AudioFlinger.h revision 04270daf50f0c602d7c57a257a693e68246cbeb7
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 audio_channel_mask_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 audio_channel_mask_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 audio_channel_mask_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_channel_mask_t channelMask() const { return mChannelMask; } 503 audio_format_t format() const { return mFormat; } 504 // Called by AudioFlinger::frameCount(audio_io_handle_t output) and effects, 505 // and returns the normal mix buffer's frame count. No API for HAL frame count. 506 size_t frameCount() const { return mNormalFrameCount; } 507 508 void wakeUp() { mWaitWorkCV.broadcast(); } 509 // Should be "virtual status_t requestExitAndWait()" and override same 510 // method in Thread, but Thread::requestExitAndWait() is not yet virtual. 511 void exit(); 512 virtual bool checkForNewParameters_l() = 0; 513 virtual status_t setParameters(const String8& keyValuePairs); 514 virtual String8 getParameters(const String8& keys) = 0; 515 virtual void audioConfigChanged_l(int event, int param = 0) = 0; 516 void sendConfigEvent(int event, int param = 0); 517 void sendConfigEvent_l(int event, int param = 0); 518 void processConfigEvents(); 519 520 // see note at declaration of mStandby and mDevice 521 bool standby() const { return mStandby; } 522 audio_devices_t device() const { return mDevice; } 523 524 virtual audio_stream_t* stream() const = 0; 525 526 sp<EffectHandle> createEffect_l( 527 const sp<AudioFlinger::Client>& client, 528 const sp<IEffectClient>& effectClient, 529 int32_t priority, 530 int sessionId, 531 effect_descriptor_t *desc, 532 int *enabled, 533 status_t *status); 534 void disconnectEffect(const sp< EffectModule>& effect, 535 EffectHandle *handle, 536 bool unpinIfLast); 537 538 // return values for hasAudioSession (bit field) 539 enum effect_state { 540 EFFECT_SESSION = 0x1, // the audio session corresponds to at least one 541 // effect 542 TRACK_SESSION = 0x2 // the audio session corresponds to at least one 543 // track 544 }; 545 546 // get effect chain corresponding to session Id. 547 sp<EffectChain> getEffectChain(int sessionId); 548 // same as getEffectChain() but must be called with ThreadBase mutex locked 549 sp<EffectChain> getEffectChain_l(int sessionId); 550 // add an effect chain to the chain list (mEffectChains) 551 virtual status_t addEffectChain_l(const sp<EffectChain>& chain) = 0; 552 // remove an effect chain from the chain list (mEffectChains) 553 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain) = 0; 554 // lock all effect chains Mutexes. Must be called before releasing the 555 // ThreadBase mutex before processing the mixer and effects. This guarantees the 556 // integrity of the chains during the process. 557 // Also sets the parameter 'effectChains' to current value of mEffectChains. 558 void lockEffectChains_l(Vector< sp<EffectChain> >& effectChains); 559 // unlock effect chains after process 560 void unlockEffectChains(const Vector< sp<EffectChain> >& effectChains); 561 // set audio mode to all effect chains 562 void setMode(audio_mode_t mode); 563 // get effect module with corresponding ID on specified audio session 564 sp<AudioFlinger::EffectModule> getEffect(int sessionId, int effectId); 565 sp<AudioFlinger::EffectModule> getEffect_l(int sessionId, int effectId); 566 // add and effect module. Also creates the effect chain is none exists for 567 // the effects audio session 568 status_t addEffect_l(const sp< EffectModule>& effect); 569 // remove and effect module. Also removes the effect chain is this was the last 570 // effect 571 void removeEffect_l(const sp< EffectModule>& effect); 572 // detach all tracks connected to an auxiliary effect 573 virtual void detachAuxEffect_l(int effectId) {} 574 // returns either EFFECT_SESSION if effects on this audio session exist in one 575 // chain, or TRACK_SESSION if tracks on this audio session exist, or both 576 virtual uint32_t hasAudioSession(int sessionId) = 0; 577 // the value returned by default implementation is not important as the 578 // strategy is only meaningful for PlaybackThread which implements this method 579 virtual uint32_t getStrategyForSession_l(int sessionId) { return 0; } 580 581 // suspend or restore effect according to the type of effect passed. a NULL 582 // type pointer means suspend all effects in the session 583 void setEffectSuspended(const effect_uuid_t *type, 584 bool suspend, 585 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 586 // check if some effects must be suspended/restored when an effect is enabled 587 // or disabled 588 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 589 bool enabled, 590 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 591 void checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 592 bool enabled, 593 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 594 595 virtual status_t setSyncEvent(const sp<SyncEvent>& event) = 0; 596 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) = 0; 597 598 599 mutable Mutex mLock; 600 601 protected: 602 603 // entry describing an effect being suspended in mSuspendedSessions keyed vector 604 class SuspendedSessionDesc : public RefBase { 605 public: 606 SuspendedSessionDesc() : mRefCount(0) {} 607 608 int mRefCount; // number of active suspend requests 609 effect_uuid_t mType; // effect type UUID 610 }; 611 612 void acquireWakeLock(); 613 void acquireWakeLock_l(); 614 void releaseWakeLock(); 615 void releaseWakeLock_l(); 616 void setEffectSuspended_l(const effect_uuid_t *type, 617 bool suspend, 618 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 619 // updated mSuspendedSessions when an effect suspended or restored 620 void updateSuspendedSessions_l(const effect_uuid_t *type, 621 bool suspend, 622 int sessionId); 623 // check if some effects must be suspended when an effect chain is added 624 void checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain); 625 626 friend class AudioFlinger; // for mEffectChains 627 628 const type_t mType; 629 630 // Used by parameters, config events, addTrack_l, exit 631 Condition mWaitWorkCV; 632 633 const sp<AudioFlinger> mAudioFlinger; 634 uint32_t mSampleRate; 635 size_t mFrameCount; // output HAL, direct output, record 636 size_t mNormalFrameCount; // normal mixer and effects 637 audio_channel_mask_t mChannelMask; 638 uint16_t mChannelCount; 639 size_t mFrameSize; 640 audio_format_t mFormat; 641 642 // Parameter sequence by client: binder thread calling setParameters(): 643 // 1. Lock mLock 644 // 2. Append to mNewParameters 645 // 3. mWaitWorkCV.signal 646 // 4. mParamCond.waitRelative with timeout 647 // 5. read mParamStatus 648 // 6. mWaitWorkCV.signal 649 // 7. Unlock 650 // 651 // Parameter sequence by server: threadLoop calling checkForNewParameters_l(): 652 // 1. Lock mLock 653 // 2. If there is an entry in mNewParameters proceed ... 654 // 2. Read first entry in mNewParameters 655 // 3. Process 656 // 4. Remove first entry from mNewParameters 657 // 5. Set mParamStatus 658 // 6. mParamCond.signal 659 // 7. mWaitWorkCV.wait with timeout (this is to avoid overwriting mParamStatus) 660 // 8. Unlock 661 Condition mParamCond; 662 Vector<String8> mNewParameters; 663 status_t mParamStatus; 664 665 Vector<ConfigEvent> mConfigEvents; 666 667 // These fields are written and read by thread itself without lock or barrier, 668 // and read by other threads without lock or barrier via standby() and device(). 669 // Because of the absence of a lock or barrier, any other thread that reads 670 // these fields must use the information in isolation, or be prepared to deal 671 // with possibility that it might be inconsistent with other information. 672 bool mStandby; // Whether thread is currently in standby. 673 audio_devices_t mDevice; // output device for PlaybackThread 674 // input + output devices for RecordThread 675 676 const audio_io_handle_t mId; 677 Vector< sp<EffectChain> > mEffectChains; 678 679 static const int kNameLength = 16; // prctl(PR_SET_NAME) limit 680 char mName[kNameLength]; 681 sp<IPowerManager> mPowerManager; 682 sp<IBinder> mWakeLockToken; 683 const sp<PMDeathRecipient> mDeathRecipient; 684 // list of suspended effects per session and per type. The first vector is 685 // keyed by session ID, the second by type UUID timeLow field 686 KeyedVector< int, KeyedVector< int, sp<SuspendedSessionDesc> > > mSuspendedSessions; 687 }; 688 689 struct stream_type_t { 690 stream_type_t() 691 : volume(1.0f), 692 mute(false) 693 { 694 } 695 float volume; 696 bool mute; 697 }; 698 699 // --- PlaybackThread --- 700 class PlaybackThread : public ThreadBase { 701 public: 702 703 enum mixer_state { 704 MIXER_IDLE, // no active tracks 705 MIXER_TRACKS_ENABLED, // at least one active track, but no track has any data ready 706 MIXER_TRACKS_READY // at least one active track, and at least one track has data 707 // standby mode does not have an enum value 708 // suspend by audio policy manager is orthogonal to mixer state 709 }; 710 711 // playback track 712 class Track : public TrackBase, public VolumeProvider { 713 public: 714 Track( PlaybackThread *thread, 715 const sp<Client>& client, 716 audio_stream_type_t streamType, 717 uint32_t sampleRate, 718 audio_format_t format, 719 audio_channel_mask_t channelMask, 720 int frameCount, 721 const sp<IMemory>& sharedBuffer, 722 int sessionId, 723 IAudioFlinger::track_flags_t flags); 724 virtual ~Track(); 725 726 static void appendDumpHeader(String8& result); 727 void dump(char* buffer, size_t size); 728 virtual status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 729 int triggerSession = 0); 730 virtual void stop(); 731 void pause(); 732 733 void flush(); 734 void destroy(); 735 void mute(bool); 736 int name() const { return mName; } 737 738 audio_stream_type_t streamType() const { 739 return mStreamType; 740 } 741 status_t attachAuxEffect(int EffectId); 742 void setAuxBuffer(int EffectId, int32_t *buffer); 743 int32_t *auxBuffer() const { return mAuxBuffer; } 744 void setMainBuffer(int16_t *buffer) { mMainBuffer = buffer; } 745 int16_t *mainBuffer() const { return mMainBuffer; } 746 int auxEffectId() const { return mAuxEffectId; } 747 748 // implement FastMixerState::VolumeProvider interface 749 virtual uint32_t getVolumeLR(); 750 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 751 752 protected: 753 // for numerous 754 friend class PlaybackThread; 755 friend class MixerThread; 756 friend class DirectOutputThread; 757 758 Track(const Track&); 759 Track& operator = (const Track&); 760 761 // AudioBufferProvider interface 762 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts = kInvalidPTS); 763 // releaseBuffer() not overridden 764 765 virtual size_t framesReady() const; 766 767 bool isMuted() const { return mMute; } 768 bool isPausing() const { 769 return mState == PAUSING; 770 } 771 bool isPaused() const { 772 return mState == PAUSED; 773 } 774 bool isResuming() const { 775 return mState == RESUMING; 776 } 777 bool isReady() const; 778 void setPaused() { mState = PAUSED; } 779 void reset(); 780 781 bool isOutputTrack() const { 782 return (mStreamType == AUDIO_STREAM_CNT); 783 } 784 785 sp<IMemory> sharedBuffer() const { return mSharedBuffer; } 786 787 bool presentationComplete(size_t framesWritten, size_t audioHalFrames); 788 789 public: 790 void triggerEvents(AudioSystem::sync_event_t type); 791 virtual bool isTimedTrack() const { return false; } 792 bool isFastTrack() const { return (mFlags & IAudioFlinger::TRACK_FAST) != 0; } 793 794 protected: 795 796 // written by Track::mute() called by binder thread(s), without a mutex or barrier. 797 // read by Track::isMuted() called by playback thread, also without a mutex or barrier. 798 // The lack of mutex or barrier is safe because the mute status is only used by itself. 799 bool mMute; 800 801 // FILLED state is used for suppressing volume ramp at begin of playing 802 enum {FS_INVALID, FS_FILLING, FS_FILLED, FS_ACTIVE}; 803 mutable uint8_t mFillingUpStatus; 804 int8_t mRetryCount; 805 const sp<IMemory> mSharedBuffer; 806 bool mResetDone; 807 const audio_stream_type_t mStreamType; 808 int mName; // track name on the normal mixer, 809 // allocated statically at track creation time, 810 // and is even allocated (though unused) for fast tracks 811 // FIXME don't allocate track name for fast tracks 812 int16_t *mMainBuffer; 813 int32_t *mAuxBuffer; 814 int mAuxEffectId; 815 bool mHasVolumeController; 816 size_t mPresentationCompleteFrames; // number of frames written to the audio HAL 817 // when this track will be fully rendered 818 private: 819 IAudioFlinger::track_flags_t mFlags; 820 821 // The following fields are only for fast tracks, and should be in a subclass 822 int mFastIndex; // index within FastMixerState::mFastTracks[]; 823 // either mFastIndex == -1 if not isFastTrack() 824 // or 0 < mFastIndex < FastMixerState::kMaxFast because 825 // index 0 is reserved for normal mixer's submix; 826 // index is allocated statically at track creation time 827 // but the slot is only used if track is active 828 FastTrackUnderruns mObservedUnderruns; // Most recently observed value of 829 // mFastMixerDumpState.mTracks[mFastIndex].mUnderruns 830 uint32_t mUnderrunCount; // Counter of total number of underruns, never reset 831 volatile float mCachedVolume; // combined master volume and stream type volume; 832 // 'volatile' means accessed without lock or 833 // barrier, but is read/written atomically 834 }; // end of Track 835 836 class TimedTrack : public Track { 837 public: 838 static sp<TimedTrack> create(PlaybackThread *thread, 839 const sp<Client>& client, 840 audio_stream_type_t streamType, 841 uint32_t sampleRate, 842 audio_format_t format, 843 audio_channel_mask_t channelMask, 844 int frameCount, 845 const sp<IMemory>& sharedBuffer, 846 int sessionId); 847 ~TimedTrack(); 848 849 class TimedBuffer { 850 public: 851 TimedBuffer(); 852 TimedBuffer(const sp<IMemory>& buffer, int64_t pts); 853 const sp<IMemory>& buffer() const { return mBuffer; } 854 int64_t pts() const { return mPTS; } 855 uint32_t position() const { return mPosition; } 856 void setPosition(uint32_t pos) { mPosition = pos; } 857 private: 858 sp<IMemory> mBuffer; 859 int64_t mPTS; 860 uint32_t mPosition; 861 }; 862 863 // Mixer facing methods. 864 virtual bool isTimedTrack() const { return true; } 865 virtual size_t framesReady() const; 866 867 // AudioBufferProvider interface 868 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, 869 int64_t pts); 870 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 871 872 // Client/App facing methods. 873 status_t allocateTimedBuffer(size_t size, 874 sp<IMemory>* buffer); 875 status_t queueTimedBuffer(const sp<IMemory>& buffer, 876 int64_t pts); 877 status_t setMediaTimeTransform(const LinearTransform& xform, 878 TimedAudioTrack::TargetTimeline target); 879 880 private: 881 TimedTrack(PlaybackThread *thread, 882 const sp<Client>& client, 883 audio_stream_type_t streamType, 884 uint32_t sampleRate, 885 audio_format_t format, 886 audio_channel_mask_t channelMask, 887 int frameCount, 888 const sp<IMemory>& sharedBuffer, 889 int sessionId); 890 891 void timedYieldSamples_l(AudioBufferProvider::Buffer* buffer); 892 void timedYieldSilence_l(uint32_t numFrames, 893 AudioBufferProvider::Buffer* buffer); 894 void trimTimedBufferQueue_l(); 895 void trimTimedBufferQueueHead_l(const char* logTag); 896 void updateFramesPendingAfterTrim_l(const TimedBuffer& buf, 897 const char* logTag); 898 899 uint64_t mLocalTimeFreq; 900 LinearTransform mLocalTimeToSampleTransform; 901 LinearTransform mMediaTimeToSampleTransform; 902 sp<MemoryDealer> mTimedMemoryDealer; 903 904 Vector<TimedBuffer> mTimedBufferQueue; 905 bool mQueueHeadInFlight; 906 bool mTrimQueueHeadOnRelease; 907 uint32_t mFramesPendingInQueue; 908 909 uint8_t* mTimedSilenceBuffer; 910 uint32_t mTimedSilenceBufferSize; 911 mutable Mutex mTimedBufferQueueLock; 912 bool mTimedAudioOutputOnTime; 913 CCHelper mCCHelper; 914 915 Mutex mMediaTimeTransformLock; 916 LinearTransform mMediaTimeTransform; 917 bool mMediaTimeTransformValid; 918 TimedAudioTrack::TargetTimeline mMediaTimeTransformTarget; 919 }; 920 921 922 // playback track 923 class OutputTrack : public Track { 924 public: 925 926 class Buffer: public AudioBufferProvider::Buffer { 927 public: 928 int16_t *mBuffer; 929 }; 930 931 OutputTrack(PlaybackThread *thread, 932 DuplicatingThread *sourceThread, 933 uint32_t sampleRate, 934 audio_format_t format, 935 audio_channel_mask_t channelMask, 936 int frameCount); 937 virtual ~OutputTrack(); 938 939 virtual status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 940 int triggerSession = 0); 941 virtual void stop(); 942 bool write(int16_t* data, uint32_t frames); 943 bool bufferQueueEmpty() const { return mBufferQueue.size() == 0; } 944 bool isActive() const { return mActive; } 945 const wp<ThreadBase>& thread() const { return mThread; } 946 947 private: 948 949 enum { 950 NO_MORE_BUFFERS = 0x80000001, // same in AudioTrack.h, ok to be different value 951 }; 952 953 status_t obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs); 954 void clearBufferQueue(); 955 956 // Maximum number of pending buffers allocated by OutputTrack::write() 957 static const uint8_t kMaxOverFlowBuffers = 10; 958 959 Vector < Buffer* > mBufferQueue; 960 AudioBufferProvider::Buffer mOutBuffer; 961 bool mActive; 962 DuplicatingThread* const mSourceThread; // for waitTimeMs() in write() 963 }; // end of OutputTrack 964 965 PlaybackThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 966 audio_io_handle_t id, uint32_t device, type_t type); 967 virtual ~PlaybackThread(); 968 969 status_t dump(int fd, const Vector<String16>& args); 970 971 // Thread virtuals 972 virtual status_t readyToRun(); 973 virtual bool threadLoop(); 974 975 // RefBase 976 virtual void onFirstRef(); 977 978protected: 979 // Code snippets that were lifted up out of threadLoop() 980 virtual void threadLoop_mix() = 0; 981 virtual void threadLoop_sleepTime() = 0; 982 virtual void threadLoop_write(); 983 virtual void threadLoop_standby(); 984 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 985 986 // prepareTracks_l reads and writes mActiveTracks, and returns 987 // the pending set of tracks to remove via Vector 'tracksToRemove'. The caller 988 // is responsible for clearing or destroying this Vector later on, when it 989 // is safe to do so. That will drop the final ref count and destroy the tracks. 990 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove) = 0; 991 992public: 993 994 virtual status_t initCheck() const { return (mOutput == NULL) ? NO_INIT : NO_ERROR; } 995 996 // return estimated latency in milliseconds, as reported by HAL 997 uint32_t latency() const; 998 // same, but lock must already be held 999 uint32_t latency_l() const; 1000 1001 void setMasterVolume(float value); 1002 void setMasterMute(bool muted); 1003 1004 void setStreamVolume(audio_stream_type_t stream, float value); 1005 void setStreamMute(audio_stream_type_t stream, bool muted); 1006 1007 float streamVolume(audio_stream_type_t stream) const; 1008 1009 sp<Track> createTrack_l( 1010 const sp<AudioFlinger::Client>& client, 1011 audio_stream_type_t streamType, 1012 uint32_t sampleRate, 1013 audio_format_t format, 1014 audio_channel_mask_t channelMask, 1015 int frameCount, 1016 const sp<IMemory>& sharedBuffer, 1017 int sessionId, 1018 IAudioFlinger::track_flags_t flags, 1019 pid_t tid, 1020 status_t *status); 1021 1022 AudioStreamOut* getOutput() const; 1023 AudioStreamOut* clearOutput(); 1024 virtual audio_stream_t* stream() const; 1025 1026 void suspend() { mSuspended++; } 1027 void restore() { if (mSuspended > 0) mSuspended--; } 1028 bool isSuspended() const { return (mSuspended > 0); } 1029 virtual String8 getParameters(const String8& keys); 1030 virtual void audioConfigChanged_l(int event, int param = 0); 1031 status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames); 1032 int16_t *mixBuffer() const { return mMixBuffer; }; 1033 1034 virtual void detachAuxEffect_l(int effectId); 1035 status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track> track, 1036 int EffectId); 1037 status_t attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track> track, 1038 int EffectId); 1039 1040 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1041 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1042 virtual uint32_t hasAudioSession(int sessionId); 1043 virtual uint32_t getStrategyForSession_l(int sessionId); 1044 1045 1046 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 1047 virtual bool isValidSyncEvent(const sp<SyncEvent>& event); 1048 void invalidateTracks(audio_stream_type_t streamType); 1049 1050 1051 protected: 1052 int16_t* mMixBuffer; 1053 uint32_t mSuspended; // suspend count, > 0 means suspended 1054 int mBytesWritten; 1055 private: 1056 // mMasterMute is in both PlaybackThread and in AudioFlinger. When a 1057 // PlaybackThread needs to find out if master-muted, it checks it's local 1058 // copy rather than the one in AudioFlinger. This optimization saves a lock. 1059 bool mMasterMute; 1060 void setMasterMute_l(bool muted) { mMasterMute = muted; } 1061 protected: 1062 SortedVector< wp<Track> > mActiveTracks; // FIXME check if this could be sp<> 1063 1064 // Allocate a track name for a given channel mask. 1065 // Returns name >= 0 if successful, -1 on failure. 1066 virtual int getTrackName_l(audio_channel_mask_t channelMask) = 0; 1067 virtual void deleteTrackName_l(int name) = 0; 1068 1069 // Time to sleep between cycles when: 1070 virtual uint32_t activeSleepTimeUs() const; // mixer state MIXER_TRACKS_ENABLED 1071 virtual uint32_t idleSleepTimeUs() const = 0; // mixer state MIXER_IDLE 1072 virtual uint32_t suspendSleepTimeUs() const = 0; // audio policy manager suspended us 1073 // No sleep when mixer state == MIXER_TRACKS_READY; relies on audio HAL stream->write() 1074 // No sleep in standby mode; waits on a condition 1075 1076 // Code snippets that are temporarily lifted up out of threadLoop() until the merge 1077 void checkSilentMode_l(); 1078 1079 // Non-trivial for DUPLICATING only 1080 virtual void saveOutputTracks() { } 1081 virtual void clearOutputTracks() { } 1082 1083 // Cache various calculated values, at threadLoop() entry and after a parameter change 1084 virtual void cacheParameters_l(); 1085 1086 virtual uint32_t correctLatency(uint32_t latency) const; 1087 1088 private: 1089 1090 friend class AudioFlinger; // for numerous 1091 1092 PlaybackThread(const Client&); 1093 PlaybackThread& operator = (const PlaybackThread&); 1094 1095 status_t addTrack_l(const sp<Track>& track); 1096 void destroyTrack_l(const sp<Track>& track); 1097 void removeTrack_l(const sp<Track>& track); 1098 1099 void readOutputParameters(); 1100 1101 virtual status_t dumpInternals(int fd, const Vector<String16>& args); 1102 status_t dumpTracks(int fd, const Vector<String16>& args); 1103 1104 SortedVector< sp<Track> > mTracks; 1105 // mStreamTypes[] uses 1 additional stream type internally for the OutputTrack used by DuplicatingThread 1106 stream_type_t mStreamTypes[AUDIO_STREAM_CNT + 1]; 1107 AudioStreamOut *mOutput; 1108 1109 float mMasterVolume; 1110 nsecs_t mLastWriteTime; 1111 int mNumWrites; 1112 int mNumDelayedWrites; 1113 bool mInWrite; 1114 1115 // FIXME rename these former local variables of threadLoop to standard "m" names 1116 nsecs_t standbyTime; 1117 size_t mixBufferSize; 1118 1119 // cached copies of activeSleepTimeUs() and idleSleepTimeUs() made by cacheParameters_l() 1120 uint32_t activeSleepTime; 1121 uint32_t idleSleepTime; 1122 1123 uint32_t sleepTime; 1124 1125 // mixer status returned by prepareTracks_l() 1126 mixer_state mMixerStatus; // current cycle 1127 // previous cycle when in prepareTracks_l() 1128 mixer_state mMixerStatusIgnoringFastTracks; 1129 // FIXME or a separate ready state per track 1130 1131 // FIXME move these declarations into the specific sub-class that needs them 1132 // MIXER only 1133 uint32_t sleepTimeShift; 1134 1135 // same as AudioFlinger::mStandbyTimeInNsecs except for DIRECT which uses a shorter value 1136 nsecs_t standbyDelay; 1137 1138 // MIXER only 1139 nsecs_t maxPeriod; 1140 1141 // DUPLICATING only 1142 uint32_t writeFrames; 1143 1144 private: 1145 // The HAL output sink is treated as non-blocking, but current implementation is blocking 1146 sp<NBAIO_Sink> mOutputSink; 1147 // If a fast mixer is present, the blocking pipe sink, otherwise clear 1148 sp<NBAIO_Sink> mPipeSink; 1149 // The current sink for the normal mixer to write it's (sub)mix, mOutputSink or mPipeSink 1150 sp<NBAIO_Sink> mNormalSink; 1151 // For dumpsys 1152 sp<NBAIO_Sink> mTeeSink; 1153 sp<NBAIO_Source> mTeeSource; 1154 uint32_t mScreenState; // cached copy of gScreenState 1155 public: 1156 virtual bool hasFastMixer() const = 0; 1157 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const 1158 { FastTrackUnderruns dummy; return dummy; } 1159 1160 protected: 1161 // accessed by both binder threads and within threadLoop(), lock on mutex needed 1162 unsigned mFastTrackAvailMask; // bit i set if fast track [i] is available 1163 1164 }; 1165 1166 class MixerThread : public PlaybackThread { 1167 public: 1168 MixerThread (const sp<AudioFlinger>& audioFlinger, 1169 AudioStreamOut* output, 1170 audio_io_handle_t id, 1171 uint32_t device, 1172 type_t type = MIXER); 1173 virtual ~MixerThread(); 1174 1175 // Thread virtuals 1176 1177 virtual bool checkForNewParameters_l(); 1178 virtual status_t dumpInternals(int fd, const Vector<String16>& args); 1179 1180 protected: 1181 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1182 virtual int getTrackName_l(audio_channel_mask_t channelMask); 1183 virtual void deleteTrackName_l(int name); 1184 virtual uint32_t idleSleepTimeUs() const; 1185 virtual uint32_t suspendSleepTimeUs() const; 1186 virtual void cacheParameters_l(); 1187 1188 // threadLoop snippets 1189 virtual void threadLoop_write(); 1190 virtual void threadLoop_standby(); 1191 virtual void threadLoop_mix(); 1192 virtual void threadLoop_sleepTime(); 1193 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 1194 virtual uint32_t correctLatency(uint32_t latency) const; 1195 1196 AudioMixer* mAudioMixer; // normal mixer 1197 private: 1198 // one-time initialization, no locks required 1199 FastMixer* mFastMixer; // non-NULL if there is also a fast mixer 1200 sp<AudioWatchdog> mAudioWatchdog; // non-0 if there is an audio watchdog thread 1201 1202 // contents are not guaranteed to be consistent, no locks required 1203 FastMixerDumpState mFastMixerDumpState; 1204#ifdef STATE_QUEUE_DUMP 1205 StateQueueObserverDump mStateQueueObserverDump; 1206 StateQueueMutatorDump mStateQueueMutatorDump; 1207#endif 1208 AudioWatchdogDump mAudioWatchdogDump; 1209 1210 // accessible only within the threadLoop(), no locks required 1211 // mFastMixer->sq() // for mutating and pushing state 1212 int32_t mFastMixerFutex; // for cold idle 1213 1214 public: 1215 virtual bool hasFastMixer() const { return mFastMixer != NULL; } 1216 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const { 1217 ALOG_ASSERT(fastIndex < FastMixerState::kMaxFastTracks); 1218 return mFastMixerDumpState.mTracks[fastIndex].mUnderruns; 1219 } 1220 }; 1221 1222 class DirectOutputThread : public PlaybackThread { 1223 public: 1224 1225 DirectOutputThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1226 audio_io_handle_t id, uint32_t device); 1227 virtual ~DirectOutputThread(); 1228 1229 // Thread virtuals 1230 1231 virtual bool checkForNewParameters_l(); 1232 1233 protected: 1234 virtual int getTrackName_l(audio_channel_mask_t channelMask); 1235 virtual void deleteTrackName_l(int name); 1236 virtual uint32_t activeSleepTimeUs() const; 1237 virtual uint32_t idleSleepTimeUs() const; 1238 virtual uint32_t suspendSleepTimeUs() const; 1239 virtual void cacheParameters_l(); 1240 1241 // threadLoop snippets 1242 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1243 virtual void threadLoop_mix(); 1244 virtual void threadLoop_sleepTime(); 1245 1246 // volumes last sent to audio HAL with stream->set_volume() 1247 float mLeftVolFloat; 1248 float mRightVolFloat; 1249 1250private: 1251 // prepareTracks_l() tells threadLoop_mix() the name of the single active track 1252 sp<Track> mActiveTrack; 1253 public: 1254 virtual bool hasFastMixer() const { return false; } 1255 }; 1256 1257 class DuplicatingThread : public MixerThread { 1258 public: 1259 DuplicatingThread (const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread, 1260 audio_io_handle_t id); 1261 virtual ~DuplicatingThread(); 1262 1263 // Thread virtuals 1264 void addOutputTrack(MixerThread* thread); 1265 void removeOutputTrack(MixerThread* thread); 1266 uint32_t waitTimeMs() const { return mWaitTimeMs; } 1267 protected: 1268 virtual uint32_t activeSleepTimeUs() const; 1269 1270 private: 1271 bool outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks); 1272 protected: 1273 // threadLoop snippets 1274 virtual void threadLoop_mix(); 1275 virtual void threadLoop_sleepTime(); 1276 virtual void threadLoop_write(); 1277 virtual void threadLoop_standby(); 1278 virtual void cacheParameters_l(); 1279 1280 private: 1281 // called from threadLoop, addOutputTrack, removeOutputTrack 1282 virtual void updateWaitTime_l(); 1283 protected: 1284 virtual void saveOutputTracks(); 1285 virtual void clearOutputTracks(); 1286 private: 1287 1288 uint32_t mWaitTimeMs; 1289 SortedVector < sp<OutputTrack> > outputTracks; 1290 SortedVector < sp<OutputTrack> > mOutputTracks; 1291 public: 1292 virtual bool hasFastMixer() const { return false; } 1293 }; 1294 1295 PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const; 1296 MixerThread *checkMixerThread_l(audio_io_handle_t output) const; 1297 RecordThread *checkRecordThread_l(audio_io_handle_t input) const; 1298 // no range check, AudioFlinger::mLock held 1299 bool streamMute_l(audio_stream_type_t stream) const 1300 { return mStreamTypes[stream].mute; } 1301 // no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held 1302 float streamVolume_l(audio_stream_type_t stream) const 1303 { return mStreamTypes[stream].volume; } 1304 void audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2); 1305 1306 // allocate an audio_io_handle_t, session ID, or effect ID 1307 uint32_t nextUniqueId(); 1308 1309 status_t moveEffectChain_l(int sessionId, 1310 PlaybackThread *srcThread, 1311 PlaybackThread *dstThread, 1312 bool reRegister); 1313 // return thread associated with primary hardware device, or NULL 1314 PlaybackThread *primaryPlaybackThread_l() const; 1315 uint32_t primaryOutputDevice_l() const; 1316 1317 sp<PlaybackThread> getEffectThread_l(int sessionId, int EffectId); 1318 1319 // server side of the client's IAudioTrack 1320 class TrackHandle : public android::BnAudioTrack { 1321 public: 1322 TrackHandle(const sp<PlaybackThread::Track>& track); 1323 virtual ~TrackHandle(); 1324 virtual sp<IMemory> getCblk() const; 1325 virtual status_t start(); 1326 virtual void stop(); 1327 virtual void flush(); 1328 virtual void mute(bool); 1329 virtual void pause(); 1330 virtual status_t attachAuxEffect(int effectId); 1331 virtual status_t allocateTimedBuffer(size_t size, 1332 sp<IMemory>* buffer); 1333 virtual status_t queueTimedBuffer(const sp<IMemory>& buffer, 1334 int64_t pts); 1335 virtual status_t setMediaTimeTransform(const LinearTransform& xform, 1336 int target); 1337 virtual status_t onTransact( 1338 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 1339 private: 1340 const sp<PlaybackThread::Track> mTrack; 1341 }; 1342 1343 void removeClient_l(pid_t pid); 1344 void removeNotificationClient(pid_t pid); 1345 1346 1347 // record thread 1348 class RecordThread : public ThreadBase, public AudioBufferProvider 1349 { 1350 public: 1351 1352 // record track 1353 class RecordTrack : public TrackBase { 1354 public: 1355 RecordTrack(RecordThread *thread, 1356 const sp<Client>& client, 1357 uint32_t sampleRate, 1358 audio_format_t format, 1359 audio_channel_mask_t channelMask, 1360 int frameCount, 1361 int sessionId); 1362 virtual ~RecordTrack(); 1363 1364 virtual status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 1365 int triggerSession = 0); 1366 virtual void stop(); 1367 1368 // clear the buffer overflow flag 1369 void clearOverflow() { mOverflow = false; } 1370 // set the buffer overflow flag and return previous value 1371 bool setOverflow() { bool tmp = mOverflow; mOverflow = true; return tmp; } 1372 1373 void dump(char* buffer, size_t size); 1374 1375 private: 1376 friend class AudioFlinger; // for mState 1377 1378 RecordTrack(const RecordTrack&); 1379 RecordTrack& operator = (const RecordTrack&); 1380 1381 // AudioBufferProvider interface 1382 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts = kInvalidPTS); 1383 // releaseBuffer() not overridden 1384 1385 bool mOverflow; // overflow on most recent attempt to fill client buffer 1386 }; 1387 1388 RecordThread(const sp<AudioFlinger>& audioFlinger, 1389 AudioStreamIn *input, 1390 uint32_t sampleRate, 1391 audio_channel_mask_t channelMask, 1392 audio_io_handle_t id, 1393 uint32_t device); 1394 virtual ~RecordThread(); 1395 1396 // Thread 1397 virtual bool threadLoop(); 1398 virtual status_t readyToRun(); 1399 1400 // RefBase 1401 virtual void onFirstRef(); 1402 1403 virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; } 1404 sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l( 1405 const sp<AudioFlinger::Client>& client, 1406 uint32_t sampleRate, 1407 audio_format_t format, 1408 audio_channel_mask_t channelMask, 1409 int frameCount, 1410 int sessionId, 1411 status_t *status); 1412 1413 status_t start(RecordTrack* recordTrack, 1414 AudioSystem::sync_event_t event, 1415 int triggerSession); 1416 void stop(RecordTrack* recordTrack); 1417 status_t dump(int fd, const Vector<String16>& args); 1418 AudioStreamIn* getInput() const; 1419 AudioStreamIn* clearInput(); 1420 virtual audio_stream_t* stream() const; 1421 1422 // AudioBufferProvider interface 1423 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts); 1424 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 1425 1426 virtual bool checkForNewParameters_l(); 1427 virtual String8 getParameters(const String8& keys); 1428 virtual void audioConfigChanged_l(int event, int param = 0); 1429 void readInputParameters(); 1430 virtual unsigned int getInputFramesLost(); 1431 1432 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1433 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1434 virtual uint32_t hasAudioSession(int sessionId); 1435 RecordTrack* track(); 1436 1437 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 1438 virtual bool isValidSyncEvent(const sp<SyncEvent>& event); 1439 1440 static void syncStartEventCallback(const wp<SyncEvent>& event); 1441 void handleSyncStartEvent(const sp<SyncEvent>& event); 1442 1443 private: 1444 void clearSyncStartEvent(); 1445 1446 RecordThread(); 1447 AudioStreamIn *mInput; 1448 RecordTrack* mTrack; 1449 sp<RecordTrack> mActiveTrack; 1450 Condition mStartStopCond; 1451 AudioResampler *mResampler; 1452 int32_t *mRsmpOutBuffer; 1453 int16_t *mRsmpInBuffer; 1454 size_t mRsmpInIndex; 1455 size_t mInputBytes; 1456 const int mReqChannelCount; 1457 const uint32_t mReqSampleRate; 1458 ssize_t mBytesRead; 1459 // sync event triggering actual audio capture. Frames read before this event will 1460 // be dropped and therefore not read by the application. 1461 sp<SyncEvent> mSyncStartEvent; 1462 // number of captured frames to drop after the start sync event has been received. 1463 // when < 0, maximum frames to drop before starting capture even if sync event is 1464 // not received 1465 ssize_t mFramestoDrop; 1466 }; 1467 1468 // server side of the client's IAudioRecord 1469 class RecordHandle : public android::BnAudioRecord { 1470 public: 1471 RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack); 1472 virtual ~RecordHandle(); 1473 virtual sp<IMemory> getCblk() const; 1474 virtual status_t start(int event, int triggerSession); 1475 virtual void stop(); 1476 virtual status_t onTransact( 1477 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 1478 private: 1479 const sp<RecordThread::RecordTrack> mRecordTrack; 1480 }; 1481 1482 //--- Audio Effect Management 1483 1484 // EffectModule and EffectChain classes both have their own mutex to protect 1485 // state changes or resource modifications. Always respect the following order 1486 // if multiple mutexes must be acquired to avoid cross deadlock: 1487 // AudioFlinger -> ThreadBase -> EffectChain -> EffectModule 1488 1489 // The EffectModule class is a wrapper object controlling the effect engine implementation 1490 // in the effect library. It prevents concurrent calls to process() and command() functions 1491 // from different client threads. It keeps a list of EffectHandle objects corresponding 1492 // to all client applications using this effect and notifies applications of effect state, 1493 // control or parameter changes. It manages the activation state machine to send appropriate 1494 // reset, enable, disable commands to effect engine and provide volume 1495 // ramping when effects are activated/deactivated. 1496 // When controlling an auxiliary effect, the EffectModule also provides an input buffer used by 1497 // the attached track(s) to accumulate their auxiliary channel. 1498 class EffectModule: public RefBase { 1499 public: 1500 EffectModule(ThreadBase *thread, 1501 const wp<AudioFlinger::EffectChain>& chain, 1502 effect_descriptor_t *desc, 1503 int id, 1504 int sessionId); 1505 virtual ~EffectModule(); 1506 1507 enum effect_state { 1508 IDLE, 1509 RESTART, 1510 STARTING, 1511 ACTIVE, 1512 STOPPING, 1513 STOPPED, 1514 DESTROYED 1515 }; 1516 1517 int id() const { return mId; } 1518 void process(); 1519 void updateState(); 1520 status_t command(uint32_t cmdCode, 1521 uint32_t cmdSize, 1522 void *pCmdData, 1523 uint32_t *replySize, 1524 void *pReplyData); 1525 1526 void reset_l(); 1527 status_t configure(); 1528 status_t init(); 1529 effect_state state() const { 1530 return mState; 1531 } 1532 uint32_t status() { 1533 return mStatus; 1534 } 1535 int sessionId() const { 1536 return mSessionId; 1537 } 1538 status_t setEnabled(bool enabled); 1539 status_t setEnabled_l(bool enabled); 1540 bool isEnabled() const; 1541 bool isProcessEnabled() const; 1542 1543 void setInBuffer(int16_t *buffer) { mConfig.inputCfg.buffer.s16 = buffer; } 1544 int16_t *inBuffer() { return mConfig.inputCfg.buffer.s16; } 1545 void setOutBuffer(int16_t *buffer) { mConfig.outputCfg.buffer.s16 = buffer; } 1546 int16_t *outBuffer() { return mConfig.outputCfg.buffer.s16; } 1547 void setChain(const wp<EffectChain>& chain) { mChain = chain; } 1548 void setThread(const wp<ThreadBase>& thread) { mThread = thread; } 1549 const wp<ThreadBase>& thread() { return mThread; } 1550 1551 status_t addHandle(EffectHandle *handle); 1552 size_t disconnect(EffectHandle *handle, bool unpinIfLast); 1553 size_t removeHandle(EffectHandle *handle); 1554 1555 effect_descriptor_t& desc() { return mDescriptor; } 1556 wp<EffectChain>& chain() { return mChain; } 1557 1558 status_t setDevice(uint32_t device); 1559 status_t setVolume(uint32_t *left, uint32_t *right, bool controller); 1560 status_t setMode(audio_mode_t mode); 1561 status_t start(); 1562 status_t stop(); 1563 void setSuspended(bool suspended); 1564 bool suspended() const; 1565 1566 EffectHandle* controlHandle_l(); 1567 1568 bool isPinned() const { return mPinned; } 1569 void unPin() { mPinned = false; } 1570 bool purgeHandles(); 1571 void lock() { mLock.lock(); } 1572 void unlock() { mLock.unlock(); } 1573 1574 status_t dump(int fd, const Vector<String16>& args); 1575 1576 protected: 1577 friend class AudioFlinger; // for mHandles 1578 bool mPinned; 1579 1580 // Maximum time allocated to effect engines to complete the turn off sequence 1581 static const uint32_t MAX_DISABLE_TIME_MS = 10000; 1582 1583 EffectModule(const EffectModule&); 1584 EffectModule& operator = (const EffectModule&); 1585 1586 status_t start_l(); 1587 status_t stop_l(); 1588 1589mutable Mutex mLock; // mutex for process, commands and handles list protection 1590 wp<ThreadBase> mThread; // parent thread 1591 wp<EffectChain> mChain; // parent effect chain 1592 const int mId; // this instance unique ID 1593 const int mSessionId; // audio session ID 1594 effect_descriptor_t mDescriptor;// effect descriptor received from effect engine 1595 effect_config_t mConfig; // input and output audio configuration 1596 effect_handle_t mEffectInterface; // Effect module C API 1597 status_t mStatus; // initialization status 1598 effect_state mState; // current activation state 1599 Vector<EffectHandle *> mHandles; // list of client handles 1600 // First handle in mHandles has highest priority and controls the effect module 1601 uint32_t mMaxDisableWaitCnt; // maximum grace period before forcing an effect off after 1602 // sending disable command. 1603 uint32_t mDisableWaitCnt; // current process() calls count during disable period. 1604 bool mSuspended; // effect is suspended: temporarily disabled by framework 1605 }; 1606 1607 // The EffectHandle class implements the IEffect interface. It provides resources 1608 // to receive parameter updates, keeps track of effect control 1609 // ownership and state and has a pointer to the EffectModule object it is controlling. 1610 // There is one EffectHandle object for each application controlling (or using) 1611 // an effect module. 1612 // The EffectHandle is obtained by calling AudioFlinger::createEffect(). 1613 class EffectHandle: public android::BnEffect { 1614 public: 1615 1616 EffectHandle(const sp<EffectModule>& effect, 1617 const sp<AudioFlinger::Client>& client, 1618 const sp<IEffectClient>& effectClient, 1619 int32_t priority); 1620 virtual ~EffectHandle(); 1621 1622 // IEffect 1623 virtual status_t enable(); 1624 virtual status_t disable(); 1625 virtual status_t command(uint32_t cmdCode, 1626 uint32_t cmdSize, 1627 void *pCmdData, 1628 uint32_t *replySize, 1629 void *pReplyData); 1630 virtual void disconnect(); 1631 private: 1632 void disconnect(bool unpinIfLast); 1633 public: 1634 virtual sp<IMemory> getCblk() const { return mCblkMemory; } 1635 virtual status_t onTransact(uint32_t code, const Parcel& data, 1636 Parcel* reply, uint32_t flags); 1637 1638 1639 // Give or take control of effect module 1640 // - hasControl: true if control is given, false if removed 1641 // - signal: true client app should be signaled of change, false otherwise 1642 // - enabled: state of the effect when control is passed 1643 void setControl(bool hasControl, bool signal, bool enabled); 1644 void commandExecuted(uint32_t cmdCode, 1645 uint32_t cmdSize, 1646 void *pCmdData, 1647 uint32_t replySize, 1648 void *pReplyData); 1649 void setEnabled(bool enabled); 1650 bool enabled() const { return mEnabled; } 1651 1652 // Getters 1653 int id() const { return mEffect->id(); } 1654 int priority() const { return mPriority; } 1655 bool hasControl() const { return mHasControl; } 1656 sp<EffectModule> effect() const { return mEffect; } 1657 // destroyed_l() must be called with the associated EffectModule mLock held 1658 bool destroyed_l() const { return mDestroyed; } 1659 1660 void dump(char* buffer, size_t size); 1661 1662 protected: 1663 friend class AudioFlinger; // for mEffect, mHasControl, mEnabled 1664 EffectHandle(const EffectHandle&); 1665 EffectHandle& operator =(const EffectHandle&); 1666 1667 sp<EffectModule> mEffect; // pointer to controlled EffectModule 1668 sp<IEffectClient> mEffectClient; // callback interface for client notifications 1669 /*const*/ sp<Client> mClient; // client for shared memory allocation, see disconnect() 1670 sp<IMemory> mCblkMemory; // shared memory for control block 1671 effect_param_cblk_t* mCblk; // control block for deferred parameter setting via shared memory 1672 uint8_t* mBuffer; // pointer to parameter area in shared memory 1673 int mPriority; // client application priority to control the effect 1674 bool mHasControl; // true if this handle is controlling the effect 1675 bool mEnabled; // cached enable state: needed when the effect is 1676 // restored after being suspended 1677 bool mDestroyed; // Set to true by destructor. Access with EffectModule 1678 // mLock held 1679 }; 1680 1681 // the EffectChain class represents a group of effects associated to one audio session. 1682 // There can be any number of EffectChain objects per output mixer thread (PlaybackThread). 1683 // The EffecChain with session ID 0 contains global effects applied to the output mix. 1684 // Effects in this chain can be insert or auxiliary. Effects in other chains (attached to tracks) 1685 // are insert only. The EffectChain maintains an ordered list of effect module, the order corresponding 1686 // in the effect process order. When attached to a track (session ID != 0), it also provide it's own 1687 // input buffer used by the track as accumulation buffer. 1688 class EffectChain: public RefBase { 1689 public: 1690 EffectChain(const wp<ThreadBase>& wThread, int sessionId); 1691 EffectChain(ThreadBase *thread, int sessionId); 1692 virtual ~EffectChain(); 1693 1694 // special key used for an entry in mSuspendedEffects keyed vector 1695 // corresponding to a suspend all request. 1696 static const int kKeyForSuspendAll = 0; 1697 1698 // minimum duration during which we force calling effect process when last track on 1699 // a session is stopped or removed to allow effect tail to be rendered 1700 static const int kProcessTailDurationMs = 1000; 1701 1702 void process_l(); 1703 1704 void lock() { 1705 mLock.lock(); 1706 } 1707 void unlock() { 1708 mLock.unlock(); 1709 } 1710 1711 status_t addEffect_l(const sp<EffectModule>& handle); 1712 size_t removeEffect_l(const sp<EffectModule>& handle); 1713 1714 int sessionId() const { return mSessionId; } 1715 void setSessionId(int sessionId) { mSessionId = sessionId; } 1716 1717 sp<EffectModule> getEffectFromDesc_l(effect_descriptor_t *descriptor); 1718 sp<EffectModule> getEffectFromId_l(int id); 1719 sp<EffectModule> getEffectFromType_l(const effect_uuid_t *type); 1720 bool setVolume_l(uint32_t *left, uint32_t *right); 1721 void setDevice_l(uint32_t device); 1722 void setMode_l(audio_mode_t mode); 1723 1724 void setInBuffer(int16_t *buffer, bool ownsBuffer = false) { 1725 mInBuffer = buffer; 1726 mOwnInBuffer = ownsBuffer; 1727 } 1728 int16_t *inBuffer() const { 1729 return mInBuffer; 1730 } 1731 void setOutBuffer(int16_t *buffer) { 1732 mOutBuffer = buffer; 1733 } 1734 int16_t *outBuffer() const { 1735 return mOutBuffer; 1736 } 1737 1738 void incTrackCnt() { android_atomic_inc(&mTrackCnt); } 1739 void decTrackCnt() { android_atomic_dec(&mTrackCnt); } 1740 int32_t trackCnt() const { return android_atomic_acquire_load(&mTrackCnt); } 1741 1742 void incActiveTrackCnt() { android_atomic_inc(&mActiveTrackCnt); 1743 mTailBufferCount = mMaxTailBuffers; } 1744 void decActiveTrackCnt() { android_atomic_dec(&mActiveTrackCnt); } 1745 int32_t activeTrackCnt() const { return android_atomic_acquire_load(&mActiveTrackCnt); } 1746 1747 uint32_t strategy() const { return mStrategy; } 1748 void setStrategy(uint32_t strategy) 1749 { mStrategy = strategy; } 1750 1751 // suspend effect of the given type 1752 void setEffectSuspended_l(const effect_uuid_t *type, 1753 bool suspend); 1754 // suspend all eligible effects 1755 void setEffectSuspendedAll_l(bool suspend); 1756 // check if effects should be suspend or restored when a given effect is enable or disabled 1757 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1758 bool enabled); 1759 1760 void clearInputBuffer(); 1761 1762 status_t dump(int fd, const Vector<String16>& args); 1763 1764 protected: 1765 friend class AudioFlinger; // for mThread, mEffects 1766 EffectChain(const EffectChain&); 1767 EffectChain& operator =(const EffectChain&); 1768 1769 class SuspendedEffectDesc : public RefBase { 1770 public: 1771 SuspendedEffectDesc() : mRefCount(0) {} 1772 1773 int mRefCount; 1774 effect_uuid_t mType; 1775 wp<EffectModule> mEffect; 1776 }; 1777 1778 // get a list of effect modules to suspend when an effect of the type 1779 // passed is enabled. 1780 void getSuspendEligibleEffects(Vector< sp<EffectModule> > &effects); 1781 1782 // get an effect module if it is currently enable 1783 sp<EffectModule> getEffectIfEnabled(const effect_uuid_t *type); 1784 // true if the effect whose descriptor is passed can be suspended 1785 // OEMs can modify the rules implemented in this method to exclude specific effect 1786 // types or implementations from the suspend/restore mechanism. 1787 bool isEffectEligibleForSuspend(const effect_descriptor_t& desc); 1788 1789 void clearInputBuffer_l(sp<ThreadBase> thread); 1790 1791 wp<ThreadBase> mThread; // parent mixer thread 1792 Mutex mLock; // mutex protecting effect list 1793 Vector< sp<EffectModule> > mEffects; // list of effect modules 1794 int mSessionId; // audio session ID 1795 int16_t *mInBuffer; // chain input buffer 1796 int16_t *mOutBuffer; // chain output buffer 1797 1798 // 'volatile' here means these are accessed with atomic operations instead of mutex 1799 volatile int32_t mActiveTrackCnt; // number of active tracks connected 1800 volatile int32_t mTrackCnt; // number of tracks connected 1801 1802 int32_t mTailBufferCount; // current effect tail buffer count 1803 int32_t mMaxTailBuffers; // maximum effect tail buffers 1804 bool mOwnInBuffer; // true if the chain owns its input buffer 1805 int mVolumeCtrlIdx; // index of insert effect having control over volume 1806 uint32_t mLeftVolume; // previous volume on left channel 1807 uint32_t mRightVolume; // previous volume on right channel 1808 uint32_t mNewLeftVolume; // new volume on left channel 1809 uint32_t mNewRightVolume; // new volume on right channel 1810 uint32_t mStrategy; // strategy for this effect chain 1811 // mSuspendedEffects lists all effects currently suspended in the chain. 1812 // Use effect type UUID timelow field as key. There is no real risk of identical 1813 // timeLow fields among effect type UUIDs. 1814 // Updated by updateSuspendedSessions_l() only. 1815 KeyedVector< int, sp<SuspendedEffectDesc> > mSuspendedEffects; 1816 }; 1817 1818 // AudioStreamOut and AudioStreamIn are immutable, so their fields are const. 1819 // For emphasis, we could also make all pointers to them be "const *", 1820 // but that would clutter the code unnecessarily. 1821 1822 struct AudioStreamOut { 1823 audio_hw_device_t* const hwDev; 1824 audio_stream_out_t* const stream; 1825 1826 AudioStreamOut(audio_hw_device_t *dev, audio_stream_out_t *out) : 1827 hwDev(dev), stream(out) {} 1828 }; 1829 1830 struct AudioStreamIn { 1831 audio_hw_device_t* const hwDev; 1832 audio_stream_in_t* const stream; 1833 1834 AudioStreamIn(audio_hw_device_t *dev, audio_stream_in_t *in) : 1835 hwDev(dev), stream(in) {} 1836 }; 1837 1838 // for mAudioSessionRefs only 1839 struct AudioSessionRef { 1840 AudioSessionRef(int sessionid, pid_t pid) : 1841 mSessionid(sessionid), mPid(pid), mCnt(1) {} 1842 const int mSessionid; 1843 const pid_t mPid; 1844 int mCnt; 1845 }; 1846 1847 enum master_volume_support { 1848 // MVS_NONE: 1849 // Audio HAL has no support for master volume, either setting or 1850 // getting. All master volume control must be implemented in SW by the 1851 // AudioFlinger mixing core. 1852 MVS_NONE, 1853 1854 // MVS_SETONLY: 1855 // Audio HAL has support for setting master volume, but not for getting 1856 // master volume (original HAL design did not include a getter). 1857 // AudioFlinger needs to keep track of the last set master volume in 1858 // addition to needing to set an initial, default, master volume at HAL 1859 // load time. 1860 MVS_SETONLY, 1861 1862 // MVS_FULL: 1863 // Audio HAL has support both for setting and getting master volume. 1864 // AudioFlinger should send all set and get master volume requests 1865 // directly to the HAL. 1866 MVS_FULL, 1867 }; 1868 1869 class AudioHwDevice { 1870 public: 1871 AudioHwDevice(const char *moduleName, audio_hw_device_t *hwDevice) : 1872 mModuleName(strdup(moduleName)), mHwDevice(hwDevice){} 1873 ~AudioHwDevice() { free((void *)mModuleName); } 1874 1875 const char *moduleName() const { return mModuleName; } 1876 audio_hw_device_t *hwDevice() const { return mHwDevice; } 1877 private: 1878 const char * const mModuleName; 1879 audio_hw_device_t * const mHwDevice; 1880 }; 1881 1882 mutable Mutex mLock; 1883 1884 DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client() 1885 1886 mutable Mutex mHardwareLock; 1887 // NOTE: If both mLock and mHardwareLock mutexes must be held, 1888 // always take mLock before mHardwareLock 1889 1890 // These two fields are immutable after onFirstRef(), so no lock needed to access 1891 audio_hw_device_t* mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL 1892 DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*> mAudioHwDevs; 1893 1894 // for dump, indicates which hardware operation is currently in progress (but not stream ops) 1895 enum hardware_call_state { 1896 AUDIO_HW_IDLE = 0, // no operation in progress 1897 AUDIO_HW_INIT, // init_check 1898 AUDIO_HW_OUTPUT_OPEN, // open_output_stream 1899 AUDIO_HW_OUTPUT_CLOSE, // unused 1900 AUDIO_HW_INPUT_OPEN, // unused 1901 AUDIO_HW_INPUT_CLOSE, // unused 1902 AUDIO_HW_STANDBY, // unused 1903 AUDIO_HW_SET_MASTER_VOLUME, // set_master_volume 1904 AUDIO_HW_GET_ROUTING, // unused 1905 AUDIO_HW_SET_ROUTING, // unused 1906 AUDIO_HW_GET_MODE, // unused 1907 AUDIO_HW_SET_MODE, // set_mode 1908 AUDIO_HW_GET_MIC_MUTE, // get_mic_mute 1909 AUDIO_HW_SET_MIC_MUTE, // set_mic_mute 1910 AUDIO_HW_SET_VOICE_VOLUME, // set_voice_volume 1911 AUDIO_HW_SET_PARAMETER, // set_parameters 1912 AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size 1913 AUDIO_HW_GET_MASTER_VOLUME, // get_master_volume 1914 AUDIO_HW_GET_PARAMETER, // get_parameters 1915 }; 1916 1917 mutable hardware_call_state mHardwareStatus; // for dump only 1918 1919 1920 DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads; 1921 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 1922 1923 // both are protected by mLock 1924 float mMasterVolume; 1925 float mMasterVolumeSW; 1926 master_volume_support mMasterVolumeSupportLvl; 1927 bool mMasterMute; 1928 1929 DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads; 1930 1931 DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients; 1932 volatile int32_t mNextUniqueId; // updated by android_atomic_inc 1933 audio_mode_t mMode; 1934 bool mBtNrecIsOff; 1935 1936 // protected by mLock 1937 Vector<AudioSessionRef*> mAudioSessionRefs; 1938 1939 float masterVolume_l() const; 1940 float masterVolumeSW_l() const { return mMasterVolumeSW; } 1941 bool masterMute_l() const { return mMasterMute; } 1942 audio_module_handle_t loadHwModule_l(const char *name); 1943 1944 Vector < sp<SyncEvent> > mPendingSyncEvents; // sync events awaiting for a session 1945 // to be created 1946 1947private: 1948 sp<Client> registerPid_l(pid_t pid); // always returns non-0 1949 1950}; 1951 1952 1953// ---------------------------------------------------------------------------- 1954 1955}; // namespace android 1956 1957#endif // ANDROID_AUDIO_FLINGER_H 1958