AudioFlinger.h revision 3acbd053c842e76e1a40fc8a0bf62de87eebf00f
1/* 2** 3** Copyright 2007, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18#ifndef ANDROID_AUDIO_FLINGER_H 19#define ANDROID_AUDIO_FLINGER_H 20 21#include <stdint.h> 22#include <sys/types.h> 23#include <limits.h> 24 25#include <common_time/cc_helper.h> 26 27#include <media/IAudioFlinger.h> 28#include <media/IAudioFlingerClient.h> 29#include <media/IAudioTrack.h> 30#include <media/IAudioRecord.h> 31#include <media/AudioSystem.h> 32#include <media/AudioTrack.h> 33 34#include <utils/Atomic.h> 35#include <utils/Errors.h> 36#include <utils/threads.h> 37#include <utils/SortedVector.h> 38#include <utils/TypeHelpers.h> 39#include <utils/Vector.h> 40 41#include <binder/BinderService.h> 42#include <binder/MemoryDealer.h> 43 44#include <system/audio.h> 45#include <hardware/audio.h> 46#include <hardware/audio_policy.h> 47 48#include "AudioBufferProvider.h" 49 50#include <powermanager/IPowerManager.h> 51 52namespace android { 53 54class audio_track_cblk_t; 55class effect_param_cblk_t; 56class AudioMixer; 57class AudioBuffer; 58class AudioResampler; 59 60// ---------------------------------------------------------------------------- 61 62// AudioFlinger has a hard-coded upper limit of 2 channels for capture and playback. 63// There is support for > 2 channel tracks down-mixed to 2 channel output via a down-mix effect. 64// Adding full support for > 2 channel capture or playback would require more than simply changing 65// this #define. There is an independent hard-coded upper limit in AudioMixer; 66// removing that AudioMixer limit would be necessary but insufficient to support > 2 channels. 67// The macro FCC_2 highlights some (but not all) places where there is are 2-channel assumptions. 68// Search also for "2", "left", "right", "[0]", "[1]", ">> 16", "<< 16", etc. 69#define FCC_2 2 // FCC_2 = Fixed Channel Count 2 70 71static const nsecs_t kDefaultStandbyTimeInNsecs = seconds(3); 72 73class AudioFlinger : 74 public BinderService<AudioFlinger>, 75 public BnAudioFlinger 76{ 77 friend class BinderService<AudioFlinger>; // for AudioFlinger() 78public: 79 static const char* getServiceName() { return "media.audio_flinger"; } 80 81 virtual status_t dump(int fd, const Vector<String16>& args); 82 83 // IAudioFlinger interface, in binder opcode order 84 virtual sp<IAudioTrack> createTrack( 85 pid_t pid, 86 audio_stream_type_t streamType, 87 uint32_t sampleRate, 88 audio_format_t format, 89 uint32_t channelMask, 90 int frameCount, 91 IAudioFlinger::track_flags_t flags, 92 const sp<IMemory>& sharedBuffer, 93 audio_io_handle_t output, 94 pid_t tid, 95 int *sessionId, 96 status_t *status); 97 98 virtual sp<IAudioRecord> openRecord( 99 pid_t pid, 100 audio_io_handle_t input, 101 uint32_t sampleRate, 102 audio_format_t format, 103 uint32_t channelMask, 104 int frameCount, 105 IAudioFlinger::track_flags_t flags, 106 int *sessionId, 107 status_t *status); 108 109 virtual uint32_t sampleRate(audio_io_handle_t output) const; 110 virtual int channelCount(audio_io_handle_t output) const; 111 virtual audio_format_t format(audio_io_handle_t output) const; 112 virtual size_t frameCount(audio_io_handle_t output) const; 113 virtual uint32_t latency(audio_io_handle_t output) const; 114 115 virtual status_t setMasterVolume(float value); 116 virtual status_t setMasterMute(bool muted); 117 118 virtual float masterVolume() const; 119 virtual float masterVolumeSW() const; 120 virtual bool masterMute() const; 121 122 virtual status_t setStreamVolume(audio_stream_type_t stream, float value, 123 audio_io_handle_t output); 124 virtual status_t setStreamMute(audio_stream_type_t stream, bool muted); 125 126 virtual float streamVolume(audio_stream_type_t stream, 127 audio_io_handle_t output) const; 128 virtual bool streamMute(audio_stream_type_t stream) const; 129 130 virtual status_t setMode(audio_mode_t mode); 131 132 virtual status_t setMicMute(bool state); 133 virtual bool getMicMute() const; 134 135 virtual status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs); 136 virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) const; 137 138 virtual void registerClient(const sp<IAudioFlingerClient>& client); 139 140 virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const; 141 142 virtual audio_io_handle_t openOutput(audio_module_handle_t module, 143 audio_devices_t *pDevices, 144 uint32_t *pSamplingRate, 145 audio_format_t *pFormat, 146 audio_channel_mask_t *pChannelMask, 147 uint32_t *pLatencyMs, 148 audio_output_flags_t flags); 149 150 virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1, 151 audio_io_handle_t output2); 152 153 virtual status_t closeOutput(audio_io_handle_t output); 154 155 virtual status_t suspendOutput(audio_io_handle_t output); 156 157 virtual status_t restoreOutput(audio_io_handle_t output); 158 159 virtual audio_io_handle_t openInput(audio_module_handle_t module, 160 audio_devices_t *pDevices, 161 uint32_t *pSamplingRate, 162 audio_format_t *pFormat, 163 audio_channel_mask_t *pChannelMask); 164 165 virtual status_t closeInput(audio_io_handle_t input); 166 167 virtual status_t setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output); 168 169 virtual status_t setVoiceVolume(float volume); 170 171 virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 172 audio_io_handle_t output) const; 173 174 virtual unsigned int getInputFramesLost(audio_io_handle_t ioHandle) const; 175 176 virtual int newAudioSessionId(); 177 178 virtual void acquireAudioSessionId(int audioSession); 179 180 virtual void releaseAudioSessionId(int audioSession); 181 182 virtual status_t queryNumberEffects(uint32_t *numEffects) const; 183 184 virtual status_t queryEffect(uint32_t index, effect_descriptor_t *descriptor) const; 185 186 virtual status_t getEffectDescriptor(const effect_uuid_t *pUuid, 187 effect_descriptor_t *descriptor) const; 188 189 virtual sp<IEffect> createEffect(pid_t pid, 190 effect_descriptor_t *pDesc, 191 const sp<IEffectClient>& effectClient, 192 int32_t priority, 193 audio_io_handle_t io, 194 int sessionId, 195 status_t *status, 196 int *id, 197 int *enabled); 198 199 virtual status_t moveEffects(int sessionId, audio_io_handle_t srcOutput, 200 audio_io_handle_t dstOutput); 201 202 virtual audio_module_handle_t loadHwModule(const char *name); 203 204 virtual status_t onTransact( 205 uint32_t code, 206 const Parcel& data, 207 Parcel* reply, 208 uint32_t flags); 209 210 // end of IAudioFlinger interface 211 212 class SyncEvent; 213 214 typedef void (*sync_event_callback_t)(const wp<SyncEvent>& event) ; 215 216 class SyncEvent : public RefBase { 217 public: 218 SyncEvent(AudioSystem::sync_event_t type, 219 int triggerSession, 220 int listenerSession, 221 sync_event_callback_t callBack, 222 void *cookie) 223 : mType(type), mTriggerSession(triggerSession), mListenerSession(listenerSession), 224 mCallback(callBack), mCookie(cookie) 225 {} 226 227 virtual ~SyncEvent() {} 228 229 void trigger() { Mutex::Autolock _l(mLock); if (mCallback) mCallback(this); } 230 void cancel() {Mutex::Autolock _l(mLock); mCallback = NULL; } 231 AudioSystem::sync_event_t type() const { return mType; } 232 int triggerSession() const { return mTriggerSession; } 233 int listenerSession() const { return mListenerSession; } 234 void *cookie() const { return mCookie; } 235 236 private: 237 const AudioSystem::sync_event_t mType; 238 const int mTriggerSession; 239 const int mListenerSession; 240 sync_event_callback_t mCallback; 241 void * const mCookie; 242 Mutex mLock; 243 }; 244 245 sp<SyncEvent> createSyncEvent(AudioSystem::sync_event_t type, 246 int triggerSession, 247 int listenerSession, 248 sync_event_callback_t callBack, 249 void *cookie); 250private: 251 audio_mode_t getMode() const { return mMode; } 252 253 bool btNrecIsOff() const { return mBtNrecIsOff; } 254 255 AudioFlinger(); 256 virtual ~AudioFlinger(); 257 258 // call in any IAudioFlinger method that accesses mPrimaryHardwareDev 259 status_t initCheck() const { return mPrimaryHardwareDev == NULL ? NO_INIT : NO_ERROR; } 260 261 // RefBase 262 virtual void onFirstRef(); 263 264 audio_hw_device_t* findSuitableHwDev_l(audio_module_handle_t module, uint32_t devices); 265 void purgeStaleEffects_l(); 266 267 // standby delay for MIXER and DUPLICATING playback threads is read from property 268 // ro.audio.flinger_standbytime_ms or defaults to kDefaultStandbyTimeInNsecs 269 static nsecs_t mStandbyTimeInNsecs; 270 271 // Internal dump utilites. 272 status_t dumpPermissionDenial(int fd, const Vector<String16>& args); 273 status_t dumpClients(int fd, const Vector<String16>& args); 274 status_t dumpInternals(int fd, const Vector<String16>& args); 275 276 // --- Client --- 277 class Client : public RefBase { 278 public: 279 Client(const sp<AudioFlinger>& audioFlinger, pid_t pid); 280 virtual ~Client(); 281 sp<MemoryDealer> heap() const; 282 pid_t pid() const { return mPid; } 283 sp<AudioFlinger> audioFlinger() const { return mAudioFlinger; } 284 285 bool reserveTimedTrack(); 286 void releaseTimedTrack(); 287 288 private: 289 Client(const Client&); 290 Client& operator = (const Client&); 291 const sp<AudioFlinger> mAudioFlinger; 292 const sp<MemoryDealer> mMemoryDealer; 293 const pid_t mPid; 294 295 Mutex mTimedTrackLock; 296 int mTimedTrackCount; 297 }; 298 299 // --- Notification Client --- 300 class NotificationClient : public IBinder::DeathRecipient { 301 public: 302 NotificationClient(const sp<AudioFlinger>& audioFlinger, 303 const sp<IAudioFlingerClient>& client, 304 pid_t pid); 305 virtual ~NotificationClient(); 306 307 sp<IAudioFlingerClient> audioFlingerClient() const { return mAudioFlingerClient; } 308 309 // IBinder::DeathRecipient 310 virtual void binderDied(const wp<IBinder>& who); 311 312 private: 313 NotificationClient(const NotificationClient&); 314 NotificationClient& operator = (const NotificationClient&); 315 316 const sp<AudioFlinger> mAudioFlinger; 317 const pid_t mPid; 318 const sp<IAudioFlingerClient> mAudioFlingerClient; 319 }; 320 321 class TrackHandle; 322 class RecordHandle; 323 class RecordThread; 324 class PlaybackThread; 325 class MixerThread; 326 class DirectOutputThread; 327 class DuplicatingThread; 328 class Track; 329 class RecordTrack; 330 class EffectModule; 331 class EffectHandle; 332 class EffectChain; 333 struct AudioStreamOut; 334 struct AudioStreamIn; 335 336 class ThreadBase : public Thread { 337 public: 338 339 enum type_t { 340 MIXER, // Thread class is MixerThread 341 DIRECT, // Thread class is DirectOutputThread 342 DUPLICATING, // Thread class is DuplicatingThread 343 RECORD // Thread class is RecordThread 344 }; 345 346 ThreadBase (const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, uint32_t device, type_t type); 347 virtual ~ThreadBase(); 348 349 status_t dumpBase(int fd, const Vector<String16>& args); 350 status_t dumpEffectChains(int fd, const Vector<String16>& args); 351 352 void clearPowerManager(); 353 354 // base for record and playback 355 class TrackBase : public AudioBufferProvider, public RefBase { 356 357 public: 358 enum track_state { 359 IDLE, 360 TERMINATED, 361 // These are order-sensitive; do not change order without reviewing the impact. 362 // In particular there are assumptions about > STOPPED. 363 STOPPED, 364 RESUMING, 365 ACTIVE, 366 PAUSING, 367 PAUSED 368 }; 369 370 TrackBase(ThreadBase *thread, 371 const sp<Client>& client, 372 uint32_t sampleRate, 373 audio_format_t format, 374 uint32_t channelMask, 375 int frameCount, 376 const sp<IMemory>& sharedBuffer, 377 int sessionId); 378 virtual ~TrackBase(); 379 380 virtual status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 381 int triggerSession = 0) = 0; 382 virtual void stop() = 0; 383 sp<IMemory> getCblk() const { return mCblkMemory; } 384 audio_track_cblk_t* cblk() const { return mCblk; } 385 int sessionId() const { return mSessionId; } 386 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 387 388 protected: 389 TrackBase(const TrackBase&); 390 TrackBase& operator = (const TrackBase&); 391 392 // AudioBufferProvider interface 393 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) = 0; 394 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 395 396 audio_format_t format() const { 397 return mFormat; 398 } 399 400 int channelCount() const { return mChannelCount; } 401 402 uint32_t channelMask() const { return mChannelMask; } 403 404 int sampleRate() const; // FIXME inline after cblk sr moved 405 406 void* getBuffer(uint32_t offset, uint32_t frames) const; 407 408 bool isStopped() const { 409 return mState == STOPPED; 410 } 411 412 bool isTerminated() const { 413 return mState == TERMINATED; 414 } 415 416 bool step(); 417 void reset(); 418 419 const wp<ThreadBase> mThread; 420 /*const*/ sp<Client> mClient; // see explanation at ~TrackBase() why not const 421 sp<IMemory> mCblkMemory; 422 audio_track_cblk_t* mCblk; 423 void* mBuffer; 424 void* mBufferEnd; 425 uint32_t mFrameCount; 426 // we don't really need a lock for these 427 track_state mState; 428 const audio_format_t mFormat; 429 bool mStepServerFailed; 430 const int mSessionId; 431 uint8_t mChannelCount; 432 uint32_t mChannelMask; 433 Vector < sp<SyncEvent> >mSyncEvents; 434 }; 435 436 class ConfigEvent { 437 public: 438 ConfigEvent() : mEvent(0), mParam(0) {} 439 440 int mEvent; 441 int mParam; 442 }; 443 444 class PMDeathRecipient : public IBinder::DeathRecipient { 445 public: 446 PMDeathRecipient(const wp<ThreadBase>& thread) : mThread(thread) {} 447 virtual ~PMDeathRecipient() {} 448 449 // IBinder::DeathRecipient 450 virtual void binderDied(const wp<IBinder>& who); 451 452 private: 453 PMDeathRecipient(const PMDeathRecipient&); 454 PMDeathRecipient& operator = (const PMDeathRecipient&); 455 456 wp<ThreadBase> mThread; 457 }; 458 459 virtual status_t initCheck() const = 0; 460 type_t type() const { return mType; } 461 uint32_t sampleRate() const { return mSampleRate; } 462 int channelCount() const { return mChannelCount; } 463 audio_format_t format() const { return mFormat; } 464 size_t frameCount() const { return mFrameCount; } 465 void wakeUp() { mWaitWorkCV.broadcast(); } 466 // Should be "virtual status_t requestExitAndWait()" and override same 467 // method in Thread, but Thread::requestExitAndWait() is not yet virtual. 468 void exit(); 469 virtual bool checkForNewParameters_l() = 0; 470 virtual status_t setParameters(const String8& keyValuePairs); 471 virtual String8 getParameters(const String8& keys) = 0; 472 virtual void audioConfigChanged_l(int event, int param = 0) = 0; 473 void sendConfigEvent(int event, int param = 0); 474 void sendConfigEvent_l(int event, int param = 0); 475 void processConfigEvents(); 476 audio_io_handle_t id() const { return mId;} 477 bool standby() const { return mStandby; } 478 uint32_t device() const { return mDevice; } 479 virtual audio_stream_t* stream() const = 0; 480 481 sp<EffectHandle> createEffect_l( 482 const sp<AudioFlinger::Client>& client, 483 const sp<IEffectClient>& effectClient, 484 int32_t priority, 485 int sessionId, 486 effect_descriptor_t *desc, 487 int *enabled, 488 status_t *status); 489 void disconnectEffect(const sp< EffectModule>& effect, 490 const wp<EffectHandle>& handle, 491 bool unpinIfLast); 492 493 // return values for hasAudioSession (bit field) 494 enum effect_state { 495 EFFECT_SESSION = 0x1, // the audio session corresponds to at least one 496 // effect 497 TRACK_SESSION = 0x2 // the audio session corresponds to at least one 498 // track 499 }; 500 501 // get effect chain corresponding to session Id. 502 sp<EffectChain> getEffectChain(int sessionId); 503 // same as getEffectChain() but must be called with ThreadBase mutex locked 504 sp<EffectChain> getEffectChain_l(int sessionId); 505 // add an effect chain to the chain list (mEffectChains) 506 virtual status_t addEffectChain_l(const sp<EffectChain>& chain) = 0; 507 // remove an effect chain from the chain list (mEffectChains) 508 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain) = 0; 509 // lock all effect chains Mutexes. Must be called before releasing the 510 // ThreadBase mutex before processing the mixer and effects. This guarantees the 511 // integrity of the chains during the process. 512 // Also sets the parameter 'effectChains' to current value of mEffectChains. 513 void lockEffectChains_l(Vector< sp<EffectChain> >& effectChains); 514 // unlock effect chains after process 515 void unlockEffectChains(const Vector< sp<EffectChain> >& effectChains); 516 // set audio mode to all effect chains 517 void setMode(audio_mode_t mode); 518 // get effect module with corresponding ID on specified audio session 519 sp<AudioFlinger::EffectModule> getEffect_l(int sessionId, int effectId); 520 // add and effect module. Also creates the effect chain is none exists for 521 // the effects audio session 522 status_t addEffect_l(const sp< EffectModule>& effect); 523 // remove and effect module. Also removes the effect chain is this was the last 524 // effect 525 void removeEffect_l(const sp< EffectModule>& effect); 526 // detach all tracks connected to an auxiliary effect 527 virtual void detachAuxEffect_l(int effectId) {} 528 // returns either EFFECT_SESSION if effects on this audio session exist in one 529 // chain, or TRACK_SESSION if tracks on this audio session exist, or both 530 virtual uint32_t hasAudioSession(int sessionId) = 0; 531 // the value returned by default implementation is not important as the 532 // strategy is only meaningful for PlaybackThread which implements this method 533 virtual uint32_t getStrategyForSession_l(int sessionId) { return 0; } 534 535 // suspend or restore effect according to the type of effect passed. a NULL 536 // type pointer means suspend all effects in the session 537 void setEffectSuspended(const effect_uuid_t *type, 538 bool suspend, 539 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 540 // check if some effects must be suspended/restored when an effect is enabled 541 // or disabled 542 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 543 bool enabled, 544 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 545 void checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 546 bool enabled, 547 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 548 549 virtual status_t setSyncEvent(const sp<SyncEvent>& event) = 0; 550 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) = 0; 551 552 553 mutable Mutex mLock; 554 555 protected: 556 557 // entry describing an effect being suspended in mSuspendedSessions keyed vector 558 class SuspendedSessionDesc : public RefBase { 559 public: 560 SuspendedSessionDesc() : mRefCount(0) {} 561 562 int mRefCount; // number of active suspend requests 563 effect_uuid_t mType; // effect type UUID 564 }; 565 566 void acquireWakeLock(); 567 void acquireWakeLock_l(); 568 void releaseWakeLock(); 569 void releaseWakeLock_l(); 570 void setEffectSuspended_l(const effect_uuid_t *type, 571 bool suspend, 572 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 573 // updated mSuspendedSessions when an effect suspended or restored 574 void updateSuspendedSessions_l(const effect_uuid_t *type, 575 bool suspend, 576 int sessionId); 577 // check if some effects must be suspended when an effect chain is added 578 void checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain); 579 580 friend class AudioFlinger; // for mEffectChains 581 582 const type_t mType; 583 584 // Used by parameters, config events, addTrack_l, exit 585 Condition mWaitWorkCV; 586 587 const sp<AudioFlinger> mAudioFlinger; 588 uint32_t mSampleRate; 589 size_t mFrameCount; 590 uint32_t mChannelMask; 591 uint16_t mChannelCount; 592 size_t mFrameSize; 593 audio_format_t mFormat; 594 595 // Parameter sequence by client: binder thread calling setParameters(): 596 // 1. Lock mLock 597 // 2. Append to mNewParameters 598 // 3. mWaitWorkCV.signal 599 // 4. mParamCond.waitRelative with timeout 600 // 5. read mParamStatus 601 // 6. mWaitWorkCV.signal 602 // 7. Unlock 603 // 604 // Parameter sequence by server: threadLoop calling checkForNewParameters_l(): 605 // 1. Lock mLock 606 // 2. If there is an entry in mNewParameters proceed ... 607 // 2. Read first entry in mNewParameters 608 // 3. Process 609 // 4. Remove first entry from mNewParameters 610 // 5. Set mParamStatus 611 // 6. mParamCond.signal 612 // 7. mWaitWorkCV.wait with timeout (this is to avoid overwriting mParamStatus) 613 // 8. Unlock 614 Condition mParamCond; 615 Vector<String8> mNewParameters; 616 status_t mParamStatus; 617 618 Vector<ConfigEvent> mConfigEvents; 619 bool mStandby; 620 const audio_io_handle_t mId; 621 Vector< sp<EffectChain> > mEffectChains; 622 uint32_t mDevice; // output device for PlaybackThread 623 // input + output devices for RecordThread 624 static const int kNameLength = 16; // prctl(PR_SET_NAME) limit 625 char mName[kNameLength]; 626 sp<IPowerManager> mPowerManager; 627 sp<IBinder> mWakeLockToken; 628 const sp<PMDeathRecipient> mDeathRecipient; 629 // list of suspended effects per session and per type. The first vector is 630 // keyed by session ID, the second by type UUID timeLow field 631 KeyedVector< int, KeyedVector< int, sp<SuspendedSessionDesc> > > mSuspendedSessions; 632 }; 633 634 struct stream_type_t { 635 stream_type_t() 636 : volume(1.0f), 637 mute(false) 638 { 639 } 640 float volume; 641 bool mute; 642 }; 643 644 // --- PlaybackThread --- 645 class PlaybackThread : public ThreadBase { 646 public: 647 648 enum mixer_state { 649 MIXER_IDLE, // no active tracks 650 MIXER_TRACKS_ENABLED, // at least one active track, but no track has any data ready 651 MIXER_TRACKS_READY // at least one active track, and at least one track has data 652 // standby mode does not have an enum value 653 // suspend by audio policy manager is orthogonal to mixer state 654 }; 655 656 // playback track 657 class Track : public TrackBase { 658 public: 659 Track( PlaybackThread *thread, 660 const sp<Client>& client, 661 audio_stream_type_t streamType, 662 uint32_t sampleRate, 663 audio_format_t format, 664 uint32_t channelMask, 665 int frameCount, 666 const sp<IMemory>& sharedBuffer, 667 int sessionId, 668 IAudioFlinger::track_flags_t flags); 669 virtual ~Track(); 670 671 void dump(char* buffer, size_t size); 672 virtual status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 673 int triggerSession = 0); 674 virtual void stop(); 675 void pause(); 676 677 void flush(); 678 void destroy(); 679 void mute(bool); 680 int name() const { 681 return mName; 682 } 683 684 audio_stream_type_t streamType() const { 685 return mStreamType; 686 } 687 status_t attachAuxEffect(int EffectId); 688 void setAuxBuffer(int EffectId, int32_t *buffer); 689 int32_t *auxBuffer() const { return mAuxBuffer; } 690 void setMainBuffer(int16_t *buffer) { mMainBuffer = buffer; } 691 int16_t *mainBuffer() const { return mMainBuffer; } 692 int auxEffectId() const { return mAuxEffectId; } 693 694 bool isFastTrack() const 695 { return (mFlags & IAudioFlinger::TRACK_FAST) != 0; } 696 697 protected: 698 // for numerous 699 friend class PlaybackThread; 700 friend class MixerThread; 701 friend class DirectOutputThread; 702 703 Track(const Track&); 704 Track& operator = (const Track&); 705 706 // AudioBufferProvider interface 707 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts = kInvalidPTS); 708 // releaseBuffer() not overridden 709 710 virtual uint32_t framesReady() const; 711 712 bool isMuted() const { return mMute; } 713 bool isPausing() const { 714 return mState == PAUSING; 715 } 716 bool isPaused() const { 717 return mState == PAUSED; 718 } 719 bool isReady() const; 720 void setPaused() { mState = PAUSED; } 721 void reset(); 722 723 bool isOutputTrack() const { 724 return (mStreamType == AUDIO_STREAM_CNT); 725 } 726 727 bool presentationComplete(size_t framesWritten, size_t audioHalFrames); 728 void triggerEvents(AudioSystem::sync_event_t type); 729 730 public: 731 virtual bool isTimedTrack() const { return false; } 732 protected: 733 734 // we don't really need a lock for these 735 volatile bool mMute; 736 // FILLED state is used for suppressing volume ramp at begin of playing 737 enum {FS_FILLING, FS_FILLED, FS_ACTIVE}; 738 mutable uint8_t mFillingUpStatus; 739 int8_t mRetryCount; 740 const sp<IMemory> mSharedBuffer; 741 bool mResetDone; 742 const audio_stream_type_t mStreamType; 743 int mName; 744 int16_t *mMainBuffer; 745 int32_t *mAuxBuffer; 746 int mAuxEffectId; 747 bool mHasVolumeController; 748 size_t mPresentationCompleteFrames; // number of frames written to the audio HAL 749 // when this track will be fully rendered 750 private: 751 IAudioFlinger::track_flags_t mFlags; 752 }; // end of Track 753 754 class TimedTrack : public Track { 755 public: 756 static sp<TimedTrack> create(PlaybackThread *thread, 757 const sp<Client>& client, 758 audio_stream_type_t streamType, 759 uint32_t sampleRate, 760 audio_format_t format, 761 uint32_t channelMask, 762 int frameCount, 763 const sp<IMemory>& sharedBuffer, 764 int sessionId); 765 ~TimedTrack(); 766 767 class TimedBuffer { 768 public: 769 TimedBuffer(); 770 TimedBuffer(const sp<IMemory>& buffer, int64_t pts); 771 const sp<IMemory>& buffer() const { return mBuffer; } 772 int64_t pts() const { return mPTS; } 773 uint32_t position() const { return mPosition; } 774 void setPosition(uint32_t pos) { mPosition = pos; } 775 private: 776 sp<IMemory> mBuffer; 777 int64_t mPTS; 778 uint32_t mPosition; 779 }; 780 781 // Mixer facing methods. 782 virtual bool isTimedTrack() const { return true; } 783 virtual uint32_t framesReady() const; 784 785 // AudioBufferProvider interface 786 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, 787 int64_t pts); 788 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 789 790 // Client/App facing methods. 791 status_t allocateTimedBuffer(size_t size, 792 sp<IMemory>* buffer); 793 status_t queueTimedBuffer(const sp<IMemory>& buffer, 794 int64_t pts); 795 status_t setMediaTimeTransform(const LinearTransform& xform, 796 TimedAudioTrack::TargetTimeline target); 797 798 private: 799 TimedTrack(PlaybackThread *thread, 800 const sp<Client>& client, 801 audio_stream_type_t streamType, 802 uint32_t sampleRate, 803 audio_format_t format, 804 uint32_t channelMask, 805 int frameCount, 806 const sp<IMemory>& sharedBuffer, 807 int sessionId); 808 809 void timedYieldSamples_l(AudioBufferProvider::Buffer* buffer); 810 void timedYieldSilence_l(uint32_t numFrames, 811 AudioBufferProvider::Buffer* buffer); 812 void trimTimedBufferQueue_l(); 813 void trimTimedBufferQueueHead_l(const char* logTag); 814 void updateFramesPendingAfterTrim_l(const TimedBuffer& buf, 815 const char* logTag); 816 817 uint64_t mLocalTimeFreq; 818 LinearTransform mLocalTimeToSampleTransform; 819 LinearTransform mMediaTimeToSampleTransform; 820 sp<MemoryDealer> mTimedMemoryDealer; 821 822 Vector<TimedBuffer> mTimedBufferQueue; 823 bool mQueueHeadInFlight; 824 bool mTrimQueueHeadOnRelease; 825 uint32_t mFramesPendingInQueue; 826 827 uint8_t* mTimedSilenceBuffer; 828 uint32_t mTimedSilenceBufferSize; 829 mutable Mutex mTimedBufferQueueLock; 830 bool mTimedAudioOutputOnTime; 831 CCHelper mCCHelper; 832 833 Mutex mMediaTimeTransformLock; 834 LinearTransform mMediaTimeTransform; 835 bool mMediaTimeTransformValid; 836 TimedAudioTrack::TargetTimeline mMediaTimeTransformTarget; 837 }; 838 839 840 // playback track 841 class OutputTrack : public Track { 842 public: 843 844 class Buffer: public AudioBufferProvider::Buffer { 845 public: 846 int16_t *mBuffer; 847 }; 848 849 OutputTrack(PlaybackThread *thread, 850 DuplicatingThread *sourceThread, 851 uint32_t sampleRate, 852 audio_format_t format, 853 uint32_t channelMask, 854 int frameCount); 855 virtual ~OutputTrack(); 856 857 virtual status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 858 int triggerSession = 0); 859 virtual void stop(); 860 bool write(int16_t* data, uint32_t frames); 861 bool bufferQueueEmpty() const { return mBufferQueue.size() == 0; } 862 bool isActive() const { return mActive; } 863 const wp<ThreadBase>& thread() const { return mThread; } 864 865 private: 866 867 enum { 868 NO_MORE_BUFFERS = 0x80000001, // same in AudioTrack.h, ok to be different value 869 }; 870 871 status_t obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs); 872 void clearBufferQueue(); 873 874 // Maximum number of pending buffers allocated by OutputTrack::write() 875 static const uint8_t kMaxOverFlowBuffers = 10; 876 877 Vector < Buffer* > mBufferQueue; 878 AudioBufferProvider::Buffer mOutBuffer; 879 bool mActive; 880 DuplicatingThread* const mSourceThread; // for waitTimeMs() in write() 881 }; // end of OutputTrack 882 883 PlaybackThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 884 audio_io_handle_t id, uint32_t device, type_t type); 885 virtual ~PlaybackThread(); 886 887 status_t dump(int fd, const Vector<String16>& args); 888 889 // Thread virtuals 890 virtual status_t readyToRun(); 891 virtual bool threadLoop(); 892 893 // RefBase 894 virtual void onFirstRef(); 895 896protected: 897 // Code snippets that were lifted up out of threadLoop() 898 virtual void threadLoop_mix() = 0; 899 virtual void threadLoop_sleepTime() = 0; 900 virtual void threadLoop_write(); 901 virtual void threadLoop_standby(); 902 903 // prepareTracks_l reads and writes mActiveTracks, and also returns the 904 // pending set of tracks to remove via Vector 'tracksToRemove'. The caller is 905 // responsible for clearing or destroying this Vector later on, when it 906 // is safe to do so. That will drop the final ref count and destroy the tracks. 907 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove) = 0; 908 909public: 910 911 virtual status_t initCheck() const { return (mOutput == NULL) ? NO_INIT : NO_ERROR; } 912 913 // return estimated latency in milliseconds, as reported by HAL 914 uint32_t latency() const; 915 916 void setMasterVolume(float value); 917 void setMasterMute(bool muted); 918 919 void setStreamVolume(audio_stream_type_t stream, float value); 920 void setStreamMute(audio_stream_type_t stream, bool muted); 921 922 float streamVolume(audio_stream_type_t stream) const; 923 924 sp<Track> createTrack_l( 925 const sp<AudioFlinger::Client>& client, 926 audio_stream_type_t streamType, 927 uint32_t sampleRate, 928 audio_format_t format, 929 uint32_t channelMask, 930 int frameCount, 931 const sp<IMemory>& sharedBuffer, 932 int sessionId, 933 IAudioFlinger::track_flags_t flags, 934 pid_t tid, 935 status_t *status); 936 937 AudioStreamOut* getOutput() const; 938 AudioStreamOut* clearOutput(); 939 virtual audio_stream_t* stream() const; 940 941 void suspend() { mSuspended++; } 942 void restore() { if (mSuspended > 0) mSuspended--; } 943 bool isSuspended() const { return (mSuspended > 0); } 944 virtual String8 getParameters(const String8& keys); 945 virtual void audioConfigChanged_l(int event, int param = 0); 946 status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames); 947 int16_t *mixBuffer() const { return mMixBuffer; }; 948 949 virtual void detachAuxEffect_l(int effectId); 950 status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track> track, 951 int EffectId); 952 status_t attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track> track, 953 int EffectId); 954 955 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 956 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 957 virtual uint32_t hasAudioSession(int sessionId); 958 virtual uint32_t getStrategyForSession_l(int sessionId); 959 960 961 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 962 virtual bool isValidSyncEvent(const sp<SyncEvent>& event); 963 964 protected: 965 int16_t* mMixBuffer; 966 uint32_t mSuspended; // suspend count, > 0 means suspended 967 int mBytesWritten; 968 private: 969 // mMasterMute is in both PlaybackThread and in AudioFlinger. When a 970 // PlaybackThread needs to find out if master-muted, it checks it's local 971 // copy rather than the one in AudioFlinger. This optimization saves a lock. 972 bool mMasterMute; 973 void setMasterMute_l(bool muted) { mMasterMute = muted; } 974 protected: 975 SortedVector< wp<Track> > mActiveTracks; 976 977 // Allocate a track name for a given channel mask. 978 // Returns name >= 0 if successful, -1 on failure. 979 virtual int getTrackName_l(audio_channel_mask_t channelMask) = 0; 980 virtual void deleteTrackName_l(int name) = 0; 981 982 // Time to sleep between cycles when: 983 virtual uint32_t activeSleepTimeUs() const; // mixer state MIXER_TRACKS_ENABLED 984 virtual uint32_t idleSleepTimeUs() const = 0; // mixer state MIXER_IDLE 985 virtual uint32_t suspendSleepTimeUs() const = 0; // audio policy manager suspended us 986 // No sleep when mixer state == MIXER_TRACKS_READY; relies on audio HAL stream->write() 987 // No sleep in standby mode; waits on a condition 988 989 // Code snippets that are temporarily lifted up out of threadLoop() until the merge 990 void checkSilentMode_l(); 991 992 // Non-trivial for DUPLICATING only 993 virtual void saveOutputTracks() { } 994 virtual void clearOutputTracks() { } 995 996 // Cache various calculated values, at threadLoop() entry and after a parameter change 997 virtual void cacheParameters_l(); 998 999 private: 1000 1001 friend class AudioFlinger; // for numerous 1002 1003 PlaybackThread(const Client&); 1004 PlaybackThread& operator = (const PlaybackThread&); 1005 1006 status_t addTrack_l(const sp<Track>& track); 1007 void destroyTrack_l(const sp<Track>& track); 1008 void removeTrack_l(const sp<Track>& track); 1009 1010 void readOutputParameters(); 1011 1012 virtual status_t dumpInternals(int fd, const Vector<String16>& args); 1013 status_t dumpTracks(int fd, const Vector<String16>& args); 1014 1015 SortedVector< sp<Track> > mTracks; 1016 // mStreamTypes[] uses 1 additional stream type internally for the OutputTrack used by DuplicatingThread 1017 stream_type_t mStreamTypes[AUDIO_STREAM_CNT + 1]; 1018 AudioStreamOut *mOutput; 1019 float mMasterVolume; 1020 nsecs_t mLastWriteTime; 1021 int mNumWrites; 1022 int mNumDelayedWrites; 1023 bool mInWrite; 1024 1025 // FIXME rename these former local variables of threadLoop to standard "m" names 1026 nsecs_t standbyTime; 1027 size_t mixBufferSize; 1028 1029 // cached copies of activeSleepTimeUs() and idleSleepTimeUs() made by cacheParameters_l() 1030 uint32_t activeSleepTime; 1031 uint32_t idleSleepTime; 1032 1033 uint32_t sleepTime; 1034 1035 // mixer status returned by prepareTracks_l() 1036 mixer_state mMixerStatus; // current cycle 1037 mixer_state mPrevMixerStatus; // previous cycle 1038 1039 // FIXME move these declarations into the specific sub-class that needs them 1040 // MIXER only 1041 bool longStandbyExit; 1042 uint32_t sleepTimeShift; 1043 1044 // same as AudioFlinger::mStandbyTimeInNsecs except for DIRECT which uses a shorter value 1045 nsecs_t standbyDelay; 1046 1047 // MIXER only 1048 nsecs_t maxPeriod; 1049 1050 // DUPLICATING only 1051 uint32_t writeFrames; 1052 }; 1053 1054 class MixerThread : public PlaybackThread { 1055 public: 1056 MixerThread (const sp<AudioFlinger>& audioFlinger, 1057 AudioStreamOut* output, 1058 audio_io_handle_t id, 1059 uint32_t device, 1060 type_t type = MIXER); 1061 virtual ~MixerThread(); 1062 1063 // Thread virtuals 1064 1065 void invalidateTracks(audio_stream_type_t streamType); 1066 virtual bool checkForNewParameters_l(); 1067 virtual status_t dumpInternals(int fd, const Vector<String16>& args); 1068 1069 protected: 1070 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1071 virtual int getTrackName_l(audio_channel_mask_t channelMask); 1072 virtual void deleteTrackName_l(int name); 1073 virtual uint32_t idleSleepTimeUs() const; 1074 virtual uint32_t suspendSleepTimeUs() const; 1075 virtual void cacheParameters_l(); 1076 1077 // threadLoop snippets 1078 virtual void threadLoop_mix(); 1079 virtual void threadLoop_sleepTime(); 1080 1081 AudioMixer* mAudioMixer; 1082 }; 1083 1084 class DirectOutputThread : public PlaybackThread { 1085 public: 1086 1087 DirectOutputThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1088 audio_io_handle_t id, uint32_t device); 1089 virtual ~DirectOutputThread(); 1090 1091 // Thread virtuals 1092 1093 virtual bool checkForNewParameters_l(); 1094 1095 protected: 1096 virtual int getTrackName_l(audio_channel_mask_t channelMask); 1097 virtual void deleteTrackName_l(int name); 1098 virtual uint32_t activeSleepTimeUs() const; 1099 virtual uint32_t idleSleepTimeUs() const; 1100 virtual uint32_t suspendSleepTimeUs() const; 1101 virtual void cacheParameters_l(); 1102 1103 // threadLoop snippets 1104 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1105 virtual void threadLoop_mix(); 1106 virtual void threadLoop_sleepTime(); 1107 1108 // volumes last sent to audio HAL with stream->set_volume() 1109 // FIXME use standard representation and names 1110 float mLeftVolFloat; 1111 float mRightVolFloat; 1112 uint16_t mLeftVolShort; 1113 uint16_t mRightVolShort; 1114 1115 // FIXME rename these former local variables of threadLoop to standard names 1116 // next 3 were local to the while !exitingPending loop 1117 bool rampVolume; 1118 uint16_t leftVol; 1119 uint16_t rightVol; 1120 1121private: 1122 // prepareTracks_l() tells threadLoop_mix() the name of the single active track 1123 sp<Track> mActiveTrack; 1124 }; 1125 1126 class DuplicatingThread : public MixerThread { 1127 public: 1128 DuplicatingThread (const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread, 1129 audio_io_handle_t id); 1130 virtual ~DuplicatingThread(); 1131 1132 // Thread virtuals 1133 void addOutputTrack(MixerThread* thread); 1134 void removeOutputTrack(MixerThread* thread); 1135 uint32_t waitTimeMs() const { return mWaitTimeMs; } 1136 protected: 1137 virtual uint32_t activeSleepTimeUs() const; 1138 1139 private: 1140 bool outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks); 1141 protected: 1142 // threadLoop snippets 1143 virtual void threadLoop_mix(); 1144 virtual void threadLoop_sleepTime(); 1145 virtual void threadLoop_write(); 1146 virtual void threadLoop_standby(); 1147 virtual void cacheParameters_l(); 1148 1149 private: 1150 // called from threadLoop, addOutputTrack, removeOutputTrack 1151 virtual void updateWaitTime_l(); 1152 protected: 1153 virtual void saveOutputTracks(); 1154 virtual void clearOutputTracks(); 1155 private: 1156 1157 uint32_t mWaitTimeMs; 1158 SortedVector < sp<OutputTrack> > outputTracks; 1159 SortedVector < sp<OutputTrack> > mOutputTracks; 1160 }; 1161 1162 PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const; 1163 MixerThread *checkMixerThread_l(audio_io_handle_t output) const; 1164 RecordThread *checkRecordThread_l(audio_io_handle_t input) const; 1165 // no range check, AudioFlinger::mLock held 1166 bool streamMute_l(audio_stream_type_t stream) const 1167 { return mStreamTypes[stream].mute; } 1168 // no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held 1169 float streamVolume_l(audio_stream_type_t stream) const 1170 { return mStreamTypes[stream].volume; } 1171 void audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2); 1172 1173 // allocate an audio_io_handle_t, session ID, or effect ID 1174 uint32_t nextUniqueId(); 1175 1176 status_t moveEffectChain_l(int sessionId, 1177 PlaybackThread *srcThread, 1178 PlaybackThread *dstThread, 1179 bool reRegister); 1180 // return thread associated with primary hardware device, or NULL 1181 PlaybackThread *primaryPlaybackThread_l() const; 1182 uint32_t primaryOutputDevice_l() const; 1183 1184 // server side of the client's IAudioTrack 1185 class TrackHandle : public android::BnAudioTrack { 1186 public: 1187 TrackHandle(const sp<PlaybackThread::Track>& track); 1188 virtual ~TrackHandle(); 1189 virtual sp<IMemory> getCblk() const; 1190 virtual status_t start(); 1191 virtual void stop(); 1192 virtual void flush(); 1193 virtual void mute(bool); 1194 virtual void pause(); 1195 virtual status_t attachAuxEffect(int effectId); 1196 virtual status_t allocateTimedBuffer(size_t size, 1197 sp<IMemory>* buffer); 1198 virtual status_t queueTimedBuffer(const sp<IMemory>& buffer, 1199 int64_t pts); 1200 virtual status_t setMediaTimeTransform(const LinearTransform& xform, 1201 int target); 1202 virtual status_t onTransact( 1203 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 1204 private: 1205 const sp<PlaybackThread::Track> mTrack; 1206 }; 1207 1208 void removeClient_l(pid_t pid); 1209 void removeNotificationClient(pid_t pid); 1210 1211 1212 // record thread 1213 class RecordThread : public ThreadBase, public AudioBufferProvider 1214 { 1215 public: 1216 1217 // record track 1218 class RecordTrack : public TrackBase { 1219 public: 1220 RecordTrack(RecordThread *thread, 1221 const sp<Client>& client, 1222 uint32_t sampleRate, 1223 audio_format_t format, 1224 uint32_t channelMask, 1225 int frameCount, 1226 int sessionId); 1227 virtual ~RecordTrack(); 1228 1229 virtual status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 1230 int triggerSession = 0); 1231 virtual void stop(); 1232 1233 bool overflow() { bool tmp = mOverflow; mOverflow = false; return tmp; } 1234 bool setOverflow() { bool tmp = mOverflow; mOverflow = true; return tmp; } 1235 1236 void dump(char* buffer, size_t size); 1237 1238 private: 1239 friend class AudioFlinger; // for mState 1240 1241 RecordTrack(const RecordTrack&); 1242 RecordTrack& operator = (const RecordTrack&); 1243 1244 // AudioBufferProvider interface 1245 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts = kInvalidPTS); 1246 // releaseBuffer() not overridden 1247 1248 bool mOverflow; 1249 }; 1250 1251 1252 RecordThread(const sp<AudioFlinger>& audioFlinger, 1253 AudioStreamIn *input, 1254 uint32_t sampleRate, 1255 uint32_t channels, 1256 audio_io_handle_t id, 1257 uint32_t device); 1258 virtual ~RecordThread(); 1259 1260 // Thread 1261 virtual bool threadLoop(); 1262 virtual status_t readyToRun(); 1263 1264 // RefBase 1265 virtual void onFirstRef(); 1266 1267 virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; } 1268 sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l( 1269 const sp<AudioFlinger::Client>& client, 1270 uint32_t sampleRate, 1271 audio_format_t format, 1272 int channelMask, 1273 int frameCount, 1274 int sessionId, 1275 status_t *status); 1276 1277 status_t start(RecordTrack* recordTrack, 1278 AudioSystem::sync_event_t event, 1279 int triggerSession); 1280 void stop(RecordTrack* recordTrack); 1281 status_t dump(int fd, const Vector<String16>& args); 1282 AudioStreamIn* getInput() const; 1283 AudioStreamIn* clearInput(); 1284 virtual audio_stream_t* stream() const; 1285 1286 // AudioBufferProvider interface 1287 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts); 1288 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 1289 1290 virtual bool checkForNewParameters_l(); 1291 virtual String8 getParameters(const String8& keys); 1292 virtual void audioConfigChanged_l(int event, int param = 0); 1293 void readInputParameters(); 1294 virtual unsigned int getInputFramesLost(); 1295 1296 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1297 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1298 virtual uint32_t hasAudioSession(int sessionId); 1299 RecordTrack* track(); 1300 1301 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 1302 virtual bool isValidSyncEvent(const sp<SyncEvent>& event); 1303 1304 static void syncStartEventCallback(const wp<SyncEvent>& event); 1305 void handleSyncStartEvent(const sp<SyncEvent>& event); 1306 1307 private: 1308 void clearSyncStartEvent(); 1309 1310 RecordThread(); 1311 AudioStreamIn *mInput; 1312 RecordTrack* mTrack; 1313 sp<RecordTrack> mActiveTrack; 1314 Condition mStartStopCond; 1315 AudioResampler *mResampler; 1316 int32_t *mRsmpOutBuffer; 1317 int16_t *mRsmpInBuffer; 1318 size_t mRsmpInIndex; 1319 size_t mInputBytes; 1320 const int mReqChannelCount; 1321 const uint32_t mReqSampleRate; 1322 ssize_t mBytesRead; 1323 // sync event triggering actual audio capture. Frames read before this event will 1324 // be dropped and therefore not read by the application. 1325 sp<SyncEvent> mSyncStartEvent; 1326 // number of captured frames to drop after the start sync event has been received. 1327 ssize_t mFramestoDrop; 1328 }; 1329 1330 // server side of the client's IAudioRecord 1331 class RecordHandle : public android::BnAudioRecord { 1332 public: 1333 RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack); 1334 virtual ~RecordHandle(); 1335 virtual sp<IMemory> getCblk() const; 1336 virtual status_t start(int event, int triggerSession); 1337 virtual void stop(); 1338 virtual status_t onTransact( 1339 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 1340 private: 1341 const sp<RecordThread::RecordTrack> mRecordTrack; 1342 }; 1343 1344 //--- Audio Effect Management 1345 1346 // EffectModule and EffectChain classes both have their own mutex to protect 1347 // state changes or resource modifications. Always respect the following order 1348 // if multiple mutexes must be acquired to avoid cross deadlock: 1349 // AudioFlinger -> ThreadBase -> EffectChain -> EffectModule 1350 1351 // The EffectModule class is a wrapper object controlling the effect engine implementation 1352 // in the effect library. It prevents concurrent calls to process() and command() functions 1353 // from different client threads. It keeps a list of EffectHandle objects corresponding 1354 // to all client applications using this effect and notifies applications of effect state, 1355 // control or parameter changes. It manages the activation state machine to send appropriate 1356 // reset, enable, disable commands to effect engine and provide volume 1357 // ramping when effects are activated/deactivated. 1358 // When controlling an auxiliary effect, the EffectModule also provides an input buffer used by 1359 // the attached track(s) to accumulate their auxiliary channel. 1360 class EffectModule: public RefBase { 1361 public: 1362 EffectModule(ThreadBase *thread, 1363 const wp<AudioFlinger::EffectChain>& chain, 1364 effect_descriptor_t *desc, 1365 int id, 1366 int sessionId); 1367 virtual ~EffectModule(); 1368 1369 enum effect_state { 1370 IDLE, 1371 RESTART, 1372 STARTING, 1373 ACTIVE, 1374 STOPPING, 1375 STOPPED, 1376 DESTROYED 1377 }; 1378 1379 int id() const { return mId; } 1380 void process(); 1381 void updateState(); 1382 status_t command(uint32_t cmdCode, 1383 uint32_t cmdSize, 1384 void *pCmdData, 1385 uint32_t *replySize, 1386 void *pReplyData); 1387 1388 void reset_l(); 1389 status_t configure(); 1390 status_t init(); 1391 effect_state state() const { 1392 return mState; 1393 } 1394 uint32_t status() { 1395 return mStatus; 1396 } 1397 int sessionId() const { 1398 return mSessionId; 1399 } 1400 status_t setEnabled(bool enabled); 1401 bool isEnabled() const; 1402 bool isProcessEnabled() const; 1403 1404 void setInBuffer(int16_t *buffer) { mConfig.inputCfg.buffer.s16 = buffer; } 1405 int16_t *inBuffer() { return mConfig.inputCfg.buffer.s16; } 1406 void setOutBuffer(int16_t *buffer) { mConfig.outputCfg.buffer.s16 = buffer; } 1407 int16_t *outBuffer() { return mConfig.outputCfg.buffer.s16; } 1408 void setChain(const wp<EffectChain>& chain) { mChain = chain; } 1409 void setThread(const wp<ThreadBase>& thread) { mThread = thread; } 1410 const wp<ThreadBase>& thread() { return mThread; } 1411 1412 status_t addHandle(const sp<EffectHandle>& handle); 1413 void disconnect(const wp<EffectHandle>& handle, bool unpinIfLast); 1414 size_t removeHandle (const wp<EffectHandle>& handle); 1415 1416 effect_descriptor_t& desc() { return mDescriptor; } 1417 wp<EffectChain>& chain() { return mChain; } 1418 1419 status_t setDevice(uint32_t device); 1420 status_t setVolume(uint32_t *left, uint32_t *right, bool controller); 1421 status_t setMode(audio_mode_t mode); 1422 status_t start(); 1423 status_t stop(); 1424 void setSuspended(bool suspended); 1425 bool suspended() const; 1426 1427 sp<EffectHandle> controlHandle(); 1428 1429 bool isPinned() const { return mPinned; } 1430 void unPin() { mPinned = false; } 1431 1432 status_t dump(int fd, const Vector<String16>& args); 1433 1434 protected: 1435 friend class AudioFlinger; // for mHandles 1436 bool mPinned; 1437 1438 // Maximum time allocated to effect engines to complete the turn off sequence 1439 static const uint32_t MAX_DISABLE_TIME_MS = 10000; 1440 1441 EffectModule(const EffectModule&); 1442 EffectModule& operator = (const EffectModule&); 1443 1444 status_t start_l(); 1445 status_t stop_l(); 1446 1447mutable Mutex mLock; // mutex for process, commands and handles list protection 1448 wp<ThreadBase> mThread; // parent thread 1449 wp<EffectChain> mChain; // parent effect chain 1450 int mId; // this instance unique ID 1451 int mSessionId; // audio session ID 1452 effect_descriptor_t mDescriptor;// effect descriptor received from effect engine 1453 effect_config_t mConfig; // input and output audio configuration 1454 effect_handle_t mEffectInterface; // Effect module C API 1455 status_t mStatus; // initialization status 1456 effect_state mState; // current activation state 1457 Vector< wp<EffectHandle> > mHandles; // list of client handles 1458 // First handle in mHandles has highest priority and controls the effect module 1459 uint32_t mMaxDisableWaitCnt; // maximum grace period before forcing an effect off after 1460 // sending disable command. 1461 uint32_t mDisableWaitCnt; // current process() calls count during disable period. 1462 bool mSuspended; // effect is suspended: temporarily disabled by framework 1463 }; 1464 1465 // The EffectHandle class implements the IEffect interface. It provides resources 1466 // to receive parameter updates, keeps track of effect control 1467 // ownership and state and has a pointer to the EffectModule object it is controlling. 1468 // There is one EffectHandle object for each application controlling (or using) 1469 // an effect module. 1470 // The EffectHandle is obtained by calling AudioFlinger::createEffect(). 1471 class EffectHandle: public android::BnEffect { 1472 public: 1473 1474 EffectHandle(const sp<EffectModule>& effect, 1475 const sp<AudioFlinger::Client>& client, 1476 const sp<IEffectClient>& effectClient, 1477 int32_t priority); 1478 virtual ~EffectHandle(); 1479 1480 // IEffect 1481 virtual status_t enable(); 1482 virtual status_t disable(); 1483 virtual status_t command(uint32_t cmdCode, 1484 uint32_t cmdSize, 1485 void *pCmdData, 1486 uint32_t *replySize, 1487 void *pReplyData); 1488 virtual void disconnect(); 1489 private: 1490 void disconnect(bool unpinIfLast); 1491 public: 1492 virtual sp<IMemory> getCblk() const { return mCblkMemory; } 1493 virtual status_t onTransact(uint32_t code, const Parcel& data, 1494 Parcel* reply, uint32_t flags); 1495 1496 1497 // Give or take control of effect module 1498 // - hasControl: true if control is given, false if removed 1499 // - signal: true client app should be signaled of change, false otherwise 1500 // - enabled: state of the effect when control is passed 1501 void setControl(bool hasControl, bool signal, bool enabled); 1502 void commandExecuted(uint32_t cmdCode, 1503 uint32_t cmdSize, 1504 void *pCmdData, 1505 uint32_t replySize, 1506 void *pReplyData); 1507 void setEnabled(bool enabled); 1508 bool enabled() const { return mEnabled; } 1509 1510 // Getters 1511 int id() const { return mEffect->id(); } 1512 int priority() const { return mPriority; } 1513 bool hasControl() const { return mHasControl; } 1514 sp<EffectModule> effect() const { return mEffect; } 1515 1516 void dump(char* buffer, size_t size); 1517 1518 protected: 1519 friend class AudioFlinger; // for mEffect, mHasControl, mEnabled 1520 EffectHandle(const EffectHandle&); 1521 EffectHandle& operator =(const EffectHandle&); 1522 1523 sp<EffectModule> mEffect; // pointer to controlled EffectModule 1524 sp<IEffectClient> mEffectClient; // callback interface for client notifications 1525 /*const*/ sp<Client> mClient; // client for shared memory allocation, see disconnect() 1526 sp<IMemory> mCblkMemory; // shared memory for control block 1527 effect_param_cblk_t* mCblk; // control block for deferred parameter setting via shared memory 1528 uint8_t* mBuffer; // pointer to parameter area in shared memory 1529 int mPriority; // client application priority to control the effect 1530 bool mHasControl; // true if this handle is controlling the effect 1531 bool mEnabled; // cached enable state: needed when the effect is 1532 // restored after being suspended 1533 }; 1534 1535 // the EffectChain class represents a group of effects associated to one audio session. 1536 // There can be any number of EffectChain objects per output mixer thread (PlaybackThread). 1537 // The EffecChain with session ID 0 contains global effects applied to the output mix. 1538 // Effects in this chain can be insert or auxiliary. Effects in other chains (attached to tracks) 1539 // are insert only. The EffectChain maintains an ordered list of effect module, the order corresponding 1540 // in the effect process order. When attached to a track (session ID != 0), it also provide it's own 1541 // input buffer used by the track as accumulation buffer. 1542 class EffectChain: public RefBase { 1543 public: 1544 EffectChain(const wp<ThreadBase>& wThread, int sessionId); 1545 EffectChain(ThreadBase *thread, int sessionId); 1546 virtual ~EffectChain(); 1547 1548 // special key used for an entry in mSuspendedEffects keyed vector 1549 // corresponding to a suspend all request. 1550 static const int kKeyForSuspendAll = 0; 1551 1552 // minimum duration during which we force calling effect process when last track on 1553 // a session is stopped or removed to allow effect tail to be rendered 1554 static const int kProcessTailDurationMs = 1000; 1555 1556 void process_l(); 1557 1558 void lock() { 1559 mLock.lock(); 1560 } 1561 void unlock() { 1562 mLock.unlock(); 1563 } 1564 1565 status_t addEffect_l(const sp<EffectModule>& handle); 1566 size_t removeEffect_l(const sp<EffectModule>& handle); 1567 1568 int sessionId() const { return mSessionId; } 1569 void setSessionId(int sessionId) { mSessionId = sessionId; } 1570 1571 sp<EffectModule> getEffectFromDesc_l(effect_descriptor_t *descriptor); 1572 sp<EffectModule> getEffectFromId_l(int id); 1573 sp<EffectModule> getEffectFromType_l(const effect_uuid_t *type); 1574 bool setVolume_l(uint32_t *left, uint32_t *right); 1575 void setDevice_l(uint32_t device); 1576 void setMode_l(audio_mode_t mode); 1577 1578 void setInBuffer(int16_t *buffer, bool ownsBuffer = false) { 1579 mInBuffer = buffer; 1580 mOwnInBuffer = ownsBuffer; 1581 } 1582 int16_t *inBuffer() const { 1583 return mInBuffer; 1584 } 1585 void setOutBuffer(int16_t *buffer) { 1586 mOutBuffer = buffer; 1587 } 1588 int16_t *outBuffer() const { 1589 return mOutBuffer; 1590 } 1591 1592 void incTrackCnt() { android_atomic_inc(&mTrackCnt); } 1593 void decTrackCnt() { android_atomic_dec(&mTrackCnt); } 1594 int32_t trackCnt() const { return mTrackCnt;} 1595 1596 void incActiveTrackCnt() { android_atomic_inc(&mActiveTrackCnt); 1597 mTailBufferCount = mMaxTailBuffers; } 1598 void decActiveTrackCnt() { android_atomic_dec(&mActiveTrackCnt); } 1599 int32_t activeTrackCnt() const { return mActiveTrackCnt;} 1600 1601 uint32_t strategy() const { return mStrategy; } 1602 void setStrategy(uint32_t strategy) 1603 { mStrategy = strategy; } 1604 1605 // suspend effect of the given type 1606 void setEffectSuspended_l(const effect_uuid_t *type, 1607 bool suspend); 1608 // suspend all eligible effects 1609 void setEffectSuspendedAll_l(bool suspend); 1610 // check if effects should be suspend or restored when a given effect is enable or disabled 1611 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1612 bool enabled); 1613 1614 status_t dump(int fd, const Vector<String16>& args); 1615 1616 protected: 1617 friend class AudioFlinger; // for mThread, mEffects 1618 EffectChain(const EffectChain&); 1619 EffectChain& operator =(const EffectChain&); 1620 1621 class SuspendedEffectDesc : public RefBase { 1622 public: 1623 SuspendedEffectDesc() : mRefCount(0) {} 1624 1625 int mRefCount; 1626 effect_uuid_t mType; 1627 wp<EffectModule> mEffect; 1628 }; 1629 1630 // get a list of effect modules to suspend when an effect of the type 1631 // passed is enabled. 1632 void getSuspendEligibleEffects(Vector< sp<EffectModule> > &effects); 1633 1634 // get an effect module if it is currently enable 1635 sp<EffectModule> getEffectIfEnabled(const effect_uuid_t *type); 1636 // true if the effect whose descriptor is passed can be suspended 1637 // OEMs can modify the rules implemented in this method to exclude specific effect 1638 // types or implementations from the suspend/restore mechanism. 1639 bool isEffectEligibleForSuspend(const effect_descriptor_t& desc); 1640 1641 wp<ThreadBase> mThread; // parent mixer thread 1642 Mutex mLock; // mutex protecting effect list 1643 Vector< sp<EffectModule> > mEffects; // list of effect modules 1644 int mSessionId; // audio session ID 1645 int16_t *mInBuffer; // chain input buffer 1646 int16_t *mOutBuffer; // chain output buffer 1647 volatile int32_t mActiveTrackCnt; // number of active tracks connected 1648 volatile int32_t mTrackCnt; // number of tracks connected 1649 int32_t mTailBufferCount; // current effect tail buffer count 1650 int32_t mMaxTailBuffers; // maximum effect tail buffers 1651 bool mOwnInBuffer; // true if the chain owns its input buffer 1652 int mVolumeCtrlIdx; // index of insert effect having control over volume 1653 uint32_t mLeftVolume; // previous volume on left channel 1654 uint32_t mRightVolume; // previous volume on right channel 1655 uint32_t mNewLeftVolume; // new volume on left channel 1656 uint32_t mNewRightVolume; // new volume on right channel 1657 uint32_t mStrategy; // strategy for this effect chain 1658 // mSuspendedEffects lists all effects currently suspended in the chain. 1659 // Use effect type UUID timelow field as key. There is no real risk of identical 1660 // timeLow fields among effect type UUIDs. 1661 // Updated by updateSuspendedSessions_l() only. 1662 KeyedVector< int, sp<SuspendedEffectDesc> > mSuspendedEffects; 1663 }; 1664 1665 // AudioStreamOut and AudioStreamIn are immutable, so their fields are const. 1666 // For emphasis, we could also make all pointers to them be "const *", 1667 // but that would clutter the code unnecessarily. 1668 1669 struct AudioStreamOut { 1670 audio_hw_device_t* const hwDev; 1671 audio_stream_out_t* const stream; 1672 1673 AudioStreamOut(audio_hw_device_t *dev, audio_stream_out_t *out) : 1674 hwDev(dev), stream(out) {} 1675 }; 1676 1677 struct AudioStreamIn { 1678 audio_hw_device_t* const hwDev; 1679 audio_stream_in_t* const stream; 1680 1681 AudioStreamIn(audio_hw_device_t *dev, audio_stream_in_t *in) : 1682 hwDev(dev), stream(in) {} 1683 }; 1684 1685 // for mAudioSessionRefs only 1686 struct AudioSessionRef { 1687 AudioSessionRef(int sessionid, pid_t pid) : 1688 mSessionid(sessionid), mPid(pid), mCnt(1) {} 1689 const int mSessionid; 1690 const pid_t mPid; 1691 int mCnt; 1692 }; 1693 1694 enum master_volume_support { 1695 // MVS_NONE: 1696 // Audio HAL has no support for master volume, either setting or 1697 // getting. All master volume control must be implemented in SW by the 1698 // AudioFlinger mixing core. 1699 MVS_NONE, 1700 1701 // MVS_SETONLY: 1702 // Audio HAL has support for setting master volume, but not for getting 1703 // master volume (original HAL design did not include a getter). 1704 // AudioFlinger needs to keep track of the last set master volume in 1705 // addition to needing to set an initial, default, master volume at HAL 1706 // load time. 1707 MVS_SETONLY, 1708 1709 // MVS_FULL: 1710 // Audio HAL has support both for setting and getting master volume. 1711 // AudioFlinger should send all set and get master volume requests 1712 // directly to the HAL. 1713 MVS_FULL, 1714 }; 1715 1716 class AudioHwDevice { 1717 public: 1718 AudioHwDevice(const char *moduleName, audio_hw_device_t *hwDevice) : 1719 mModuleName(strdup(moduleName)), mHwDevice(hwDevice){} 1720 ~AudioHwDevice() { free((void *)mModuleName); } 1721 1722 const char *moduleName() const { return mModuleName; } 1723 audio_hw_device_t *hwDevice() const { return mHwDevice; } 1724 private: 1725 const char * const mModuleName; 1726 audio_hw_device_t * const mHwDevice; 1727 }; 1728 1729 mutable Mutex mLock; 1730 1731 DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client() 1732 1733 mutable Mutex mHardwareLock; 1734 // NOTE: If both mLock and mHardwareLock mutexes must be held, 1735 // always take mLock before mHardwareLock 1736 1737 // These two fields are immutable after onFirstRef(), so no lock needed to access 1738 audio_hw_device_t* mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL 1739 DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*> mAudioHwDevs; 1740 1741 // for dump, indicates which hardware operation is currently in progress (but not stream ops) 1742 enum hardware_call_state { 1743 AUDIO_HW_IDLE = 0, // no operation in progress 1744 AUDIO_HW_INIT, // init_check 1745 AUDIO_HW_OUTPUT_OPEN, // open_output_stream 1746 AUDIO_HW_OUTPUT_CLOSE, // unused 1747 AUDIO_HW_INPUT_OPEN, // unused 1748 AUDIO_HW_INPUT_CLOSE, // unused 1749 AUDIO_HW_STANDBY, // unused 1750 AUDIO_HW_SET_MASTER_VOLUME, // set_master_volume 1751 AUDIO_HW_GET_ROUTING, // unused 1752 AUDIO_HW_SET_ROUTING, // unused 1753 AUDIO_HW_GET_MODE, // unused 1754 AUDIO_HW_SET_MODE, // set_mode 1755 AUDIO_HW_GET_MIC_MUTE, // get_mic_mute 1756 AUDIO_HW_SET_MIC_MUTE, // set_mic_mute 1757 AUDIO_HW_SET_VOICE_VOLUME, // set_voice_volume 1758 AUDIO_HW_SET_PARAMETER, // set_parameters 1759 AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size 1760 AUDIO_HW_GET_MASTER_VOLUME, // get_master_volume 1761 AUDIO_HW_GET_PARAMETER, // get_parameters 1762 }; 1763 1764 mutable hardware_call_state mHardwareStatus; // for dump only 1765 1766 1767 DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads; 1768 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 1769 1770 // both are protected by mLock 1771 float mMasterVolume; 1772 float mMasterVolumeSW; 1773 master_volume_support mMasterVolumeSupportLvl; 1774 bool mMasterMute; 1775 1776 DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads; 1777 1778 DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients; 1779 volatile int32_t mNextUniqueId; // updated by android_atomic_inc 1780 audio_mode_t mMode; 1781 bool mBtNrecIsOff; 1782 1783 // protected by mLock 1784 Vector<AudioSessionRef*> mAudioSessionRefs; 1785 1786 float masterVolume_l() const; 1787 float masterVolumeSW_l() const { return mMasterVolumeSW; } 1788 bool masterMute_l() const { return mMasterMute; } 1789 audio_module_handle_t loadHwModule_l(const char *name); 1790 1791 Vector < sp<SyncEvent> > mPendingSyncEvents; // sync events awaiting for a session 1792 // to be created 1793 1794private: 1795 sp<Client> registerPid_l(pid_t pid); // always returns non-0 1796 1797}; 1798 1799 1800// ---------------------------------------------------------------------------- 1801 1802}; // namespace android 1803 1804#endif // ANDROID_AUDIO_FLINGER_H 1805