AudioFlinger.h revision 7d5b26230a179cd7bcc01f6578cd80d8c15a92a5
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 int *sessionId, 95 status_t *status); 96 97 virtual sp<IAudioRecord> openRecord( 98 pid_t pid, 99 audio_io_handle_t input, 100 uint32_t sampleRate, 101 audio_format_t format, 102 uint32_t channelMask, 103 int frameCount, 104 IAudioFlinger::track_flags_t flags, 105 int *sessionId, 106 status_t *status); 107 108 virtual uint32_t sampleRate(audio_io_handle_t output) const; 109 virtual int channelCount(audio_io_handle_t output) const; 110 virtual audio_format_t format(audio_io_handle_t output) const; 111 virtual size_t frameCount(audio_io_handle_t output) const; 112 virtual uint32_t latency(audio_io_handle_t output) const; 113 114 virtual status_t setMasterVolume(float value); 115 virtual status_t setMasterMute(bool muted); 116 117 virtual float masterVolume() const; 118 virtual float masterVolumeSW() const; 119 virtual bool masterMute() const; 120 121 virtual status_t setStreamVolume(audio_stream_type_t stream, float value, 122 audio_io_handle_t output); 123 virtual status_t setStreamMute(audio_stream_type_t stream, bool muted); 124 125 virtual float streamVolume(audio_stream_type_t stream, 126 audio_io_handle_t output) const; 127 virtual bool streamMute(audio_stream_type_t stream) const; 128 129 virtual status_t setMode(audio_mode_t mode); 130 131 virtual status_t setMicMute(bool state); 132 virtual bool getMicMute() const; 133 134 virtual status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs); 135 virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) const; 136 137 virtual void registerClient(const sp<IAudioFlingerClient>& client); 138 139 virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const; 140 141 virtual audio_io_handle_t openOutput(audio_module_handle_t module, 142 audio_devices_t *pDevices, 143 uint32_t *pSamplingRate, 144 audio_format_t *pFormat, 145 audio_channel_mask_t *pChannelMask, 146 uint32_t *pLatencyMs, 147 audio_policy_output_flags_t flags); 148 149 virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1, 150 audio_io_handle_t output2); 151 152 virtual status_t closeOutput(audio_io_handle_t output); 153 154 virtual status_t suspendOutput(audio_io_handle_t output); 155 156 virtual status_t restoreOutput(audio_io_handle_t output); 157 158 virtual audio_io_handle_t openInput(audio_module_handle_t module, 159 audio_devices_t *pDevices, 160 uint32_t *pSamplingRate, 161 audio_format_t *pFormat, 162 audio_channel_mask_t *pChannelMask); 163 164 virtual status_t closeInput(audio_io_handle_t input); 165 166 virtual status_t setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output); 167 168 virtual status_t setVoiceVolume(float volume); 169 170 virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 171 audio_io_handle_t output) const; 172 173 virtual unsigned int getInputFramesLost(audio_io_handle_t ioHandle) const; 174 175 virtual int newAudioSessionId(); 176 177 virtual void acquireAudioSessionId(int audioSession); 178 179 virtual void releaseAudioSessionId(int audioSession); 180 181 virtual status_t queryNumberEffects(uint32_t *numEffects) const; 182 183 virtual status_t queryEffect(uint32_t index, effect_descriptor_t *descriptor) const; 184 185 virtual status_t getEffectDescriptor(const effect_uuid_t *pUuid, 186 effect_descriptor_t *descriptor) const; 187 188 virtual sp<IEffect> createEffect(pid_t pid, 189 effect_descriptor_t *pDesc, 190 const sp<IEffectClient>& effectClient, 191 int32_t priority, 192 audio_io_handle_t io, 193 int sessionId, 194 status_t *status, 195 int *id, 196 int *enabled); 197 198 virtual status_t moveEffects(int sessionId, audio_io_handle_t srcOutput, 199 audio_io_handle_t dstOutput); 200 201 virtual audio_module_handle_t loadHwModule(const char *name); 202 203 virtual status_t onTransact( 204 uint32_t code, 205 const Parcel& data, 206 Parcel* reply, 207 uint32_t flags); 208 209 // end of IAudioFlinger interface 210 211 class SyncEvent; 212 213 typedef void (*sync_event_callback_t)(const wp<SyncEvent>& event) ; 214 215 class SyncEvent : public RefBase { 216 public: 217 SyncEvent(AudioSystem::sync_event_t type, 218 int triggerSession, 219 int listenerSession, 220 sync_event_callback_t callBack, 221 void *cookie) 222 : mType(type), mTriggerSession(triggerSession), mListenerSession(listenerSession), 223 mCallback(callBack), mCookie(cookie) 224 {} 225 226 virtual ~SyncEvent() {} 227 228 void trigger() { Mutex::Autolock _l(mLock); if (mCallback) mCallback(this); } 229 void cancel() {Mutex::Autolock _l(mLock); mCallback = NULL; } 230 AudioSystem::sync_event_t type() const { return mType; } 231 int triggerSession() const { return mTriggerSession; } 232 int listenerSession() const { return mListenerSession; } 233 void *cookie() const { return mCookie; } 234 235 private: 236 const AudioSystem::sync_event_t mType; 237 const int mTriggerSession; 238 const int mListenerSession; 239 sync_event_callback_t mCallback; 240 void * const mCookie; 241 Mutex mLock; 242 }; 243 244 sp<SyncEvent> createSyncEvent(AudioSystem::sync_event_t type, 245 int triggerSession, 246 int listenerSession, 247 sync_event_callback_t callBack, 248 void *cookie); 249private: 250 audio_mode_t getMode() const { return mMode; } 251 252 bool btNrecIsOff() const { return mBtNrecIsOff; } 253 254 AudioFlinger(); 255 virtual ~AudioFlinger(); 256 257 // call in any IAudioFlinger method that accesses mPrimaryHardwareDev 258 status_t initCheck() const { return mPrimaryHardwareDev == NULL ? NO_INIT : NO_ERROR; } 259 260 // RefBase 261 virtual void onFirstRef(); 262 263 audio_hw_device_t* findSuitableHwDev_l(audio_module_handle_t module, uint32_t devices); 264 void purgeStaleEffects_l(); 265 266 // standby delay for MIXER and DUPLICATING playback threads is read from property 267 // ro.audio.flinger_standbytime_ms or defaults to kDefaultStandbyTimeInNsecs 268 static nsecs_t mStandbyTimeInNsecs; 269 270 // Internal dump utilites. 271 status_t dumpPermissionDenial(int fd, const Vector<String16>& args); 272 status_t dumpClients(int fd, const Vector<String16>& args); 273 status_t dumpInternals(int fd, const Vector<String16>& args); 274 275 // --- Client --- 276 class Client : public RefBase { 277 public: 278 Client(const sp<AudioFlinger>& audioFlinger, pid_t pid); 279 virtual ~Client(); 280 sp<MemoryDealer> heap() const; 281 pid_t pid() const { return mPid; } 282 sp<AudioFlinger> audioFlinger() const { return mAudioFlinger; } 283 284 bool reserveTimedTrack(); 285 void releaseTimedTrack(); 286 287 private: 288 Client(const Client&); 289 Client& operator = (const Client&); 290 const sp<AudioFlinger> mAudioFlinger; 291 const sp<MemoryDealer> mMemoryDealer; 292 const pid_t mPid; 293 294 Mutex mTimedTrackLock; 295 int mTimedTrackCount; 296 }; 297 298 // --- Notification Client --- 299 class NotificationClient : public IBinder::DeathRecipient { 300 public: 301 NotificationClient(const sp<AudioFlinger>& audioFlinger, 302 const sp<IAudioFlingerClient>& client, 303 pid_t pid); 304 virtual ~NotificationClient(); 305 306 sp<IAudioFlingerClient> audioFlingerClient() const { return mAudioFlingerClient; } 307 308 // IBinder::DeathRecipient 309 virtual void binderDied(const wp<IBinder>& who); 310 311 private: 312 NotificationClient(const NotificationClient&); 313 NotificationClient& operator = (const NotificationClient&); 314 315 const sp<AudioFlinger> mAudioFlinger; 316 const pid_t mPid; 317 const sp<IAudioFlingerClient> mAudioFlingerClient; 318 }; 319 320 class TrackHandle; 321 class RecordHandle; 322 class RecordThread; 323 class PlaybackThread; 324 class MixerThread; 325 class DirectOutputThread; 326 class DuplicatingThread; 327 class Track; 328 class RecordTrack; 329 class EffectModule; 330 class EffectHandle; 331 class EffectChain; 332 struct AudioStreamOut; 333 struct AudioStreamIn; 334 335 class ThreadBase : public Thread { 336 public: 337 338 enum type_t { 339 MIXER, // Thread class is MixerThread 340 DIRECT, // Thread class is DirectOutputThread 341 DUPLICATING, // Thread class is DuplicatingThread 342 RECORD // Thread class is RecordThread 343 }; 344 345 ThreadBase (const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, uint32_t device, type_t type); 346 virtual ~ThreadBase(); 347 348 status_t dumpBase(int fd, const Vector<String16>& args); 349 status_t dumpEffectChains(int fd, const Vector<String16>& args); 350 351 void clearPowerManager(); 352 353 // base for record and playback 354 class TrackBase : public AudioBufferProvider, public RefBase { 355 356 public: 357 enum track_state { 358 IDLE, 359 TERMINATED, 360 // These are order-sensitive; do not change order without reviewing the impact. 361 // In particular there are assumptions about > STOPPED. 362 STOPPED, 363 RESUMING, 364 ACTIVE, 365 PAUSING, 366 PAUSED 367 }; 368 369 TrackBase(ThreadBase *thread, 370 const sp<Client>& client, 371 uint32_t sampleRate, 372 audio_format_t format, 373 uint32_t channelMask, 374 int frameCount, 375 const sp<IMemory>& sharedBuffer, 376 int sessionId); 377 virtual ~TrackBase(); 378 379 virtual status_t start(pid_t tid, 380 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(pid_t tid, 673 AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 674 int triggerSession = 0); 675 virtual void stop(); 676 void pause(); 677 678 void flush(); 679 void destroy(); 680 void mute(bool); 681 int name() const { 682 return mName; 683 } 684 685 audio_stream_type_t streamType() const { 686 return mStreamType; 687 } 688 status_t attachAuxEffect(int EffectId); 689 void setAuxBuffer(int EffectId, int32_t *buffer); 690 int32_t *auxBuffer() const { return mAuxBuffer; } 691 void setMainBuffer(int16_t *buffer) { mMainBuffer = buffer; } 692 int16_t *mainBuffer() const { return mMainBuffer; } 693 int auxEffectId() const { return mAuxEffectId; } 694 695 bool isFastTrack() const 696 { return (mFlags & IAudioFlinger::TRACK_FAST) != 0; } 697 698 protected: 699 // for numerous 700 friend class PlaybackThread; 701 friend class MixerThread; 702 friend class DirectOutputThread; 703 704 Track(const Track&); 705 Track& operator = (const Track&); 706 707 // AudioBufferProvider interface 708 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts = kInvalidPTS); 709 // releaseBuffer() not overridden 710 711 virtual uint32_t framesReady() const; 712 713 bool isMuted() const { return mMute; } 714 bool isPausing() const { 715 return mState == PAUSING; 716 } 717 bool isPaused() const { 718 return mState == PAUSED; 719 } 720 bool isReady() const; 721 void setPaused() { mState = PAUSED; } 722 void reset(); 723 724 bool isOutputTrack() const { 725 return (mStreamType == AUDIO_STREAM_CNT); 726 } 727 728 bool presentationComplete(size_t framesWritten, size_t audioHalFrames); 729 void triggerEvents(AudioSystem::sync_event_t type); 730 731 public: 732 virtual bool isTimedTrack() const { return false; } 733 protected: 734 735 // we don't really need a lock for these 736 volatile bool mMute; 737 // FILLED state is used for suppressing volume ramp at begin of playing 738 enum {FS_FILLING, FS_FILLED, FS_ACTIVE}; 739 mutable uint8_t mFillingUpStatus; 740 int8_t mRetryCount; 741 const sp<IMemory> mSharedBuffer; 742 bool mResetDone; 743 const audio_stream_type_t mStreamType; 744 int mName; 745 int16_t *mMainBuffer; 746 int32_t *mAuxBuffer; 747 int mAuxEffectId; 748 bool mHasVolumeController; 749 size_t mPresentationCompleteFrames; // number of frames written to the audio HAL 750 // when this track will be fully rendered 751 private: 752 IAudioFlinger::track_flags_t mFlags; 753 }; // end of Track 754 755 class TimedTrack : public Track { 756 public: 757 static sp<TimedTrack> create(PlaybackThread *thread, 758 const sp<Client>& client, 759 audio_stream_type_t streamType, 760 uint32_t sampleRate, 761 audio_format_t format, 762 uint32_t channelMask, 763 int frameCount, 764 const sp<IMemory>& sharedBuffer, 765 int sessionId); 766 ~TimedTrack(); 767 768 class TimedBuffer { 769 public: 770 TimedBuffer(); 771 TimedBuffer(const sp<IMemory>& buffer, int64_t pts); 772 const sp<IMemory>& buffer() const { return mBuffer; } 773 int64_t pts() const { return mPTS; } 774 int position() const { return mPosition; } 775 void setPosition(int pos) { mPosition = pos; } 776 private: 777 sp<IMemory> mBuffer; 778 int64_t mPTS; 779 int mPosition; 780 }; 781 782 virtual bool isTimedTrack() const { return true; } 783 784 virtual uint32_t framesReady() const; 785 786 // AudioBufferProvider interface 787 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts); 788 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 789 790 void timedYieldSamples(AudioBufferProvider::Buffer* buffer); 791 void timedYieldSilence(uint32_t numFrames, 792 AudioBufferProvider::Buffer* buffer); 793 794 status_t allocateTimedBuffer(size_t size, 795 sp<IMemory>* buffer); 796 status_t queueTimedBuffer(const sp<IMemory>& buffer, 797 int64_t pts); 798 status_t setMediaTimeTransform(const LinearTransform& xform, 799 TimedAudioTrack::TargetTimeline target); 800 void trimTimedBufferQueue_l(); 801 802 private: 803 TimedTrack(PlaybackThread *thread, 804 const sp<Client>& client, 805 audio_stream_type_t streamType, 806 uint32_t sampleRate, 807 audio_format_t format, 808 uint32_t channelMask, 809 int frameCount, 810 const sp<IMemory>& sharedBuffer, 811 int sessionId); 812 813 uint64_t mLocalTimeFreq; 814 LinearTransform mLocalTimeToSampleTransform; 815 sp<MemoryDealer> mTimedMemoryDealer; 816 Vector<TimedBuffer> mTimedBufferQueue; 817 uint8_t* mTimedSilenceBuffer; 818 uint32_t mTimedSilenceBufferSize; 819 mutable Mutex mTimedBufferQueueLock; 820 bool mTimedAudioOutputOnTime; 821 CCHelper mCCHelper; 822 823 Mutex mMediaTimeTransformLock; 824 LinearTransform mMediaTimeTransform; 825 bool mMediaTimeTransformValid; 826 TimedAudioTrack::TargetTimeline mMediaTimeTransformTarget; 827 }; 828 829 830 // playback track 831 class OutputTrack : public Track { 832 public: 833 834 class Buffer: public AudioBufferProvider::Buffer { 835 public: 836 int16_t *mBuffer; 837 }; 838 839 OutputTrack(PlaybackThread *thread, 840 DuplicatingThread *sourceThread, 841 uint32_t sampleRate, 842 audio_format_t format, 843 uint32_t channelMask, 844 int frameCount); 845 virtual ~OutputTrack(); 846 847 virtual status_t start(pid_t tid, 848 AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 849 int triggerSession = 0); 850 virtual void stop(); 851 bool write(int16_t* data, uint32_t frames); 852 bool bufferQueueEmpty() const { return mBufferQueue.size() == 0; } 853 bool isActive() const { return mActive; } 854 const wp<ThreadBase>& thread() const { return mThread; } 855 856 private: 857 858 enum { 859 NO_MORE_BUFFERS = 0x80000001, // same in AudioTrack.h, ok to be different value 860 }; 861 862 status_t obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs); 863 void clearBufferQueue(); 864 865 // Maximum number of pending buffers allocated by OutputTrack::write() 866 static const uint8_t kMaxOverFlowBuffers = 10; 867 868 Vector < Buffer* > mBufferQueue; 869 AudioBufferProvider::Buffer mOutBuffer; 870 bool mActive; 871 DuplicatingThread* const mSourceThread; // for waitTimeMs() in write() 872 }; // end of OutputTrack 873 874 PlaybackThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 875 audio_io_handle_t id, uint32_t device, type_t type); 876 virtual ~PlaybackThread(); 877 878 status_t dump(int fd, const Vector<String16>& args); 879 880 // Thread virtuals 881 virtual status_t readyToRun(); 882 virtual bool threadLoop(); 883 884 // RefBase 885 virtual void onFirstRef(); 886 887protected: 888 // Code snippets that were lifted up out of threadLoop() 889 virtual void threadLoop_mix() = 0; 890 virtual void threadLoop_sleepTime() = 0; 891 virtual void threadLoop_write(); 892 virtual void threadLoop_standby(); 893 894 // prepareTracks_l reads and writes mActiveTracks, and also returns the 895 // pending set of tracks to remove via Vector 'tracksToRemove'. The caller is 896 // responsible for clearing or destroying this Vector later on, when it 897 // is safe to do so. That will drop the final ref count and destroy the tracks. 898 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove) = 0; 899 900public: 901 902 virtual status_t initCheck() const { return (mOutput == NULL) ? NO_INIT : NO_ERROR; } 903 904 // return estimated latency in milliseconds, as reported by HAL 905 uint32_t latency() const; 906 907 void setMasterVolume(float value); 908 void setMasterMute(bool muted); 909 910 void setStreamVolume(audio_stream_type_t stream, float value); 911 void setStreamMute(audio_stream_type_t stream, bool muted); 912 913 float streamVolume(audio_stream_type_t stream) const; 914 915 sp<Track> createTrack_l( 916 const sp<AudioFlinger::Client>& client, 917 audio_stream_type_t streamType, 918 uint32_t sampleRate, 919 audio_format_t format, 920 uint32_t channelMask, 921 int frameCount, 922 const sp<IMemory>& sharedBuffer, 923 int sessionId, 924 IAudioFlinger::track_flags_t flags, 925 status_t *status); 926 927 AudioStreamOut* getOutput() const; 928 AudioStreamOut* clearOutput(); 929 virtual audio_stream_t* stream() const; 930 931 void suspend() { mSuspended++; } 932 void restore() { if (mSuspended > 0) mSuspended--; } 933 bool isSuspended() const { return (mSuspended > 0); } 934 virtual String8 getParameters(const String8& keys); 935 virtual void audioConfigChanged_l(int event, int param = 0); 936 status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames); 937 int16_t *mixBuffer() const { return mMixBuffer; }; 938 939 virtual void detachAuxEffect_l(int effectId); 940 status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track> track, 941 int EffectId); 942 status_t attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track> track, 943 int EffectId); 944 945 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 946 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 947 virtual uint32_t hasAudioSession(int sessionId); 948 virtual uint32_t getStrategyForSession_l(int sessionId); 949 950 951 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 952 virtual bool isValidSyncEvent(const sp<SyncEvent>& event); 953 954 protected: 955 int16_t* mMixBuffer; 956 uint32_t mSuspended; // suspend count, > 0 means suspended 957 int mBytesWritten; 958 private: 959 // mMasterMute is in both PlaybackThread and in AudioFlinger. When a 960 // PlaybackThread needs to find out if master-muted, it checks it's local 961 // copy rather than the one in AudioFlinger. This optimization saves a lock. 962 bool mMasterMute; 963 void setMasterMute_l(bool muted) { mMasterMute = muted; } 964 protected: 965 SortedVector< wp<Track> > mActiveTracks; 966 967 // Allocate a track name for a given channel mask. 968 // Returns name >= 0 if successful, -1 on failure. 969 virtual int getTrackName_l(audio_channel_mask_t channelMask) = 0; 970 virtual void deleteTrackName_l(int name) = 0; 971 972 // Time to sleep between cycles when: 973 virtual uint32_t activeSleepTimeUs() const; // mixer state MIXER_TRACKS_ENABLED 974 virtual uint32_t idleSleepTimeUs() const = 0; // mixer state MIXER_IDLE 975 virtual uint32_t suspendSleepTimeUs() const = 0; // audio policy manager suspended us 976 // No sleep when mixer state == MIXER_TRACKS_READY; relies on audio HAL stream->write() 977 // No sleep in standby mode; waits on a condition 978 979 // Code snippets that are temporarily lifted up out of threadLoop() until the merge 980 void checkSilentMode_l(); 981 982 // Non-trivial for DUPLICATING only 983 virtual void saveOutputTracks() { } 984 virtual void clearOutputTracks() { } 985 986 // Cache various calculated values, at threadLoop() entry and after a parameter change 987 virtual void cacheParameters_l(); 988 989 private: 990 991 friend class AudioFlinger; // for numerous 992 993 PlaybackThread(const Client&); 994 PlaybackThread& operator = (const PlaybackThread&); 995 996 status_t addTrack_l(const sp<Track>& track); 997 void destroyTrack_l(const sp<Track>& track); 998 void removeTrack_l(const sp<Track>& track); 999 1000 void readOutputParameters(); 1001 1002 virtual status_t dumpInternals(int fd, const Vector<String16>& args); 1003 status_t dumpTracks(int fd, const Vector<String16>& args); 1004 1005 SortedVector< sp<Track> > mTracks; 1006 // mStreamTypes[] uses 1 additional stream type internally for the OutputTrack used by DuplicatingThread 1007 stream_type_t mStreamTypes[AUDIO_STREAM_CNT + 1]; 1008 AudioStreamOut *mOutput; 1009 float mMasterVolume; 1010 nsecs_t mLastWriteTime; 1011 int mNumWrites; 1012 int mNumDelayedWrites; 1013 bool mInWrite; 1014 1015 // FIXME rename these former local variables of threadLoop to standard "m" names 1016 nsecs_t standbyTime; 1017 size_t mixBufferSize; 1018 1019 // cached copies of activeSleepTimeUs() and idleSleepTimeUs() made by cacheParameters_l() 1020 uint32_t activeSleepTime; 1021 uint32_t idleSleepTime; 1022 1023 uint32_t sleepTime; 1024 1025 // mixer status returned by prepareTracks_l() 1026 mixer_state mMixerStatus; // current cycle 1027 mixer_state mPrevMixerStatus; // previous cycle 1028 1029 // FIXME move these declarations into the specific sub-class that needs them 1030 // MIXER only 1031 bool longStandbyExit; 1032 uint32_t sleepTimeShift; 1033 1034 // same as AudioFlinger::mStandbyTimeInNsecs except for DIRECT which uses a shorter value 1035 nsecs_t standbyDelay; 1036 1037 // MIXER only 1038 nsecs_t maxPeriod; 1039 1040 // DUPLICATING only 1041 uint32_t writeFrames; 1042 }; 1043 1044 class MixerThread : public PlaybackThread { 1045 public: 1046 MixerThread (const sp<AudioFlinger>& audioFlinger, 1047 AudioStreamOut* output, 1048 audio_io_handle_t id, 1049 uint32_t device, 1050 type_t type = MIXER); 1051 virtual ~MixerThread(); 1052 1053 // Thread virtuals 1054 1055 void invalidateTracks(audio_stream_type_t streamType); 1056 virtual bool checkForNewParameters_l(); 1057 virtual status_t dumpInternals(int fd, const Vector<String16>& args); 1058 1059 protected: 1060 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1061 virtual int getTrackName_l(audio_channel_mask_t channelMask); 1062 virtual void deleteTrackName_l(int name); 1063 virtual uint32_t idleSleepTimeUs() const; 1064 virtual uint32_t suspendSleepTimeUs() const; 1065 virtual void cacheParameters_l(); 1066 1067 // threadLoop snippets 1068 virtual void threadLoop_mix(); 1069 virtual void threadLoop_sleepTime(); 1070 1071 AudioMixer* mAudioMixer; 1072 }; 1073 1074 class DirectOutputThread : public PlaybackThread { 1075 public: 1076 1077 DirectOutputThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1078 audio_io_handle_t id, uint32_t device); 1079 virtual ~DirectOutputThread(); 1080 1081 // Thread virtuals 1082 1083 virtual bool checkForNewParameters_l(); 1084 1085 protected: 1086 virtual int getTrackName_l(audio_channel_mask_t channelMask); 1087 virtual void deleteTrackName_l(int name); 1088 virtual uint32_t activeSleepTimeUs() const; 1089 virtual uint32_t idleSleepTimeUs() const; 1090 virtual uint32_t suspendSleepTimeUs() const; 1091 virtual void cacheParameters_l(); 1092 1093 // threadLoop snippets 1094 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1095 virtual void threadLoop_mix(); 1096 virtual void threadLoop_sleepTime(); 1097 1098 // volumes last sent to audio HAL with stream->set_volume() 1099 // FIXME use standard representation and names 1100 float mLeftVolFloat; 1101 float mRightVolFloat; 1102 uint16_t mLeftVolShort; 1103 uint16_t mRightVolShort; 1104 1105 // FIXME rename these former local variables of threadLoop to standard names 1106 // next 3 were local to the while !exitingPending loop 1107 bool rampVolume; 1108 uint16_t leftVol; 1109 uint16_t rightVol; 1110 1111private: 1112 // prepareTracks_l() tells threadLoop_mix() the name of the single active track 1113 sp<Track> mActiveTrack; 1114 }; 1115 1116 class DuplicatingThread : public MixerThread { 1117 public: 1118 DuplicatingThread (const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread, 1119 audio_io_handle_t id); 1120 virtual ~DuplicatingThread(); 1121 1122 // Thread virtuals 1123 void addOutputTrack(MixerThread* thread); 1124 void removeOutputTrack(MixerThread* thread); 1125 uint32_t waitTimeMs() const { return mWaitTimeMs; } 1126 protected: 1127 virtual uint32_t activeSleepTimeUs() const; 1128 1129 private: 1130 bool outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks); 1131 protected: 1132 // threadLoop snippets 1133 virtual void threadLoop_mix(); 1134 virtual void threadLoop_sleepTime(); 1135 virtual void threadLoop_write(); 1136 virtual void threadLoop_standby(); 1137 virtual void cacheParameters_l(); 1138 1139 private: 1140 // called from threadLoop, addOutputTrack, removeOutputTrack 1141 virtual void updateWaitTime_l(); 1142 protected: 1143 virtual void saveOutputTracks(); 1144 virtual void clearOutputTracks(); 1145 private: 1146 1147 uint32_t mWaitTimeMs; 1148 SortedVector < sp<OutputTrack> > outputTracks; 1149 SortedVector < sp<OutputTrack> > mOutputTracks; 1150 }; 1151 1152 PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const; 1153 MixerThread *checkMixerThread_l(audio_io_handle_t output) const; 1154 RecordThread *checkRecordThread_l(audio_io_handle_t input) const; 1155 // no range check, AudioFlinger::mLock held 1156 bool streamMute_l(audio_stream_type_t stream) const 1157 { return mStreamTypes[stream].mute; } 1158 // no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held 1159 float streamVolume_l(audio_stream_type_t stream) const 1160 { return mStreamTypes[stream].volume; } 1161 void audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2); 1162 1163 // allocate an audio_io_handle_t, session ID, or effect ID 1164 uint32_t nextUniqueId(); 1165 1166 status_t moveEffectChain_l(int sessionId, 1167 PlaybackThread *srcThread, 1168 PlaybackThread *dstThread, 1169 bool reRegister); 1170 // return thread associated with primary hardware device, or NULL 1171 PlaybackThread *primaryPlaybackThread_l() const; 1172 uint32_t primaryOutputDevice_l() const; 1173 1174 // server side of the client's IAudioTrack 1175 class TrackHandle : public android::BnAudioTrack { 1176 public: 1177 TrackHandle(const sp<PlaybackThread::Track>& track); 1178 virtual ~TrackHandle(); 1179 virtual sp<IMemory> getCblk() const; 1180 virtual status_t start(pid_t tid); 1181 virtual void stop(); 1182 virtual void flush(); 1183 virtual void mute(bool); 1184 virtual void pause(); 1185 virtual status_t attachAuxEffect(int effectId); 1186 virtual status_t allocateTimedBuffer(size_t size, 1187 sp<IMemory>* buffer); 1188 virtual status_t queueTimedBuffer(const sp<IMemory>& buffer, 1189 int64_t pts); 1190 virtual status_t setMediaTimeTransform(const LinearTransform& xform, 1191 int target); 1192 virtual status_t onTransact( 1193 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 1194 private: 1195 const sp<PlaybackThread::Track> mTrack; 1196 }; 1197 1198 void removeClient_l(pid_t pid); 1199 void removeNotificationClient(pid_t pid); 1200 1201 1202 // record thread 1203 class RecordThread : public ThreadBase, public AudioBufferProvider 1204 { 1205 public: 1206 1207 // record track 1208 class RecordTrack : public TrackBase { 1209 public: 1210 RecordTrack(RecordThread *thread, 1211 const sp<Client>& client, 1212 uint32_t sampleRate, 1213 audio_format_t format, 1214 uint32_t channelMask, 1215 int frameCount, 1216 int sessionId); 1217 virtual ~RecordTrack(); 1218 1219 virtual status_t start(pid_t tid, 1220 AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 1221 int triggerSession = 0); 1222 virtual void stop(); 1223 1224 bool overflow() { bool tmp = mOverflow; mOverflow = false; return tmp; } 1225 bool setOverflow() { bool tmp = mOverflow; mOverflow = true; return tmp; } 1226 1227 void dump(char* buffer, size_t size); 1228 1229 private: 1230 friend class AudioFlinger; // for mState 1231 1232 RecordTrack(const RecordTrack&); 1233 RecordTrack& operator = (const RecordTrack&); 1234 1235 // AudioBufferProvider interface 1236 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts = kInvalidPTS); 1237 // releaseBuffer() not overridden 1238 1239 bool mOverflow; 1240 }; 1241 1242 1243 RecordThread(const sp<AudioFlinger>& audioFlinger, 1244 AudioStreamIn *input, 1245 uint32_t sampleRate, 1246 uint32_t channels, 1247 audio_io_handle_t id, 1248 uint32_t device); 1249 virtual ~RecordThread(); 1250 1251 // Thread 1252 virtual bool threadLoop(); 1253 virtual status_t readyToRun(); 1254 1255 // RefBase 1256 virtual void onFirstRef(); 1257 1258 virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; } 1259 sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l( 1260 const sp<AudioFlinger::Client>& client, 1261 uint32_t sampleRate, 1262 audio_format_t format, 1263 int channelMask, 1264 int frameCount, 1265 int sessionId, 1266 status_t *status); 1267 1268 status_t start(RecordTrack* recordTrack, pid_t tid, 1269 AudioSystem::sync_event_t event, 1270 int triggerSession); 1271 void stop(RecordTrack* recordTrack); 1272 status_t dump(int fd, const Vector<String16>& args); 1273 AudioStreamIn* getInput() const; 1274 AudioStreamIn* clearInput(); 1275 virtual audio_stream_t* stream() const; 1276 1277 // AudioBufferProvider interface 1278 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts); 1279 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 1280 1281 virtual bool checkForNewParameters_l(); 1282 virtual String8 getParameters(const String8& keys); 1283 virtual void audioConfigChanged_l(int event, int param = 0); 1284 void readInputParameters(); 1285 virtual unsigned int getInputFramesLost(); 1286 1287 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1288 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1289 virtual uint32_t hasAudioSession(int sessionId); 1290 RecordTrack* track(); 1291 1292 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 1293 virtual bool isValidSyncEvent(const sp<SyncEvent>& event); 1294 1295 static void syncStartEventCallback(const wp<SyncEvent>& event); 1296 void handleSyncStartEvent(const sp<SyncEvent>& event); 1297 1298 private: 1299 void clearSyncStartEvent(); 1300 1301 RecordThread(); 1302 AudioStreamIn *mInput; 1303 RecordTrack* mTrack; 1304 sp<RecordTrack> mActiveTrack; 1305 Condition mStartStopCond; 1306 AudioResampler *mResampler; 1307 int32_t *mRsmpOutBuffer; 1308 int16_t *mRsmpInBuffer; 1309 size_t mRsmpInIndex; 1310 size_t mInputBytes; 1311 const int mReqChannelCount; 1312 const uint32_t mReqSampleRate; 1313 ssize_t mBytesRead; 1314 // sync event triggering actual audio capture. Frames read before this event will 1315 // be dropped and therefore not read by the application. 1316 sp<SyncEvent> mSyncStartEvent; 1317 // number of captured frames to drop after the start sync event has been received. 1318 ssize_t mFramestoDrop; 1319 }; 1320 1321 // server side of the client's IAudioRecord 1322 class RecordHandle : public android::BnAudioRecord { 1323 public: 1324 RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack); 1325 virtual ~RecordHandle(); 1326 virtual sp<IMemory> getCblk() const; 1327 virtual status_t start(pid_t tid, int event, int triggerSession); 1328 virtual void stop(); 1329 virtual status_t onTransact( 1330 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 1331 private: 1332 const sp<RecordThread::RecordTrack> mRecordTrack; 1333 }; 1334 1335 //--- Audio Effect Management 1336 1337 // EffectModule and EffectChain classes both have their own mutex to protect 1338 // state changes or resource modifications. Always respect the following order 1339 // if multiple mutexes must be acquired to avoid cross deadlock: 1340 // AudioFlinger -> ThreadBase -> EffectChain -> EffectModule 1341 1342 // The EffectModule class is a wrapper object controlling the effect engine implementation 1343 // in the effect library. It prevents concurrent calls to process() and command() functions 1344 // from different client threads. It keeps a list of EffectHandle objects corresponding 1345 // to all client applications using this effect and notifies applications of effect state, 1346 // control or parameter changes. It manages the activation state machine to send appropriate 1347 // reset, enable, disable commands to effect engine and provide volume 1348 // ramping when effects are activated/deactivated. 1349 // When controlling an auxiliary effect, the EffectModule also provides an input buffer used by 1350 // the attached track(s) to accumulate their auxiliary channel. 1351 class EffectModule: public RefBase { 1352 public: 1353 EffectModule(ThreadBase *thread, 1354 const wp<AudioFlinger::EffectChain>& chain, 1355 effect_descriptor_t *desc, 1356 int id, 1357 int sessionId); 1358 virtual ~EffectModule(); 1359 1360 enum effect_state { 1361 IDLE, 1362 RESTART, 1363 STARTING, 1364 ACTIVE, 1365 STOPPING, 1366 STOPPED, 1367 DESTROYED 1368 }; 1369 1370 int id() const { return mId; } 1371 void process(); 1372 void updateState(); 1373 status_t command(uint32_t cmdCode, 1374 uint32_t cmdSize, 1375 void *pCmdData, 1376 uint32_t *replySize, 1377 void *pReplyData); 1378 1379 void reset_l(); 1380 status_t configure(); 1381 status_t init(); 1382 effect_state state() const { 1383 return mState; 1384 } 1385 uint32_t status() { 1386 return mStatus; 1387 } 1388 int sessionId() const { 1389 return mSessionId; 1390 } 1391 status_t setEnabled(bool enabled); 1392 bool isEnabled() const; 1393 bool isProcessEnabled() const; 1394 1395 void setInBuffer(int16_t *buffer) { mConfig.inputCfg.buffer.s16 = buffer; } 1396 int16_t *inBuffer() { return mConfig.inputCfg.buffer.s16; } 1397 void setOutBuffer(int16_t *buffer) { mConfig.outputCfg.buffer.s16 = buffer; } 1398 int16_t *outBuffer() { return mConfig.outputCfg.buffer.s16; } 1399 void setChain(const wp<EffectChain>& chain) { mChain = chain; } 1400 void setThread(const wp<ThreadBase>& thread) { mThread = thread; } 1401 const wp<ThreadBase>& thread() { return mThread; } 1402 1403 status_t addHandle(const sp<EffectHandle>& handle); 1404 void disconnect(const wp<EffectHandle>& handle, bool unpinIfLast); 1405 size_t removeHandle (const wp<EffectHandle>& handle); 1406 1407 effect_descriptor_t& desc() { return mDescriptor; } 1408 wp<EffectChain>& chain() { return mChain; } 1409 1410 status_t setDevice(uint32_t device); 1411 status_t setVolume(uint32_t *left, uint32_t *right, bool controller); 1412 status_t setMode(audio_mode_t mode); 1413 status_t start(); 1414 status_t stop(); 1415 void setSuspended(bool suspended); 1416 bool suspended() const; 1417 1418 sp<EffectHandle> controlHandle(); 1419 1420 bool isPinned() const { return mPinned; } 1421 void unPin() { mPinned = false; } 1422 1423 status_t dump(int fd, const Vector<String16>& args); 1424 1425 protected: 1426 friend class AudioFlinger; // for mHandles 1427 bool mPinned; 1428 1429 // Maximum time allocated to effect engines to complete the turn off sequence 1430 static const uint32_t MAX_DISABLE_TIME_MS = 10000; 1431 1432 EffectModule(const EffectModule&); 1433 EffectModule& operator = (const EffectModule&); 1434 1435 status_t start_l(); 1436 status_t stop_l(); 1437 1438mutable Mutex mLock; // mutex for process, commands and handles list protection 1439 wp<ThreadBase> mThread; // parent thread 1440 wp<EffectChain> mChain; // parent effect chain 1441 int mId; // this instance unique ID 1442 int mSessionId; // audio session ID 1443 effect_descriptor_t mDescriptor;// effect descriptor received from effect engine 1444 effect_config_t mConfig; // input and output audio configuration 1445 effect_handle_t mEffectInterface; // Effect module C API 1446 status_t mStatus; // initialization status 1447 effect_state mState; // current activation state 1448 Vector< wp<EffectHandle> > mHandles; // list of client handles 1449 // First handle in mHandles has highest priority and controls the effect module 1450 uint32_t mMaxDisableWaitCnt; // maximum grace period before forcing an effect off after 1451 // sending disable command. 1452 uint32_t mDisableWaitCnt; // current process() calls count during disable period. 1453 bool mSuspended; // effect is suspended: temporarily disabled by framework 1454 }; 1455 1456 // The EffectHandle class implements the IEffect interface. It provides resources 1457 // to receive parameter updates, keeps track of effect control 1458 // ownership and state and has a pointer to the EffectModule object it is controlling. 1459 // There is one EffectHandle object for each application controlling (or using) 1460 // an effect module. 1461 // The EffectHandle is obtained by calling AudioFlinger::createEffect(). 1462 class EffectHandle: public android::BnEffect { 1463 public: 1464 1465 EffectHandle(const sp<EffectModule>& effect, 1466 const sp<AudioFlinger::Client>& client, 1467 const sp<IEffectClient>& effectClient, 1468 int32_t priority); 1469 virtual ~EffectHandle(); 1470 1471 // IEffect 1472 virtual status_t enable(); 1473 virtual status_t disable(); 1474 virtual status_t command(uint32_t cmdCode, 1475 uint32_t cmdSize, 1476 void *pCmdData, 1477 uint32_t *replySize, 1478 void *pReplyData); 1479 virtual void disconnect(); 1480 private: 1481 void disconnect(bool unpinIfLast); 1482 public: 1483 virtual sp<IMemory> getCblk() const { return mCblkMemory; } 1484 virtual status_t onTransact(uint32_t code, const Parcel& data, 1485 Parcel* reply, uint32_t flags); 1486 1487 1488 // Give or take control of effect module 1489 // - hasControl: true if control is given, false if removed 1490 // - signal: true client app should be signaled of change, false otherwise 1491 // - enabled: state of the effect when control is passed 1492 void setControl(bool hasControl, bool signal, bool enabled); 1493 void commandExecuted(uint32_t cmdCode, 1494 uint32_t cmdSize, 1495 void *pCmdData, 1496 uint32_t replySize, 1497 void *pReplyData); 1498 void setEnabled(bool enabled); 1499 bool enabled() const { return mEnabled; } 1500 1501 // Getters 1502 int id() const { return mEffect->id(); } 1503 int priority() const { return mPriority; } 1504 bool hasControl() const { return mHasControl; } 1505 sp<EffectModule> effect() const { return mEffect; } 1506 1507 void dump(char* buffer, size_t size); 1508 1509 protected: 1510 friend class AudioFlinger; // for mEffect, mHasControl, mEnabled 1511 EffectHandle(const EffectHandle&); 1512 EffectHandle& operator =(const EffectHandle&); 1513 1514 sp<EffectModule> mEffect; // pointer to controlled EffectModule 1515 sp<IEffectClient> mEffectClient; // callback interface for client notifications 1516 /*const*/ sp<Client> mClient; // client for shared memory allocation, see disconnect() 1517 sp<IMemory> mCblkMemory; // shared memory for control block 1518 effect_param_cblk_t* mCblk; // control block for deferred parameter setting via shared memory 1519 uint8_t* mBuffer; // pointer to parameter area in shared memory 1520 int mPriority; // client application priority to control the effect 1521 bool mHasControl; // true if this handle is controlling the effect 1522 bool mEnabled; // cached enable state: needed when the effect is 1523 // restored after being suspended 1524 }; 1525 1526 // the EffectChain class represents a group of effects associated to one audio session. 1527 // There can be any number of EffectChain objects per output mixer thread (PlaybackThread). 1528 // The EffecChain with session ID 0 contains global effects applied to the output mix. 1529 // Effects in this chain can be insert or auxiliary. Effects in other chains (attached to tracks) 1530 // are insert only. The EffectChain maintains an ordered list of effect module, the order corresponding 1531 // in the effect process order. When attached to a track (session ID != 0), it also provide it's own 1532 // input buffer used by the track as accumulation buffer. 1533 class EffectChain: public RefBase { 1534 public: 1535 EffectChain(const wp<ThreadBase>& wThread, int sessionId); 1536 EffectChain(ThreadBase *thread, int sessionId); 1537 virtual ~EffectChain(); 1538 1539 // special key used for an entry in mSuspendedEffects keyed vector 1540 // corresponding to a suspend all request. 1541 static const int kKeyForSuspendAll = 0; 1542 1543 // minimum duration during which we force calling effect process when last track on 1544 // a session is stopped or removed to allow effect tail to be rendered 1545 static const int kProcessTailDurationMs = 1000; 1546 1547 void process_l(); 1548 1549 void lock() { 1550 mLock.lock(); 1551 } 1552 void unlock() { 1553 mLock.unlock(); 1554 } 1555 1556 status_t addEffect_l(const sp<EffectModule>& handle); 1557 size_t removeEffect_l(const sp<EffectModule>& handle); 1558 1559 int sessionId() const { return mSessionId; } 1560 void setSessionId(int sessionId) { mSessionId = sessionId; } 1561 1562 sp<EffectModule> getEffectFromDesc_l(effect_descriptor_t *descriptor); 1563 sp<EffectModule> getEffectFromId_l(int id); 1564 sp<EffectModule> getEffectFromType_l(const effect_uuid_t *type); 1565 bool setVolume_l(uint32_t *left, uint32_t *right); 1566 void setDevice_l(uint32_t device); 1567 void setMode_l(audio_mode_t mode); 1568 1569 void setInBuffer(int16_t *buffer, bool ownsBuffer = false) { 1570 mInBuffer = buffer; 1571 mOwnInBuffer = ownsBuffer; 1572 } 1573 int16_t *inBuffer() const { 1574 return mInBuffer; 1575 } 1576 void setOutBuffer(int16_t *buffer) { 1577 mOutBuffer = buffer; 1578 } 1579 int16_t *outBuffer() const { 1580 return mOutBuffer; 1581 } 1582 1583 void incTrackCnt() { android_atomic_inc(&mTrackCnt); } 1584 void decTrackCnt() { android_atomic_dec(&mTrackCnt); } 1585 int32_t trackCnt() const { return mTrackCnt;} 1586 1587 void incActiveTrackCnt() { android_atomic_inc(&mActiveTrackCnt); 1588 mTailBufferCount = mMaxTailBuffers; } 1589 void decActiveTrackCnt() { android_atomic_dec(&mActiveTrackCnt); } 1590 int32_t activeTrackCnt() const { return mActiveTrackCnt;} 1591 1592 uint32_t strategy() const { return mStrategy; } 1593 void setStrategy(uint32_t strategy) 1594 { mStrategy = strategy; } 1595 1596 // suspend effect of the given type 1597 void setEffectSuspended_l(const effect_uuid_t *type, 1598 bool suspend); 1599 // suspend all eligible effects 1600 void setEffectSuspendedAll_l(bool suspend); 1601 // check if effects should be suspend or restored when a given effect is enable or disabled 1602 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1603 bool enabled); 1604 1605 status_t dump(int fd, const Vector<String16>& args); 1606 1607 protected: 1608 friend class AudioFlinger; // for mThread, mEffects 1609 EffectChain(const EffectChain&); 1610 EffectChain& operator =(const EffectChain&); 1611 1612 class SuspendedEffectDesc : public RefBase { 1613 public: 1614 SuspendedEffectDesc() : mRefCount(0) {} 1615 1616 int mRefCount; 1617 effect_uuid_t mType; 1618 wp<EffectModule> mEffect; 1619 }; 1620 1621 // get a list of effect modules to suspend when an effect of the type 1622 // passed is enabled. 1623 void getSuspendEligibleEffects(Vector< sp<EffectModule> > &effects); 1624 1625 // get an effect module if it is currently enable 1626 sp<EffectModule> getEffectIfEnabled(const effect_uuid_t *type); 1627 // true if the effect whose descriptor is passed can be suspended 1628 // OEMs can modify the rules implemented in this method to exclude specific effect 1629 // types or implementations from the suspend/restore mechanism. 1630 bool isEffectEligibleForSuspend(const effect_descriptor_t& desc); 1631 1632 wp<ThreadBase> mThread; // parent mixer thread 1633 Mutex mLock; // mutex protecting effect list 1634 Vector< sp<EffectModule> > mEffects; // list of effect modules 1635 int mSessionId; // audio session ID 1636 int16_t *mInBuffer; // chain input buffer 1637 int16_t *mOutBuffer; // chain output buffer 1638 volatile int32_t mActiveTrackCnt; // number of active tracks connected 1639 volatile int32_t mTrackCnt; // number of tracks connected 1640 int32_t mTailBufferCount; // current effect tail buffer count 1641 int32_t mMaxTailBuffers; // maximum effect tail buffers 1642 bool mOwnInBuffer; // true if the chain owns its input buffer 1643 int mVolumeCtrlIdx; // index of insert effect having control over volume 1644 uint32_t mLeftVolume; // previous volume on left channel 1645 uint32_t mRightVolume; // previous volume on right channel 1646 uint32_t mNewLeftVolume; // new volume on left channel 1647 uint32_t mNewRightVolume; // new volume on right channel 1648 uint32_t mStrategy; // strategy for this effect chain 1649 // mSuspendedEffects lists all effects currently suspended in the chain. 1650 // Use effect type UUID timelow field as key. There is no real risk of identical 1651 // timeLow fields among effect type UUIDs. 1652 // Updated by updateSuspendedSessions_l() only. 1653 KeyedVector< int, sp<SuspendedEffectDesc> > mSuspendedEffects; 1654 }; 1655 1656 // AudioStreamOut and AudioStreamIn are immutable, so their fields are const. 1657 // For emphasis, we could also make all pointers to them be "const *", 1658 // but that would clutter the code unnecessarily. 1659 1660 struct AudioStreamOut { 1661 audio_hw_device_t* const hwDev; 1662 audio_stream_out_t* const stream; 1663 1664 AudioStreamOut(audio_hw_device_t *dev, audio_stream_out_t *out) : 1665 hwDev(dev), stream(out) {} 1666 }; 1667 1668 struct AudioStreamIn { 1669 audio_hw_device_t* const hwDev; 1670 audio_stream_in_t* const stream; 1671 1672 AudioStreamIn(audio_hw_device_t *dev, audio_stream_in_t *in) : 1673 hwDev(dev), stream(in) {} 1674 }; 1675 1676 // for mAudioSessionRefs only 1677 struct AudioSessionRef { 1678 AudioSessionRef(int sessionid, pid_t pid) : 1679 mSessionid(sessionid), mPid(pid), mCnt(1) {} 1680 const int mSessionid; 1681 const pid_t mPid; 1682 int mCnt; 1683 }; 1684 1685 enum master_volume_support { 1686 // MVS_NONE: 1687 // Audio HAL has no support for master volume, either setting or 1688 // getting. All master volume control must be implemented in SW by the 1689 // AudioFlinger mixing core. 1690 MVS_NONE, 1691 1692 // MVS_SETONLY: 1693 // Audio HAL has support for setting master volume, but not for getting 1694 // master volume (original HAL design did not include a getter). 1695 // AudioFlinger needs to keep track of the last set master volume in 1696 // addition to needing to set an initial, default, master volume at HAL 1697 // load time. 1698 MVS_SETONLY, 1699 1700 // MVS_FULL: 1701 // Audio HAL has support both for setting and getting master volume. 1702 // AudioFlinger should send all set and get master volume requests 1703 // directly to the HAL. 1704 MVS_FULL, 1705 }; 1706 1707 class AudioHwDevice { 1708 public: 1709 AudioHwDevice(const char *moduleName, audio_hw_device_t *hwDevice) : 1710 mModuleName(strdup(moduleName)), mHwDevice(hwDevice){} 1711 ~AudioHwDevice() { free((void *)mModuleName); } 1712 1713 const char *moduleName() const { return mModuleName; } 1714 audio_hw_device_t *hwDevice() const { return mHwDevice; } 1715 private: 1716 const char * const mModuleName; 1717 audio_hw_device_t * const mHwDevice; 1718 }; 1719 1720 mutable Mutex mLock; 1721 1722 DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client() 1723 1724 mutable Mutex mHardwareLock; 1725 // NOTE: If both mLock and mHardwareLock mutexes must be held, 1726 // always take mLock before mHardwareLock 1727 1728 // These two fields are immutable after onFirstRef(), so no lock needed to access 1729 audio_hw_device_t* mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL 1730 DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*> mAudioHwDevs; 1731 1732 // for dump, indicates which hardware operation is currently in progress (but not stream ops) 1733 enum hardware_call_state { 1734 AUDIO_HW_IDLE = 0, // no operation in progress 1735 AUDIO_HW_INIT, // init_check 1736 AUDIO_HW_OUTPUT_OPEN, // open_output_stream 1737 AUDIO_HW_OUTPUT_CLOSE, // unused 1738 AUDIO_HW_INPUT_OPEN, // unused 1739 AUDIO_HW_INPUT_CLOSE, // unused 1740 AUDIO_HW_STANDBY, // unused 1741 AUDIO_HW_SET_MASTER_VOLUME, // set_master_volume 1742 AUDIO_HW_GET_ROUTING, // unused 1743 AUDIO_HW_SET_ROUTING, // unused 1744 AUDIO_HW_GET_MODE, // unused 1745 AUDIO_HW_SET_MODE, // set_mode 1746 AUDIO_HW_GET_MIC_MUTE, // get_mic_mute 1747 AUDIO_HW_SET_MIC_MUTE, // set_mic_mute 1748 AUDIO_HW_SET_VOICE_VOLUME, // set_voice_volume 1749 AUDIO_HW_SET_PARAMETER, // set_parameters 1750 AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size 1751 AUDIO_HW_GET_MASTER_VOLUME, // get_master_volume 1752 AUDIO_HW_GET_PARAMETER, // get_parameters 1753 }; 1754 1755 mutable hardware_call_state mHardwareStatus; // for dump only 1756 1757 1758 DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads; 1759 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 1760 1761 // both are protected by mLock 1762 float mMasterVolume; 1763 float mMasterVolumeSW; 1764 master_volume_support mMasterVolumeSupportLvl; 1765 bool mMasterMute; 1766 1767 DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads; 1768 1769 DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients; 1770 volatile int32_t mNextUniqueId; // updated by android_atomic_inc 1771 audio_mode_t mMode; 1772 bool mBtNrecIsOff; 1773 1774 // protected by mLock 1775 Vector<AudioSessionRef*> mAudioSessionRefs; 1776 1777 float masterVolume_l() const; 1778 float masterVolumeSW_l() const { return mMasterVolumeSW; } 1779 bool masterMute_l() const { return mMasterMute; } 1780 audio_module_handle_t loadHwModule_l(const char *name); 1781 1782 Vector < sp<SyncEvent> > mPendingSyncEvents; // sync events awaiting for a session 1783 // to be created 1784 1785private: 1786 sp<Client> registerPid_l(pid_t pid); // always returns non-0 1787 1788}; 1789 1790 1791// ---------------------------------------------------------------------------- 1792 1793}; // namespace android 1794 1795#endif // ANDROID_AUDIO_FLINGER_H 1796