AudioFlinger.h revision a075db4ff9b086ac2885df77bb6da0869293df92
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 207private: 208 audio_mode_t getMode() const { return mMode; } 209 210 bool btNrecIsOff() const { return mBtNrecIsOff; } 211 212 AudioFlinger(); 213 virtual ~AudioFlinger(); 214 215 // call in any IAudioFlinger method that accesses mPrimaryHardwareDev 216 status_t initCheck() const { return mPrimaryHardwareDev == NULL ? NO_INIT : NO_ERROR; } 217 218 // RefBase 219 virtual void onFirstRef(); 220 221 audio_hw_device_t* findSuitableHwDev_l(uint32_t devices); 222 void purgeStaleEffects_l(); 223 224 // standby delay for MIXER and DUPLICATING playback threads is read from property 225 // ro.audio.flinger_standbytime_ms or defaults to kDefaultStandbyTimeInNsecs 226 static nsecs_t mStandbyTimeInNsecs; 227 228 // Internal dump utilites. 229 status_t dumpPermissionDenial(int fd, const Vector<String16>& args); 230 status_t dumpClients(int fd, const Vector<String16>& args); 231 status_t dumpInternals(int fd, const Vector<String16>& args); 232 233 // --- Client --- 234 class Client : public RefBase { 235 public: 236 Client(const sp<AudioFlinger>& audioFlinger, pid_t pid); 237 virtual ~Client(); 238 sp<MemoryDealer> heap() const; 239 pid_t pid() const { return mPid; } 240 sp<AudioFlinger> audioFlinger() const { return mAudioFlinger; } 241 242 bool reserveTimedTrack(); 243 void releaseTimedTrack(); 244 245 private: 246 Client(const Client&); 247 Client& operator = (const Client&); 248 const sp<AudioFlinger> mAudioFlinger; 249 const sp<MemoryDealer> mMemoryDealer; 250 const pid_t mPid; 251 252 Mutex mTimedTrackLock; 253 int mTimedTrackCount; 254 }; 255 256 // --- Notification Client --- 257 class NotificationClient : public IBinder::DeathRecipient { 258 public: 259 NotificationClient(const sp<AudioFlinger>& audioFlinger, 260 const sp<IAudioFlingerClient>& client, 261 pid_t pid); 262 virtual ~NotificationClient(); 263 264 sp<IAudioFlingerClient> audioFlingerClient() const { return mAudioFlingerClient; } 265 266 // IBinder::DeathRecipient 267 virtual void binderDied(const wp<IBinder>& who); 268 269 private: 270 NotificationClient(const NotificationClient&); 271 NotificationClient& operator = (const NotificationClient&); 272 273 const sp<AudioFlinger> mAudioFlinger; 274 const pid_t mPid; 275 const sp<IAudioFlingerClient> mAudioFlingerClient; 276 }; 277 278 class TrackHandle; 279 class RecordHandle; 280 class RecordThread; 281 class PlaybackThread; 282 class MixerThread; 283 class DirectOutputThread; 284 class DuplicatingThread; 285 class Track; 286 class RecordTrack; 287 class EffectModule; 288 class EffectHandle; 289 class EffectChain; 290 struct AudioStreamOut; 291 struct AudioStreamIn; 292 293 class ThreadBase : public Thread { 294 public: 295 296 enum type_t { 297 MIXER, // Thread class is MixerThread 298 DIRECT, // Thread class is DirectOutputThread 299 DUPLICATING, // Thread class is DuplicatingThread 300 RECORD // Thread class is RecordThread 301 }; 302 303 ThreadBase (const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, uint32_t device, type_t type); 304 virtual ~ThreadBase(); 305 306 status_t dumpBase(int fd, const Vector<String16>& args); 307 status_t dumpEffectChains(int fd, const Vector<String16>& args); 308 309 void clearPowerManager(); 310 311 // base for record and playback 312 class TrackBase : public AudioBufferProvider, public RefBase { 313 314 public: 315 enum track_state { 316 IDLE, 317 TERMINATED, 318 // These are order-sensitive; do not change order without reviewing the impact. 319 // In particular there are assumptions about > STOPPED. 320 STOPPED, 321 RESUMING, 322 ACTIVE, 323 PAUSING, 324 PAUSED 325 }; 326 327 TrackBase(ThreadBase *thread, 328 const sp<Client>& client, 329 uint32_t sampleRate, 330 audio_format_t format, 331 uint32_t channelMask, 332 int frameCount, 333 const sp<IMemory>& sharedBuffer, 334 int sessionId); 335 virtual ~TrackBase(); 336 337 virtual status_t start(pid_t tid) = 0; 338 virtual void stop() = 0; 339 sp<IMemory> getCblk() const { return mCblkMemory; } 340 audio_track_cblk_t* cblk() const { return mCblk; } 341 int sessionId() const { return mSessionId; } 342 343 protected: 344 TrackBase(const TrackBase&); 345 TrackBase& operator = (const TrackBase&); 346 347 // AudioBufferProvider interface 348 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) = 0; 349 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 350 351 audio_format_t format() const { 352 return mFormat; 353 } 354 355 int channelCount() const { return mChannelCount; } 356 357 uint32_t channelMask() const { return mChannelMask; } 358 359 int sampleRate() const; // FIXME inline after cblk sr moved 360 361 void* getBuffer(uint32_t offset, uint32_t frames) const; 362 363 bool isStopped() const { 364 return mState == STOPPED; 365 } 366 367 bool isTerminated() const { 368 return mState == TERMINATED; 369 } 370 371 bool step(); 372 void reset(); 373 374 const wp<ThreadBase> mThread; 375 /*const*/ sp<Client> mClient; // see explanation at ~TrackBase() why not const 376 sp<IMemory> mCblkMemory; 377 audio_track_cblk_t* mCblk; 378 void* mBuffer; 379 void* mBufferEnd; 380 uint32_t mFrameCount; 381 // we don't really need a lock for these 382 track_state mState; 383 const audio_format_t mFormat; 384 bool mStepServerFailed; 385 const int mSessionId; 386 uint8_t mChannelCount; 387 uint32_t mChannelMask; 388 }; 389 390 class ConfigEvent { 391 public: 392 ConfigEvent() : mEvent(0), mParam(0) {} 393 394 int mEvent; 395 int mParam; 396 }; 397 398 class PMDeathRecipient : public IBinder::DeathRecipient { 399 public: 400 PMDeathRecipient(const wp<ThreadBase>& thread) : mThread(thread) {} 401 virtual ~PMDeathRecipient() {} 402 403 // IBinder::DeathRecipient 404 virtual void binderDied(const wp<IBinder>& who); 405 406 private: 407 PMDeathRecipient(const PMDeathRecipient&); 408 PMDeathRecipient& operator = (const PMDeathRecipient&); 409 410 wp<ThreadBase> mThread; 411 }; 412 413 virtual status_t initCheck() const = 0; 414 type_t type() const { return mType; } 415 uint32_t sampleRate() const { return mSampleRate; } 416 int channelCount() const { return mChannelCount; } 417 audio_format_t format() const { return mFormat; } 418 size_t frameCount() const { return mFrameCount; } 419 void wakeUp() { mWaitWorkCV.broadcast(); } 420 // Should be "virtual status_t requestExitAndWait()" and override same 421 // method in Thread, but Thread::requestExitAndWait() is not yet virtual. 422 void exit(); 423 virtual bool checkForNewParameters_l() = 0; 424 virtual status_t setParameters(const String8& keyValuePairs); 425 virtual String8 getParameters(const String8& keys) = 0; 426 virtual void audioConfigChanged_l(int event, int param = 0) = 0; 427 void sendConfigEvent(int event, int param = 0); 428 void sendConfigEvent_l(int event, int param = 0); 429 void processConfigEvents(); 430 audio_io_handle_t id() const { return mId;} 431 bool standby() const { return mStandby; } 432 uint32_t device() const { return mDevice; } 433 virtual audio_stream_t* stream() = 0; 434 435 sp<EffectHandle> createEffect_l( 436 const sp<AudioFlinger::Client>& client, 437 const sp<IEffectClient>& effectClient, 438 int32_t priority, 439 int sessionId, 440 effect_descriptor_t *desc, 441 int *enabled, 442 status_t *status); 443 void disconnectEffect(const sp< EffectModule>& effect, 444 const wp<EffectHandle>& handle, 445 bool unpinIfLast); 446 447 // return values for hasAudioSession (bit field) 448 enum effect_state { 449 EFFECT_SESSION = 0x1, // the audio session corresponds to at least one 450 // effect 451 TRACK_SESSION = 0x2 // the audio session corresponds to at least one 452 // track 453 }; 454 455 // get effect chain corresponding to session Id. 456 sp<EffectChain> getEffectChain(int sessionId); 457 // same as getEffectChain() but must be called with ThreadBase mutex locked 458 sp<EffectChain> getEffectChain_l(int sessionId); 459 // add an effect chain to the chain list (mEffectChains) 460 virtual status_t addEffectChain_l(const sp<EffectChain>& chain) = 0; 461 // remove an effect chain from the chain list (mEffectChains) 462 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain) = 0; 463 // lock all effect chains Mutexes. Must be called before releasing the 464 // ThreadBase mutex before processing the mixer and effects. This guarantees the 465 // integrity of the chains during the process. 466 // Also sets the parameter 'effectChains' to current value of mEffectChains. 467 void lockEffectChains_l(Vector< sp<EffectChain> >& effectChains); 468 // unlock effect chains after process 469 void unlockEffectChains(const Vector< sp<EffectChain> >& effectChains); 470 // set audio mode to all effect chains 471 void setMode(audio_mode_t mode); 472 // get effect module with corresponding ID on specified audio session 473 sp<AudioFlinger::EffectModule> getEffect_l(int sessionId, int effectId); 474 // add and effect module. Also creates the effect chain is none exists for 475 // the effects audio session 476 status_t addEffect_l(const sp< EffectModule>& effect); 477 // remove and effect module. Also removes the effect chain is this was the last 478 // effect 479 void removeEffect_l(const sp< EffectModule>& effect); 480 // detach all tracks connected to an auxiliary effect 481 virtual void detachAuxEffect_l(int effectId) {} 482 // returns either EFFECT_SESSION if effects on this audio session exist in one 483 // chain, or TRACK_SESSION if tracks on this audio session exist, or both 484 virtual uint32_t hasAudioSession(int sessionId) = 0; 485 // the value returned by default implementation is not important as the 486 // strategy is only meaningful for PlaybackThread which implements this method 487 virtual uint32_t getStrategyForSession_l(int sessionId) { return 0; } 488 489 // suspend or restore effect according to the type of effect passed. a NULL 490 // type pointer means suspend all effects in the session 491 void setEffectSuspended(const effect_uuid_t *type, 492 bool suspend, 493 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 494 // check if some effects must be suspended/restored when an effect is enabled 495 // or disabled 496 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 497 bool enabled, 498 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 499 void checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 500 bool enabled, 501 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 502 mutable Mutex mLock; 503 504 protected: 505 506 // entry describing an effect being suspended in mSuspendedSessions keyed vector 507 class SuspendedSessionDesc : public RefBase { 508 public: 509 SuspendedSessionDesc() : mRefCount(0) {} 510 511 int mRefCount; // number of active suspend requests 512 effect_uuid_t mType; // effect type UUID 513 }; 514 515 void acquireWakeLock(); 516 void acquireWakeLock_l(); 517 void releaseWakeLock(); 518 void releaseWakeLock_l(); 519 void setEffectSuspended_l(const effect_uuid_t *type, 520 bool suspend, 521 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 522 // updated mSuspendedSessions when an effect suspended or restored 523 void updateSuspendedSessions_l(const effect_uuid_t *type, 524 bool suspend, 525 int sessionId); 526 // check if some effects must be suspended when an effect chain is added 527 void checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain); 528 529 friend class AudioFlinger; // for mEffectChains 530 531 const type_t mType; 532 533 // Used by parameters, config events, addTrack_l, exit 534 Condition mWaitWorkCV; 535 536 const sp<AudioFlinger> mAudioFlinger; 537 uint32_t mSampleRate; 538 size_t mFrameCount; 539 uint32_t mChannelMask; 540 uint16_t mChannelCount; 541 size_t mFrameSize; 542 audio_format_t mFormat; 543 544 // Parameter sequence by client: binder thread calling setParameters(): 545 // 1. Lock mLock 546 // 2. Append to mNewParameters 547 // 3. mWaitWorkCV.signal 548 // 4. mParamCond.waitRelative with timeout 549 // 5. read mParamStatus 550 // 6. mWaitWorkCV.signal 551 // 7. Unlock 552 // 553 // Parameter sequence by server: threadLoop calling checkForNewParameters_l(): 554 // 1. Lock mLock 555 // 2. If there is an entry in mNewParameters proceed ... 556 // 2. Read first entry in mNewParameters 557 // 3. Process 558 // 4. Remove first entry from mNewParameters 559 // 5. Set mParamStatus 560 // 6. mParamCond.signal 561 // 7. mWaitWorkCV.wait with timeout (this is to avoid overwriting mParamStatus) 562 // 8. Unlock 563 Condition mParamCond; 564 Vector<String8> mNewParameters; 565 status_t mParamStatus; 566 567 Vector<ConfigEvent> mConfigEvents; 568 bool mStandby; 569 const audio_io_handle_t mId; 570 Vector< sp<EffectChain> > mEffectChains; 571 uint32_t mDevice; // output device for PlaybackThread 572 // input + output devices for RecordThread 573 static const int kNameLength = 16; // prctl(PR_SET_NAME) limit 574 char mName[kNameLength]; 575 sp<IPowerManager> mPowerManager; 576 sp<IBinder> mWakeLockToken; 577 const sp<PMDeathRecipient> mDeathRecipient; 578 // list of suspended effects per session and per type. The first vector is 579 // keyed by session ID, the second by type UUID timeLow field 580 KeyedVector< int, KeyedVector< int, sp<SuspendedSessionDesc> > > mSuspendedSessions; 581 }; 582 583 struct stream_type_t { 584 stream_type_t() 585 : volume(1.0f), 586 mute(false), 587 valid(true) 588 { 589 } 590 float volume; 591 bool mute; 592 bool valid; 593 }; 594 595 // --- PlaybackThread --- 596 class PlaybackThread : public ThreadBase { 597 public: 598 599 enum mixer_state { 600 MIXER_IDLE, // no active tracks 601 MIXER_TRACKS_ENABLED, // at least one active track, but no track has any data ready 602 MIXER_TRACKS_READY // at least one active track, and at least one track has data 603 // standby mode does not have an enum value 604 // suspend by audio policy manager is orthogonal to mixer state 605 }; 606 607 // playback track 608 class Track : public TrackBase { 609 public: 610 Track( PlaybackThread *thread, 611 const sp<Client>& client, 612 audio_stream_type_t streamType, 613 uint32_t sampleRate, 614 audio_format_t format, 615 uint32_t channelMask, 616 int frameCount, 617 const sp<IMemory>& sharedBuffer, 618 int sessionId); 619 virtual ~Track(); 620 621 void dump(char* buffer, size_t size); 622 virtual status_t start(pid_t tid); 623 virtual void stop(); 624 void pause(); 625 626 void flush(); 627 void destroy(); 628 void mute(bool); 629 int name() const { 630 return mName; 631 } 632 633 audio_stream_type_t streamType() const { 634 return mStreamType; 635 } 636 status_t attachAuxEffect(int EffectId); 637 void setAuxBuffer(int EffectId, int32_t *buffer); 638 int32_t *auxBuffer() const { return mAuxBuffer; } 639 void setMainBuffer(int16_t *buffer) { mMainBuffer = buffer; } 640 int16_t *mainBuffer() const { return mMainBuffer; } 641 int auxEffectId() const { return mAuxEffectId; } 642 643 protected: 644 // for numerous 645 friend class PlaybackThread; 646 friend class MixerThread; 647 friend class DirectOutputThread; 648 649 Track(const Track&); 650 Track& operator = (const Track&); 651 652 // AudioBufferProvider interface 653 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts = kInvalidPTS); 654 // releaseBuffer() not overridden 655 656 virtual uint32_t framesReady() const; 657 658 bool isMuted() const { return mMute; } 659 bool isPausing() const { 660 return mState == PAUSING; 661 } 662 bool isPaused() const { 663 return mState == PAUSED; 664 } 665 bool isReady() const; 666 void setPaused() { mState = PAUSED; } 667 void reset(); 668 669 bool isOutputTrack() const { 670 return (mStreamType == AUDIO_STREAM_CNT); 671 } 672 673 public: 674 virtual bool isTimedTrack() const { return false; } 675 protected: 676 677 // we don't really need a lock for these 678 volatile bool mMute; 679 // FILLED state is used for suppressing volume ramp at begin of playing 680 enum {FS_FILLING, FS_FILLED, FS_ACTIVE}; 681 mutable uint8_t mFillingUpStatus; 682 int8_t mRetryCount; 683 sp<IMemory> mSharedBuffer; 684 bool mResetDone; 685 audio_stream_type_t mStreamType; 686 int mName; 687 int16_t *mMainBuffer; 688 int32_t *mAuxBuffer; 689 int mAuxEffectId; 690 bool mHasVolumeController; 691 }; // end of Track 692 693 class TimedTrack : public Track { 694 public: 695 static sp<TimedTrack> create(PlaybackThread *thread, 696 const sp<Client>& client, 697 audio_stream_type_t streamType, 698 uint32_t sampleRate, 699 audio_format_t format, 700 uint32_t channelMask, 701 int frameCount, 702 const sp<IMemory>& sharedBuffer, 703 int sessionId); 704 ~TimedTrack(); 705 706 class TimedBuffer { 707 public: 708 TimedBuffer(); 709 TimedBuffer(const sp<IMemory>& buffer, int64_t pts); 710 const sp<IMemory>& buffer() const { return mBuffer; } 711 int64_t pts() const { return mPTS; } 712 int position() const { return mPosition; } 713 void setPosition(int pos) { mPosition = pos; } 714 private: 715 sp<IMemory> mBuffer; 716 int64_t mPTS; 717 int mPosition; 718 }; 719 720 virtual bool isTimedTrack() const { return true; } 721 722 virtual uint32_t framesReady() const; 723 724 // AudioBufferProvider interface 725 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts); 726 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 727 728 void timedYieldSamples(AudioBufferProvider::Buffer* buffer); 729 void timedYieldSilence(uint32_t numFrames, 730 AudioBufferProvider::Buffer* buffer); 731 732 status_t allocateTimedBuffer(size_t size, 733 sp<IMemory>* buffer); 734 status_t queueTimedBuffer(const sp<IMemory>& buffer, 735 int64_t pts); 736 status_t setMediaTimeTransform(const LinearTransform& xform, 737 TimedAudioTrack::TargetTimeline target); 738 void trimTimedBufferQueue_l(); 739 740 private: 741 TimedTrack(PlaybackThread *thread, 742 const sp<Client>& client, 743 audio_stream_type_t streamType, 744 uint32_t sampleRate, 745 audio_format_t format, 746 uint32_t channelMask, 747 int frameCount, 748 const sp<IMemory>& sharedBuffer, 749 int sessionId); 750 751 uint64_t mLocalTimeFreq; 752 LinearTransform mLocalTimeToSampleTransform; 753 sp<MemoryDealer> mTimedMemoryDealer; 754 Vector<TimedBuffer> mTimedBufferQueue; 755 uint8_t* mTimedSilenceBuffer; 756 uint32_t mTimedSilenceBufferSize; 757 mutable Mutex mTimedBufferQueueLock; 758 bool mTimedAudioOutputOnTime; 759 CCHelper mCCHelper; 760 761 Mutex mMediaTimeTransformLock; 762 LinearTransform mMediaTimeTransform; 763 bool mMediaTimeTransformValid; 764 TimedAudioTrack::TargetTimeline mMediaTimeTransformTarget; 765 }; 766 767 768 // playback track 769 class OutputTrack : public Track { 770 public: 771 772 class Buffer: public AudioBufferProvider::Buffer { 773 public: 774 int16_t *mBuffer; 775 }; 776 777 OutputTrack(PlaybackThread *thread, 778 DuplicatingThread *sourceThread, 779 uint32_t sampleRate, 780 audio_format_t format, 781 uint32_t channelMask, 782 int frameCount); 783 virtual ~OutputTrack(); 784 785 virtual status_t start(pid_t tid); 786 virtual void stop(); 787 bool write(int16_t* data, uint32_t frames); 788 bool bufferQueueEmpty() const { return mBufferQueue.size() == 0; } 789 bool isActive() const { return mActive; } 790 const wp<ThreadBase>& thread() const { return mThread; } 791 792 private: 793 794 enum { 795 NO_MORE_BUFFERS = 0x80000001, // same in AudioTrack.h, ok to be different value 796 }; 797 798 status_t obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs); 799 void clearBufferQueue(); 800 801 // Maximum number of pending buffers allocated by OutputTrack::write() 802 static const uint8_t kMaxOverFlowBuffers = 10; 803 804 Vector < Buffer* > mBufferQueue; 805 AudioBufferProvider::Buffer mOutBuffer; 806 bool mActive; 807 DuplicatingThread* const mSourceThread; // for waitTimeMs() in write() 808 }; // end of OutputTrack 809 810 PlaybackThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 811 audio_io_handle_t id, uint32_t device, type_t type); 812 virtual ~PlaybackThread(); 813 814 status_t dump(int fd, const Vector<String16>& args); 815 816 // Thread virtuals 817 virtual status_t readyToRun(); 818 virtual bool threadLoop(); 819 820 // RefBase 821 virtual void onFirstRef(); 822 823protected: 824 // Code snippets that were lifted up out of threadLoop() 825 virtual void threadLoop_mix() = 0; 826 virtual void threadLoop_sleepTime() = 0; 827 virtual void threadLoop_write(); 828 virtual void threadLoop_standby(); 829 830 // prepareTracks_l reads and writes mActiveTracks, and also returns the 831 // pending set of tracks to remove via Vector 'tracksToRemove'. The caller is 832 // responsible for clearing or destroying this Vector later on, when it 833 // is safe to do so. That will drop the final ref count and destroy the tracks. 834 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove) = 0; 835 836public: 837 838 virtual status_t initCheck() const { return (mOutput == NULL) ? NO_INIT : NO_ERROR; } 839 840 // return estimated latency in milliseconds, as reported by HAL 841 uint32_t latency() const; 842 843 void setMasterVolume(float value); 844 void setMasterMute(bool muted); 845 846 void setStreamVolume(audio_stream_type_t stream, float value); 847 void setStreamMute(audio_stream_type_t stream, bool muted); 848 849 float streamVolume(audio_stream_type_t stream) const; 850 851 sp<Track> createTrack_l( 852 const sp<AudioFlinger::Client>& client, 853 audio_stream_type_t streamType, 854 uint32_t sampleRate, 855 audio_format_t format, 856 uint32_t channelMask, 857 int frameCount, 858 const sp<IMemory>& sharedBuffer, 859 int sessionId, 860 bool isTimed, 861 status_t *status); 862 863 AudioStreamOut* getOutput() const; 864 AudioStreamOut* clearOutput(); 865 virtual audio_stream_t* stream(); 866 867 void suspend() { mSuspended++; } 868 void restore() { if (mSuspended > 0) mSuspended--; } 869 bool isSuspended() const { return (mSuspended > 0); } 870 virtual String8 getParameters(const String8& keys); 871 virtual void audioConfigChanged_l(int event, int param = 0); 872 status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames); 873 int16_t *mixBuffer() const { return mMixBuffer; }; 874 875 virtual void detachAuxEffect_l(int effectId); 876 status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track> track, 877 int EffectId); 878 status_t attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track> track, 879 int EffectId); 880 881 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 882 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 883 virtual uint32_t hasAudioSession(int sessionId); 884 virtual uint32_t getStrategyForSession_l(int sessionId); 885 886 void setStreamValid(audio_stream_type_t streamType, bool valid); 887 888 protected: 889 int16_t* mMixBuffer; 890 uint32_t mSuspended; // suspend count, > 0 means suspended 891 int mBytesWritten; 892 private: 893 // mMasterMute is in both PlaybackThread and in AudioFlinger. When a 894 // PlaybackThread needs to find out if master-muted, it checks it's local 895 // copy rather than the one in AudioFlinger. This optimization saves a lock. 896 bool mMasterMute; 897 void setMasterMute_l(bool muted) { mMasterMute = muted; } 898 protected: 899 SortedVector< wp<Track> > mActiveTracks; 900 901 virtual int getTrackName_l() = 0; 902 virtual void deleteTrackName_l(int name) = 0; 903 virtual uint32_t activeSleepTimeUs(); 904 virtual uint32_t idleSleepTimeUs() = 0; 905 virtual uint32_t suspendSleepTimeUs() = 0; 906 907 // Code snippets that are temporarily lifted up out of threadLoop() until the merge 908 void checkSilentMode_l(); 909 910 // Non-trivial for DUPLICATING only 911 virtual void saveOutputTracks() { } 912 virtual void clearOutputTracks() { } 913 914 // Cache various calculated values, at threadLoop() entry and after a parameter change 915 virtual void cacheParameters_l(); 916 917 private: 918 919 friend class AudioFlinger; // for numerous 920 921 PlaybackThread(const Client&); 922 PlaybackThread& operator = (const PlaybackThread&); 923 924 status_t addTrack_l(const sp<Track>& track); 925 void destroyTrack_l(const sp<Track>& track); 926 void removeTrack_l(const sp<Track>& track); 927 928 void readOutputParameters(); 929 930 virtual status_t dumpInternals(int fd, const Vector<String16>& args); 931 status_t dumpTracks(int fd, const Vector<String16>& args); 932 933 SortedVector< sp<Track> > mTracks; 934 // mStreamTypes[] uses 1 additional stream type internally for the OutputTrack used by DuplicatingThread 935 stream_type_t mStreamTypes[AUDIO_STREAM_CNT + 1]; 936 AudioStreamOut *mOutput; 937 float mMasterVolume; 938 nsecs_t mLastWriteTime; 939 int mNumWrites; 940 int mNumDelayedWrites; 941 bool mInWrite; 942 943 // FIXME rename these former local variables of threadLoop to standard "m" names 944 nsecs_t standbyTime; 945 size_t mixBufferSize; 946 947 // cached copies of activeSleepTimeUs() and idleSleepTimeUs() made by cacheParameters_l() 948 uint32_t activeSleepTime; 949 uint32_t idleSleepTime; 950 951 uint32_t sleepTime; 952 953 // mixer status returned by prepareTracks_l() 954 mixer_state mMixerStatus; // current cycle 955 mixer_state mPrevMixerStatus; // previous cycle 956 957 // FIXME move these declarations into the specific sub-class that needs them 958 // MIXER only 959 bool longStandbyExit; 960 uint32_t sleepTimeShift; 961 962 // same as AudioFlinger::mStandbyTimeInNsecs except for DIRECT which uses a shorter value 963 nsecs_t standbyDelay; 964 965 // MIXER only 966 nsecs_t maxPeriod; 967 968 // DUPLICATING only 969 uint32_t writeFrames; 970 }; 971 972 class MixerThread : public PlaybackThread { 973 public: 974 MixerThread (const sp<AudioFlinger>& audioFlinger, 975 AudioStreamOut* output, 976 audio_io_handle_t id, 977 uint32_t device, 978 type_t type = MIXER); 979 virtual ~MixerThread(); 980 981 // Thread virtuals 982 983 void invalidateTracks(audio_stream_type_t streamType); 984 virtual bool checkForNewParameters_l(); 985 virtual status_t dumpInternals(int fd, const Vector<String16>& args); 986 987 protected: 988 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 989 virtual int getTrackName_l(); 990 virtual void deleteTrackName_l(int name); 991 virtual uint32_t idleSleepTimeUs(); 992 virtual uint32_t suspendSleepTimeUs(); 993 virtual void cacheParameters_l(); 994 995 // threadLoop snippets 996 virtual void threadLoop_mix(); 997 virtual void threadLoop_sleepTime(); 998 999 AudioMixer* mAudioMixer; 1000 }; 1001 1002 class DirectOutputThread : public PlaybackThread { 1003 public: 1004 1005 DirectOutputThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1006 audio_io_handle_t id, uint32_t device); 1007 virtual ~DirectOutputThread(); 1008 1009 // Thread virtuals 1010 1011 virtual bool checkForNewParameters_l(); 1012 1013 protected: 1014 virtual int getTrackName_l(); 1015 virtual void deleteTrackName_l(int name); 1016 virtual uint32_t activeSleepTimeUs(); 1017 virtual uint32_t idleSleepTimeUs(); 1018 virtual uint32_t suspendSleepTimeUs(); 1019 virtual void cacheParameters_l(); 1020 1021 // threadLoop snippets 1022 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1023 virtual void threadLoop_mix(); 1024 virtual void threadLoop_sleepTime(); 1025 1026 // volumes last sent to audio HAL with stream->set_volume() 1027 // FIXME use standard representation and names 1028 float mLeftVolFloat; 1029 float mRightVolFloat; 1030 uint16_t mLeftVolShort; 1031 uint16_t mRightVolShort; 1032 1033 // FIXME rename these former local variables of threadLoop to standard names 1034 // next 3 were local to the while !exitingPending loop 1035 bool rampVolume; 1036 uint16_t leftVol; 1037 uint16_t rightVol; 1038 1039private: 1040 // prepareTracks_l() tells threadLoop_mix() the name of the single active track 1041 sp<Track> mActiveTrack; 1042 }; 1043 1044 class DuplicatingThread : public MixerThread { 1045 public: 1046 DuplicatingThread (const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread, 1047 audio_io_handle_t id); 1048 virtual ~DuplicatingThread(); 1049 1050 // Thread virtuals 1051 void addOutputTrack(MixerThread* thread); 1052 void removeOutputTrack(MixerThread* thread); 1053 uint32_t waitTimeMs() { return mWaitTimeMs; } 1054 protected: 1055 virtual uint32_t activeSleepTimeUs(); 1056 1057 private: 1058 bool outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks); 1059 protected: 1060 // threadLoop snippets 1061 virtual void threadLoop_mix(); 1062 virtual void threadLoop_sleepTime(); 1063 virtual void threadLoop_write(); 1064 virtual void threadLoop_standby(); 1065 virtual void cacheParameters_l(); 1066 1067 private: 1068 // called from threadLoop, addOutputTrack, removeOutputTrack 1069 virtual void updateWaitTime_l(); 1070 protected: 1071 virtual void saveOutputTracks(); 1072 virtual void clearOutputTracks(); 1073 private: 1074 1075 uint32_t mWaitTimeMs; 1076 SortedVector < sp<OutputTrack> > outputTracks; 1077 SortedVector < sp<OutputTrack> > mOutputTracks; 1078 }; 1079 1080 PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const; 1081 MixerThread *checkMixerThread_l(audio_io_handle_t output) const; 1082 RecordThread *checkRecordThread_l(audio_io_handle_t input) const; 1083 // no range check, AudioFlinger::mLock held 1084 bool streamMute_l(audio_stream_type_t stream) const 1085 { return mStreamTypes[stream].mute; } 1086 // no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held 1087 float streamVolume_l(audio_stream_type_t stream) const 1088 { return mStreamTypes[stream].volume; } 1089 void audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2); 1090 1091 // allocate an audio_io_handle_t, session ID, or effect ID 1092 uint32_t nextUniqueId(); 1093 1094 status_t moveEffectChain_l(int sessionId, 1095 PlaybackThread *srcThread, 1096 PlaybackThread *dstThread, 1097 bool reRegister); 1098 // return thread associated with primary hardware device, or NULL 1099 PlaybackThread *primaryPlaybackThread_l() const; 1100 uint32_t primaryOutputDevice_l() const; 1101 1102 // server side of the client's IAudioTrack 1103 class TrackHandle : public android::BnAudioTrack { 1104 public: 1105 TrackHandle(const sp<PlaybackThread::Track>& track); 1106 virtual ~TrackHandle(); 1107 virtual sp<IMemory> getCblk() const; 1108 virtual status_t start(pid_t tid); 1109 virtual void stop(); 1110 virtual void flush(); 1111 virtual void mute(bool); 1112 virtual void pause(); 1113 virtual status_t attachAuxEffect(int effectId); 1114 virtual status_t allocateTimedBuffer(size_t size, 1115 sp<IMemory>* buffer); 1116 virtual status_t queueTimedBuffer(const sp<IMemory>& buffer, 1117 int64_t pts); 1118 virtual status_t setMediaTimeTransform(const LinearTransform& xform, 1119 int target); 1120 virtual status_t onTransact( 1121 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 1122 private: 1123 const sp<PlaybackThread::Track> mTrack; 1124 }; 1125 1126 void removeClient_l(pid_t pid); 1127 void removeNotificationClient(pid_t pid); 1128 1129 1130 // record thread 1131 class RecordThread : public ThreadBase, public AudioBufferProvider 1132 { 1133 public: 1134 1135 // record track 1136 class RecordTrack : public TrackBase { 1137 public: 1138 RecordTrack(RecordThread *thread, 1139 const sp<Client>& client, 1140 uint32_t sampleRate, 1141 audio_format_t format, 1142 uint32_t channelMask, 1143 int frameCount, 1144 int sessionId); 1145 virtual ~RecordTrack(); 1146 1147 virtual status_t start(pid_t tid); 1148 virtual void stop(); 1149 1150 bool overflow() { bool tmp = mOverflow; mOverflow = false; return tmp; } 1151 bool setOverflow() { bool tmp = mOverflow; mOverflow = true; return tmp; } 1152 1153 void dump(char* buffer, size_t size); 1154 1155 private: 1156 friend class AudioFlinger; // for mState 1157 1158 RecordTrack(const RecordTrack&); 1159 RecordTrack& operator = (const RecordTrack&); 1160 1161 // AudioBufferProvider interface 1162 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts = kInvalidPTS); 1163 // releaseBuffer() not overridden 1164 1165 bool mOverflow; 1166 }; 1167 1168 1169 RecordThread(const sp<AudioFlinger>& audioFlinger, 1170 AudioStreamIn *input, 1171 uint32_t sampleRate, 1172 uint32_t channels, 1173 audio_io_handle_t id, 1174 uint32_t device); 1175 virtual ~RecordThread(); 1176 1177 // Thread 1178 virtual bool threadLoop(); 1179 virtual status_t readyToRun(); 1180 1181 // RefBase 1182 virtual void onFirstRef(); 1183 1184 virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; } 1185 sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l( 1186 const sp<AudioFlinger::Client>& client, 1187 uint32_t sampleRate, 1188 audio_format_t format, 1189 int channelMask, 1190 int frameCount, 1191 int sessionId, 1192 status_t *status); 1193 1194 status_t start(RecordTrack* recordTrack); 1195 status_t start(RecordTrack* recordTrack, pid_t tid); 1196 void stop(RecordTrack* recordTrack); 1197 status_t dump(int fd, const Vector<String16>& args); 1198 AudioStreamIn* getInput() const; 1199 AudioStreamIn* clearInput(); 1200 virtual audio_stream_t* stream(); 1201 1202 // AudioBufferProvider interface 1203 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts); 1204 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 1205 1206 virtual bool checkForNewParameters_l(); 1207 virtual String8 getParameters(const String8& keys); 1208 virtual void audioConfigChanged_l(int event, int param = 0); 1209 void readInputParameters(); 1210 virtual unsigned int getInputFramesLost(); 1211 1212 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1213 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1214 virtual uint32_t hasAudioSession(int sessionId); 1215 RecordTrack* track(); 1216 1217 private: 1218 RecordThread(); 1219 AudioStreamIn *mInput; 1220 RecordTrack* mTrack; 1221 sp<RecordTrack> mActiveTrack; 1222 Condition mStartStopCond; 1223 AudioResampler *mResampler; 1224 int32_t *mRsmpOutBuffer; 1225 int16_t *mRsmpInBuffer; 1226 size_t mRsmpInIndex; 1227 size_t mInputBytes; 1228 const int mReqChannelCount; 1229 const uint32_t mReqSampleRate; 1230 ssize_t mBytesRead; 1231 }; 1232 1233 // server side of the client's IAudioRecord 1234 class RecordHandle : public android::BnAudioRecord { 1235 public: 1236 RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack); 1237 virtual ~RecordHandle(); 1238 virtual sp<IMemory> getCblk() const; 1239 virtual status_t start(pid_t tid); 1240 virtual void stop(); 1241 virtual status_t onTransact( 1242 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 1243 private: 1244 const sp<RecordThread::RecordTrack> mRecordTrack; 1245 }; 1246 1247 //--- Audio Effect Management 1248 1249 // EffectModule and EffectChain classes both have their own mutex to protect 1250 // state changes or resource modifications. Always respect the following order 1251 // if multiple mutexes must be acquired to avoid cross deadlock: 1252 // AudioFlinger -> ThreadBase -> EffectChain -> EffectModule 1253 1254 // The EffectModule class is a wrapper object controlling the effect engine implementation 1255 // in the effect library. It prevents concurrent calls to process() and command() functions 1256 // from different client threads. It keeps a list of EffectHandle objects corresponding 1257 // to all client applications using this effect and notifies applications of effect state, 1258 // control or parameter changes. It manages the activation state machine to send appropriate 1259 // reset, enable, disable commands to effect engine and provide volume 1260 // ramping when effects are activated/deactivated. 1261 // When controlling an auxiliary effect, the EffectModule also provides an input buffer used by 1262 // the attached track(s) to accumulate their auxiliary channel. 1263 class EffectModule: public RefBase { 1264 public: 1265 EffectModule(ThreadBase *thread, 1266 const wp<AudioFlinger::EffectChain>& chain, 1267 effect_descriptor_t *desc, 1268 int id, 1269 int sessionId); 1270 virtual ~EffectModule(); 1271 1272 enum effect_state { 1273 IDLE, 1274 RESTART, 1275 STARTING, 1276 ACTIVE, 1277 STOPPING, 1278 STOPPED, 1279 DESTROYED 1280 }; 1281 1282 int id() const { return mId; } 1283 void process(); 1284 void updateState(); 1285 status_t command(uint32_t cmdCode, 1286 uint32_t cmdSize, 1287 void *pCmdData, 1288 uint32_t *replySize, 1289 void *pReplyData); 1290 1291 void reset_l(); 1292 status_t configure(); 1293 status_t init(); 1294 effect_state state() const { 1295 return mState; 1296 } 1297 uint32_t status() { 1298 return mStatus; 1299 } 1300 int sessionId() const { 1301 return mSessionId; 1302 } 1303 status_t setEnabled(bool enabled); 1304 bool isEnabled() const; 1305 bool isProcessEnabled() const; 1306 1307 void setInBuffer(int16_t *buffer) { mConfig.inputCfg.buffer.s16 = buffer; } 1308 int16_t *inBuffer() { return mConfig.inputCfg.buffer.s16; } 1309 void setOutBuffer(int16_t *buffer) { mConfig.outputCfg.buffer.s16 = buffer; } 1310 int16_t *outBuffer() { return mConfig.outputCfg.buffer.s16; } 1311 void setChain(const wp<EffectChain>& chain) { mChain = chain; } 1312 void setThread(const wp<ThreadBase>& thread) { mThread = thread; } 1313 const wp<ThreadBase>& thread() { return mThread; } 1314 1315 status_t addHandle(const sp<EffectHandle>& handle); 1316 void disconnect(const wp<EffectHandle>& handle, bool unpinIfLast); 1317 size_t removeHandle (const wp<EffectHandle>& handle); 1318 1319 effect_descriptor_t& desc() { return mDescriptor; } 1320 wp<EffectChain>& chain() { return mChain; } 1321 1322 status_t setDevice(uint32_t device); 1323 status_t setVolume(uint32_t *left, uint32_t *right, bool controller); 1324 status_t setMode(audio_mode_t mode); 1325 status_t start(); 1326 status_t stop(); 1327 void setSuspended(bool suspended); 1328 bool suspended() const; 1329 1330 sp<EffectHandle> controlHandle(); 1331 1332 bool isPinned() const { return mPinned; } 1333 void unPin() { mPinned = false; } 1334 1335 status_t dump(int fd, const Vector<String16>& args); 1336 1337 protected: 1338 friend class AudioFlinger; // for mHandles 1339 bool mPinned; 1340 1341 // Maximum time allocated to effect engines to complete the turn off sequence 1342 static const uint32_t MAX_DISABLE_TIME_MS = 10000; 1343 1344 EffectModule(const EffectModule&); 1345 EffectModule& operator = (const EffectModule&); 1346 1347 status_t start_l(); 1348 status_t stop_l(); 1349 1350mutable Mutex mLock; // mutex for process, commands and handles list protection 1351 wp<ThreadBase> mThread; // parent thread 1352 wp<EffectChain> mChain; // parent effect chain 1353 int mId; // this instance unique ID 1354 int mSessionId; // audio session ID 1355 effect_descriptor_t mDescriptor;// effect descriptor received from effect engine 1356 effect_config_t mConfig; // input and output audio configuration 1357 effect_handle_t mEffectInterface; // Effect module C API 1358 status_t mStatus; // initialization status 1359 effect_state mState; // current activation state 1360 Vector< wp<EffectHandle> > mHandles; // list of client handles 1361 // First handle in mHandles has highest priority and controls the effect module 1362 uint32_t mMaxDisableWaitCnt; // maximum grace period before forcing an effect off after 1363 // sending disable command. 1364 uint32_t mDisableWaitCnt; // current process() calls count during disable period. 1365 bool mSuspended; // effect is suspended: temporarily disabled by framework 1366 }; 1367 1368 // The EffectHandle class implements the IEffect interface. It provides resources 1369 // to receive parameter updates, keeps track of effect control 1370 // ownership and state and has a pointer to the EffectModule object it is controlling. 1371 // There is one EffectHandle object for each application controlling (or using) 1372 // an effect module. 1373 // The EffectHandle is obtained by calling AudioFlinger::createEffect(). 1374 class EffectHandle: public android::BnEffect { 1375 public: 1376 1377 EffectHandle(const sp<EffectModule>& effect, 1378 const sp<AudioFlinger::Client>& client, 1379 const sp<IEffectClient>& effectClient, 1380 int32_t priority); 1381 virtual ~EffectHandle(); 1382 1383 // IEffect 1384 virtual status_t enable(); 1385 virtual status_t disable(); 1386 virtual status_t command(uint32_t cmdCode, 1387 uint32_t cmdSize, 1388 void *pCmdData, 1389 uint32_t *replySize, 1390 void *pReplyData); 1391 virtual void disconnect(); 1392 private: 1393 void disconnect(bool unpinIfLast); 1394 public: 1395 virtual sp<IMemory> getCblk() const { return mCblkMemory; } 1396 virtual status_t onTransact(uint32_t code, const Parcel& data, 1397 Parcel* reply, uint32_t flags); 1398 1399 1400 // Give or take control of effect module 1401 // - hasControl: true if control is given, false if removed 1402 // - signal: true client app should be signaled of change, false otherwise 1403 // - enabled: state of the effect when control is passed 1404 void setControl(bool hasControl, bool signal, bool enabled); 1405 void commandExecuted(uint32_t cmdCode, 1406 uint32_t cmdSize, 1407 void *pCmdData, 1408 uint32_t replySize, 1409 void *pReplyData); 1410 void setEnabled(bool enabled); 1411 bool enabled() const { return mEnabled; } 1412 1413 // Getters 1414 int id() const { return mEffect->id(); } 1415 int priority() const { return mPriority; } 1416 bool hasControl() const { return mHasControl; } 1417 sp<EffectModule> effect() const { return mEffect; } 1418 1419 void dump(char* buffer, size_t size); 1420 1421 protected: 1422 friend class AudioFlinger; // for mEffect, mHasControl, mEnabled 1423 EffectHandle(const EffectHandle&); 1424 EffectHandle& operator =(const EffectHandle&); 1425 1426 sp<EffectModule> mEffect; // pointer to controlled EffectModule 1427 sp<IEffectClient> mEffectClient; // callback interface for client notifications 1428 /*const*/ sp<Client> mClient; // client for shared memory allocation, see disconnect() 1429 sp<IMemory> mCblkMemory; // shared memory for control block 1430 effect_param_cblk_t* mCblk; // control block for deferred parameter setting via shared memory 1431 uint8_t* mBuffer; // pointer to parameter area in shared memory 1432 int mPriority; // client application priority to control the effect 1433 bool mHasControl; // true if this handle is controlling the effect 1434 bool mEnabled; // cached enable state: needed when the effect is 1435 // restored after being suspended 1436 }; 1437 1438 // the EffectChain class represents a group of effects associated to one audio session. 1439 // There can be any number of EffectChain objects per output mixer thread (PlaybackThread). 1440 // The EffecChain with session ID 0 contains global effects applied to the output mix. 1441 // Effects in this chain can be insert or auxiliary. Effects in other chains (attached to tracks) 1442 // are insert only. The EffectChain maintains an ordered list of effect module, the order corresponding 1443 // in the effect process order. When attached to a track (session ID != 0), it also provide it's own 1444 // input buffer used by the track as accumulation buffer. 1445 class EffectChain: public RefBase { 1446 public: 1447 EffectChain(const wp<ThreadBase>& wThread, int sessionId); 1448 EffectChain(ThreadBase *thread, int sessionId); 1449 virtual ~EffectChain(); 1450 1451 // special key used for an entry in mSuspendedEffects keyed vector 1452 // corresponding to a suspend all request. 1453 static const int kKeyForSuspendAll = 0; 1454 1455 // minimum duration during which we force calling effect process when last track on 1456 // a session is stopped or removed to allow effect tail to be rendered 1457 static const int kProcessTailDurationMs = 1000; 1458 1459 void process_l(); 1460 1461 void lock() { 1462 mLock.lock(); 1463 } 1464 void unlock() { 1465 mLock.unlock(); 1466 } 1467 1468 status_t addEffect_l(const sp<EffectModule>& handle); 1469 size_t removeEffect_l(const sp<EffectModule>& handle); 1470 1471 int sessionId() const { return mSessionId; } 1472 void setSessionId(int sessionId) { mSessionId = sessionId; } 1473 1474 sp<EffectModule> getEffectFromDesc_l(effect_descriptor_t *descriptor); 1475 sp<EffectModule> getEffectFromId_l(int id); 1476 sp<EffectModule> getEffectFromType_l(const effect_uuid_t *type); 1477 bool setVolume_l(uint32_t *left, uint32_t *right); 1478 void setDevice_l(uint32_t device); 1479 void setMode_l(audio_mode_t mode); 1480 1481 void setInBuffer(int16_t *buffer, bool ownsBuffer = false) { 1482 mInBuffer = buffer; 1483 mOwnInBuffer = ownsBuffer; 1484 } 1485 int16_t *inBuffer() const { 1486 return mInBuffer; 1487 } 1488 void setOutBuffer(int16_t *buffer) { 1489 mOutBuffer = buffer; 1490 } 1491 int16_t *outBuffer() const { 1492 return mOutBuffer; 1493 } 1494 1495 void incTrackCnt() { android_atomic_inc(&mTrackCnt); } 1496 void decTrackCnt() { android_atomic_dec(&mTrackCnt); } 1497 int32_t trackCnt() const { return mTrackCnt;} 1498 1499 void incActiveTrackCnt() { android_atomic_inc(&mActiveTrackCnt); 1500 mTailBufferCount = mMaxTailBuffers; } 1501 void decActiveTrackCnt() { android_atomic_dec(&mActiveTrackCnt); } 1502 int32_t activeTrackCnt() const { return mActiveTrackCnt;} 1503 1504 uint32_t strategy() const { return mStrategy; } 1505 void setStrategy(uint32_t strategy) 1506 { mStrategy = strategy; } 1507 1508 // suspend effect of the given type 1509 void setEffectSuspended_l(const effect_uuid_t *type, 1510 bool suspend); 1511 // suspend all eligible effects 1512 void setEffectSuspendedAll_l(bool suspend); 1513 // check if effects should be suspend or restored when a given effect is enable or disabled 1514 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1515 bool enabled); 1516 1517 status_t dump(int fd, const Vector<String16>& args); 1518 1519 protected: 1520 friend class AudioFlinger; // for mThread, mEffects 1521 EffectChain(const EffectChain&); 1522 EffectChain& operator =(const EffectChain&); 1523 1524 class SuspendedEffectDesc : public RefBase { 1525 public: 1526 SuspendedEffectDesc() : mRefCount(0) {} 1527 1528 int mRefCount; 1529 effect_uuid_t mType; 1530 wp<EffectModule> mEffect; 1531 }; 1532 1533 // get a list of effect modules to suspend when an effect of the type 1534 // passed is enabled. 1535 void getSuspendEligibleEffects(Vector< sp<EffectModule> > &effects); 1536 1537 // get an effect module if it is currently enable 1538 sp<EffectModule> getEffectIfEnabled(const effect_uuid_t *type); 1539 // true if the effect whose descriptor is passed can be suspended 1540 // OEMs can modify the rules implemented in this method to exclude specific effect 1541 // types or implementations from the suspend/restore mechanism. 1542 bool isEffectEligibleForSuspend(const effect_descriptor_t& desc); 1543 1544 wp<ThreadBase> mThread; // parent mixer thread 1545 Mutex mLock; // mutex protecting effect list 1546 Vector< sp<EffectModule> > mEffects; // list of effect modules 1547 int mSessionId; // audio session ID 1548 int16_t *mInBuffer; // chain input buffer 1549 int16_t *mOutBuffer; // chain output buffer 1550 volatile int32_t mActiveTrackCnt; // number of active tracks connected 1551 volatile int32_t mTrackCnt; // number of tracks connected 1552 int32_t mTailBufferCount; // current effect tail buffer count 1553 int32_t mMaxTailBuffers; // maximum effect tail buffers 1554 bool mOwnInBuffer; // true if the chain owns its input buffer 1555 int mVolumeCtrlIdx; // index of insert effect having control over volume 1556 uint32_t mLeftVolume; // previous volume on left channel 1557 uint32_t mRightVolume; // previous volume on right channel 1558 uint32_t mNewLeftVolume; // new volume on left channel 1559 uint32_t mNewRightVolume; // new volume on right channel 1560 uint32_t mStrategy; // strategy for this effect chain 1561 // mSuspendedEffects lists all effect currently suspended in the chain 1562 // use effect type UUID timelow field as key. There is no real risk of identical 1563 // timeLow fields among effect type UUIDs. 1564 KeyedVector< int, sp<SuspendedEffectDesc> > mSuspendedEffects; 1565 }; 1566 1567 // AudioStreamOut and AudioStreamIn are immutable, so their fields are const. 1568 // For emphasis, we could also make all pointers to them be "const *", 1569 // but that would clutter the code unnecessarily. 1570 1571 struct AudioStreamOut { 1572 audio_hw_device_t* const hwDev; 1573 audio_stream_out_t* const stream; 1574 1575 AudioStreamOut(audio_hw_device_t *dev, audio_stream_out_t *out) : 1576 hwDev(dev), stream(out) {} 1577 }; 1578 1579 struct AudioStreamIn { 1580 audio_hw_device_t* const hwDev; 1581 audio_stream_in_t* const stream; 1582 1583 AudioStreamIn(audio_hw_device_t *dev, audio_stream_in_t *in) : 1584 hwDev(dev), stream(in) {} 1585 }; 1586 1587 // for mAudioSessionRefs only 1588 struct AudioSessionRef { 1589 AudioSessionRef(int sessionid, pid_t pid) : 1590 mSessionid(sessionid), mPid(pid), mCnt(1) {} 1591 const int mSessionid; 1592 const pid_t mPid; 1593 int mCnt; 1594 }; 1595 1596 enum master_volume_support { 1597 // MVS_NONE: 1598 // Audio HAL has no support for master volume, either setting or 1599 // getting. All master volume control must be implemented in SW by the 1600 // AudioFlinger mixing core. 1601 MVS_NONE, 1602 1603 // MVS_SETONLY: 1604 // Audio HAL has support for setting master volume, but not for getting 1605 // master volume (original HAL design did not include a getter). 1606 // AudioFlinger needs to keep track of the last set master volume in 1607 // addition to needing to set an initial, default, master volume at HAL 1608 // load time. 1609 MVS_SETONLY, 1610 1611 // MVS_FULL: 1612 // Audio HAL has support both for setting and getting master volume. 1613 // AudioFlinger should send all set and get master volume requests 1614 // directly to the HAL. 1615 MVS_FULL, 1616 }; 1617 1618 mutable Mutex mLock; 1619 1620 DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client() 1621 1622 mutable Mutex mHardwareLock; 1623 1624 // These two fields are immutable after onFirstRef(), so no lock needed to access 1625 audio_hw_device_t* mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL 1626 Vector<audio_hw_device_t*> mAudioHwDevs; 1627 1628 // for dump, indicates which hardware operation is currently in progress (but not stream ops) 1629 enum hardware_call_state { 1630 AUDIO_HW_IDLE = 0, // no operation in progress 1631 AUDIO_HW_INIT, // init_check 1632 AUDIO_HW_OUTPUT_OPEN, // open_output_stream 1633 AUDIO_HW_OUTPUT_CLOSE, // unused 1634 AUDIO_HW_INPUT_OPEN, // unused 1635 AUDIO_HW_INPUT_CLOSE, // unused 1636 AUDIO_HW_STANDBY, // unused 1637 AUDIO_HW_SET_MASTER_VOLUME, // set_master_volume 1638 AUDIO_HW_GET_ROUTING, // unused 1639 AUDIO_HW_SET_ROUTING, // unused 1640 AUDIO_HW_GET_MODE, // unused 1641 AUDIO_HW_SET_MODE, // set_mode 1642 AUDIO_HW_GET_MIC_MUTE, // get_mic_mute 1643 AUDIO_HW_SET_MIC_MUTE, // set_mic_mute 1644 AUDIO_HW_SET_VOICE_VOLUME, // set_voice_volume 1645 AUDIO_HW_SET_PARAMETER, // set_parameters 1646 AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size 1647 AUDIO_HW_GET_MASTER_VOLUME, // get_master_volume 1648 AUDIO_HW_GET_PARAMETER, // get_parameters 1649 }; 1650 1651 mutable hardware_call_state mHardwareStatus; // for dump only 1652 1653 1654 DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads; 1655 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 1656 1657 // both are protected by mLock 1658 float mMasterVolume; 1659 float mMasterVolumeSW; 1660 master_volume_support mMasterVolumeSupportLvl; 1661 bool mMasterMute; 1662 1663 DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads; 1664 1665 DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients; 1666 volatile int32_t mNextUniqueId; // updated by android_atomic_inc 1667 audio_mode_t mMode; 1668 bool mBtNrecIsOff; 1669 1670 // protected by mLock 1671 Vector<AudioSessionRef*> mAudioSessionRefs; 1672 1673 float masterVolume_l() const; 1674 float masterVolumeSW_l() const { return mMasterVolumeSW; } 1675 bool masterMute_l() const { return mMasterMute; } 1676 1677private: 1678 sp<Client> registerPid_l(pid_t pid); // always returns non-0 1679 1680}; 1681 1682 1683// ---------------------------------------------------------------------------- 1684 1685}; // namespace android 1686 1687#endif // ANDROID_AUDIO_FLINGER_H 1688