AudioFlinger.h revision 73f4bc33e0d458933460250a47c64aa868c05f97
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 uint32_t flags, 91 const sp<IMemory>& sharedBuffer, 92 audio_io_handle_t output, 93 bool isTimed, 94 int *sessionId, 95 status_t *status); 96 97 virtual sp<IAudioRecord> openRecord( 98 pid_t pid, 99 audio_io_handle_t input, 100 uint32_t sampleRate, 101 audio_format_t format, 102 uint32_t channelMask, 103 int frameCount, 104 uint32_t flags, 105 int *sessionId, 106 status_t *status); 107 108 virtual uint32_t sampleRate(audio_io_handle_t output) const; 109 virtual int channelCount(audio_io_handle_t output) const; 110 virtual audio_format_t format(audio_io_handle_t output) const; 111 virtual size_t frameCount(audio_io_handle_t output) const; 112 virtual uint32_t latency(audio_io_handle_t output) const; 113 114 virtual status_t setMasterVolume(float value); 115 virtual status_t setMasterMute(bool muted); 116 117 virtual float masterVolume() const; 118 virtual float masterVolumeSW() const; 119 virtual bool masterMute() const; 120 121 virtual status_t setStreamVolume(audio_stream_type_t stream, float value, 122 audio_io_handle_t output); 123 virtual status_t setStreamMute(audio_stream_type_t stream, bool muted); 124 125 virtual float streamVolume(audio_stream_type_t stream, 126 audio_io_handle_t output) const; 127 virtual bool streamMute(audio_stream_type_t stream) const; 128 129 virtual status_t setMode(audio_mode_t mode); 130 131 virtual status_t setMicMute(bool state); 132 virtual bool getMicMute() const; 133 134 virtual status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs); 135 virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) const; 136 137 virtual void registerClient(const sp<IAudioFlingerClient>& client); 138 139 virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const; 140 141 virtual audio_io_handle_t openOutput(uint32_t *pDevices, 142 uint32_t *pSamplingRate, 143 audio_format_t *pFormat, 144 uint32_t *pChannels, 145 uint32_t *pLatencyMs, 146 audio_policy_output_flags_t flags); 147 148 virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1, 149 audio_io_handle_t output2); 150 151 virtual status_t closeOutput(audio_io_handle_t output); 152 153 virtual status_t suspendOutput(audio_io_handle_t output); 154 155 virtual status_t restoreOutput(audio_io_handle_t output); 156 157 virtual audio_io_handle_t openInput(uint32_t *pDevices, 158 uint32_t *pSamplingRate, 159 audio_format_t *pFormat, 160 uint32_t *pChannels, 161 audio_in_acoustics_t acoustics); 162 163 virtual status_t closeInput(audio_io_handle_t input); 164 165 virtual status_t setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output); 166 167 virtual status_t setVoiceVolume(float volume); 168 169 virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 170 audio_io_handle_t output) const; 171 172 virtual unsigned int getInputFramesLost(audio_io_handle_t ioHandle) const; 173 174 virtual int newAudioSessionId(); 175 176 virtual void acquireAudioSessionId(int audioSession); 177 178 virtual void releaseAudioSessionId(int audioSession); 179 180 virtual status_t queryNumberEffects(uint32_t *numEffects) const; 181 182 virtual status_t queryEffect(uint32_t index, effect_descriptor_t *descriptor) const; 183 184 virtual status_t getEffectDescriptor(const effect_uuid_t *pUuid, 185 effect_descriptor_t *descriptor) const; 186 187 virtual sp<IEffect> createEffect(pid_t pid, 188 effect_descriptor_t *pDesc, 189 const sp<IEffectClient>& effectClient, 190 int32_t priority, 191 audio_io_handle_t io, 192 int sessionId, 193 status_t *status, 194 int *id, 195 int *enabled); 196 197 virtual status_t moveEffects(int sessionId, audio_io_handle_t srcOutput, 198 audio_io_handle_t dstOutput); 199 200 virtual status_t onTransact( 201 uint32_t code, 202 const Parcel& data, 203 Parcel* reply, 204 uint32_t flags); 205 206 // end of IAudioFlinger interface 207 208private: 209 audio_mode_t getMode() const { return mMode; } 210 211 bool btNrecIsOff() const { return mBtNrecIsOff; } 212 213 AudioFlinger(); 214 virtual ~AudioFlinger(); 215 216 // call in any IAudioFlinger method that accesses mPrimaryHardwareDev 217 status_t initCheck() const { return mPrimaryHardwareDev == NULL ? NO_INIT : NO_ERROR; } 218 219 // RefBase 220 virtual void onFirstRef(); 221 222 audio_hw_device_t* findSuitableHwDev_l(uint32_t devices); 223 void purgeStaleEffects_l(); 224 225 // standby delay for MIXER and DUPLICATING playback threads is read from property 226 // ro.audio.flinger_standbytime_ms or defaults to kDefaultStandbyTimeInNsecs 227 static nsecs_t mStandbyTimeInNsecs; 228 229 // Internal dump utilites. 230 status_t dumpPermissionDenial(int fd, const Vector<String16>& args); 231 status_t dumpClients(int fd, const Vector<String16>& args); 232 status_t dumpInternals(int fd, const Vector<String16>& args); 233 234 // --- Client --- 235 class Client : public RefBase { 236 public: 237 Client(const sp<AudioFlinger>& audioFlinger, pid_t pid); 238 virtual ~Client(); 239 sp<MemoryDealer> heap() const; 240 pid_t pid() const { return mPid; } 241 sp<AudioFlinger> audioFlinger() const { return mAudioFlinger; } 242 243 bool reserveTimedTrack(); 244 void releaseTimedTrack(); 245 246 private: 247 Client(const Client&); 248 Client& operator = (const Client&); 249 const sp<AudioFlinger> mAudioFlinger; 250 const sp<MemoryDealer> mMemoryDealer; 251 const pid_t mPid; 252 253 Mutex mTimedTrackLock; 254 int mTimedTrackCount; 255 }; 256 257 // --- Notification Client --- 258 class NotificationClient : public IBinder::DeathRecipient { 259 public: 260 NotificationClient(const sp<AudioFlinger>& audioFlinger, 261 const sp<IAudioFlingerClient>& client, 262 pid_t pid); 263 virtual ~NotificationClient(); 264 265 sp<IAudioFlingerClient> audioFlingerClient() const { return mAudioFlingerClient; } 266 267 // IBinder::DeathRecipient 268 virtual void binderDied(const wp<IBinder>& who); 269 270 private: 271 NotificationClient(const NotificationClient&); 272 NotificationClient& operator = (const NotificationClient&); 273 274 const sp<AudioFlinger> mAudioFlinger; 275 const pid_t mPid; 276 const sp<IAudioFlingerClient> mAudioFlingerClient; 277 }; 278 279 class TrackHandle; 280 class RecordHandle; 281 class RecordThread; 282 class PlaybackThread; 283 class MixerThread; 284 class DirectOutputThread; 285 class DuplicatingThread; 286 class Track; 287 class RecordTrack; 288 class EffectModule; 289 class EffectHandle; 290 class EffectChain; 291 struct AudioStreamOut; 292 struct AudioStreamIn; 293 294 class ThreadBase : public Thread { 295 public: 296 297 enum type_t { 298 MIXER, // Thread class is MixerThread 299 DIRECT, // Thread class is DirectOutputThread 300 DUPLICATING, // Thread class is DuplicatingThread 301 RECORD // Thread class is RecordThread 302 }; 303 304 ThreadBase (const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, uint32_t device, type_t type); 305 virtual ~ThreadBase(); 306 307 status_t dumpBase(int fd, const Vector<String16>& args); 308 status_t dumpEffectChains(int fd, const Vector<String16>& args); 309 310 void clearPowerManager(); 311 312 // base for record and playback 313 class TrackBase : public AudioBufferProvider, public RefBase { 314 315 public: 316 enum track_state { 317 IDLE, 318 TERMINATED, 319 // These are order-sensitive; do not change order without reviewing the impact. 320 // In particular there are assumptions about > STOPPED. 321 STOPPED, 322 RESUMING, 323 ACTIVE, 324 PAUSING, 325 PAUSED 326 }; 327 328 TrackBase(ThreadBase *thread, 329 const sp<Client>& client, 330 uint32_t sampleRate, 331 audio_format_t format, 332 uint32_t channelMask, 333 int frameCount, 334 const sp<IMemory>& sharedBuffer, 335 int sessionId); 336 virtual ~TrackBase(); 337 338 virtual status_t start(pid_t tid) = 0; 339 virtual void stop() = 0; 340 sp<IMemory> getCblk() const { return mCblkMemory; } 341 audio_track_cblk_t* cblk() const { return mCblk; } 342 int sessionId() const { return mSessionId; } 343 344 protected: 345 TrackBase(const TrackBase&); 346 TrackBase& operator = (const TrackBase&); 347 348 // AudioBufferProvider interface 349 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) = 0; 350 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 351 352 audio_format_t format() const { 353 return mFormat; 354 } 355 356 int channelCount() const { return mChannelCount; } 357 358 uint32_t channelMask() const { return mChannelMask; } 359 360 int sampleRate() const; // FIXME inline after cblk sr moved 361 362 void* getBuffer(uint32_t offset, uint32_t frames) const; 363 364 bool isStopped() const { 365 return mState == STOPPED; 366 } 367 368 bool isTerminated() const { 369 return mState == TERMINATED; 370 } 371 372 bool step(); 373 void reset(); 374 375 const wp<ThreadBase> mThread; 376 /*const*/ sp<Client> mClient; // see explanation at ~TrackBase() why not const 377 sp<IMemory> mCblkMemory; 378 audio_track_cblk_t* mCblk; 379 void* mBuffer; 380 void* mBufferEnd; 381 uint32_t mFrameCount; 382 // we don't really need a lock for these 383 track_state mState; 384 const audio_format_t mFormat; 385 bool mStepServerFailed; 386 const int mSessionId; 387 uint8_t mChannelCount; 388 uint32_t mChannelMask; 389 }; 390 391 class ConfigEvent { 392 public: 393 ConfigEvent() : mEvent(0), mParam(0) {} 394 395 int mEvent; 396 int mParam; 397 }; 398 399 class PMDeathRecipient : public IBinder::DeathRecipient { 400 public: 401 PMDeathRecipient(const wp<ThreadBase>& thread) : mThread(thread) {} 402 virtual ~PMDeathRecipient() {} 403 404 // IBinder::DeathRecipient 405 virtual void binderDied(const wp<IBinder>& who); 406 407 private: 408 PMDeathRecipient(const PMDeathRecipient&); 409 PMDeathRecipient& operator = (const PMDeathRecipient&); 410 411 wp<ThreadBase> mThread; 412 }; 413 414 virtual status_t initCheck() const = 0; 415 type_t type() const { return mType; } 416 uint32_t sampleRate() const { return mSampleRate; } 417 int channelCount() const { return mChannelCount; } 418 audio_format_t format() const { return mFormat; } 419 size_t frameCount() const { return mFrameCount; } 420 void wakeUp() { mWaitWorkCV.broadcast(); } 421 // Should be "virtual status_t requestExitAndWait()" and override same 422 // method in Thread, but Thread::requestExitAndWait() is not yet virtual. 423 void exit(); 424 virtual bool checkForNewParameters_l() = 0; 425 virtual status_t setParameters(const String8& keyValuePairs); 426 virtual String8 getParameters(const String8& keys) = 0; 427 virtual void audioConfigChanged_l(int event, int param = 0) = 0; 428 void sendConfigEvent(int event, int param = 0); 429 void sendConfigEvent_l(int event, int param = 0); 430 void processConfigEvents(); 431 audio_io_handle_t id() const { return mId;} 432 bool standby() const { return mStandby; } 433 uint32_t device() const { return mDevice; } 434 virtual audio_stream_t* stream() = 0; 435 436 sp<EffectHandle> createEffect_l( 437 const sp<AudioFlinger::Client>& client, 438 const sp<IEffectClient>& effectClient, 439 int32_t priority, 440 int sessionId, 441 effect_descriptor_t *desc, 442 int *enabled, 443 status_t *status); 444 void disconnectEffect(const sp< EffectModule>& effect, 445 const wp<EffectHandle>& handle, 446 bool unpinIfLast); 447 448 // return values for hasAudioSession (bit field) 449 enum effect_state { 450 EFFECT_SESSION = 0x1, // the audio session corresponds to at least one 451 // effect 452 TRACK_SESSION = 0x2 // the audio session corresponds to at least one 453 // track 454 }; 455 456 // get effect chain corresponding to session Id. 457 sp<EffectChain> getEffectChain(int sessionId); 458 // same as getEffectChain() but must be called with ThreadBase mutex locked 459 sp<EffectChain> getEffectChain_l(int sessionId); 460 // add an effect chain to the chain list (mEffectChains) 461 virtual status_t addEffectChain_l(const sp<EffectChain>& chain) = 0; 462 // remove an effect chain from the chain list (mEffectChains) 463 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain) = 0; 464 // lock all effect chains Mutexes. Must be called before releasing the 465 // ThreadBase mutex before processing the mixer and effects. This guarantees the 466 // integrity of the chains during the process. 467 // Also sets the parameter 'effectChains' to current value of mEffectChains. 468 void lockEffectChains_l(Vector<sp <EffectChain> >& effectChains); 469 // unlock effect chains after process 470 void unlockEffectChains(const Vector<sp<EffectChain> >& effectChains); 471 // set audio mode to all effect chains 472 void setMode(audio_mode_t mode); 473 // get effect module with corresponding ID on specified audio session 474 sp<AudioFlinger::EffectModule> getEffect_l(int sessionId, int effectId); 475 // add and effect module. Also creates the effect chain is none exists for 476 // the effects audio session 477 status_t addEffect_l(const sp< EffectModule>& effect); 478 // remove and effect module. Also removes the effect chain is this was the last 479 // effect 480 void removeEffect_l(const sp< EffectModule>& effect); 481 // detach all tracks connected to an auxiliary effect 482 virtual void detachAuxEffect_l(int effectId) {} 483 // returns either EFFECT_SESSION if effects on this audio session exist in one 484 // chain, or TRACK_SESSION if tracks on this audio session exist, or both 485 virtual uint32_t hasAudioSession(int sessionId) = 0; 486 // the value returned by default implementation is not important as the 487 // strategy is only meaningful for PlaybackThread which implements this method 488 virtual uint32_t getStrategyForSession_l(int sessionId) { return 0; } 489 490 // suspend or restore effect according to the type of effect passed. a NULL 491 // type pointer means suspend all effects in the session 492 void setEffectSuspended(const effect_uuid_t *type, 493 bool suspend, 494 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 495 // check if some effects must be suspended/restored when an effect is enabled 496 // or disabled 497 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 498 bool enabled, 499 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 500 void checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 501 bool enabled, 502 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 503 mutable Mutex mLock; 504 505 protected: 506 507 // entry describing an effect being suspended in mSuspendedSessions keyed vector 508 class SuspendedSessionDesc : public RefBase { 509 public: 510 SuspendedSessionDesc() : mRefCount(0) {} 511 512 int mRefCount; // number of active suspend requests 513 effect_uuid_t mType; // effect type UUID 514 }; 515 516 void acquireWakeLock(); 517 void acquireWakeLock_l(); 518 void releaseWakeLock(); 519 void releaseWakeLock_l(); 520 void setEffectSuspended_l(const effect_uuid_t *type, 521 bool suspend, 522 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 523 // updated mSuspendedSessions when an effect suspended or restored 524 void updateSuspendedSessions_l(const effect_uuid_t *type, 525 bool suspend, 526 int sessionId); 527 // check if some effects must be suspended when an effect chain is added 528 void checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain); 529 530 friend class AudioFlinger; // for mEffectChains 531 532 const type_t mType; 533 534 // Used by parameters, config events, addTrack_l, exit 535 Condition mWaitWorkCV; 536 537 const sp<AudioFlinger> mAudioFlinger; 538 uint32_t mSampleRate; 539 size_t mFrameCount; 540 uint32_t mChannelMask; 541 uint16_t mChannelCount; 542 size_t mFrameSize; 543 audio_format_t mFormat; 544 545 // Parameter sequence by client: binder thread calling setParameters(): 546 // 1. Lock mLock 547 // 2. Append to mNewParameters 548 // 3. mWaitWorkCV.signal 549 // 4. mParamCond.waitRelative with timeout 550 // 5. read mParamStatus 551 // 6. mWaitWorkCV.signal 552 // 7. Unlock 553 // 554 // Parameter sequence by server: threadLoop calling checkForNewParameters_l(): 555 // 1. Lock mLock 556 // 2. If there is an entry in mNewParameters proceed ... 557 // 2. Read first entry in mNewParameters 558 // 3. Process 559 // 4. Remove first entry from mNewParameters 560 // 5. Set mParamStatus 561 // 6. mParamCond.signal 562 // 7. mWaitWorkCV.wait with timeout (this is to avoid overwriting mParamStatus) 563 // 8. Unlock 564 Condition mParamCond; 565 Vector<String8> mNewParameters; 566 status_t mParamStatus; 567 568 Vector<ConfigEvent> mConfigEvents; 569 bool mStandby; 570 const audio_io_handle_t mId; 571 Vector< sp<EffectChain> > mEffectChains; 572 uint32_t mDevice; // output device for PlaybackThread 573 // input + output devices for RecordThread 574 static const int kNameLength = 16; // prctl(PR_SET_NAME) limit 575 char mName[kNameLength]; 576 sp<IPowerManager> mPowerManager; 577 sp<IBinder> mWakeLockToken; 578 const sp<PMDeathRecipient> mDeathRecipient; 579 // list of suspended effects per session and per type. The first vector is 580 // keyed by session ID, the second by type UUID timeLow field 581 KeyedVector< int, KeyedVector< int, sp<SuspendedSessionDesc> > > mSuspendedSessions; 582 }; 583 584 struct stream_type_t { 585 stream_type_t() 586 : volume(1.0f), 587 mute(false), 588 valid(true) 589 { 590 } 591 float volume; 592 bool mute; 593 bool valid; 594 }; 595 596 // --- PlaybackThread --- 597 class PlaybackThread : public ThreadBase { 598 public: 599 600 enum mixer_state { 601 MIXER_IDLE, // no active tracks 602 MIXER_TRACKS_ENABLED, // at least one active track, but no track has any data ready 603 MIXER_TRACKS_READY // at least one active track, and at least one track has data 604 // standby mode does not have an enum value 605 // suspend by audio policy manager is orthogonal to mixer state 606 }; 607 608 // playback track 609 class Track : public TrackBase { 610 public: 611 Track( PlaybackThread *thread, 612 const sp<Client>& client, 613 audio_stream_type_t streamType, 614 uint32_t sampleRate, 615 audio_format_t format, 616 uint32_t channelMask, 617 int frameCount, 618 const sp<IMemory>& sharedBuffer, 619 int sessionId); 620 virtual ~Track(); 621 622 void dump(char* buffer, size_t size); 623 virtual status_t start(pid_t tid); 624 virtual void stop(); 625 void pause(); 626 627 void flush(); 628 void destroy(); 629 void mute(bool); 630 int name() const { 631 return mName; 632 } 633 634 audio_stream_type_t streamType() const { 635 return mStreamType; 636 } 637 status_t attachAuxEffect(int EffectId); 638 void setAuxBuffer(int EffectId, int32_t *buffer); 639 int32_t *auxBuffer() const { return mAuxBuffer; } 640 void setMainBuffer(int16_t *buffer) { mMainBuffer = buffer; } 641 int16_t *mainBuffer() const { return mMainBuffer; } 642 int auxEffectId() const { return mAuxEffectId; } 643 644 protected: 645 // for numerous 646 friend class PlaybackThread; 647 friend class MixerThread; 648 friend class DirectOutputThread; 649 650 Track(const Track&); 651 Track& operator = (const Track&); 652 653 // AudioBufferProvider interface 654 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts = kInvalidPTS); 655 // releaseBuffer() not overridden 656 657 virtual uint32_t framesReady() const; 658 659 bool isMuted() const { return mMute; } 660 bool isPausing() const { 661 return mState == PAUSING; 662 } 663 bool isPaused() const { 664 return mState == PAUSED; 665 } 666 bool isReady() const; 667 void setPaused() { mState = PAUSED; } 668 void reset(); 669 670 bool isOutputTrack() const { 671 return (mStreamType == AUDIO_STREAM_CNT); 672 } 673 674 public: 675 virtual bool isTimedTrack() const { return false; } 676 protected: 677 678 // we don't really need a lock for these 679 volatile bool mMute; 680 // FILLED state is used for suppressing volume ramp at begin of playing 681 enum {FS_FILLING, FS_FILLED, FS_ACTIVE}; 682 mutable uint8_t mFillingUpStatus; 683 int8_t mRetryCount; 684 sp<IMemory> mSharedBuffer; 685 bool mResetDone; 686 audio_stream_type_t mStreamType; 687 int mName; 688 int16_t *mMainBuffer; 689 int32_t *mAuxBuffer; 690 int mAuxEffectId; 691 bool mHasVolumeController; 692 }; // end of Track 693 694 class TimedTrack : public Track { 695 public: 696 static sp<TimedTrack> create(PlaybackThread *thread, 697 const sp<Client>& client, 698 audio_stream_type_t streamType, 699 uint32_t sampleRate, 700 audio_format_t format, 701 uint32_t channelMask, 702 int frameCount, 703 const sp<IMemory>& sharedBuffer, 704 int sessionId); 705 ~TimedTrack(); 706 707 class TimedBuffer { 708 public: 709 TimedBuffer(); 710 TimedBuffer(const sp<IMemory>& buffer, int64_t pts); 711 const sp<IMemory>& buffer() const { return mBuffer; } 712 int64_t pts() const { return mPTS; } 713 int position() const { return mPosition; } 714 void setPosition(int pos) { mPosition = pos; } 715 private: 716 sp<IMemory> mBuffer; 717 int64_t mPTS; 718 int mPosition; 719 }; 720 721 virtual bool isTimedTrack() const { return true; } 722 723 virtual uint32_t framesReady() const; 724 725 // AudioBufferProvider interface 726 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts); 727 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 728 729 void timedYieldSamples(AudioBufferProvider::Buffer* buffer); 730 void timedYieldSilence(uint32_t numFrames, 731 AudioBufferProvider::Buffer* buffer); 732 733 status_t allocateTimedBuffer(size_t size, 734 sp<IMemory>* buffer); 735 status_t queueTimedBuffer(const sp<IMemory>& buffer, 736 int64_t pts); 737 status_t setMediaTimeTransform(const LinearTransform& xform, 738 TimedAudioTrack::TargetTimeline target); 739 void trimTimedBufferQueue_l(); 740 741 private: 742 TimedTrack(PlaybackThread *thread, 743 const sp<Client>& client, 744 audio_stream_type_t streamType, 745 uint32_t sampleRate, 746 audio_format_t format, 747 uint32_t channelMask, 748 int frameCount, 749 const sp<IMemory>& sharedBuffer, 750 int sessionId); 751 752 uint64_t mLocalTimeFreq; 753 LinearTransform mLocalTimeToSampleTransform; 754 sp<MemoryDealer> mTimedMemoryDealer; 755 Vector<TimedBuffer> mTimedBufferQueue; 756 uint8_t* mTimedSilenceBuffer; 757 uint32_t mTimedSilenceBufferSize; 758 mutable Mutex mTimedBufferQueueLock; 759 bool mTimedAudioOutputOnTime; 760 CCHelper mCCHelper; 761 762 Mutex mMediaTimeTransformLock; 763 LinearTransform mMediaTimeTransform; 764 bool mMediaTimeTransformValid; 765 TimedAudioTrack::TargetTimeline mMediaTimeTransformTarget; 766 }; 767 768 769 // playback track 770 class OutputTrack : public Track { 771 public: 772 773 class Buffer: public AudioBufferProvider::Buffer { 774 public: 775 int16_t *mBuffer; 776 }; 777 778 OutputTrack(PlaybackThread *thread, 779 DuplicatingThread *sourceThread, 780 uint32_t sampleRate, 781 audio_format_t format, 782 uint32_t channelMask, 783 int frameCount); 784 virtual ~OutputTrack(); 785 786 virtual status_t start(pid_t tid); 787 virtual void stop(); 788 bool write(int16_t* data, uint32_t frames); 789 bool bufferQueueEmpty() const { return mBufferQueue.size() == 0; } 790 bool isActive() const { return mActive; } 791 const wp<ThreadBase>& thread() const { return mThread; } 792 793 private: 794 795 enum { 796 NO_MORE_BUFFERS = 0x80000001, // same in AudioTrack.h, ok to be different value 797 }; 798 799 status_t obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs); 800 void clearBufferQueue(); 801 802 // Maximum number of pending buffers allocated by OutputTrack::write() 803 static const uint8_t kMaxOverFlowBuffers = 10; 804 805 Vector < Buffer* > mBufferQueue; 806 AudioBufferProvider::Buffer mOutBuffer; 807 bool mActive; 808 DuplicatingThread* const mSourceThread; // for waitTimeMs() in write() 809 }; // end of OutputTrack 810 811 PlaybackThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 812 audio_io_handle_t id, uint32_t device, type_t type); 813 virtual ~PlaybackThread(); 814 815 status_t dump(int fd, const Vector<String16>& args); 816 817 // Thread virtuals 818 virtual status_t readyToRun(); 819 virtual bool threadLoop(); 820 821 // RefBase 822 virtual void onFirstRef(); 823 824protected: 825 // Code snippets that were lifted up out of threadLoop() 826 virtual void threadLoop_mix() = 0; 827 virtual void threadLoop_sleepTime() = 0; 828 virtual void threadLoop_write(); 829 virtual void threadLoop_standby(); 830 831 // prepareTracks_l reads and writes mActiveTracks, and also returns the 832 // pending set of tracks to remove via Vector 'tracksToRemove'. The caller is 833 // responsible for clearing or destroying this Vector later on, when it 834 // is safe to do so. That will drop the final ref count and destroy the tracks. 835 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove) = 0; 836 837public: 838 839 virtual status_t initCheck() const { return (mOutput == NULL) ? NO_INIT : NO_ERROR; } 840 841 // return estimated latency in milliseconds, as reported by HAL 842 uint32_t latency() const; 843 844 void setMasterVolume(float value); 845 void setMasterMute(bool muted); 846 847 void setStreamVolume(audio_stream_type_t stream, float value); 848 void setStreamMute(audio_stream_type_t stream, bool muted); 849 850 float streamVolume(audio_stream_type_t stream) const; 851 852 sp<Track> createTrack_l( 853 const sp<AudioFlinger::Client>& client, 854 audio_stream_type_t streamType, 855 uint32_t sampleRate, 856 audio_format_t format, 857 uint32_t channelMask, 858 int frameCount, 859 const sp<IMemory>& sharedBuffer, 860 int sessionId, 861 bool isTimed, 862 status_t *status); 863 864 AudioStreamOut* getOutput() const; 865 AudioStreamOut* clearOutput(); 866 virtual audio_stream_t* stream(); 867 868 void suspend() { mSuspended++; } 869 void restore() { if (mSuspended > 0) mSuspended--; } 870 bool isSuspended() const { return (mSuspended > 0); } 871 virtual String8 getParameters(const String8& keys); 872 virtual void audioConfigChanged_l(int event, int param = 0); 873 status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames); 874 int16_t *mixBuffer() const { return mMixBuffer; }; 875 876 virtual void detachAuxEffect_l(int effectId); 877 status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track> track, 878 int EffectId); 879 status_t attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track> track, 880 int EffectId); 881 882 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 883 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 884 virtual uint32_t hasAudioSession(int sessionId); 885 virtual uint32_t getStrategyForSession_l(int sessionId); 886 887 void setStreamValid(audio_stream_type_t streamType, bool valid); 888 889 protected: 890 int16_t* mMixBuffer; 891 uint32_t mSuspended; // suspend count, > 0 means suspended 892 int mBytesWritten; 893 private: 894 // mMasterMute is in both PlaybackThread and in AudioFlinger. When a 895 // PlaybackThread needs to find out if master-muted, it checks it's local 896 // copy rather than the one in AudioFlinger. This optimization saves a lock. 897 bool mMasterMute; 898 void setMasterMute_l(bool muted) { mMasterMute = muted; } 899 protected: 900 SortedVector< wp<Track> > mActiveTracks; 901 902 virtual int getTrackName_l() = 0; 903 virtual void deleteTrackName_l(int name) = 0; 904 virtual uint32_t activeSleepTimeUs(); 905 virtual uint32_t idleSleepTimeUs() = 0; 906 virtual uint32_t suspendSleepTimeUs() = 0; 907 908 // Code snippets that are temporarily lifted up out of threadLoop() until the merge 909 void checkSilentMode_l(); 910 911 // Non-trivial for DUPLICATING only 912 virtual void saveOutputTracks() { } 913 virtual void clearOutputTracks() { } 914 915 // Cache various calculated values, at threadLoop() entry and after a parameter change 916 virtual void cacheParameters_l(); 917 918 private: 919 920 friend class AudioFlinger; // for numerous 921 922 PlaybackThread(const Client&); 923 PlaybackThread& operator = (const PlaybackThread&); 924 925 status_t addTrack_l(const sp<Track>& track); 926 void destroyTrack_l(const sp<Track>& track); 927 void removeTrack_l(const sp<Track>& track); 928 929 void readOutputParameters(); 930 931 virtual status_t dumpInternals(int fd, const Vector<String16>& args); 932 status_t dumpTracks(int fd, const Vector<String16>& args); 933 934 SortedVector< sp<Track> > mTracks; 935 // mStreamTypes[] uses 1 additional stream type internally for the OutputTrack used by DuplicatingThread 936 stream_type_t mStreamTypes[AUDIO_STREAM_CNT + 1]; 937 AudioStreamOut *mOutput; 938 float mMasterVolume; 939 nsecs_t mLastWriteTime; 940 int mNumWrites; 941 int mNumDelayedWrites; 942 bool mInWrite; 943 944 // FIXME rename these former local variables of threadLoop to standard "m" names 945 nsecs_t standbyTime; 946 size_t mixBufferSize; 947 948 // cached copies of activeSleepTimeUs() and idleSleepTimeUs() made by cacheParameters_l() 949 uint32_t activeSleepTime; 950 uint32_t idleSleepTime; 951 952 uint32_t sleepTime; 953 954 // mixer status returned by prepareTracks_l() 955 mixer_state mMixerStatus; // current cycle 956 mixer_state mPrevMixerStatus; // previous cycle 957 958 // FIXME move these declarations into the specific sub-class that needs them 959 // MIXER only 960 bool longStandbyExit; 961 uint32_t sleepTimeShift; 962 963 // same as AudioFlinger::mStandbyTimeInNsecs except for DIRECT which uses a shorter value 964 nsecs_t standbyDelay; 965 966 // MIXER only 967 nsecs_t maxPeriod; 968 969 // DUPLICATING only 970 uint32_t writeFrames; 971 }; 972 973 class MixerThread : public PlaybackThread { 974 public: 975 MixerThread (const sp<AudioFlinger>& audioFlinger, 976 AudioStreamOut* output, 977 audio_io_handle_t id, 978 uint32_t device, 979 type_t type = MIXER); 980 virtual ~MixerThread(); 981 982 // Thread virtuals 983 984 void invalidateTracks(audio_stream_type_t streamType); 985 virtual bool checkForNewParameters_l(); 986 virtual status_t dumpInternals(int fd, const Vector<String16>& args); 987 988 protected: 989 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 990 virtual int getTrackName_l(); 991 virtual void deleteTrackName_l(int name); 992 virtual uint32_t idleSleepTimeUs(); 993 virtual uint32_t suspendSleepTimeUs(); 994 virtual void cacheParameters_l(); 995 996 // threadLoop snippets 997 virtual void threadLoop_mix(); 998 virtual void threadLoop_sleepTime(); 999 1000 AudioMixer* mAudioMixer; 1001 }; 1002 1003 class DirectOutputThread : public PlaybackThread { 1004 public: 1005 1006 DirectOutputThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1007 audio_io_handle_t id, uint32_t device); 1008 virtual ~DirectOutputThread(); 1009 1010 // Thread virtuals 1011 1012 virtual bool checkForNewParameters_l(); 1013 1014 protected: 1015 virtual int getTrackName_l(); 1016 virtual void deleteTrackName_l(int name); 1017 virtual uint32_t activeSleepTimeUs(); 1018 virtual uint32_t idleSleepTimeUs(); 1019 virtual uint32_t suspendSleepTimeUs(); 1020 virtual void cacheParameters_l(); 1021 1022 // threadLoop snippets 1023 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1024 virtual void threadLoop_mix(); 1025 virtual void threadLoop_sleepTime(); 1026 1027 // volumes last sent to audio HAL with stream->set_volume() 1028 // FIXME use standard representation and names 1029 float mLeftVolFloat; 1030 float mRightVolFloat; 1031 uint16_t mLeftVolShort; 1032 uint16_t mRightVolShort; 1033 1034 // FIXME rename these former local variables of threadLoop to standard names 1035 // next 3 were local to the while !exitingPending loop 1036 bool rampVolume; 1037 uint16_t leftVol; 1038 uint16_t rightVol; 1039 1040private: 1041 // prepareTracks_l() tells threadLoop_mix() the name of the single active track 1042 sp<Track> mActiveTrack; 1043 }; 1044 1045 class DuplicatingThread : public MixerThread { 1046 public: 1047 DuplicatingThread (const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread, 1048 audio_io_handle_t id); 1049 virtual ~DuplicatingThread(); 1050 1051 // Thread virtuals 1052 void addOutputTrack(MixerThread* thread); 1053 void removeOutputTrack(MixerThread* thread); 1054 uint32_t waitTimeMs() { return mWaitTimeMs; } 1055 protected: 1056 virtual uint32_t activeSleepTimeUs(); 1057 1058 private: 1059 bool outputsReady(const SortedVector<sp<OutputTrack> > &outputTracks); 1060 protected: 1061 // threadLoop snippets 1062 virtual void threadLoop_mix(); 1063 virtual void threadLoop_sleepTime(); 1064 virtual void threadLoop_write(); 1065 virtual void threadLoop_standby(); 1066 virtual void cacheParameters_l(); 1067 1068 private: 1069 // called from threadLoop, addOutputTrack, removeOutputTrack 1070 virtual void updateWaitTime_l(); 1071 protected: 1072 virtual void saveOutputTracks(); 1073 virtual void clearOutputTracks(); 1074 private: 1075 1076 uint32_t mWaitTimeMs; 1077 SortedVector < sp<OutputTrack> > outputTracks; 1078 SortedVector < sp<OutputTrack> > mOutputTracks; 1079 }; 1080 1081 PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const; 1082 MixerThread *checkMixerThread_l(audio_io_handle_t output) const; 1083 RecordThread *checkRecordThread_l(audio_io_handle_t input) const; 1084 // no range check, AudioFlinger::mLock held 1085 bool streamMute_l(audio_stream_type_t stream) const 1086 { return mStreamTypes[stream].mute; } 1087 // no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held 1088 float streamVolume_l(audio_stream_type_t stream) const 1089 { return mStreamTypes[stream].volume; } 1090 void audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2); 1091 1092 // allocate an audio_io_handle_t, session ID, or effect ID 1093 uint32_t nextUniqueId(); 1094 1095 status_t moveEffectChain_l(int sessionId, 1096 PlaybackThread *srcThread, 1097 PlaybackThread *dstThread, 1098 bool reRegister); 1099 // return thread associated with primary hardware device, or NULL 1100 PlaybackThread *primaryPlaybackThread_l() const; 1101 uint32_t primaryOutputDevice_l() const; 1102 1103 // server side of the client's IAudioTrack 1104 class TrackHandle : public android::BnAudioTrack { 1105 public: 1106 TrackHandle(const sp<PlaybackThread::Track>& track); 1107 virtual ~TrackHandle(); 1108 virtual sp<IMemory> getCblk() const; 1109 virtual status_t start(pid_t tid); 1110 virtual void stop(); 1111 virtual void flush(); 1112 virtual void mute(bool); 1113 virtual void pause(); 1114 virtual status_t attachAuxEffect(int effectId); 1115 virtual status_t allocateTimedBuffer(size_t size, 1116 sp<IMemory>* buffer); 1117 virtual status_t queueTimedBuffer(const sp<IMemory>& buffer, 1118 int64_t pts); 1119 virtual status_t setMediaTimeTransform(const LinearTransform& xform, 1120 int target); 1121 virtual status_t onTransact( 1122 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 1123 private: 1124 const sp<PlaybackThread::Track> mTrack; 1125 }; 1126 1127 void removeClient_l(pid_t pid); 1128 void removeNotificationClient(pid_t pid); 1129 1130 1131 // record thread 1132 class RecordThread : public ThreadBase, public AudioBufferProvider 1133 { 1134 public: 1135 1136 // record track 1137 class RecordTrack : public TrackBase { 1138 public: 1139 RecordTrack(RecordThread *thread, 1140 const sp<Client>& client, 1141 uint32_t sampleRate, 1142 audio_format_t format, 1143 uint32_t channelMask, 1144 int frameCount, 1145 int sessionId); 1146 virtual ~RecordTrack(); 1147 1148 virtual status_t start(pid_t tid); 1149 virtual void stop(); 1150 1151 bool overflow() { bool tmp = mOverflow; mOverflow = false; return tmp; } 1152 bool setOverflow() { bool tmp = mOverflow; mOverflow = true; return tmp; } 1153 1154 void dump(char* buffer, size_t size); 1155 1156 private: 1157 friend class AudioFlinger; // for mState 1158 1159 RecordTrack(const RecordTrack&); 1160 RecordTrack& operator = (const RecordTrack&); 1161 1162 // AudioBufferProvider interface 1163 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts = kInvalidPTS); 1164 // releaseBuffer() not overridden 1165 1166 bool mOverflow; 1167 }; 1168 1169 1170 RecordThread(const sp<AudioFlinger>& audioFlinger, 1171 AudioStreamIn *input, 1172 uint32_t sampleRate, 1173 uint32_t channels, 1174 audio_io_handle_t id, 1175 uint32_t device); 1176 virtual ~RecordThread(); 1177 1178 // Thread 1179 virtual bool threadLoop(); 1180 virtual status_t readyToRun(); 1181 1182 // RefBase 1183 virtual void onFirstRef(); 1184 1185 virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; } 1186 sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l( 1187 const sp<AudioFlinger::Client>& client, 1188 uint32_t sampleRate, 1189 audio_format_t format, 1190 int channelMask, 1191 int frameCount, 1192 int sessionId, 1193 status_t *status); 1194 1195 status_t start(RecordTrack* recordTrack); 1196 status_t start(RecordTrack* recordTrack, pid_t tid); 1197 void stop(RecordTrack* recordTrack); 1198 status_t dump(int fd, const Vector<String16>& args); 1199 AudioStreamIn* getInput() const; 1200 AudioStreamIn* clearInput(); 1201 virtual audio_stream_t* stream(); 1202 1203 // AudioBufferProvider interface 1204 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts); 1205 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 1206 1207 virtual bool checkForNewParameters_l(); 1208 virtual String8 getParameters(const String8& keys); 1209 virtual void audioConfigChanged_l(int event, int param = 0); 1210 void readInputParameters(); 1211 virtual unsigned int getInputFramesLost(); 1212 1213 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1214 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1215 virtual uint32_t hasAudioSession(int sessionId); 1216 RecordTrack* track(); 1217 1218 private: 1219 RecordThread(); 1220 AudioStreamIn *mInput; 1221 RecordTrack* mTrack; 1222 sp<RecordTrack> mActiveTrack; 1223 Condition mStartStopCond; 1224 AudioResampler *mResampler; 1225 int32_t *mRsmpOutBuffer; 1226 int16_t *mRsmpInBuffer; 1227 size_t mRsmpInIndex; 1228 size_t mInputBytes; 1229 const int mReqChannelCount; 1230 const uint32_t mReqSampleRate; 1231 ssize_t mBytesRead; 1232 }; 1233 1234 // server side of the client's IAudioRecord 1235 class RecordHandle : public android::BnAudioRecord { 1236 public: 1237 RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack); 1238 virtual ~RecordHandle(); 1239 virtual sp<IMemory> getCblk() const; 1240 virtual status_t start(pid_t tid); 1241 virtual void stop(); 1242 virtual status_t onTransact( 1243 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 1244 private: 1245 const sp<RecordThread::RecordTrack> mRecordTrack; 1246 }; 1247 1248 //--- Audio Effect Management 1249 1250 // EffectModule and EffectChain classes both have their own mutex to protect 1251 // state changes or resource modifications. Always respect the following order 1252 // if multiple mutexes must be acquired to avoid cross deadlock: 1253 // AudioFlinger -> ThreadBase -> EffectChain -> EffectModule 1254 1255 // The EffectModule class is a wrapper object controlling the effect engine implementation 1256 // in the effect library. It prevents concurrent calls to process() and command() functions 1257 // from different client threads. It keeps a list of EffectHandle objects corresponding 1258 // to all client applications using this effect and notifies applications of effect state, 1259 // control or parameter changes. It manages the activation state machine to send appropriate 1260 // reset, enable, disable commands to effect engine and provide volume 1261 // ramping when effects are activated/deactivated. 1262 // When controlling an auxiliary effect, the EffectModule also provides an input buffer used by 1263 // the attached track(s) to accumulate their auxiliary channel. 1264 class EffectModule: public RefBase { 1265 public: 1266 EffectModule(ThreadBase *thread, 1267 const wp<AudioFlinger::EffectChain>& chain, 1268 effect_descriptor_t *desc, 1269 int id, 1270 int sessionId); 1271 virtual ~EffectModule(); 1272 1273 enum effect_state { 1274 IDLE, 1275 RESTART, 1276 STARTING, 1277 ACTIVE, 1278 STOPPING, 1279 STOPPED, 1280 DESTROYED 1281 }; 1282 1283 int id() const { return mId; } 1284 void process(); 1285 void updateState(); 1286 status_t command(uint32_t cmdCode, 1287 uint32_t cmdSize, 1288 void *pCmdData, 1289 uint32_t *replySize, 1290 void *pReplyData); 1291 1292 void reset_l(); 1293 status_t configure(); 1294 status_t init(); 1295 effect_state state() const { 1296 return mState; 1297 } 1298 uint32_t status() { 1299 return mStatus; 1300 } 1301 int sessionId() const { 1302 return mSessionId; 1303 } 1304 status_t setEnabled(bool enabled); 1305 bool isEnabled() const; 1306 bool isProcessEnabled() const; 1307 1308 void setInBuffer(int16_t *buffer) { mConfig.inputCfg.buffer.s16 = buffer; } 1309 int16_t *inBuffer() { return mConfig.inputCfg.buffer.s16; } 1310 void setOutBuffer(int16_t *buffer) { mConfig.outputCfg.buffer.s16 = buffer; } 1311 int16_t *outBuffer() { return mConfig.outputCfg.buffer.s16; } 1312 void setChain(const wp<EffectChain>& chain) { mChain = chain; } 1313 void setThread(const wp<ThreadBase>& thread) { mThread = thread; } 1314 const wp<ThreadBase>& thread() { return mThread; } 1315 1316 status_t addHandle(const sp<EffectHandle>& handle); 1317 void disconnect(const wp<EffectHandle>& handle, bool unpinIfLast); 1318 size_t removeHandle (const wp<EffectHandle>& handle); 1319 1320 effect_descriptor_t& desc() { return mDescriptor; } 1321 wp<EffectChain>& chain() { return mChain; } 1322 1323 status_t setDevice(uint32_t device); 1324 status_t setVolume(uint32_t *left, uint32_t *right, bool controller); 1325 status_t setMode(audio_mode_t mode); 1326 status_t start(); 1327 status_t stop(); 1328 void setSuspended(bool suspended); 1329 bool suspended() const; 1330 1331 sp<EffectHandle> controlHandle(); 1332 1333 bool isPinned() const { return mPinned; } 1334 void unPin() { mPinned = false; } 1335 1336 status_t dump(int fd, const Vector<String16>& args); 1337 1338 protected: 1339 friend class AudioFlinger; // for mHandles 1340 bool mPinned; 1341 1342 // Maximum time allocated to effect engines to complete the turn off sequence 1343 static const uint32_t MAX_DISABLE_TIME_MS = 10000; 1344 1345 EffectModule(const EffectModule&); 1346 EffectModule& operator = (const EffectModule&); 1347 1348 status_t start_l(); 1349 status_t stop_l(); 1350 1351mutable Mutex mLock; // mutex for process, commands and handles list protection 1352 wp<ThreadBase> mThread; // parent thread 1353 wp<EffectChain> mChain; // parent effect chain 1354 int mId; // this instance unique ID 1355 int mSessionId; // audio session ID 1356 effect_descriptor_t mDescriptor;// effect descriptor received from effect engine 1357 effect_config_t mConfig; // input and output audio configuration 1358 effect_handle_t mEffectInterface; // Effect module C API 1359 status_t mStatus; // initialization status 1360 effect_state mState; // current activation state 1361 Vector< wp<EffectHandle> > mHandles; // list of client handles 1362 // First handle in mHandles has highest priority and controls the effect module 1363 uint32_t mMaxDisableWaitCnt; // maximum grace period before forcing an effect off after 1364 // sending disable command. 1365 uint32_t mDisableWaitCnt; // current process() calls count during disable period. 1366 bool mSuspended; // effect is suspended: temporarily disabled by framework 1367 }; 1368 1369 // The EffectHandle class implements the IEffect interface. It provides resources 1370 // to receive parameter updates, keeps track of effect control 1371 // ownership and state and has a pointer to the EffectModule object it is controlling. 1372 // There is one EffectHandle object for each application controlling (or using) 1373 // an effect module. 1374 // The EffectHandle is obtained by calling AudioFlinger::createEffect(). 1375 class EffectHandle: public android::BnEffect { 1376 public: 1377 1378 EffectHandle(const sp<EffectModule>& effect, 1379 const sp<AudioFlinger::Client>& client, 1380 const sp<IEffectClient>& effectClient, 1381 int32_t priority); 1382 virtual ~EffectHandle(); 1383 1384 // IEffect 1385 virtual status_t enable(); 1386 virtual status_t disable(); 1387 virtual status_t command(uint32_t cmdCode, 1388 uint32_t cmdSize, 1389 void *pCmdData, 1390 uint32_t *replySize, 1391 void *pReplyData); 1392 virtual void disconnect(); 1393 private: 1394 void disconnect(bool unpinIfLast); 1395 public: 1396 virtual sp<IMemory> getCblk() const { return mCblkMemory; } 1397 virtual status_t onTransact(uint32_t code, const Parcel& data, 1398 Parcel* reply, uint32_t flags); 1399 1400 1401 // Give or take control of effect module 1402 // - hasControl: true if control is given, false if removed 1403 // - signal: true client app should be signaled of change, false otherwise 1404 // - enabled: state of the effect when control is passed 1405 void setControl(bool hasControl, bool signal, bool enabled); 1406 void commandExecuted(uint32_t cmdCode, 1407 uint32_t cmdSize, 1408 void *pCmdData, 1409 uint32_t replySize, 1410 void *pReplyData); 1411 void setEnabled(bool enabled); 1412 bool enabled() const { return mEnabled; } 1413 1414 // Getters 1415 int id() const { return mEffect->id(); } 1416 int priority() const { return mPriority; } 1417 bool hasControl() const { return mHasControl; } 1418 sp<EffectModule> effect() const { return mEffect; } 1419 1420 void dump(char* buffer, size_t size); 1421 1422 protected: 1423 friend class AudioFlinger; // for mEffect, mHasControl, mEnabled 1424 EffectHandle(const EffectHandle&); 1425 EffectHandle& operator =(const EffectHandle&); 1426 1427 sp<EffectModule> mEffect; // pointer to controlled EffectModule 1428 sp<IEffectClient> mEffectClient; // callback interface for client notifications 1429 /*const*/ sp<Client> mClient; // client for shared memory allocation, see disconnect() 1430 sp<IMemory> mCblkMemory; // shared memory for control block 1431 effect_param_cblk_t* mCblk; // control block for deferred parameter setting via shared memory 1432 uint8_t* mBuffer; // pointer to parameter area in shared memory 1433 int mPriority; // client application priority to control the effect 1434 bool mHasControl; // true if this handle is controlling the effect 1435 bool mEnabled; // cached enable state: needed when the effect is 1436 // restored after being suspended 1437 }; 1438 1439 // the EffectChain class represents a group of effects associated to one audio session. 1440 // There can be any number of EffectChain objects per output mixer thread (PlaybackThread). 1441 // The EffecChain with session ID 0 contains global effects applied to the output mix. 1442 // Effects in this chain can be insert or auxiliary. Effects in other chains (attached to tracks) 1443 // are insert only. The EffectChain maintains an ordered list of effect module, the order corresponding 1444 // in the effect process order. When attached to a track (session ID != 0), it also provide it's own 1445 // input buffer used by the track as accumulation buffer. 1446 class EffectChain: public RefBase { 1447 public: 1448 EffectChain(const wp<ThreadBase>& wThread, int sessionId); 1449 EffectChain(ThreadBase *thread, int sessionId); 1450 virtual ~EffectChain(); 1451 1452 // special key used for an entry in mSuspendedEffects keyed vector 1453 // corresponding to a suspend all request. 1454 static const int kKeyForSuspendAll = 0; 1455 1456 // minimum duration during which we force calling effect process when last track on 1457 // a session is stopped or removed to allow effect tail to be rendered 1458 static const int kProcessTailDurationMs = 1000; 1459 1460 void process_l(); 1461 1462 void lock() { 1463 mLock.lock(); 1464 } 1465 void unlock() { 1466 mLock.unlock(); 1467 } 1468 1469 status_t addEffect_l(const sp<EffectModule>& handle); 1470 size_t removeEffect_l(const sp<EffectModule>& handle); 1471 1472 int sessionId() const { return mSessionId; } 1473 void setSessionId(int sessionId) { mSessionId = sessionId; } 1474 1475 sp<EffectModule> getEffectFromDesc_l(effect_descriptor_t *descriptor); 1476 sp<EffectModule> getEffectFromId_l(int id); 1477 sp<EffectModule> getEffectFromType_l(const effect_uuid_t *type); 1478 bool setVolume_l(uint32_t *left, uint32_t *right); 1479 void setDevice_l(uint32_t device); 1480 void setMode_l(audio_mode_t mode); 1481 1482 void setInBuffer(int16_t *buffer, bool ownsBuffer = false) { 1483 mInBuffer = buffer; 1484 mOwnInBuffer = ownsBuffer; 1485 } 1486 int16_t *inBuffer() const { 1487 return mInBuffer; 1488 } 1489 void setOutBuffer(int16_t *buffer) { 1490 mOutBuffer = buffer; 1491 } 1492 int16_t *outBuffer() const { 1493 return mOutBuffer; 1494 } 1495 1496 void incTrackCnt() { android_atomic_inc(&mTrackCnt); } 1497 void decTrackCnt() { android_atomic_dec(&mTrackCnt); } 1498 int32_t trackCnt() const { return mTrackCnt;} 1499 1500 void incActiveTrackCnt() { android_atomic_inc(&mActiveTrackCnt); 1501 mTailBufferCount = mMaxTailBuffers; } 1502 void decActiveTrackCnt() { android_atomic_dec(&mActiveTrackCnt); } 1503 int32_t activeTrackCnt() const { return mActiveTrackCnt;} 1504 1505 uint32_t strategy() const { return mStrategy; } 1506 void setStrategy(uint32_t strategy) 1507 { mStrategy = strategy; } 1508 1509 // suspend effect of the given type 1510 void setEffectSuspended_l(const effect_uuid_t *type, 1511 bool suspend); 1512 // suspend all eligible effects 1513 void setEffectSuspendedAll_l(bool suspend); 1514 // check if effects should be suspend or restored when a given effect is enable or disabled 1515 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1516 bool enabled); 1517 1518 status_t dump(int fd, const Vector<String16>& args); 1519 1520 protected: 1521 friend class AudioFlinger; // for mThread, mEffects 1522 EffectChain(const EffectChain&); 1523 EffectChain& operator =(const EffectChain&); 1524 1525 class SuspendedEffectDesc : public RefBase { 1526 public: 1527 SuspendedEffectDesc() : mRefCount(0) {} 1528 1529 int mRefCount; 1530 effect_uuid_t mType; 1531 wp<EffectModule> mEffect; 1532 }; 1533 1534 // get a list of effect modules to suspend when an effect of the type 1535 // passed is enabled. 1536 void getSuspendEligibleEffects(Vector< sp<EffectModule> > &effects); 1537 1538 // get an effect module if it is currently enable 1539 sp<EffectModule> getEffectIfEnabled(const effect_uuid_t *type); 1540 // true if the effect whose descriptor is passed can be suspended 1541 // OEMs can modify the rules implemented in this method to exclude specific effect 1542 // types or implementations from the suspend/restore mechanism. 1543 bool isEffectEligibleForSuspend(const effect_descriptor_t& desc); 1544 1545 wp<ThreadBase> mThread; // parent mixer thread 1546 Mutex mLock; // mutex protecting effect list 1547 Vector<sp<EffectModule> > mEffects; // list of effect modules 1548 int mSessionId; // audio session ID 1549 int16_t *mInBuffer; // chain input buffer 1550 int16_t *mOutBuffer; // chain output buffer 1551 volatile int32_t mActiveTrackCnt; // number of active tracks connected 1552 volatile int32_t mTrackCnt; // number of tracks connected 1553 int32_t mTailBufferCount; // current effect tail buffer count 1554 int32_t mMaxTailBuffers; // maximum effect tail buffers 1555 bool mOwnInBuffer; // true if the chain owns its input buffer 1556 int mVolumeCtrlIdx; // index of insert effect having control over volume 1557 uint32_t mLeftVolume; // previous volume on left channel 1558 uint32_t mRightVolume; // previous volume on right channel 1559 uint32_t mNewLeftVolume; // new volume on left channel 1560 uint32_t mNewRightVolume; // new volume on right channel 1561 uint32_t mStrategy; // strategy for this effect chain 1562 // mSuspendedEffects lists all effect currently suspended in the chain 1563 // use effect type UUID timelow field as key. There is no real risk of identical 1564 // timeLow fields among effect type UUIDs. 1565 KeyedVector< int, sp<SuspendedEffectDesc> > mSuspendedEffects; 1566 }; 1567 1568 // AudioStreamOut and AudioStreamIn are immutable, so their fields are const. 1569 // For emphasis, we could also make all pointers to them be "const *", 1570 // but that would clutter the code unnecessarily. 1571 1572 struct AudioStreamOut { 1573 audio_hw_device_t* const hwDev; 1574 audio_stream_out_t* const stream; 1575 1576 AudioStreamOut(audio_hw_device_t *dev, audio_stream_out_t *out) : 1577 hwDev(dev), stream(out) {} 1578 }; 1579 1580 struct AudioStreamIn { 1581 audio_hw_device_t* const hwDev; 1582 audio_stream_in_t* const stream; 1583 1584 AudioStreamIn(audio_hw_device_t *dev, audio_stream_in_t *in) : 1585 hwDev(dev), stream(in) {} 1586 }; 1587 1588 // for mAudioSessionRefs only 1589 struct AudioSessionRef { 1590 AudioSessionRef(int sessionid, pid_t pid) : 1591 mSessionid(sessionid), mPid(pid), mCnt(1) {} 1592 const int mSessionid; 1593 const pid_t mPid; 1594 int mCnt; 1595 }; 1596 1597 enum master_volume_support { 1598 // MVS_NONE: 1599 // Audio HAL has no support for master volume, either setting or 1600 // getting. All master volume control must be implemented in SW by the 1601 // AudioFlinger mixing core. 1602 MVS_NONE, 1603 1604 // MVS_SETONLY: 1605 // Audio HAL has support for setting master volume, but not for getting 1606 // master volume (original HAL design did not include a getter). 1607 // AudioFlinger needs to keep track of the last set master volume in 1608 // addition to needing to set an initial, default, master volume at HAL 1609 // load time. 1610 MVS_SETONLY, 1611 1612 // MVS_FULL: 1613 // Audio HAL has support both for setting and getting master volume. 1614 // AudioFlinger should send all set and get master volume requests 1615 // directly to the HAL. 1616 MVS_FULL, 1617 }; 1618 1619 mutable Mutex mLock; 1620 1621 DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client() 1622 1623 mutable Mutex mHardwareLock; 1624 1625 // These two fields are immutable after onFirstRef(), so no lock needed to access 1626 audio_hw_device_t* mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL 1627 Vector<audio_hw_device_t*> mAudioHwDevs; 1628 1629 // for dump, indicates which hardware operation is currently in progress (but not stream ops) 1630 enum hardware_call_state { 1631 AUDIO_HW_IDLE = 0, // no operation in progress 1632 AUDIO_HW_INIT, // init_check 1633 AUDIO_HW_OUTPUT_OPEN, // open_output_stream 1634 AUDIO_HW_OUTPUT_CLOSE, // unused 1635 AUDIO_HW_INPUT_OPEN, // unused 1636 AUDIO_HW_INPUT_CLOSE, // unused 1637 AUDIO_HW_STANDBY, // unused 1638 AUDIO_HW_SET_MASTER_VOLUME, // set_master_volume 1639 AUDIO_HW_GET_ROUTING, // unused 1640 AUDIO_HW_SET_ROUTING, // unused 1641 AUDIO_HW_GET_MODE, // unused 1642 AUDIO_HW_SET_MODE, // set_mode 1643 AUDIO_HW_GET_MIC_MUTE, // get_mic_mute 1644 AUDIO_HW_SET_MIC_MUTE, // set_mic_mute 1645 AUDIO_HW_SET_VOICE_VOLUME, // set_voice_volume 1646 AUDIO_HW_SET_PARAMETER, // set_parameters 1647 AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size 1648 AUDIO_HW_GET_MASTER_VOLUME, // get_master_volume 1649 AUDIO_HW_GET_PARAMETER, // get_parameters 1650 }; 1651 1652 mutable hardware_call_state mHardwareStatus; // for dump only 1653 1654 1655 DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads; 1656 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 1657 1658 // both are protected by mLock 1659 float mMasterVolume; 1660 float mMasterVolumeSW; 1661 master_volume_support mMasterVolumeSupportLvl; 1662 bool mMasterMute; 1663 1664 DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads; 1665 1666 DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients; 1667 volatile int32_t mNextUniqueId; // updated by android_atomic_inc 1668 audio_mode_t mMode; 1669 bool mBtNrecIsOff; 1670 1671 // protected by mLock 1672 Vector<AudioSessionRef*> mAudioSessionRefs; 1673 1674 float masterVolume_l() const; 1675 float masterVolumeSW_l() const { return mMasterVolumeSW; } 1676 bool masterMute_l() const { return mMasterMute; } 1677 1678private: 1679 sp<Client> registerPid_l(pid_t pid); // always returns non-0 1680 1681}; 1682 1683 1684// ---------------------------------------------------------------------------- 1685 1686}; // namespace android 1687 1688#endif // ANDROID_AUDIO_FLINGER_H 1689