLING, FS_FILLED, FS_ACTIVE}; 798 mutable uint8_t mFillingUpStatus; 799 int8_t mRetryCount; 800 const sp<IMemory> mSharedBuffer; 801 bool mResetDone; 802 const audio_stream_type_t mStreamType; 803 int mName; // track name on the normal mixer, 804 // allocated statically at track creation time, 805 // and is even allocated (though unused) for fast tracks 806 // FIXME don't allocate track name for fast tracks 807 int16_t *mMainBuffer; 808 int32_t *mAuxBuffer; 809 int mAuxEffectId; 810 bool mHasVolumeController; 811 size_t mPresentationCompleteFrames; // number of frames written to the audio HAL 812 // when this track will be fully rendered 813 private: 814 IAudioFlinger::track_flags_t mFlags; 815 816 // The following fields are only for fast tracks, and should be in a subclass 817 int mFastIndex; // index within FastMixerState::mFastTracks[]; 818 // either mFastIndex == -1 if not isFastTrack() 819 // or 0 < mFastIndex < FastMixerState::kMaxFast because 820 // index 0 is reserved for normal mixer's submix; 821 // index is allocated statically at track creation time 822 // but the slot is only used if track is active 823 FastTrackUnderruns mObservedUnderruns; // Most recently observed value of 824 // mFastMixerDumpState.mTracks[mFastIndex].mUnderruns 825 uint32_t mUnderrunCount; // Counter of total number of underruns, never reset 826 volatile float mCachedVolume; // combined master volume and stream type volume; 827 // 'volatile' means accessed without lock or 828 // barrier, but is read/written atomically 829 }; // end of Track 830 831 class TimedTrack : public Track { 832 public: 833 static sp<TimedTrack> create(PlaybackThread *thread, 834 const sp<Client>& client, 835 audio_stream_type_t streamType, 836 uint32_t sampleRate, 837 audio_format_t format, 838 uint32_t channelMask, 839 int frameCount, 840 const sp<IMemory>& sharedBuffer, 841 int sessionId); 842 ~TimedTrack(); 843 844 class TimedBuffer { 845 public: 846 TimedBuffer(); 847 TimedBuffer(const sp<IMemory>& buffer, int64_t pts); 848 const sp<IMemory>& buffer() const { return mBuffer; } 849 int64_t pts() const { return mPTS; } 850 uint32_t position() const { return mPosition; } 851 void setPosition(uint32_t pos) { mPosition = pos; } 852 private: 853 sp<IMemory> mBuffer; 854 int64_t mPTS; 855 uint32_t mPosition; 856 }; 857 858 // Mixer facing methods. 859 virtual bool isTimedTrack() const { return true; } 860 virtual size_t framesReady() const; 861 862 // AudioBufferProvider interface 863 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, 864 int64_t pts); 865 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 866 867 // Client/App facing methods. 868 status_t allocateTimedBuffer(size_t size, 869 sp<IMemory>* buffer); 870 status_t queueTimedBuffer(const sp<IMemory>& buffer, 871 int64_t pts); 872 status_t setMediaTimeTransform(const LinearTransform& xform, 873 TimedAudioTrack::TargetTimeline target); 874 875 private: 876 TimedTrack(PlaybackThread *thread, 877 const sp<Client>& client, 878 audio_stream_type_t streamType, 879 uint32_t sampleRate, 880 audio_format_t format, 881 uint32_t channelMask, 882 int frameCount, 883 const sp<IMemory>& sharedBuffer, 884 int sessionId); 885 886 void timedYieldSamples_l(AudioBufferProvider::Buffer* buffer); 887 void timedYieldSilence_l(uint32_t numFrames, 888 AudioBufferProvider::Buffer* buffer); 889 void trimTimedBufferQueue_l(); 890 void trimTimedBufferQueueHead_l(const char* logTag); 891 void updateFramesPendingAfterTrim_l(const TimedBuffer& buf, 892 const char* logTag); 893 894 uint64_t mLocalTimeFreq; 895 LinearTransform mLocalTimeToSampleTransform; 896 LinearTransform mMediaTimeToSampleTransform; 897 sp<MemoryDealer> mTimedMemoryDealer; 898 899 Vector<TimedBuffer> mTimedBufferQueue; 900 bool mQueueHeadInFlight; 901 bool mTrimQueueHeadOnRelease; 902 uint32_t mFramesPendingInQueue; 903 904 uint8_t* mTimedSilenceBuffer; 905 uint32_t mTimedSilenceBufferSize; 906 mutable Mutex mTimedBufferQueueLock; 907 bool mTimedAudioOutputOnTime; 908 CCHelper mCCHelper; 909 910 Mutex mMediaTimeTransformLock; 911 LinearTransform mMediaTimeTransform; 912 bool mMediaTimeTransformValid; 913 TimedAudioTrack::TargetTimeline mMediaTimeTransformTarget; 914 }; 915 916 917 // playback track 918 class OutputTrack : public Track { 919 public: 920 921 class Buffer: public AudioBufferProvider::Buffer { 922 public: 923 int16_t *mBuffer; 924 }; 925 926 OutputTrack(PlaybackThread *thread, 927 DuplicatingThread *sourceThread, 928 uint32_t sampleRate, 929 audio_format_t format, 930 uint32_t channelMask, 931 int frameCount); 932 virtual ~OutputTrack(); 933 934 virtual status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 935 int triggerSession = 0); 936 virtual void stop(); 937 bool write(int16_t* data, uint32_t frames); 938 bool bufferQueueEmpty() const { return mBufferQueue.size() == 0; } 939 bool isActive() const { return mActive; } 940 const wp<ThreadBase>& thread() const { return mThread; } 941 942 private: 943 944 enum { 945 NO_MORE_BUFFERS = 0x80000001, // same in AudioTrack.h, ok to be different value 946 }; 947 948 status_t obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs); 949 void clearBufferQueue(); 950 951 // Maximum number of pending buffers allocated by OutputTrack::write() 952 static const uint8_t kMaxOverFlowBuffers = 10; 953 954 Vector < Buffer* > mBufferQueue; 955 AudioBufferProvider::Buffer mOutBuffer; 956 bool mActive; 957 DuplicatingThread* const mSourceThread; // for waitTimeMs() in write() 958 }; // end of OutputTrack 959 960 PlaybackThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 961 audio_io_handle_t id, uint32_t device, type_t type); 962 virtual ~PlaybackThread(); 963 964 status_t dump(int fd, const Vector<String16>& args); 965 966 // Thread virtuals 967 virtual status_t readyToRun(); 968 virtual bool threadLoop(); 969 970 // RefBase 971 virtual void onFirstRef(); 972 973protected: 974 // Code snippets that were lifted up out of threadLoop() 975 virtual void threadLoop_mix() = 0; 976 virtual void threadLoop_sleepTime() = 0; 977 virtual void threadLoop_write(); 978 virtual void threadLoop_standby(); 979 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 980 981 // prepareTracks_l reads and writes mActiveTracks, and returns 982 // the pending set of tracks to remove via Vector 'tracksToRemove'. The caller 983 // is responsible for clearing or destroying this Vector later on, when it 984 // is safe to do so. That will drop the final ref count and destroy the tracks. 985 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove) = 0; 986 987public: 988 989 virtual status_t initCheck() const { return (mOutput == NULL) ? NO_INIT : NO_ERROR; } 990 991 // return estimated latency in milliseconds, as reported by HAL 992 uint32_t latency() const; 993 // same, but lock must already be held 994 uint32_t latency_l() const; 995 996 void setMasterVolume(float value); 997 void setMasterMute(bool muted); 998 999 void setStreamVolume(audio_stream_type_t stream, float value); 1000 void setStreamMute(audio_stream_type_t stream, bool muted); 1001 1002 float streamVolume(audio_stream_type_t stream) const; 1003 1004 sp<Track> createTrack_l( 1005 const sp<AudioFlinger::Client>& client, 1006 audio_stream_type_t streamType, 1007 uint32_t sampleRate, 1008 audio_format_t format, 1009 uint32_t channelMask, 1010 int frameCount, 1011 const sp<IMemory>& sharedBuffer, 1012 int sessionId, 1013 IAudioFlinger::track_flags_t flags, 1014 pid_t tid, 1015 status_t *status); 1016 1017 AudioStreamOut* getOutput() const; 1018 AudioStreamOut* clearOutput(); 1019 virtual audio_stream_t* stream() const; 1020 1021 void suspend() { mSuspended++; } 1022 void restore() { if (mSuspended > 0) mSuspended--; } 1023 bool isSuspended() const { return (mSuspended > 0); } 1024 virtual String8 getParameters(const String8& keys); 1025 virtual void audioConfigChanged_l(int event, int param = 0); 1026 status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames); 1027 int16_t *mixBuffer() const { return mMixBuffer; }; 1028 1029 virtual void detachAuxEffect_l(int effectId); 1030 status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track> track, 1031 int EffectId); 1032 status_t attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track> track, 1033 int EffectId); 1034 1035 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1036 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1037 virtual uint32_t hasAudioSession(int sessionId); 1038 virtual uint32_t getStrategyForSession_l(int sessionId); 1039 1040 1041 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 1042 virtual bool isValidSyncEvent(const sp<SyncEvent>& event); 1043 void invalidateTracks(audio_stream_type_t streamType); 1044 1045 1046 protected: 1047 int16_t* mMixBuffer; 1048 uint32_t mSuspended; // suspend count, > 0 means suspended 1049 int mBytesWritten; 1050 private: 1051 // mMasterMute is in both PlaybackThread and in AudioFlinger. When a 1052 // PlaybackThread needs to find out if master-muted, it checks it's local 1053 // copy rather than the one in AudioFlinger. This optimization saves a lock. 1054 bool mMasterMute; 1055 void setMasterMute_l(bool muted) { mMasterMute = muted; } 1056 protected: 1057 SortedVector< wp<Track> > mActiveTracks; // FIXME check if this could be sp<> 1058 1059 // Allocate a track name for a given channel mask. 1060 // Returns name >= 0 if successful, -1 on failure. 1061 virtual int getTrackName_l(audio_channel_mask_t channelMask) = 0; 1062 virtual void deleteTrackName_l(int name) = 0; 1063 1064 // Time to sleep between cycles when: 1065 virtual uint32_t activeSleepTimeUs() const; // mixer state MIXER_TRACKS_ENABLED 1066 virtual uint32_t idleSleepTimeUs() const = 0; // mixer state MIXER_IDLE 1067 virtual uint32_t suspendSleepTimeUs() const = 0; // audio policy manager suspended us 1068 // No sleep when mixer state == MIXER_TRACKS_READY; relies on audio HAL stream->write() 1069 // No sleep in standby mode; waits on a condition 1070 1071 // Code snippets that are temporarily lifted up out of threadLoop() until the merge 1072 void checkSilentMode_l(); 1073 1074 // Non-trivial for DUPLICATING only 1075 virtual void saveOutputTracks() { } 1076 virtual void clearOutputTracks() { } 1077 1078 // Cache various calculated values, at threadLoop() entry and after a parameter change 1079 virtual void cacheParameters_l(); 1080 1081 virtual uint32_t correctLatency(uint32_t latency) const; 1082 1083 private: 1084 1085 friend class AudioFlinger; // for numerous 1086 1087 PlaybackThread(const Client&); 1088 PlaybackThread& operator = (const PlaybackThread&); 1089 1090 status_t addTrack_l(const sp<Track>& track); 1091 void destroyTrack_l(const sp<Track>& track); 1092 void removeTrack_l(const sp<Track>& track); 1093 1094 void readOutputParameters(); 1095 1096 virtual status_t dumpInternals(int fd, const Vector<String16>& args); 1097 status_t dumpTracks(int fd, const Vector<String16>& args); 1098 1099 SortedVector< sp<Track> > mTracks; 1100 // mStreamTypes[] uses 1 additional stream type internally for the OutputTrack used by DuplicatingThread 1101 stream_type_t mStreamTypes[AUDIO_STREAM_CNT + 1]; 1102 AudioStreamOut *mOutput; 1103 1104 float mMasterVolume; 1105 nsecs_t mLastWriteTime; 1106 int mNumWrites; 1107 int mNumDelayedWrites; 1108 bool mInWrite; 1109 1110 // FIXME rename these former local variables of threadLoop to standard "m" names 1111 nsecs_t standbyTime; 1112 size_t mixBufferSize; 1113 1114 // cached copies of activeSleepTimeUs() and idleSleepTimeUs() made by cacheParameters_l() 1115 uint32_t activeSleepTime; 1116 uint32_t idleSleepTime; 1117 1118 uint32_t sleepTime; 1119 1120 // mixer status returned by prepareTracks_l() 1121 mixer_state mMixerStatus; // current cycle 1122 // previous cycle when in prepareTracks_l() 1123 mixer_state mMixerStatusIgnoringFastTracks; 1124 // FIXME or a separate ready state per track 1125 1126 // FIXME move these declarations into the specific sub-class that needs them 1127 // MIXER only 1128 uint32_t sleepTimeShift; 1129 1130 // same as AudioFlinger::mStandbyTimeInNsecs except for DIRECT which uses a shorter value 1131 nsecs_t standbyDelay; 1132 1133 // MIXER only 1134 nsecs_t maxPeriod; 1135 1136 // DUPLICATING only 1137 uint32_t writeFrames; 1138 1139 private: 1140 // The HAL output sink is treated as non-blocking, but current implementation is blocking 1141 sp<NBAIO_Sink> mOutputSink; 1142 // If a fast mixer is present, the blocking pipe sink, otherwise clear 1143 sp<NBAIO_Sink> mPipeSink; 1144 // The current sink for the normal mixer to write it's (sub)mix, mOutputSink or mPipeSink 1145 sp<NBAIO_Sink> mNormalSink; 1146 // For dumpsys 1147 sp<NBAIO_Sink> mTeeSink; 1148 sp<NBAIO_Source> mTeeSource; 1149 uint32_t mScreenState; // cached copy of gScreenState 1150 public: 1151 virtual bool hasFastMixer() const = 0; 1152 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const 1153 { FastTrackUnderruns dummy; return dummy; } 1154 1155 protected: 1156 // accessed by both binder threads and within threadLoop(), lock on mutex needed 1157 unsigned mFastTrackAvailMask; // bit i set if fast track [i] is available 1158 1159 }; 1160 1161 class MixerThread : public PlaybackThread { 1162 public: 1163 MixerThread (const sp<AudioFlinger>& audioFlinger, 1164 AudioStreamOut* output, 1165 audio_io_handle_t id, 1166 uint32_t device, 1167 type_t type = MIXER); 1168 virtual ~MixerThread(); 1169 1170 // Thread virtuals 1171 1172 virtual bool checkForNewParameters_l(); 1173 virtual status_t dumpInternals(int fd, const Vector<String16>& args); 1174 1175 protected: 1176 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1177 virtual int getTrackName_l(audio_channel_mask_t channelMask); 1178 virtual void deleteTrackName_l(int name); 1179 virtual uint32_t idleSleepTimeUs() const; 1180 virtual uint32_t suspendSleepTimeUs() const; 1181 virtual void cacheParameters_l(); 1182 1183 // threadLoop snippets 1184 virtual void threadLoop_write(); 1185 virtual void threadLoop_standby(); 1186 virtual void threadLoop_mix(); 1187 virtual void threadLoop_sleepTime(); 1188 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 1189 virtual uint32_t correctLatency(uint32_t latency) const; 1190 1191 AudioMixer* mAudioMixer; // normal mixer 1192 private: 1193 // one-time initialization, no locks required 1194 FastMixer* mFastMixer; // non-NULL if there is also a fast mixer 1195 sp<AudioWatchdog> mAudioWatchdog; // non-0 if there is an audio watchdog thread 1196 1197 // contents are not guaranteed to be consistent, no locks required 1198 FastMixerDumpState mFastMixerDumpState; 1199#ifdef STATE_QUEUE_DUMP 1200 StateQueueObserverDump mStateQueueObserverDump; 1201 StateQueueMutatorDump mStateQueueMutatorDump; 1202#endif 1203 AudioWatchdogDump mAudioWatchdogDump; 1204 1205 // accessible only within the threadLoop(), no locks required 1206 // mFastMixer->sq() // for mutating and pushing state 1207 int32_t mFastMixerFutex; // for cold idle 1208 1209 public: 1210 virtual bool hasFastMixer() const { return mFastMixer != NULL; } 1211 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const { 1212 ALOG_ASSERT(fastIndex < FastMixerState::kMaxFastTracks); 1213 return mFastMixerDumpState.mTracks[fastIndex].mUnderruns; 1214 } 1215 }; 1216 1217 class DirectOutputThread : public PlaybackThread { 1218 public: 1219 1220 DirectOutputThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1221 audio_io_handle_t id, uint32_t device); 1222 virtual ~DirectOutputThread(); 1223 1224 // Thread virtuals 1225 1226 virtual bool checkForNewParameters_l(); 1227 1228 protected: 1229 virtual int getTrackName_l(audio_channel_mask_t channelMask); 1230 virtual void deleteTrackName_l(int name); 1231 virtual uint32_t activeSleepTimeUs() const; 1232 virtual uint32_t idleSleepTimeUs() const; 1233 virtual uint32_t suspendSleepTimeUs() const; 1234 virtual void cacheParameters_l(); 1235 1236 // threadLoop snippets 1237 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1238 virtual void threadLoop_mix(); 1239 virtual void threadLoop_sleepTime(); 1240 1241 // volumes last sent to audio HAL with stream->set_volume() 1242 float mLeftVolFloat; 1243 float mRightVolFloat; 1244 1245private: 1246 // prepareTracks_l() tells threadLoop_mix() the name of the single active track 1247 sp<Track> mActiveTrack; 1248 public: 1249 virtual bool hasFastMixer() const { return false; } 1250 }; 1251 1252 class DuplicatingThread : public MixerThread { 1253 public: 1254 DuplicatingThread (const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread, 1255 audio_io_handle_t id); 1256 virtual ~DuplicatingThread(); 1257 1258 // Thread virtuals 1259 void addOutputTrack(MixerThread* thread); 1260 void removeOutputTrack(MixerThread* thread); 1261 uint32_t waitTimeMs() const { return mWaitTimeMs; } 1262 protected: 1263 virtual uint32_t activeSleepTimeUs() const; 1264 1265 private: 1266 bool outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks); 1267 protected: 1268 // threadLoop snippets 1269 virtual void threadLoop_mix(); 1270 virtual void threadLoop_sleepTime(); 1271 virtual void threadLoop_write(); 1272 virtual void threadLoop_standby(); 1273 virtual void cacheParameters_l(); 1274 1275 private: 1276 // called from threadLoop, addOutputTrack, removeOutputTrack 1277 virtual void updateWaitTime_l(); 1278 protected: 1279 virtual void saveOutputTracks(); 1280 virtual void clearOutputTracks(); 1281 private: 1282 1283 uint32_t mWaitTimeMs; 1284 SortedVector < sp<OutputTrack> > outputTracks; 1285 SortedVector < sp<OutputTrack> > mOutputTracks; 1286 public: 1287 virtual bool hasFastMixer() const { return false; } 1288 }; 1289 1290 PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const; 1291 MixerThread *checkMixerThread_l(audio_io_handle_t output) const; 1292 RecordThread *checkRecordThread_l(audio_io_handle_t input) const; 1293 // no range check, AudioFlinger::mLock held 1294 bool streamMute_l(audio_stream_type_t stream) const 1295 { return mStreamTypes[stream].mute; } 1296 // no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held 1297 float streamVolume_l(audio_stream_type_t stream) const 1298 { return mStreamTypes[stream].volume; } 1299 void audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2); 1300 1301 // allocate an audio_io_handle_t, session ID, or effect ID 1302 uint32_t nextUniqueId(); 1303 1304 status_t moveEffectChain_l(int sessionId, 1305 PlaybackThread *srcThread, 1306 PlaybackThread *dstThread, 1307 bool reRegister); 1308 // return thread associated with primary hardware device, or NULL 1309 PlaybackThread *primaryPlaybackThread_l() const; 1310 uint32_t primaryOutputDevice_l() const; 1311 1312 sp<PlaybackThread> getEffectThread_l(int sessionId, int EffectId); 1313 1314 // server side of the client's IAudioTrack 1315 class TrackHandle : public android::BnAudioTrack { 1316 public: 1317 TrackHandle(const sp<PlaybackThread::Track>& track); 1318 virtual ~TrackHandle(); 1319 virtual sp<IMemory> getCblk() const; 1320 virtual status_t start(); 1321 virtual void stop(); 1322 virtual void flush(); 1323 virtual void mute(bool); 1324 virtual void pause(); 1325 virtual status_t attachAuxEffect(int effectId); 1326 virtual status_t allocateTimedBuffer(size_t size, 1327 sp<IMemory>* buffer); 1328 virtual status_t queueTimedBuffer(const sp<IMemory>& buffer, 1329 int64_t pts); 1330 virtual status_t setMediaTimeTransform(const LinearTransform& xform, 1331 int target); 1332 virtual status_t onTransact( 1333 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 1334 private: 1335 const sp<PlaybackThread::Track> mTrack; 1336 }; 1337 1338 void removeClient_l(pid_t pid); 1339 void removeNotificationClient(pid_t pid); 1340 1341 1342 // record thread 1343 class RecordThread : public ThreadBase, public AudioBufferProvider 1344 { 1345 public: 1346 1347 // record track 1348 class RecordTrack : public TrackBase { 1349 public: 1350 RecordTrack(RecordThread *thread, 1351 const sp<Client>& client, 1352 uint32_t sampleRate, 1353 audio_format_t format, 1354 uint32_t channelMask, 1355 int frameCount, 1356 int sessionId); 1357 virtual ~RecordTrack(); 1358 1359 virtual status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 1360 int triggerSession = 0); 1361 virtual void stop(); 1362 1363 bool overflow() { bool tmp = mOverflow; mOverflow = false; return tmp; } 1364 bool setOverflow() { bool tmp = mOverflow; mOverflow = true; return tmp; } 1365 1366 void dump(char* buffer, size_t size); 1367 1368 private: 1369 friend class AudioFlinger; // for mState 1370 1371 RecordTrack(const RecordTrack&); 1372 RecordTrack& operator = (const RecordTrack&); 1373 1374 // AudioBufferProvider interface 1375 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts = kInvalidPTS); 1376 // releaseBuffer() not overridden 1377 1378 bool mOverflow; 1379 }; 1380 1381 1382 RecordThread(const sp<AudioFlinger>& audioFlinger, 1383 AudioStreamIn *input, 1384 uint32_t sampleRate, 1385 uint32_t channels, 1386 audio_io_handle_t id, 1387 uint32_t device); 1388 virtual ~RecordThread(); 1389 1390 // Thread 1391 virtual bool threadLoop(); 1392 virtual status_t readyToRun(); 1393 1394 // RefBase 1395 virtual void onFirstRef(); 1396 1397 virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; } 1398 sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l( 1399 const sp<AudioFlinger::Client>& client, 1400 uint32_t sampleRate, 1401 audio_format_t format, 1402 int channelMask, 1403 int frameCount, 1404 int sessionId, 1405 status_t *status); 1406 1407 status_t start(RecordTrack* recordTrack, 1408 AudioSystem::sync_event_t event, 1409 int triggerSession); 1410 void stop(RecordTrack* recordTrack); 1411 status_t dump(int fd, const Vector<String16>& args); 1412 AudioStreamIn* getInput() const; 1413 AudioStreamIn* clearInput(); 1414 virtual audio_stream_t* stream() const; 1415 1416 // AudioBufferProvider interface 1417 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts); 1418 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 1419 1420 virtual bool checkForNewParameters_l(); 1421 virtual String8 getParameters(const String8& keys); 1422 virtual void audioConfigChanged_l(int event, int param = 0); 1423 void readInputParameters(); 1424 virtual unsigned int getInputFramesLost(); 1425 1426 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1427 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1428 virtual uint32_t hasAudioSession(int sessionId); 1429 RecordTrack* track(); 1430 1431 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 1432 virtual bool isValidSyncEvent(const sp<SyncEvent>& event); 1433 1434 static void syncStartEventCallback(const wp<SyncEvent>& event); 1435 void handleSyncStartEvent(const sp<SyncEvent>& event); 1436 1437 private: 1438 void clearSyncStartEvent(); 1439 1440 RecordThread(); 1441 AudioStreamIn *mInput; 1442 RecordTrack* mTrack; 1443 sp<RecordTrack> mActiveTrack; 1444 Condition mStartStopCond; 1445 AudioResampler *mResampler; 1446 int32_t *mRsmpOutBuffer; 1447 int16_t *mRsmpInBuffer; 1448 size_t mRsmpInIndex; 1449 size_t mInputBytes; 1450 const int mReqChannelCount; 1451 const uint32_t mReqSampleRate; 1452 ssize_t mBytesRead; 1453 // sync event triggering actual audio capture. Frames read before this event will 1454 // be dropped and therefore not read by the application. 1455 sp<SyncEvent> mSyncStartEvent; 1456 // number of captured frames to drop after the start sync event has been received. 1457 // when < 0, maximum frames to drop before starting capture even if sync event is 1458 // not received 1459 ssize_t mFramestoDrop; 1460 }; 1461 1462 // server side of the client's IAudioRecord 1463 class RecordHandle : public android::BnAudioRecord { 1464 public: 1465 RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack); 1466 virtual ~RecordHandle(); 1467 virtual sp<IMemory> getCblk() const; 1468 virtual status_t start(int event, int triggerSession); 1469 virtual void stop(); 1470 virtual status_t onTransact( 1471 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 1472 private: 1473 const sp<RecordThread::RecordTrack> mRecordTrack; 1474 }; 1475 1476 //--- Audio Effect Management 1477 1478 // EffectModule and EffectChain classes both have their own mutex to protect 1479 // state changes or resource modifications. Always respect the following order 1480 // if multiple mutexes must be acquired to avoid cross deadlock: 1481 // AudioFlinger -> ThreadBase -> EffectChain -> EffectModule 1482 1483 // The EffectModule class is a wrapper object controlling the effect engine implementation 1484 // in the effect library. It prevents concurrent calls to process() and command() functions 1485 // from different client threads. It keeps a list of EffectHandle objects corresponding 1486 // to all client applications using this effect and notifies applications of effect state, 1487 // control or parameter changes. It manages the activation state machine to send appropriate 1488 // reset, enable, disable commands to effect engine and provide volume 1489 // ramping when effects are activated/deactivated. 1490 // When controlling an auxiliary effect, the EffectModule also provides an input buffer used by 1491 // the attached track(s) to accumulate their auxiliary channel. 1492 class EffectModule: public RefBase { 1493 public: 1494 EffectModule(ThreadBase *thread, 1495 const wp<AudioFlinger::EffectChain>& chain, 1496 effect_descriptor_t *desc, 1497 int id, 1498 int sessionId); 1499 virtual ~EffectModule(); 1500 1501 enum effect_state { 1502 IDLE, 1503 RESTART, 1504 STARTING, 1505 ACTIVE, 1506 STOPPING, 1507 STOPPED, 1508 DESTROYED 1509 }; 1510 1511 int id() const { return mId; } 1512 void process(); 1513 void updateState(); 1514 status_t command(uint32_t cmdCode, 1515 uint32_t cmdSize, 1516 void *pCmdData, 1517 uint32_t *replySize, 1518 void *pReplyData); 1519 1520 void reset_l(); 1521 status_t configure(); 1522 status_t init(); 1523 effect_state state() const { 1524 return mState; 1525 } 1526 uint32_t status() { 1527 return mStatus; 1528 } 1529 int sessionId() const { 1530 return mSessionId; 1531 } 1532 status_t setEnabled(bool enabled); 1533 status_t setEnabled_l(bool enabled); 1534 bool isEnabled() const; 1535 bool isProcessEnabled() const; 1536 1537 void setInBuffer(int16_t *buffer) { mConfig.inputCfg.buffer.s16 = buffer; } 1538 int16_t *inBuffer() { return mConfig.inputCfg.buffer.s16; } 1539 void setOutBuffer(int16_t *buffer) { mConfig.outputCfg.buffer.s16 = buffer; } 1540 int16_t *outBuffer() { return mConfig.outputCfg.buffer.s16; } 1541 void setChain(const wp<EffectChain>& chain) { mChain = chain; } 1542 void setThread(const wp<ThreadBase>& thread) { mThread = thread; } 1543 const wp<ThreadBase>& thread() { return mThread; } 1544 1545 status_t addHandle(EffectHandle *handle); 1546 size_t disconnect(EffectHandle *handle, bool unpinIfLast); 1547 size_t removeHandle(EffectHandle *handle); 1548 1549 effect_descriptor_t& desc() { return mDescriptor; } 1550 wp<EffectChain>& chain() { return mChain; } 1551 1552 status_t setDevice(uint32_t device); 1553 status_t setVolume(uint32_t *left, uint32_t *right, bool controller); 1554 status_t setMode(audio_mode_t mode); 1555 status_t start(); 1556 status_t stop(); 1557 void setSuspended(bool suspended); 1558 bool suspended() const; 1559 1560 EffectHandle* controlHandle_l(); 1561 1562 bool isPinned() const { return mPinned; } 1563 void unPin() { mPinned = false; } 1564 bool purgeHandles(); 1565 void lock() { mLock.lock(); } 1566 void unlock() { mLock.unlock(); } 1567 1568 status_t dump(int fd, const Vector<String16>& args); 1569 1570 protected: 1571 friend class AudioFlinger; // for mHandles 1572 bool mPinned; 1573 1574 // Maximum time allocated to effect engines to complete the turn off sequence 1575 static const uint32_t MAX_DISABLE_TIME_MS = 10000; 1576 1577 EffectModule(const EffectModule&); 1578 EffectModule& operator = (const EffectModule&); 1579 1580 status_t start_l(); 1581 status_t stop_l(); 1582 1583mutable Mutex mLock; // mutex for process, commands and handles list protection 1584 wp<ThreadBase> mThread; // parent thread 1585 wp<EffectChain> mChain; // parent effect chain 1586 const int mId; // this instance unique ID 1587 const int mSessionId; // audio session ID 1588 effect_descriptor_t mDescriptor;// effect descriptor received from effect engine 1589 effect_config_t mConfig; // input and output audio configuration 1590 effect_handle_t mEffectInterface; // Effect module C API 1591 status_t mStatus; // initialization status 1592 effect_state mState; // current activation state 1593 Vector<EffectHandle *> mHandles; // list of client handles 1594 // First handle in mHandles has highest priority and controls the effect module 1595 uint32_t mMaxDisableWaitCnt; // maximum grace period before forcing an effect off after 1596 // sending disable command. 1597 uint32_t mDisableWaitCnt; // current process() calls count during disable period. 1598 bool mSuspended; // effect is suspended: temporarily disabled by framework 1599 }; 1600 1601 // The EffectHandle class implements the IEffect interface. It provides resources 1602 // to receive parameter updates, keeps track of effect control 1603 // ownership and state and has a pointer to the EffectModule object it is controlling. 1604 // There is one EffectHandle object for each application controlling (or using) 1605 // an effect module. 1606 // The EffectHandle is obtained by calling AudioFlinger::createEffect(). 1607 class EffectHandle: public android::BnEffect { 1608 public: 1609 1610 EffectHandle(const sp<EffectModule>& effect, 1611 const sp<AudioFlinger::Client>& client, 1612 const sp<IEffectClient>& effectClient, 1613 int32_t priority); 1614 virtual ~EffectHandle(); 1615 1616 // IEffect 1617 virtual status_t enable(); 1618 virtual status_t disable(); 1619 virtual status_t command(uint32_t cmdCode, 1620 uint32_t cmdSize, 1621 void *pCmdData, 1622 uint32_t *replySize, 1623 void *pReplyData); 1624 virtual void disconnect(); 1625 private: 1626 void disconnect(bool unpinIfLast); 1627 public: 1628 virtual sp<IMemory> getCblk() const { return mCblkMemory; } 1629 virtual status_t onTransact(uint32_t code, const Parcel& data, 1630 Parcel* reply, uint32_t flags); 1631 1632 1633 // Give or take control of effect module 1634 // - hasControl: true if control is given, false if removed 1635 // - signal: true client app should be signaled of change, false otherwise 1636 // - enabled: state of the effect when control is passed 1637 void setControl(bool hasControl, bool signal, bool enabled); 1638 void commandExecuted(uint32_t cmdCode, 1639 uint32_t cmdSize, 1640 void *pCmdData, 1641 uint32_t replySize, 1642 void *pReplyData); 1643 void setEnabled(bool enabled); 1644 bool enabled() const { return mEnabled; } 1645 1646 // Getters 1647 int id() const { return mEffect->id(); } 1648 int priority() const { return mPriority; } 1649 bool hasControl() const { return mHasControl; } 1650 sp<EffectModule> effect() const { return mEffect; } 1651 // destroyed_l() must be called with the associated EffectModule mLock held 1652 bool destroyed_l() const { return mDestroyed; } 1653 1654 void dump(char* buffer, size_t size); 1655 1656 protected: 1657 friend class AudioFlinger; // for mEffect, mHasControl, mEnabled 1658 EffectHandle(const EffectHandle&); 1659 EffectHandle& operator =(const EffectHandle&); 1660 1661 sp<EffectModule> mEffect; // pointer to controlled EffectModule 1662 sp<IEffectClient> mEffectClient; // callback interface for client notifications 1663 /*const*/ sp<Client> mClient; // client for shared memory allocation, see disconnect() 1664 sp<IMemory> mCblkMemory; // shared memory for control block 1665 effect_param_cblk_t* mCblk; // control block for deferred parameter setting via shared memory 1666 uint8_t* mBuffer; // pointer to parameter area in shared memory 1667 int mPriority; // client application priority to control the effect 1668 bool mHasControl; // true if this handle is controlling the effect 1669 bool mEnabled; // cached enable state: needed when the effect is 1670 // restored after being suspended 1671 bool mDestroyed; // Set to true by destructor. Access with EffectModule 1672 // mLock held 1673 }; 1674 1675 // the EffectChain class represents a group of effects associated to one audio session. 1676 // There can be any number of EffectChain objects per output mixer thread (PlaybackThread). 1677 // The EffecChain with session ID 0 contains global effects applied to the output mix. 1678 // Effects in this chain can be insert or auxiliary. Effects in other chains (attached to tracks) 1679 // are insert only. The EffectChain maintains an ordered list of effect module, the order corresponding 1680 // in the effect process order. When attached to a track (session ID != 0), it also provide it's own 1681 // input buffer used by the track as accumulation buffer. 1682 class EffectChain: public RefBase { 1683 public: 1684 EffectChain(const wp<ThreadBase>& wThread, int sessionId); 1685 EffectChain(ThreadBase *thread, int sessionId); 1686 virtual ~EffectChain(); 1687 1688 // special key used for an entry in mSuspendedEffects keyed vector 1689 // corresponding to a suspend all request. 1690 static const int kKeyForSuspendAll = 0; 1691 1692 // minimum duration during which we force calling effect process when last track on 1693 // a session is stopped or removed to allow effect tail to be rendered 1694 static const int kProcessTailDurationMs = 1000; 1695 1696 void process_l(); 1697 1698 void lock() { 1699 mLock.lock(); 1700 } 1701 void unlock() { 1702 mLock.unlock(); 1703 } 1704 1705 status_t addEffect_l(const sp<EffectModule>& handle); 1706 size_t removeEffect_l(const sp<EffectModule>& handle); 1707 1708 int sessionId() const { return mSessionId; } 1709 void setSessionId(int sessionId) { mSessionId = sessionId; } 1710 1711 sp<EffectModule> getEffectFromDesc_l(effect_descriptor_t *descriptor); 1712 sp<EffectModule> getEffectFromId_l(int id); 1713 sp<EffectModule> getEffectFromType_l(const effect_uuid_t *type); 1714 bool setVolume_l(uint32_t *left, uint32_t *right); 1715 void setDevice_l(uint32_t device); 1716 void setMode_l(audio_mode_t mode); 1717 1718 void setInBuffer(int16_t *buffer, bool ownsBuffer = false) { 1719 mInBuffer = buffer; 1720 mOwnInBuffer = ownsBuffer; 1721 } 1722 int16_t *inBuffer() const { 1723 return mInBuffer; 1724 } 1725 void setOutBuffer(int16_t *buffer) { 1726 mOutBuffer = buffer; 1727 } 1728 int16_t *outBuffer() const { 1729 return mOutBuffer; 1730 } 1731 1732 void incTrackCnt() { android_atomic_inc(&mTrackCnt); } 1733 void decTrackCnt() { android_atomic_dec(&mTrackCnt); } 1734 int32_t trackCnt() const { return mTrackCnt;} 1735 1736 void incActiveTrackCnt() { android_atomic_inc(&mActiveTrackCnt); 1737 mTailBufferCount = mMaxTailBuffers; } 1738 void decActiveTrackCnt() { android_atomic_dec(&mActiveTrackCnt); } 1739 int32_t activeTrackCnt() const { return mActiveTrackCnt;} 1740 1741 uint32_t strategy() const { return mStrategy; } 1742 void setStrategy(uint32_t strategy) 1743 { mStrategy = strategy; } 1744 1745 // suspend effect of the given type 1746 void setEffectSuspended_l(const effect_uuid_t *type, 1747 bool suspend); 1748 // suspend all eligible effects 1749 void setEffectSuspendedAll_l(bool suspend); 1750 // check if effects should be suspend or restored when a given effect is enable or disabled 1751 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1752 bool enabled); 1753 1754 void clearInputBuffer(); 1755 1756 status_t dump(int fd, const Vector<String16>& args); 1757 1758 protected: 1759 friend class AudioFlinger; // for mThread, mEffects 1760 EffectChain(const EffectChain&); 1761 EffectChain& operator =(const EffectChain&); 1762 1763 class SuspendedEffectDesc : public RefBase { 1764 public: 1765 SuspendedEffectDesc() : mRefCount(0) {} 1766 1767 int mRefCount; 1768 effect_uuid_t mType; 1769 wp<EffectModule> mEffect; 1770 }; 1771 1772 // get a list of effect modules to suspend when an effect of the type 1773 // passed is enabled. 1774 void getSuspendEligibleEffects(Vector< sp<EffectModule> > &effects); 1775 1776 // get an effect module if it is currently enable 1777 sp<EffectModule> getEffectIfEnabled(const effect_uuid_t *type); 1778 // true if the effect whose descriptor is passed can be suspended 1779 // OEMs can modify the rules implemented in this method to exclude specific effect 1780 // types or implementations from the suspend/restore mechanism. 1781 bool isEffectEligibleForSuspend(const effect_descriptor_t& desc); 1782 1783 void clearInputBuffer_l(sp<ThreadBase> thread); 1784 1785 wp<ThreadBase> mThread; // parent mixer thread 1786 Mutex mLock; // mutex protecting effect list 1787 Vector< sp<EffectModule> > mEffects; // list of effect modules 1788 int mSessionId; // audio session ID 1789 int16_t *mInBuffer; // chain input buffer 1790 int16_t *mOutBuffer; // chain output buffer 1791 volatile int32_t mActiveTrackCnt; // number of active tracks connected 1792 volatile int32_t mTrackCnt; // number of tracks connected 1793 int32_t mTailBufferCount; // current effect tail buffer count 1794 int32_t mMaxTailBuffers; // maximum effect tail buffers 1795 bool mOwnInBuffer; // true if the chain owns its input buffer 1796 int mVolumeCtrlIdx; // index of insert effect having control over volume 1797 uint32_t mLeftVolume; // previous volume on left channel 1798 uint32_t mRightVolume; // previous volume on right channel 1799 uint32_t mNewLeftVolume; // new volume on left channel 1800 uint32_t mNewRightVolume; // new volume on right channel 1801 uint32_t mStrategy; // strategy for this effect chain 1802 // mSuspendedEffects lists all effects currently suspended in the chain. 1803 // Use effect type UUID timelow field as key. There is no real risk of identical 1804 // timeLow fields among effect type UUIDs. 1805 // Updated by updateSuspendedSessions_l() only. 1806 KeyedVector< int, sp<SuspendedEffectDesc> > mSuspendedEffects; 1807 }; 1808 1809 // AudioStreamOut and AudioStreamIn are immutable, so their fields are const. 1810 // For emphasis, we could also make all pointers to them be "const *", 1811 // but that would clutter the code unnecessarily. 1812 1813 struct AudioStreamOut { 1814 audio_hw_device_t* const hwDev; 1815 audio_stream_out_t* const stream; 1816 1817 AudioStreamOut(audio_hw_device_t *dev, audio_stream_out_t *out) : 1818 hwDev(dev), stream(out) {} 1819 }; 1820 1821 struct AudioStreamIn { 1822 audio_hw_device_t* const hwDev; 1823 audio_stream_in_t* const stream; 1824 1825 AudioStreamIn(audio_hw_device_t *dev, audio_stream_in_t *in) : 1826 hwDev(dev), stream(in) {} 1827 }; 1828 1829 // for mAudioSessionRefs only 1830 struct AudioSessionRef { 1831 AudioSessionRef(int sessionid, pid_t pid) : 1832 mSessionid(sessionid), mPid(pid), mCnt(1) {} 1833 const int mSessionid; 1834 const pid_t mPid; 1835 int mCnt; 1836 }; 1837 1838 enum master_volume_support { 1839 // MVS_NONE: 1840 // Audio HAL has no support for master volume, either setting or 1841 // getting. All master volume control must be implemented in SW by the 1842 // AudioFlinger mixing core. 1843 MVS_NONE, 1844 1845 // MVS_SETONLY: 1846 // Audio HAL has support for setting master volume, but not for getting 1847 // master volume (original HAL design did not include a getter). 1848 // AudioFlinger needs to keep track of the last set master volume in 1849 // addition to needing to set an initial, default, master volume at HAL 1850 // load time. 1851 MVS_SETONLY, 1852 1853 // MVS_FULL: 1854 // Audio HAL has support both for setting and getting master volume. 1855 // AudioFlinger should send all set and get master volume requests 1856 // directly to the HAL. 1857 MVS_FULL, 1858 }; 1859 1860 class AudioHwDevice { 1861 public: 1862 AudioHwDevice(const char *moduleName, audio_hw_device_t *hwDevice) : 1863 mModuleName(strdup(moduleName)), mHwDevice(hwDevice){} 1864 ~AudioHwDevice() { free((void *)mModuleName); } 1865 1866 const char *moduleName() const { return mModuleName; } 1867 audio_hw_device_t *hwDevice() const { return mHwDevice; } 1868 private: 1869 const char * const mModuleName; 1870 audio_hw_device_t * const mHwDevice; 1871 }; 1872 1873 mutable Mutex mLock; 1874 1875 DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client() 1876 1877 mutable Mutex mHardwareLock; 1878 // NOTE: If both mLock and mHardwareLock mutexes must be held, 1879 // always take mLock before mHardwareLock 1880 1881 // These two fields are immutable after onFirstRef(), so no lock needed to access 1882 audio_hw_device_t* mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL 1883 DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*> mAudioHwDevs; 1884 1885 // for dump, indicates which hardware operation is currently in progress (but not stream ops) 1886 enum hardware_call_state { 1887 AUDIO_HW_IDLE = 0, // no operation in progress 1888 AUDIO_HW_INIT, // init_check 1889 AUDIO_HW_OUTPUT_OPEN, // open_output_stream 1890 AUDIO_HW_OUTPUT_CLOSE, // unused 1891 AUDIO_HW_INPUT_OPEN, // unused 1892 AUDIO_HW_INPUT_CLOSE, // unused 1893 AUDIO_HW_STANDBY, // unused 1894 AUDIO_HW_SET_MASTER_VOLUME, // set_master_volume 1895 AUDIO_HW_GET_ROUTING, // unused 1896 AUDIO_HW_SET_ROUTING, // unused 1897 AUDIO_HW_GET_MODE, // unused 1898 AUDIO_HW_SET_MODE, // set_mode 1899 AUDIO_HW_GET_MIC_MUTE, // get_mic_mute 1900 AUDIO_HW_SET_MIC_MUTE, // set_mic_mute 1901 AUDIO_HW_SET_VOICE_VOLUME, // set_voice_volume 1902 AUDIO_HW_SET_PARAMETER, // set_parameters 1903 AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size 1904 AUDIO_HW_GET_MASTER_VOLUME, // get_master_volume 1905 AUDIO_HW_GET_PARAMETER, // get_parameters 1906 }; 1907 1908 mutable hardware_call_state mHardwareStatus; // for dump only 1909 1910 1911 DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads; 1912 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 1913 1914 // both are protected by mLock 1915 float mMasterVolume; 1916 float mMasterVolumeSW; 1917 master_volume_support mMasterVolumeSupportLvl; 1918 bool mMasterMute; 1919 1920 DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads; 1921 1922 DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients; 1923 volatile int32_t mNextUniqueId; // updated by android_atomic_inc 1924 audio_mode_t mMode; 1925 bool mBtNrecIsOff; 1926 1927 // protected by mLock 1928 Vector<AudioSessionRef*> mAudioSessionRefs; 1929 1930 float masterVolume_l() const; 1931 float masterVolumeSW_l() const { return mMasterVolumeSW; } 1932 bool masterMute_l() const { return mMasterMute; } 1933 audio_module_handle_t loadHwModule_l(const char *name); 1934 1935 Vector < sp<SyncEvent> > mPendingSyncEvents; // sync events awaiting for a session 1936 // to be created 1937 1938private: 1939 sp<Client> registerPid_l(pid_t pid); // always returns non-0 1940 1941}; 1942 1943 1944// ---------------------------------------------------------------------------- 1945 1946}; // namespace android 1947 1948#endif // ANDROID_AUDIO_FLINGER_H 1949