AudioFlinger.h revision 0bf65bdde04b8e66c998ff37e2b2afafddddfa33
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 47#include "AudioBufferProvider.h" 48 49#include <powermanager/IPowerManager.h> 50 51namespace android { 52 53class audio_track_cblk_t; 54class effect_param_cblk_t; 55class AudioMixer; 56class AudioBuffer; 57class AudioResampler; 58 59// ---------------------------------------------------------------------------- 60 61// AudioFlinger has a hard-coded upper limit of 2 channels for capture and playback. 62// There is support for > 2 channel tracks down-mixed to 2 channel output via a down-mix effect. 63// Adding full support for > 2 channel capture or playback would require more than simply changing 64// this #define. There is an independent hard-coded upper limit in AudioMixer; 65// removing that AudioMixer limit would be necessary but insufficient to support > 2 channels. 66// The macro FCC_2 highlights some (but not all) places where there is are 2-channel assumptions. 67// Search also for "2", "left", "right", "[0]", "[1]", ">> 16", "<< 16", etc. 68#define FCC_2 2 // FCC_2 = Fixed Channel Count 2 69 70static const nsecs_t kDefaultStandbyTimeInNsecs = seconds(3); 71 72class AudioFlinger : 73 public BinderService<AudioFlinger>, 74 public BnAudioFlinger 75{ 76 friend class BinderService<AudioFlinger>; // for AudioFlinger() 77public: 78 static const char* getServiceName() { return "media.audio_flinger"; } 79 80 virtual status_t dump(int fd, const Vector<String16>& args); 81 82 // IAudioFlinger interface, in binder opcode order 83 virtual sp<IAudioTrack> createTrack( 84 pid_t pid, 85 audio_stream_type_t streamType, 86 uint32_t sampleRate, 87 audio_format_t format, 88 uint32_t channelMask, 89 int frameCount, 90 IAudioFlinger::track_flags_t flags, 91 const sp<IMemory>& sharedBuffer, 92 audio_io_handle_t output, 93 int *sessionId, 94 status_t *status); 95 96 virtual sp<IAudioRecord> openRecord( 97 pid_t pid, 98 audio_io_handle_t input, 99 uint32_t sampleRate, 100 audio_format_t format, 101 uint32_t channelMask, 102 int frameCount, 103 IAudioFlinger::track_flags_t flags, 104 int *sessionId, 105 status_t *status); 106 107 virtual uint32_t sampleRate(audio_io_handle_t output) const; 108 virtual int channelCount(audio_io_handle_t output) const; 109 virtual audio_format_t format(audio_io_handle_t output) const; 110 virtual size_t frameCount(audio_io_handle_t output) const; 111 virtual uint32_t latency(audio_io_handle_t output) const; 112 113 virtual status_t setMasterVolume(float value); 114 virtual status_t setMasterMute(bool muted); 115 116 virtual float masterVolume() const; 117 virtual float masterVolumeSW() const; 118 virtual bool masterMute() const; 119 120 virtual status_t setStreamVolume(audio_stream_type_t stream, float value, 121 audio_io_handle_t output); 122 virtual status_t setStreamMute(audio_stream_type_t stream, bool muted); 123 124 virtual float streamVolume(audio_stream_type_t stream, 125 audio_io_handle_t output) const; 126 virtual bool streamMute(audio_stream_type_t stream) const; 127 128 virtual status_t setMode(audio_mode_t mode); 129 130 virtual status_t setMicMute(bool state); 131 virtual bool getMicMute() const; 132 133 virtual status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs); 134 virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) const; 135 136 virtual void registerClient(const sp<IAudioFlingerClient>& client); 137 138 virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const; 139 140 virtual audio_io_handle_t openOutput(uint32_t *pDevices, 141 uint32_t *pSamplingRate, 142 audio_format_t *pFormat, 143 uint32_t *pChannels, 144 uint32_t *pLatencyMs, 145 audio_policy_output_flags_t flags); 146 147 virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1, 148 audio_io_handle_t output2); 149 150 virtual status_t closeOutput(audio_io_handle_t output); 151 152 virtual status_t suspendOutput(audio_io_handle_t output); 153 154 virtual status_t restoreOutput(audio_io_handle_t output); 155 156 virtual audio_io_handle_t openInput(uint32_t *pDevices, 157 uint32_t *pSamplingRate, 158 audio_format_t *pFormat, 159 uint32_t *pChannels, 160 audio_in_acoustics_t acoustics); 161 162 virtual status_t closeInput(audio_io_handle_t input); 163 164 virtual status_t setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output); 165 166 virtual status_t setVoiceVolume(float volume); 167 168 virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 169 audio_io_handle_t output) const; 170 171 virtual unsigned int getInputFramesLost(audio_io_handle_t ioHandle) const; 172 173 virtual int newAudioSessionId(); 174 175 virtual void acquireAudioSessionId(int audioSession); 176 177 virtual void releaseAudioSessionId(int audioSession); 178 179 virtual status_t queryNumberEffects(uint32_t *numEffects) const; 180 181 virtual status_t queryEffect(uint32_t index, effect_descriptor_t *descriptor) const; 182 183 virtual status_t getEffectDescriptor(const effect_uuid_t *pUuid, 184 effect_descriptor_t *descriptor) const; 185 186 virtual sp<IEffect> createEffect(pid_t pid, 187 effect_descriptor_t *pDesc, 188 const sp<IEffectClient>& effectClient, 189 int32_t priority, 190 audio_io_handle_t io, 191 int sessionId, 192 status_t *status, 193 int *id, 194 int *enabled); 195 196 virtual status_t moveEffects(int sessionId, audio_io_handle_t srcOutput, 197 audio_io_handle_t dstOutput); 198 199 virtual status_t onTransact( 200 uint32_t code, 201 const Parcel& data, 202 Parcel* reply, 203 uint32_t flags); 204 205 // end of IAudioFlinger interface 206 207 class SyncEvent; 208 209 typedef void (*sync_event_callback_t)(const wp<SyncEvent>& event) ; 210 211 class SyncEvent : public RefBase { 212 public: 213 SyncEvent(AudioSystem::sync_event_t type, 214 int triggerSession, 215 int listenerSession, 216 sync_event_callback_t callBack, 217 void *cookie) 218 : mType(type), mTriggerSession(triggerSession), mListenerSession(listenerSession), 219 mCallback(callBack), mCookie(cookie) 220 {} 221 222 virtual ~SyncEvent() {} 223 224 void trigger() { Mutex::Autolock _l(mLock); if (mCallback) mCallback(this); } 225 void cancel() {Mutex::Autolock _l(mLock); mCallback = NULL; } 226 AudioSystem::sync_event_t type() const { return mType; } 227 int triggerSession() const { return mTriggerSession; } 228 int listenerSession() const { return mListenerSession; } 229 void *cookie() const { return mCookie; } 230 231 private: 232 const AudioSystem::sync_event_t mType; 233 const int mTriggerSession; 234 const int mListenerSession; 235 sync_event_callback_t mCallback; 236 void * const mCookie; 237 Mutex mLock; 238 }; 239 240 sp<SyncEvent> createSyncEvent(AudioSystem::sync_event_t type, 241 int triggerSession, 242 int listenerSession, 243 sync_event_callback_t callBack, 244 void *cookie); 245private: 246 audio_mode_t getMode() const { return mMode; } 247 248 bool btNrecIsOff() const { return mBtNrecIsOff; } 249 250 AudioFlinger(); 251 virtual ~AudioFlinger(); 252 253 // call in any IAudioFlinger method that accesses mPrimaryHardwareDev 254 status_t initCheck() const { return mPrimaryHardwareDev == NULL ? NO_INIT : NO_ERROR; } 255 256 // RefBase 257 virtual void onFirstRef(); 258 259 audio_hw_device_t* findSuitableHwDev_l(uint32_t devices); 260 void purgeStaleEffects_l(); 261 262 // standby delay for MIXER and DUPLICATING playback threads is read from property 263 // ro.audio.flinger_standbytime_ms or defaults to kDefaultStandbyTimeInNsecs 264 static nsecs_t mStandbyTimeInNsecs; 265 266 // Internal dump utilites. 267 status_t dumpPermissionDenial(int fd, const Vector<String16>& args); 268 status_t dumpClients(int fd, const Vector<String16>& args); 269 status_t dumpInternals(int fd, const Vector<String16>& args); 270 271 // --- Client --- 272 class Client : public RefBase { 273 public: 274 Client(const sp<AudioFlinger>& audioFlinger, pid_t pid); 275 virtual ~Client(); 276 sp<MemoryDealer> heap() const; 277 pid_t pid() const { return mPid; } 278 sp<AudioFlinger> audioFlinger() const { return mAudioFlinger; } 279 280 bool reserveTimedTrack(); 281 void releaseTimedTrack(); 282 283 private: 284 Client(const Client&); 285 Client& operator = (const Client&); 286 const sp<AudioFlinger> mAudioFlinger; 287 const sp<MemoryDealer> mMemoryDealer; 288 const pid_t mPid; 289 290 Mutex mTimedTrackLock; 291 int mTimedTrackCount; 292 }; 293 294 // --- Notification Client --- 295 class NotificationClient : public IBinder::DeathRecipient { 296 public: 297 NotificationClient(const sp<AudioFlinger>& audioFlinger, 298 const sp<IAudioFlingerClient>& client, 299 pid_t pid); 300 virtual ~NotificationClient(); 301 302 sp<IAudioFlingerClient> audioFlingerClient() const { return mAudioFlingerClient; } 303 304 // IBinder::DeathRecipient 305 virtual void binderDied(const wp<IBinder>& who); 306 307 private: 308 NotificationClient(const NotificationClient&); 309 NotificationClient& operator = (const NotificationClient&); 310 311 const sp<AudioFlinger> mAudioFlinger; 312 const pid_t mPid; 313 const sp<IAudioFlingerClient> mAudioFlingerClient; 314 }; 315 316 class TrackHandle; 317 class RecordHandle; 318 class RecordThread; 319 class PlaybackThread; 320 class MixerThread; 321 class DirectOutputThread; 322 class DuplicatingThread; 323 class Track; 324 class RecordTrack; 325 class EffectModule; 326 class EffectHandle; 327 class EffectChain; 328 struct AudioStreamOut; 329 struct AudioStreamIn; 330 331 class ThreadBase : public Thread { 332 public: 333 334 enum type_t { 335 MIXER, // Thread class is MixerThread 336 DIRECT, // Thread class is DirectOutputThread 337 DUPLICATING, // Thread class is DuplicatingThread 338 RECORD // Thread class is RecordThread 339 }; 340 341 ThreadBase (const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, uint32_t device, type_t type); 342 virtual ~ThreadBase(); 343 344 status_t dumpBase(int fd, const Vector<String16>& args); 345 status_t dumpEffectChains(int fd, const Vector<String16>& args); 346 347 void clearPowerManager(); 348 349 // base for record and playback 350 class TrackBase : public AudioBufferProvider, public RefBase { 351 352 public: 353 enum track_state { 354 IDLE, 355 TERMINATED, 356 // These are order-sensitive; do not change order without reviewing the impact. 357 // In particular there are assumptions about > STOPPED. 358 STOPPED, 359 RESUMING, 360 ACTIVE, 361 PAUSING, 362 PAUSED 363 }; 364 365 TrackBase(ThreadBase *thread, 366 const sp<Client>& client, 367 uint32_t sampleRate, 368 audio_format_t format, 369 uint32_t channelMask, 370 int frameCount, 371 const sp<IMemory>& sharedBuffer, 372 int sessionId); 373 virtual ~TrackBase(); 374 375 virtual status_t start(pid_t tid, 376 AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 377 int triggerSession = 0) = 0; 378 virtual void stop() = 0; 379 sp<IMemory> getCblk() const { return mCblkMemory; } 380 audio_track_cblk_t* cblk() const { return mCblk; } 381 int sessionId() const { return mSessionId; } 382 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 383 384 protected: 385 TrackBase(const TrackBase&); 386 TrackBase& operator = (const TrackBase&); 387 388 // AudioBufferProvider interface 389 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) = 0; 390 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 391 392 audio_format_t format() const { 393 return mFormat; 394 } 395 396 int channelCount() const { return mChannelCount; } 397 398 uint32_t channelMask() const { return mChannelMask; } 399 400 int sampleRate() const; // FIXME inline after cblk sr moved 401 402 void* getBuffer(uint32_t offset, uint32_t frames) const; 403 404 bool isStopped() const { 405 return mState == STOPPED; 406 } 407 408 bool isTerminated() const { 409 return mState == TERMINATED; 410 } 411 412 bool step(); 413 void reset(); 414 415 const wp<ThreadBase> mThread; 416 /*const*/ sp<Client> mClient; // see explanation at ~TrackBase() why not const 417 sp<IMemory> mCblkMemory; 418 audio_track_cblk_t* mCblk; 419 void* mBuffer; 420 void* mBufferEnd; 421 uint32_t mFrameCount; 422 // we don't really need a lock for these 423 track_state mState; 424 const audio_format_t mFormat; 425 bool mStepServerFailed; 426 const int mSessionId; 427 uint8_t mChannelCount; 428 uint32_t mChannelMask; 429 Vector < sp<SyncEvent> >mSyncEvents; 430 }; 431 432 class ConfigEvent { 433 public: 434 ConfigEvent() : mEvent(0), mParam(0) {} 435 436 int mEvent; 437 int mParam; 438 }; 439 440 class PMDeathRecipient : public IBinder::DeathRecipient { 441 public: 442 PMDeathRecipient(const wp<ThreadBase>& thread) : mThread(thread) {} 443 virtual ~PMDeathRecipient() {} 444 445 // IBinder::DeathRecipient 446 virtual void binderDied(const wp<IBinder>& who); 447 448 private: 449 PMDeathRecipient(const PMDeathRecipient&); 450 PMDeathRecipient& operator = (const PMDeathRecipient&); 451 452 wp<ThreadBase> mThread; 453 }; 454 455 virtual status_t initCheck() const = 0; 456 type_t type() const { return mType; } 457 uint32_t sampleRate() const { return mSampleRate; } 458 int channelCount() const { return mChannelCount; } 459 audio_format_t format() const { return mFormat; } 460 size_t frameCount() const { return mFrameCount; } 461 void wakeUp() { mWaitWorkCV.broadcast(); } 462 // Should be "virtual status_t requestExitAndWait()" and override same 463 // method in Thread, but Thread::requestExitAndWait() is not yet virtual. 464 void exit(); 465 virtual bool checkForNewParameters_l() = 0; 466 virtual status_t setParameters(const String8& keyValuePairs); 467 virtual String8 getParameters(const String8& keys) = 0; 468 virtual void audioConfigChanged_l(int event, int param = 0) = 0; 469 void sendConfigEvent(int event, int param = 0); 470 void sendConfigEvent_l(int event, int param = 0); 471 void processConfigEvents(); 472 audio_io_handle_t id() const { return mId;} 473 bool standby() const { return mStandby; } 474 uint32_t device() const { return mDevice; } 475 virtual audio_stream_t* stream() const = 0; 476 477 sp<EffectHandle> createEffect_l( 478 const sp<AudioFlinger::Client>& client, 479 const sp<IEffectClient>& effectClient, 480 int32_t priority, 481 int sessionId, 482 effect_descriptor_t *desc, 483 int *enabled, 484 status_t *status); 485 void disconnectEffect(const sp< EffectModule>& effect, 486 const wp<EffectHandle>& handle, 487 bool unpinIfLast); 488 489 // return values for hasAudioSession (bit field) 490 enum effect_state { 491 EFFECT_SESSION = 0x1, // the audio session corresponds to at least one 492 // effect 493 TRACK_SESSION = 0x2 // the audio session corresponds to at least one 494 // track 495 }; 496 497 // get effect chain corresponding to session Id. 498 sp<EffectChain> getEffectChain(int sessionId); 499 // same as getEffectChain() but must be called with ThreadBase mutex locked 500 sp<EffectChain> getEffectChain_l(int sessionId); 501 // add an effect chain to the chain list (mEffectChains) 502 virtual status_t addEffectChain_l(const sp<EffectChain>& chain) = 0; 503 // remove an effect chain from the chain list (mEffectChains) 504 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain) = 0; 505 // lock all effect chains Mutexes. Must be called before releasing the 506 // ThreadBase mutex before processing the mixer and effects. This guarantees the 507 // integrity of the chains during the process. 508 // Also sets the parameter 'effectChains' to current value of mEffectChains. 509 void lockEffectChains_l(Vector< sp<EffectChain> >& effectChains); 510 // unlock effect chains after process 511 void unlockEffectChains(const Vector< sp<EffectChain> >& effectChains); 512 // set audio mode to all effect chains 513 void setMode(audio_mode_t mode); 514 // get effect module with corresponding ID on specified audio session 515 sp<AudioFlinger::EffectModule> getEffect_l(int sessionId, int effectId); 516 // add and effect module. Also creates the effect chain is none exists for 517 // the effects audio session 518 status_t addEffect_l(const sp< EffectModule>& effect); 519 // remove and effect module. Also removes the effect chain is this was the last 520 // effect 521 void removeEffect_l(const sp< EffectModule>& effect); 522 // detach all tracks connected to an auxiliary effect 523 virtual void detachAuxEffect_l(int effectId) {} 524 // returns either EFFECT_SESSION if effects on this audio session exist in one 525 // chain, or TRACK_SESSION if tracks on this audio session exist, or both 526 virtual uint32_t hasAudioSession(int sessionId) = 0; 527 // the value returned by default implementation is not important as the 528 // strategy is only meaningful for PlaybackThread which implements this method 529 virtual uint32_t getStrategyForSession_l(int sessionId) { return 0; } 530 531 // suspend or restore effect according to the type of effect passed. a NULL 532 // type pointer means suspend all effects in the session 533 void setEffectSuspended(const effect_uuid_t *type, 534 bool suspend, 535 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 536 // check if some effects must be suspended/restored when an effect is enabled 537 // or disabled 538 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 539 bool enabled, 540 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 541 void checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 542 bool enabled, 543 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 544 545 virtual status_t setSyncEvent(const sp<SyncEvent>& event) = 0; 546 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) = 0; 547 548 549 mutable Mutex mLock; 550 551 protected: 552 553 // entry describing an effect being suspended in mSuspendedSessions keyed vector 554 class SuspendedSessionDesc : public RefBase { 555 public: 556 SuspendedSessionDesc() : mRefCount(0) {} 557 558 int mRefCount; // number of active suspend requests 559 effect_uuid_t mType; // effect type UUID 560 }; 561 562 void acquireWakeLock(); 563 void acquireWakeLock_l(); 564 void releaseWakeLock(); 565 void releaseWakeLock_l(); 566 void setEffectSuspended_l(const effect_uuid_t *type, 567 bool suspend, 568 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 569 // updated mSuspendedSessions when an effect suspended or restored 570 void updateSuspendedSessions_l(const effect_uuid_t *type, 571 bool suspend, 572 int sessionId); 573 // check if some effects must be suspended when an effect chain is added 574 void checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain); 575 576 friend class AudioFlinger; // for mEffectChains 577 578 const type_t mType; 579 580 // Used by parameters, config events, addTrack_l, exit 581 Condition mWaitWorkCV; 582 583 const sp<AudioFlinger> mAudioFlinger; 584 uint32_t mSampleRate; 585 size_t mFrameCount; 586 uint32_t mChannelMask; 587 uint16_t mChannelCount; 588 size_t mFrameSize; 589 audio_format_t mFormat; 590 591 // Parameter sequence by client: binder thread calling setParameters(): 592 // 1. Lock mLock 593 // 2. Append to mNewParameters 594 // 3. mWaitWorkCV.signal 595 // 4. mParamCond.waitRelative with timeout 596 // 5. read mParamStatus 597 // 6. mWaitWorkCV.signal 598 // 7. Unlock 599 // 600 // Parameter sequence by server: threadLoop calling checkForNewParameters_l(): 601 // 1. Lock mLock 602 // 2. If there is an entry in mNewParameters proceed ... 603 // 2. Read first entry in mNewParameters 604 // 3. Process 605 // 4. Remove first entry from mNewParameters 606 // 5. Set mParamStatus 607 // 6. mParamCond.signal 608 // 7. mWaitWorkCV.wait with timeout (this is to avoid overwriting mParamStatus) 609 // 8. Unlock 610 Condition mParamCond; 611 Vector<String8> mNewParameters; 612 status_t mParamStatus; 613 614 Vector<ConfigEvent> mConfigEvents; 615 bool mStandby; 616 const audio_io_handle_t mId; 617 Vector< sp<EffectChain> > mEffectChains; 618 uint32_t mDevice; // output device for PlaybackThread 619 // input + output devices for RecordThread 620 static const int kNameLength = 16; // prctl(PR_SET_NAME) limit 621 char mName[kNameLength]; 622 sp<IPowerManager> mPowerManager; 623 sp<IBinder> mWakeLockToken; 624 const sp<PMDeathRecipient> mDeathRecipient; 625 // list of suspended effects per session and per type. The first vector is 626 // keyed by session ID, the second by type UUID timeLow field 627 KeyedVector< int, KeyedVector< int, sp<SuspendedSessionDesc> > > mSuspendedSessions; 628 }; 629 630 struct stream_type_t { 631 stream_type_t() 632 : volume(1.0f), 633 mute(false), 634 valid(true) 635 { 636 } 637 float volume; 638 bool mute; 639 bool valid; 640 }; 641 642 // --- PlaybackThread --- 643 class PlaybackThread : public ThreadBase { 644 public: 645 646 enum mixer_state { 647 MIXER_IDLE, // no active tracks 648 MIXER_TRACKS_ENABLED, // at least one active track, but no track has any data ready 649 MIXER_TRACKS_READY // at least one active track, and at least one track has data 650 // standby mode does not have an enum value 651 // suspend by audio policy manager is orthogonal to mixer state 652 }; 653 654 // playback track 655 class Track : public TrackBase { 656 public: 657 Track( PlaybackThread *thread, 658 const sp<Client>& client, 659 audio_stream_type_t streamType, 660 uint32_t sampleRate, 661 audio_format_t format, 662 uint32_t channelMask, 663 int frameCount, 664 const sp<IMemory>& sharedBuffer, 665 int sessionId); 666 virtual ~Track(); 667 668 void dump(char* buffer, size_t size); 669 virtual status_t start(pid_t tid, 670 AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 671 int triggerSession = 0); 672 virtual void stop(); 673 void pause(); 674 675 void flush(); 676 void destroy(); 677 void mute(bool); 678 int name() const { 679 return mName; 680 } 681 682 audio_stream_type_t streamType() const { 683 return mStreamType; 684 } 685 status_t attachAuxEffect(int EffectId); 686 void setAuxBuffer(int EffectId, int32_t *buffer); 687 int32_t *auxBuffer() const { return mAuxBuffer; } 688 void setMainBuffer(int16_t *buffer) { mMainBuffer = buffer; } 689 int16_t *mainBuffer() const { return mMainBuffer; } 690 int auxEffectId() const { return mAuxEffectId; } 691 692 protected: 693 // for numerous 694 friend class PlaybackThread; 695 friend class MixerThread; 696 friend class DirectOutputThread; 697 698 Track(const Track&); 699 Track& operator = (const Track&); 700 701 // AudioBufferProvider interface 702 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts = kInvalidPTS); 703 // releaseBuffer() not overridden 704 705 virtual uint32_t framesReady() const; 706 707 bool isMuted() const { return mMute; } 708 bool isPausing() const { 709 return mState == PAUSING; 710 } 711 bool isPaused() const { 712 return mState == PAUSED; 713 } 714 bool isReady() const; 715 void setPaused() { mState = PAUSED; } 716 void reset(); 717 718 bool isOutputTrack() const { 719 return (mStreamType == AUDIO_STREAM_CNT); 720 } 721 722 bool presentationComplete(size_t framesWritten, size_t audioHalFrames); 723 void triggerEvents(AudioSystem::sync_event_t type); 724 725 public: 726 virtual bool isTimedTrack() const { return false; } 727 protected: 728 729 // we don't really need a lock for these 730 volatile bool mMute; 731 // FILLED state is used for suppressing volume ramp at begin of playing 732 enum {FS_FILLING, FS_FILLED, FS_ACTIVE}; 733 mutable uint8_t mFillingUpStatus; 734 int8_t mRetryCount; 735 const sp<IMemory> mSharedBuffer; 736 bool mResetDone; 737 const audio_stream_type_t mStreamType; 738 int mName; 739 int16_t *mMainBuffer; 740 int32_t *mAuxBuffer; 741 int mAuxEffectId; 742 bool mHasVolumeController; 743 size_t mPresentationCompleteFrames; // number of frames written to the audio HAL 744 // when this track will be fully rendered 745 }; // end of Track 746 747 class TimedTrack : public Track { 748 public: 749 static sp<TimedTrack> create(PlaybackThread *thread, 750 const sp<Client>& client, 751 audio_stream_type_t streamType, 752 uint32_t sampleRate, 753 audio_format_t format, 754 uint32_t channelMask, 755 int frameCount, 756 const sp<IMemory>& sharedBuffer, 757 int sessionId); 758 ~TimedTrack(); 759 760 class TimedBuffer { 761 public: 762 TimedBuffer(); 763 TimedBuffer(const sp<IMemory>& buffer, int64_t pts); 764 const sp<IMemory>& buffer() const { return mBuffer; } 765 int64_t pts() const { return mPTS; } 766 int position() const { return mPosition; } 767 void setPosition(int pos) { mPosition = pos; } 768 private: 769 sp<IMemory> mBuffer; 770 int64_t mPTS; 771 int mPosition; 772 }; 773 774 virtual bool isTimedTrack() const { return true; } 775 776 virtual uint32_t framesReady() const; 777 778 // AudioBufferProvider interface 779 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts); 780 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 781 782 void timedYieldSamples(AudioBufferProvider::Buffer* buffer); 783 void timedYieldSilence(uint32_t numFrames, 784 AudioBufferProvider::Buffer* buffer); 785 786 status_t allocateTimedBuffer(size_t size, 787 sp<IMemory>* buffer); 788 status_t queueTimedBuffer(const sp<IMemory>& buffer, 789 int64_t pts); 790 status_t setMediaTimeTransform(const LinearTransform& xform, 791 TimedAudioTrack::TargetTimeline target); 792 void trimTimedBufferQueue_l(); 793 794 private: 795 TimedTrack(PlaybackThread *thread, 796 const sp<Client>& client, 797 audio_stream_type_t streamType, 798 uint32_t sampleRate, 799 audio_format_t format, 800 uint32_t channelMask, 801 int frameCount, 802 const sp<IMemory>& sharedBuffer, 803 int sessionId); 804 805 uint64_t mLocalTimeFreq; 806 LinearTransform mLocalTimeToSampleTransform; 807 sp<MemoryDealer> mTimedMemoryDealer; 808 Vector<TimedBuffer> mTimedBufferQueue; 809 uint8_t* mTimedSilenceBuffer; 810 uint32_t mTimedSilenceBufferSize; 811 mutable Mutex mTimedBufferQueueLock; 812 bool mTimedAudioOutputOnTime; 813 CCHelper mCCHelper; 814 815 Mutex mMediaTimeTransformLock; 816 LinearTransform mMediaTimeTransform; 817 bool mMediaTimeTransformValid; 818 TimedAudioTrack::TargetTimeline mMediaTimeTransformTarget; 819 }; 820 821 822 // playback track 823 class OutputTrack : public Track { 824 public: 825 826 class Buffer: public AudioBufferProvider::Buffer { 827 public: 828 int16_t *mBuffer; 829 }; 830 831 OutputTrack(PlaybackThread *thread, 832 DuplicatingThread *sourceThread, 833 uint32_t sampleRate, 834 audio_format_t format, 835 uint32_t channelMask, 836 int frameCount); 837 virtual ~OutputTrack(); 838 839 virtual status_t start(pid_t tid, 840 AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 841 int triggerSession = 0); 842 virtual void stop(); 843 bool write(int16_t* data, uint32_t frames); 844 bool bufferQueueEmpty() const { return mBufferQueue.size() == 0; } 845 bool isActive() const { return mActive; } 846 const wp<ThreadBase>& thread() const { return mThread; } 847 848 private: 849 850 enum { 851 NO_MORE_BUFFERS = 0x80000001, // same in AudioTrack.h, ok to be different value 852 }; 853 854 status_t obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs); 855 void clearBufferQueue(); 856 857 // Maximum number of pending buffers allocated by OutputTrack::write() 858 static const uint8_t kMaxOverFlowBuffers = 10; 859 860 Vector < Buffer* > mBufferQueue; 861 AudioBufferProvider::Buffer mOutBuffer; 862 bool mActive; 863 DuplicatingThread* const mSourceThread; // for waitTimeMs() in write() 864 }; // end of OutputTrack 865 866 PlaybackThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 867 audio_io_handle_t id, uint32_t device, type_t type); 868 virtual ~PlaybackThread(); 869 870 status_t dump(int fd, const Vector<String16>& args); 871 872 // Thread virtuals 873 virtual status_t readyToRun(); 874 virtual bool threadLoop(); 875 876 // RefBase 877 virtual void onFirstRef(); 878 879protected: 880 // Code snippets that were lifted up out of threadLoop() 881 virtual void threadLoop_mix() = 0; 882 virtual void threadLoop_sleepTime() = 0; 883 virtual void threadLoop_write(); 884 virtual void threadLoop_standby(); 885 886 // prepareTracks_l reads and writes mActiveTracks, and also returns the 887 // pending set of tracks to remove via Vector 'tracksToRemove'. The caller is 888 // responsible for clearing or destroying this Vector later on, when it 889 // is safe to do so. That will drop the final ref count and destroy the tracks. 890 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove) = 0; 891 892public: 893 894 virtual status_t initCheck() const { return (mOutput == NULL) ? NO_INIT : NO_ERROR; } 895 896 // return estimated latency in milliseconds, as reported by HAL 897 uint32_t latency() const; 898 899 void setMasterVolume(float value); 900 void setMasterMute(bool muted); 901 902 void setStreamVolume(audio_stream_type_t stream, float value); 903 void setStreamMute(audio_stream_type_t stream, bool muted); 904 905 float streamVolume(audio_stream_type_t stream) const; 906 907 sp<Track> createTrack_l( 908 const sp<AudioFlinger::Client>& client, 909 audio_stream_type_t streamType, 910 uint32_t sampleRate, 911 audio_format_t format, 912 uint32_t channelMask, 913 int frameCount, 914 const sp<IMemory>& sharedBuffer, 915 int sessionId, 916 bool isTimed, 917 status_t *status); 918 919 AudioStreamOut* getOutput() const; 920 AudioStreamOut* clearOutput(); 921 virtual audio_stream_t* stream() const; 922 923 void suspend() { mSuspended++; } 924 void restore() { if (mSuspended > 0) mSuspended--; } 925 bool isSuspended() const { return (mSuspended > 0); } 926 virtual String8 getParameters(const String8& keys); 927 virtual void audioConfigChanged_l(int event, int param = 0); 928 status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames); 929 int16_t *mixBuffer() const { return mMixBuffer; }; 930 931 virtual void detachAuxEffect_l(int effectId); 932 status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track> track, 933 int EffectId); 934 status_t attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track> track, 935 int EffectId); 936 937 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 938 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 939 virtual uint32_t hasAudioSession(int sessionId); 940 virtual uint32_t getStrategyForSession_l(int sessionId); 941 942 void setStreamValid(audio_stream_type_t streamType, bool valid); 943 944 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 945 virtual bool isValidSyncEvent(const sp<SyncEvent>& event); 946 947 protected: 948 int16_t* mMixBuffer; 949 uint32_t mSuspended; // suspend count, > 0 means suspended 950 int mBytesWritten; 951 private: 952 // mMasterMute is in both PlaybackThread and in AudioFlinger. When a 953 // PlaybackThread needs to find out if master-muted, it checks it's local 954 // copy rather than the one in AudioFlinger. This optimization saves a lock. 955 bool mMasterMute; 956 void setMasterMute_l(bool muted) { mMasterMute = muted; } 957 protected: 958 SortedVector< wp<Track> > mActiveTracks; 959 960 // Allocate a track name. Returns name >= 0 if successful, -1 on failure. 961 virtual int getTrackName_l() = 0; 962 virtual void deleteTrackName_l(int name) = 0; 963 964 // Time to sleep between cycles when: 965 virtual uint32_t activeSleepTimeUs() const; // mixer state MIXER_TRACKS_ENABLED 966 virtual uint32_t idleSleepTimeUs() const = 0; // mixer state MIXER_IDLE 967 virtual uint32_t suspendSleepTimeUs() const = 0; // audio policy manager suspended us 968 // No sleep when mixer state == MIXER_TRACKS_READY; relies on audio HAL stream->write() 969 // No sleep in standby mode; waits on a condition 970 971 // Code snippets that are temporarily lifted up out of threadLoop() until the merge 972 void checkSilentMode_l(); 973 974 // Non-trivial for DUPLICATING only 975 virtual void saveOutputTracks() { } 976 virtual void clearOutputTracks() { } 977 978 // Cache various calculated values, at threadLoop() entry and after a parameter change 979 virtual void cacheParameters_l(); 980 981 private: 982 983 friend class AudioFlinger; // for numerous 984 985 PlaybackThread(const Client&); 986 PlaybackThread& operator = (const PlaybackThread&); 987 988 status_t addTrack_l(const sp<Track>& track); 989 void destroyTrack_l(const sp<Track>& track); 990 void removeTrack_l(const sp<Track>& track); 991 992 void readOutputParameters(); 993 994 virtual status_t dumpInternals(int fd, const Vector<String16>& args); 995 status_t dumpTracks(int fd, const Vector<String16>& args); 996 997 SortedVector< sp<Track> > mTracks; 998 // mStreamTypes[] uses 1 additional stream type internally for the OutputTrack used by DuplicatingThread 999 stream_type_t mStreamTypes[AUDIO_STREAM_CNT + 1]; 1000 AudioStreamOut *mOutput; 1001 float mMasterVolume; 1002 nsecs_t mLastWriteTime; 1003 int mNumWrites; 1004 int mNumDelayedWrites; 1005 bool mInWrite; 1006 1007 // FIXME rename these former local variables of threadLoop to standard "m" names 1008 nsecs_t standbyTime; 1009 size_t mixBufferSize; 1010 1011 // cached copies of activeSleepTimeUs() and idleSleepTimeUs() made by cacheParameters_l() 1012 uint32_t activeSleepTime; 1013 uint32_t idleSleepTime; 1014 1015 uint32_t sleepTime; 1016 1017 // mixer status returned by prepareTracks_l() 1018 mixer_state mMixerStatus; // current cycle 1019 mixer_state mPrevMixerStatus; // previous cycle 1020 1021 // FIXME move these declarations into the specific sub-class that needs them 1022 // MIXER only 1023 bool longStandbyExit; 1024 uint32_t sleepTimeShift; 1025 1026 // same as AudioFlinger::mStandbyTimeInNsecs except for DIRECT which uses a shorter value 1027 nsecs_t standbyDelay; 1028 1029 // MIXER only 1030 nsecs_t maxPeriod; 1031 1032 // DUPLICATING only 1033 uint32_t writeFrames; 1034 }; 1035 1036 class MixerThread : public PlaybackThread { 1037 public: 1038 MixerThread (const sp<AudioFlinger>& audioFlinger, 1039 AudioStreamOut* output, 1040 audio_io_handle_t id, 1041 uint32_t device, 1042 type_t type = MIXER); 1043 virtual ~MixerThread(); 1044 1045 // Thread virtuals 1046 1047 void invalidateTracks(audio_stream_type_t streamType); 1048 virtual bool checkForNewParameters_l(); 1049 virtual status_t dumpInternals(int fd, const Vector<String16>& args); 1050 1051 protected: 1052 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1053 virtual int getTrackName_l(); 1054 virtual void deleteTrackName_l(int name); 1055 virtual uint32_t idleSleepTimeUs() const; 1056 virtual uint32_t suspendSleepTimeUs() const; 1057 virtual void cacheParameters_l(); 1058 1059 // threadLoop snippets 1060 virtual void threadLoop_mix(); 1061 virtual void threadLoop_sleepTime(); 1062 1063 AudioMixer* mAudioMixer; 1064 }; 1065 1066 class DirectOutputThread : public PlaybackThread { 1067 public: 1068 1069 DirectOutputThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1070 audio_io_handle_t id, uint32_t device); 1071 virtual ~DirectOutputThread(); 1072 1073 // Thread virtuals 1074 1075 virtual bool checkForNewParameters_l(); 1076 1077 protected: 1078 virtual int getTrackName_l(); 1079 virtual void deleteTrackName_l(int name); 1080 virtual uint32_t activeSleepTimeUs() const; 1081 virtual uint32_t idleSleepTimeUs() const; 1082 virtual uint32_t suspendSleepTimeUs() const; 1083 virtual void cacheParameters_l(); 1084 1085 // threadLoop snippets 1086 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1087 virtual void threadLoop_mix(); 1088 virtual void threadLoop_sleepTime(); 1089 1090 // volumes last sent to audio HAL with stream->set_volume() 1091 // FIXME use standard representation and names 1092 float mLeftVolFloat; 1093 float mRightVolFloat; 1094 uint16_t mLeftVolShort; 1095 uint16_t mRightVolShort; 1096 1097 // FIXME rename these former local variables of threadLoop to standard names 1098 // next 3 were local to the while !exitingPending loop 1099 bool rampVolume; 1100 uint16_t leftVol; 1101 uint16_t rightVol; 1102 1103private: 1104 // prepareTracks_l() tells threadLoop_mix() the name of the single active track 1105 sp<Track> mActiveTrack; 1106 }; 1107 1108 class DuplicatingThread : public MixerThread { 1109 public: 1110 DuplicatingThread (const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread, 1111 audio_io_handle_t id); 1112 virtual ~DuplicatingThread(); 1113 1114 // Thread virtuals 1115 void addOutputTrack(MixerThread* thread); 1116 void removeOutputTrack(MixerThread* thread); 1117 uint32_t waitTimeMs() const { return mWaitTimeMs; } 1118 protected: 1119 virtual uint32_t activeSleepTimeUs() const; 1120 1121 private: 1122 bool outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks); 1123 protected: 1124 // threadLoop snippets 1125 virtual void threadLoop_mix(); 1126 virtual void threadLoop_sleepTime(); 1127 virtual void threadLoop_write(); 1128 virtual void threadLoop_standby(); 1129 virtual void cacheParameters_l(); 1130 1131 private: 1132 // called from threadLoop, addOutputTrack, removeOutputTrack 1133 virtual void updateWaitTime_l(); 1134 protected: 1135 virtual void saveOutputTracks(); 1136 virtual void clearOutputTracks(); 1137 private: 1138 1139 uint32_t mWaitTimeMs; 1140 SortedVector < sp<OutputTrack> > outputTracks; 1141 SortedVector < sp<OutputTrack> > mOutputTracks; 1142 }; 1143 1144 PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const; 1145 MixerThread *checkMixerThread_l(audio_io_handle_t output) const; 1146 RecordThread *checkRecordThread_l(audio_io_handle_t input) const; 1147 // no range check, AudioFlinger::mLock held 1148 bool streamMute_l(audio_stream_type_t stream) const 1149 { return mStreamTypes[stream].mute; } 1150 // no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held 1151 float streamVolume_l(audio_stream_type_t stream) const 1152 { return mStreamTypes[stream].volume; } 1153 void audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2); 1154 1155 // allocate an audio_io_handle_t, session ID, or effect ID 1156 uint32_t nextUniqueId(); 1157 1158 status_t moveEffectChain_l(int sessionId, 1159 PlaybackThread *srcThread, 1160 PlaybackThread *dstThread, 1161 bool reRegister); 1162 // return thread associated with primary hardware device, or NULL 1163 PlaybackThread *primaryPlaybackThread_l() const; 1164 uint32_t primaryOutputDevice_l() const; 1165 1166 // server side of the client's IAudioTrack 1167 class TrackHandle : public android::BnAudioTrack { 1168 public: 1169 TrackHandle(const sp<PlaybackThread::Track>& track); 1170 virtual ~TrackHandle(); 1171 virtual sp<IMemory> getCblk() const; 1172 virtual status_t start(pid_t tid); 1173 virtual void stop(); 1174 virtual void flush(); 1175 virtual void mute(bool); 1176 virtual void pause(); 1177 virtual status_t attachAuxEffect(int effectId); 1178 virtual status_t allocateTimedBuffer(size_t size, 1179 sp<IMemory>* buffer); 1180 virtual status_t queueTimedBuffer(const sp<IMemory>& buffer, 1181 int64_t pts); 1182 virtual status_t setMediaTimeTransform(const LinearTransform& xform, 1183 int target); 1184 virtual status_t onTransact( 1185 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 1186 private: 1187 const sp<PlaybackThread::Track> mTrack; 1188 }; 1189 1190 void removeClient_l(pid_t pid); 1191 void removeNotificationClient(pid_t pid); 1192 1193 1194 // record thread 1195 class RecordThread : public ThreadBase, public AudioBufferProvider 1196 { 1197 public: 1198 1199 // record track 1200 class RecordTrack : public TrackBase { 1201 public: 1202 RecordTrack(RecordThread *thread, 1203 const sp<Client>& client, 1204 uint32_t sampleRate, 1205 audio_format_t format, 1206 uint32_t channelMask, 1207 int frameCount, 1208 int sessionId); 1209 virtual ~RecordTrack(); 1210 1211 virtual status_t start(pid_t tid, 1212 AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 1213 int triggerSession = 0); 1214 virtual void stop(); 1215 1216 bool overflow() { bool tmp = mOverflow; mOverflow = false; return tmp; } 1217 bool setOverflow() { bool tmp = mOverflow; mOverflow = true; return tmp; } 1218 1219 void dump(char* buffer, size_t size); 1220 1221 private: 1222 friend class AudioFlinger; // for mState 1223 1224 RecordTrack(const RecordTrack&); 1225 RecordTrack& operator = (const RecordTrack&); 1226 1227 // AudioBufferProvider interface 1228 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts = kInvalidPTS); 1229 // releaseBuffer() not overridden 1230 1231 bool mOverflow; 1232 }; 1233 1234 1235 RecordThread(const sp<AudioFlinger>& audioFlinger, 1236 AudioStreamIn *input, 1237 uint32_t sampleRate, 1238 uint32_t channels, 1239 audio_io_handle_t id, 1240 uint32_t device); 1241 virtual ~RecordThread(); 1242 1243 // Thread 1244 virtual bool threadLoop(); 1245 virtual status_t readyToRun(); 1246 1247 // RefBase 1248 virtual void onFirstRef(); 1249 1250 virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; } 1251 sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l( 1252 const sp<AudioFlinger::Client>& client, 1253 uint32_t sampleRate, 1254 audio_format_t format, 1255 int channelMask, 1256 int frameCount, 1257 int sessionId, 1258 status_t *status); 1259 1260 status_t start(RecordTrack* recordTrack, pid_t tid, 1261 AudioSystem::sync_event_t event, 1262 int triggerSession); 1263 void stop(RecordTrack* recordTrack); 1264 status_t dump(int fd, const Vector<String16>& args); 1265 AudioStreamIn* getInput() const; 1266 AudioStreamIn* clearInput(); 1267 virtual audio_stream_t* stream() const; 1268 1269 // AudioBufferProvider interface 1270 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts); 1271 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 1272 1273 virtual bool checkForNewParameters_l(); 1274 virtual String8 getParameters(const String8& keys); 1275 virtual void audioConfigChanged_l(int event, int param = 0); 1276 void readInputParameters(); 1277 virtual unsigned int getInputFramesLost(); 1278 1279 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1280 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1281 virtual uint32_t hasAudioSession(int sessionId); 1282 RecordTrack* track(); 1283 1284 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 1285 virtual bool isValidSyncEvent(const sp<SyncEvent>& event); 1286 1287 static void syncStartEventCallback(const wp<SyncEvent>& event); 1288 void handleSyncStartEvent(const sp<SyncEvent>& event); 1289 1290 private: 1291 void clearSyncStartEvent(); 1292 1293 RecordThread(); 1294 AudioStreamIn *mInput; 1295 RecordTrack* mTrack; 1296 sp<RecordTrack> mActiveTrack; 1297 Condition mStartStopCond; 1298 AudioResampler *mResampler; 1299 int32_t *mRsmpOutBuffer; 1300 int16_t *mRsmpInBuffer; 1301 size_t mRsmpInIndex; 1302 size_t mInputBytes; 1303 const int mReqChannelCount; 1304 const uint32_t mReqSampleRate; 1305 ssize_t mBytesRead; 1306 // sync event triggering actual audio capture. Frames read before this event will 1307 // be dropped and therefore not read by the application. 1308 sp<SyncEvent> mSyncStartEvent; 1309 // number of captured frames to drop after the start sync event has been received. 1310 ssize_t mFramestoDrop; 1311 }; 1312 1313 // server side of the client's IAudioRecord 1314 class RecordHandle : public android::BnAudioRecord { 1315 public: 1316 RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack); 1317 virtual ~RecordHandle(); 1318 virtual sp<IMemory> getCblk() const; 1319 virtual status_t start(pid_t tid, int event, int triggerSession); 1320 virtual void stop(); 1321 virtual status_t onTransact( 1322 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 1323 private: 1324 const sp<RecordThread::RecordTrack> mRecordTrack; 1325 }; 1326 1327 //--- Audio Effect Management 1328 1329 // EffectModule and EffectChain classes both have their own mutex to protect 1330 // state changes or resource modifications. Always respect the following order 1331 // if multiple mutexes must be acquired to avoid cross deadlock: 1332 // AudioFlinger -> ThreadBase -> EffectChain -> EffectModule 1333 1334 // The EffectModule class is a wrapper object controlling the effect engine implementation 1335 // in the effect library. It prevents concurrent calls to process() and command() functions 1336 // from different client threads. It keeps a list of EffectHandle objects corresponding 1337 // to all client applications using this effect and notifies applications of effect state, 1338 // control or parameter changes. It manages the activation state machine to send appropriate 1339 // reset, enable, disable commands to effect engine and provide volume 1340 // ramping when effects are activated/deactivated. 1341 // When controlling an auxiliary effect, the EffectModule also provides an input buffer used by 1342 // the attached track(s) to accumulate their auxiliary channel. 1343 class EffectModule: public RefBase { 1344 public: 1345 EffectModule(ThreadBase *thread, 1346 const wp<AudioFlinger::EffectChain>& chain, 1347 effect_descriptor_t *desc, 1348 int id, 1349 int sessionId); 1350 virtual ~EffectModule(); 1351 1352 enum effect_state { 1353 IDLE, 1354 RESTART, 1355 STARTING, 1356 ACTIVE, 1357 STOPPING, 1358 STOPPED, 1359 DESTROYED 1360 }; 1361 1362 int id() const { return mId; } 1363 void process(); 1364 void updateState(); 1365 status_t command(uint32_t cmdCode, 1366 uint32_t cmdSize, 1367 void *pCmdData, 1368 uint32_t *replySize, 1369 void *pReplyData); 1370 1371 void reset_l(); 1372 status_t configure(); 1373 status_t init(); 1374 effect_state state() const { 1375 return mState; 1376 } 1377 uint32_t status() { 1378 return mStatus; 1379 } 1380 int sessionId() const { 1381 return mSessionId; 1382 } 1383 status_t setEnabled(bool enabled); 1384 bool isEnabled() const; 1385 bool isProcessEnabled() const; 1386 1387 void setInBuffer(int16_t *buffer) { mConfig.inputCfg.buffer.s16 = buffer; } 1388 int16_t *inBuffer() { return mConfig.inputCfg.buffer.s16; } 1389 void setOutBuffer(int16_t *buffer) { mConfig.outputCfg.buffer.s16 = buffer; } 1390 int16_t *outBuffer() { return mConfig.outputCfg.buffer.s16; } 1391 void setChain(const wp<EffectChain>& chain) { mChain = chain; } 1392 void setThread(const wp<ThreadBase>& thread) { mThread = thread; } 1393 const wp<ThreadBase>& thread() { return mThread; } 1394 1395 status_t addHandle(const sp<EffectHandle>& handle); 1396 void disconnect(const wp<EffectHandle>& handle, bool unpinIfLast); 1397 size_t removeHandle (const wp<EffectHandle>& handle); 1398 1399 effect_descriptor_t& desc() { return mDescriptor; } 1400 wp<EffectChain>& chain() { return mChain; } 1401 1402 status_t setDevice(uint32_t device); 1403 status_t setVolume(uint32_t *left, uint32_t *right, bool controller); 1404 status_t setMode(audio_mode_t mode); 1405 status_t start(); 1406 status_t stop(); 1407 void setSuspended(bool suspended); 1408 bool suspended() const; 1409 1410 sp<EffectHandle> controlHandle(); 1411 1412 bool isPinned() const { return mPinned; } 1413 void unPin() { mPinned = false; } 1414 1415 status_t dump(int fd, const Vector<String16>& args); 1416 1417 protected: 1418 friend class AudioFlinger; // for mHandles 1419 bool mPinned; 1420 1421 // Maximum time allocated to effect engines to complete the turn off sequence 1422 static const uint32_t MAX_DISABLE_TIME_MS = 10000; 1423 1424 EffectModule(const EffectModule&); 1425 EffectModule& operator = (const EffectModule&); 1426 1427 status_t start_l(); 1428 status_t stop_l(); 1429 1430mutable Mutex mLock; // mutex for process, commands and handles list protection 1431 wp<ThreadBase> mThread; // parent thread 1432 wp<EffectChain> mChain; // parent effect chain 1433 int mId; // this instance unique ID 1434 int mSessionId; // audio session ID 1435 effect_descriptor_t mDescriptor;// effect descriptor received from effect engine 1436 effect_config_t mConfig; // input and output audio configuration 1437 effect_handle_t mEffectInterface; // Effect module C API 1438 status_t mStatus; // initialization status 1439 effect_state mState; // current activation state 1440 Vector< wp<EffectHandle> > mHandles; // list of client handles 1441 // First handle in mHandles has highest priority and controls the effect module 1442 uint32_t mMaxDisableWaitCnt; // maximum grace period before forcing an effect off after 1443 // sending disable command. 1444 uint32_t mDisableWaitCnt; // current process() calls count during disable period. 1445 bool mSuspended; // effect is suspended: temporarily disabled by framework 1446 }; 1447 1448 // The EffectHandle class implements the IEffect interface. It provides resources 1449 // to receive parameter updates, keeps track of effect control 1450 // ownership and state and has a pointer to the EffectModule object it is controlling. 1451 // There is one EffectHandle object for each application controlling (or using) 1452 // an effect module. 1453 // The EffectHandle is obtained by calling AudioFlinger::createEffect(). 1454 class EffectHandle: public android::BnEffect { 1455 public: 1456 1457 EffectHandle(const sp<EffectModule>& effect, 1458 const sp<AudioFlinger::Client>& client, 1459 const sp<IEffectClient>& effectClient, 1460 int32_t priority); 1461 virtual ~EffectHandle(); 1462 1463 // IEffect 1464 virtual status_t enable(); 1465 virtual status_t disable(); 1466 virtual status_t command(uint32_t cmdCode, 1467 uint32_t cmdSize, 1468 void *pCmdData, 1469 uint32_t *replySize, 1470 void *pReplyData); 1471 virtual void disconnect(); 1472 private: 1473 void disconnect(bool unpinIfLast); 1474 public: 1475 virtual sp<IMemory> getCblk() const { return mCblkMemory; } 1476 virtual status_t onTransact(uint32_t code, const Parcel& data, 1477 Parcel* reply, uint32_t flags); 1478 1479 1480 // Give or take control of effect module 1481 // - hasControl: true if control is given, false if removed 1482 // - signal: true client app should be signaled of change, false otherwise 1483 // - enabled: state of the effect when control is passed 1484 void setControl(bool hasControl, bool signal, bool enabled); 1485 void commandExecuted(uint32_t cmdCode, 1486 uint32_t cmdSize, 1487 void *pCmdData, 1488 uint32_t replySize, 1489 void *pReplyData); 1490 void setEnabled(bool enabled); 1491 bool enabled() const { return mEnabled; } 1492 1493 // Getters 1494 int id() const { return mEffect->id(); } 1495 int priority() const { return mPriority; } 1496 bool hasControl() const { return mHasControl; } 1497 sp<EffectModule> effect() const { return mEffect; } 1498 1499 void dump(char* buffer, size_t size); 1500 1501 protected: 1502 friend class AudioFlinger; // for mEffect, mHasControl, mEnabled 1503 EffectHandle(const EffectHandle&); 1504 EffectHandle& operator =(const EffectHandle&); 1505 1506 sp<EffectModule> mEffect; // pointer to controlled EffectModule 1507 sp<IEffectClient> mEffectClient; // callback interface for client notifications 1508 /*const*/ sp<Client> mClient; // client for shared memory allocation, see disconnect() 1509 sp<IMemory> mCblkMemory; // shared memory for control block 1510 effect_param_cblk_t* mCblk; // control block for deferred parameter setting via shared memory 1511 uint8_t* mBuffer; // pointer to parameter area in shared memory 1512 int mPriority; // client application priority to control the effect 1513 bool mHasControl; // true if this handle is controlling the effect 1514 bool mEnabled; // cached enable state: needed when the effect is 1515 // restored after being suspended 1516 }; 1517 1518 // the EffectChain class represents a group of effects associated to one audio session. 1519 // There can be any number of EffectChain objects per output mixer thread (PlaybackThread). 1520 // The EffecChain with session ID 0 contains global effects applied to the output mix. 1521 // Effects in this chain can be insert or auxiliary. Effects in other chains (attached to tracks) 1522 // are insert only. The EffectChain maintains an ordered list of effect module, the order corresponding 1523 // in the effect process order. When attached to a track (session ID != 0), it also provide it's own 1524 // input buffer used by the track as accumulation buffer. 1525 class EffectChain: public RefBase { 1526 public: 1527 EffectChain(const wp<ThreadBase>& wThread, int sessionId); 1528 EffectChain(ThreadBase *thread, int sessionId); 1529 virtual ~EffectChain(); 1530 1531 // special key used for an entry in mSuspendedEffects keyed vector 1532 // corresponding to a suspend all request. 1533 static const int kKeyForSuspendAll = 0; 1534 1535 // minimum duration during which we force calling effect process when last track on 1536 // a session is stopped or removed to allow effect tail to be rendered 1537 static const int kProcessTailDurationMs = 1000; 1538 1539 void process_l(); 1540 1541 void lock() { 1542 mLock.lock(); 1543 } 1544 void unlock() { 1545 mLock.unlock(); 1546 } 1547 1548 status_t addEffect_l(const sp<EffectModule>& handle); 1549 size_t removeEffect_l(const sp<EffectModule>& handle); 1550 1551 int sessionId() const { return mSessionId; } 1552 void setSessionId(int sessionId) { mSessionId = sessionId; } 1553 1554 sp<EffectModule> getEffectFromDesc_l(effect_descriptor_t *descriptor); 1555 sp<EffectModule> getEffectFromId_l(int id); 1556 sp<EffectModule> getEffectFromType_l(const effect_uuid_t *type); 1557 bool setVolume_l(uint32_t *left, uint32_t *right); 1558 void setDevice_l(uint32_t device); 1559 void setMode_l(audio_mode_t mode); 1560 1561 void setInBuffer(int16_t *buffer, bool ownsBuffer = false) { 1562 mInBuffer = buffer; 1563 mOwnInBuffer = ownsBuffer; 1564 } 1565 int16_t *inBuffer() const { 1566 return mInBuffer; 1567 } 1568 void setOutBuffer(int16_t *buffer) { 1569 mOutBuffer = buffer; 1570 } 1571 int16_t *outBuffer() const { 1572 return mOutBuffer; 1573 } 1574 1575 void incTrackCnt() { android_atomic_inc(&mTrackCnt); } 1576 void decTrackCnt() { android_atomic_dec(&mTrackCnt); } 1577 int32_t trackCnt() const { return mTrackCnt;} 1578 1579 void incActiveTrackCnt() { android_atomic_inc(&mActiveTrackCnt); 1580 mTailBufferCount = mMaxTailBuffers; } 1581 void decActiveTrackCnt() { android_atomic_dec(&mActiveTrackCnt); } 1582 int32_t activeTrackCnt() const { return mActiveTrackCnt;} 1583 1584 uint32_t strategy() const { return mStrategy; } 1585 void setStrategy(uint32_t strategy) 1586 { mStrategy = strategy; } 1587 1588 // suspend effect of the given type 1589 void setEffectSuspended_l(const effect_uuid_t *type, 1590 bool suspend); 1591 // suspend all eligible effects 1592 void setEffectSuspendedAll_l(bool suspend); 1593 // check if effects should be suspend or restored when a given effect is enable or disabled 1594 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1595 bool enabled); 1596 1597 status_t dump(int fd, const Vector<String16>& args); 1598 1599 protected: 1600 friend class AudioFlinger; // for mThread, mEffects 1601 EffectChain(const EffectChain&); 1602 EffectChain& operator =(const EffectChain&); 1603 1604 class SuspendedEffectDesc : public RefBase { 1605 public: 1606 SuspendedEffectDesc() : mRefCount(0) {} 1607 1608 int mRefCount; 1609 effect_uuid_t mType; 1610 wp<EffectModule> mEffect; 1611 }; 1612 1613 // get a list of effect modules to suspend when an effect of the type 1614 // passed is enabled. 1615 void getSuspendEligibleEffects(Vector< sp<EffectModule> > &effects); 1616 1617 // get an effect module if it is currently enable 1618 sp<EffectModule> getEffectIfEnabled(const effect_uuid_t *type); 1619 // true if the effect whose descriptor is passed can be suspended 1620 // OEMs can modify the rules implemented in this method to exclude specific effect 1621 // types or implementations from the suspend/restore mechanism. 1622 bool isEffectEligibleForSuspend(const effect_descriptor_t& desc); 1623 1624 wp<ThreadBase> mThread; // parent mixer thread 1625 Mutex mLock; // mutex protecting effect list 1626 Vector< sp<EffectModule> > mEffects; // list of effect modules 1627 int mSessionId; // audio session ID 1628 int16_t *mInBuffer; // chain input buffer 1629 int16_t *mOutBuffer; // chain output buffer 1630 volatile int32_t mActiveTrackCnt; // number of active tracks connected 1631 volatile int32_t mTrackCnt; // number of tracks connected 1632 int32_t mTailBufferCount; // current effect tail buffer count 1633 int32_t mMaxTailBuffers; // maximum effect tail buffers 1634 bool mOwnInBuffer; // true if the chain owns its input buffer 1635 int mVolumeCtrlIdx; // index of insert effect having control over volume 1636 uint32_t mLeftVolume; // previous volume on left channel 1637 uint32_t mRightVolume; // previous volume on right channel 1638 uint32_t mNewLeftVolume; // new volume on left channel 1639 uint32_t mNewRightVolume; // new volume on right channel 1640 uint32_t mStrategy; // strategy for this effect chain 1641 // mSuspendedEffects lists all effects currently suspended in the chain. 1642 // Use effect type UUID timelow field as key. There is no real risk of identical 1643 // timeLow fields among effect type UUIDs. 1644 // Updated by updateSuspendedSessions_l() only. 1645 KeyedVector< int, sp<SuspendedEffectDesc> > mSuspendedEffects; 1646 }; 1647 1648 // AudioStreamOut and AudioStreamIn are immutable, so their fields are const. 1649 // For emphasis, we could also make all pointers to them be "const *", 1650 // but that would clutter the code unnecessarily. 1651 1652 struct AudioStreamOut { 1653 audio_hw_device_t* const hwDev; 1654 audio_stream_out_t* const stream; 1655 1656 AudioStreamOut(audio_hw_device_t *dev, audio_stream_out_t *out) : 1657 hwDev(dev), stream(out) {} 1658 }; 1659 1660 struct AudioStreamIn { 1661 audio_hw_device_t* const hwDev; 1662 audio_stream_in_t* const stream; 1663 1664 AudioStreamIn(audio_hw_device_t *dev, audio_stream_in_t *in) : 1665 hwDev(dev), stream(in) {} 1666 }; 1667 1668 // for mAudioSessionRefs only 1669 struct AudioSessionRef { 1670 AudioSessionRef(int sessionid, pid_t pid) : 1671 mSessionid(sessionid), mPid(pid), mCnt(1) {} 1672 const int mSessionid; 1673 const pid_t mPid; 1674 int mCnt; 1675 }; 1676 1677 enum master_volume_support { 1678 // MVS_NONE: 1679 // Audio HAL has no support for master volume, either setting or 1680 // getting. All master volume control must be implemented in SW by the 1681 // AudioFlinger mixing core. 1682 MVS_NONE, 1683 1684 // MVS_SETONLY: 1685 // Audio HAL has support for setting master volume, but not for getting 1686 // master volume (original HAL design did not include a getter). 1687 // AudioFlinger needs to keep track of the last set master volume in 1688 // addition to needing to set an initial, default, master volume at HAL 1689 // load time. 1690 MVS_SETONLY, 1691 1692 // MVS_FULL: 1693 // Audio HAL has support both for setting and getting master volume. 1694 // AudioFlinger should send all set and get master volume requests 1695 // directly to the HAL. 1696 MVS_FULL, 1697 }; 1698 1699 mutable Mutex mLock; 1700 1701 DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client() 1702 1703 mutable Mutex mHardwareLock; 1704 1705 // These two fields are immutable after onFirstRef(), so no lock needed to access 1706 audio_hw_device_t* mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL 1707 Vector<audio_hw_device_t*> mAudioHwDevs; 1708 1709 // for dump, indicates which hardware operation is currently in progress (but not stream ops) 1710 enum hardware_call_state { 1711 AUDIO_HW_IDLE = 0, // no operation in progress 1712 AUDIO_HW_INIT, // init_check 1713 AUDIO_HW_OUTPUT_OPEN, // open_output_stream 1714 AUDIO_HW_OUTPUT_CLOSE, // unused 1715 AUDIO_HW_INPUT_OPEN, // unused 1716 AUDIO_HW_INPUT_CLOSE, // unused 1717 AUDIO_HW_STANDBY, // unused 1718 AUDIO_HW_SET_MASTER_VOLUME, // set_master_volume 1719 AUDIO_HW_GET_ROUTING, // unused 1720 AUDIO_HW_SET_ROUTING, // unused 1721 AUDIO_HW_GET_MODE, // unused 1722 AUDIO_HW_SET_MODE, // set_mode 1723 AUDIO_HW_GET_MIC_MUTE, // get_mic_mute 1724 AUDIO_HW_SET_MIC_MUTE, // set_mic_mute 1725 AUDIO_HW_SET_VOICE_VOLUME, // set_voice_volume 1726 AUDIO_HW_SET_PARAMETER, // set_parameters 1727 AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size 1728 AUDIO_HW_GET_MASTER_VOLUME, // get_master_volume 1729 AUDIO_HW_GET_PARAMETER, // get_parameters 1730 }; 1731 1732 mutable hardware_call_state mHardwareStatus; // for dump only 1733 1734 1735 DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads; 1736 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 1737 1738 // both are protected by mLock 1739 float mMasterVolume; 1740 float mMasterVolumeSW; 1741 master_volume_support mMasterVolumeSupportLvl; 1742 bool mMasterMute; 1743 1744 DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads; 1745 1746 DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients; 1747 volatile int32_t mNextUniqueId; // updated by android_atomic_inc 1748 audio_mode_t mMode; 1749 bool mBtNrecIsOff; 1750 1751 // protected by mLock 1752 Vector<AudioSessionRef*> mAudioSessionRefs; 1753 1754 float masterVolume_l() const; 1755 float masterVolumeSW_l() const { return mMasterVolumeSW; } 1756 bool masterMute_l() const { return mMasterMute; } 1757 1758 Vector < sp<SyncEvent> > mPendingSyncEvents; // sync events awaiting for a session 1759 // to be created 1760 1761private: 1762 sp<Client> registerPid_l(pid_t pid); // always returns non-0 1763 1764}; 1765 1766 1767// ---------------------------------------------------------------------------- 1768 1769}; // namespace android 1770 1771#endif // ANDROID_AUDIO_FLINGER_H 1772