AudioFlinger.h revision cc0f1cfb69ce8b8985fc2c0984847a06a13ad22d
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 <media/AudioBufferProvider.h> 49#include <media/ExtendedAudioBufferProvider.h> 50#include "FastMixer.h" 51#include <media/nbaio/NBAIO.h> 52#include "AudioWatchdog.h" 53 54#include <powermanager/IPowerManager.h> 55 56namespace android { 57 58class audio_track_cblk_t; 59class effect_param_cblk_t; 60class AudioMixer; 61class AudioBuffer; 62class AudioResampler; 63class FastMixer; 64 65// ---------------------------------------------------------------------------- 66 67// AudioFlinger has a hard-coded upper limit of 2 channels for capture and playback. 68// There is support for > 2 channel tracks down-mixed to 2 channel output via a down-mix effect. 69// Adding full support for > 2 channel capture or playback would require more than simply changing 70// this #define. There is an independent hard-coded upper limit in AudioMixer; 71// removing that AudioMixer limit would be necessary but insufficient to support > 2 channels. 72// The macro FCC_2 highlights some (but not all) places where there is are 2-channel assumptions. 73// Search also for "2", "left", "right", "[0]", "[1]", ">> 16", "<< 16", etc. 74#define FCC_2 2 // FCC_2 = Fixed Channel Count 2 75 76static const nsecs_t kDefaultStandbyTimeInNsecs = seconds(3); 77 78class AudioFlinger : 79 public BinderService<AudioFlinger>, 80 public BnAudioFlinger 81{ 82 friend class BinderService<AudioFlinger>; // for AudioFlinger() 83public: 84 static const char* getServiceName() { return "media.audio_flinger"; } 85 86 virtual status_t dump(int fd, const Vector<String16>& args); 87 88 // IAudioFlinger interface, in binder opcode order 89 virtual sp<IAudioTrack> createTrack( 90 pid_t pid, 91 audio_stream_type_t streamType, 92 uint32_t sampleRate, 93 audio_format_t format, 94 audio_channel_mask_t channelMask, 95 int frameCount, 96 IAudioFlinger::track_flags_t flags, 97 const sp<IMemory>& sharedBuffer, 98 audio_io_handle_t output, 99 pid_t tid, 100 int *sessionId, 101 status_t *status); 102 103 virtual sp<IAudioRecord> openRecord( 104 pid_t pid, 105 audio_io_handle_t input, 106 uint32_t sampleRate, 107 audio_format_t format, 108 audio_channel_mask_t channelMask, 109 int frameCount, 110 IAudioFlinger::track_flags_t flags, 111 pid_t tid, 112 int *sessionId, 113 status_t *status); 114 115 virtual uint32_t sampleRate(audio_io_handle_t output) const; 116 virtual int channelCount(audio_io_handle_t output) const; 117 virtual audio_format_t format(audio_io_handle_t output) const; 118 virtual size_t frameCount(audio_io_handle_t output) const; 119 virtual uint32_t latency(audio_io_handle_t output) const; 120 121 virtual status_t setMasterVolume(float value); 122 virtual status_t setMasterMute(bool muted); 123 124 virtual float masterVolume() const; 125 virtual bool masterMute() const; 126 127 virtual status_t setStreamVolume(audio_stream_type_t stream, float value, 128 audio_io_handle_t output); 129 virtual status_t setStreamMute(audio_stream_type_t stream, bool muted); 130 131 virtual float streamVolume(audio_stream_type_t stream, 132 audio_io_handle_t output) const; 133 virtual bool streamMute(audio_stream_type_t stream) const; 134 135 virtual status_t setMode(audio_mode_t mode); 136 137 virtual status_t setMicMute(bool state); 138 virtual bool getMicMute() const; 139 140 virtual status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs); 141 virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) const; 142 143 virtual void registerClient(const sp<IAudioFlingerClient>& client); 144 145 virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, 146 audio_channel_mask_t channelMask) const; 147 148 virtual audio_io_handle_t openOutput(audio_module_handle_t module, 149 audio_devices_t *pDevices, 150 uint32_t *pSamplingRate, 151 audio_format_t *pFormat, 152 audio_channel_mask_t *pChannelMask, 153 uint32_t *pLatencyMs, 154 audio_output_flags_t flags); 155 156 virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1, 157 audio_io_handle_t output2); 158 159 virtual status_t closeOutput(audio_io_handle_t output); 160 161 virtual status_t suspendOutput(audio_io_handle_t output); 162 163 virtual status_t restoreOutput(audio_io_handle_t output); 164 165 virtual audio_io_handle_t openInput(audio_module_handle_t module, 166 audio_devices_t *pDevices, 167 uint32_t *pSamplingRate, 168 audio_format_t *pFormat, 169 audio_channel_mask_t *pChannelMask); 170 171 virtual status_t closeInput(audio_io_handle_t input); 172 173 virtual status_t setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output); 174 175 virtual status_t setVoiceVolume(float volume); 176 177 virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 178 audio_io_handle_t output) const; 179 180 virtual unsigned int getInputFramesLost(audio_io_handle_t ioHandle) const; 181 182 virtual int newAudioSessionId(); 183 184 virtual void acquireAudioSessionId(int audioSession); 185 186 virtual void releaseAudioSessionId(int audioSession); 187 188 virtual status_t queryNumberEffects(uint32_t *numEffects) const; 189 190 virtual status_t queryEffect(uint32_t index, effect_descriptor_t *descriptor) const; 191 192 virtual status_t getEffectDescriptor(const effect_uuid_t *pUuid, 193 effect_descriptor_t *descriptor) const; 194 195 virtual sp<IEffect> createEffect(pid_t pid, 196 effect_descriptor_t *pDesc, 197 const sp<IEffectClient>& effectClient, 198 int32_t priority, 199 audio_io_handle_t io, 200 int sessionId, 201 status_t *status, 202 int *id, 203 int *enabled); 204 205 virtual status_t moveEffects(int sessionId, audio_io_handle_t srcOutput, 206 audio_io_handle_t dstOutput); 207 208 virtual audio_module_handle_t loadHwModule(const char *name); 209 210 virtual int32_t getPrimaryOutputSamplingRate(); 211 virtual int32_t getPrimaryOutputFrameCount(); 212 213 virtual status_t onTransact( 214 uint32_t code, 215 const Parcel& data, 216 Parcel* reply, 217 uint32_t flags); 218 219 // end of IAudioFlinger interface 220 221 class SyncEvent; 222 223 typedef void (*sync_event_callback_t)(const wp<SyncEvent>& event) ; 224 225 class SyncEvent : public RefBase { 226 public: 227 SyncEvent(AudioSystem::sync_event_t type, 228 int triggerSession, 229 int listenerSession, 230 sync_event_callback_t callBack, 231 void *cookie) 232 : mType(type), mTriggerSession(triggerSession), mListenerSession(listenerSession), 233 mCallback(callBack), mCookie(cookie) 234 {} 235 236 virtual ~SyncEvent() {} 237 238 void trigger() { Mutex::Autolock _l(mLock); if (mCallback) mCallback(this); } 239 bool isCancelled() const { Mutex::Autolock _l(mLock); return (mCallback == NULL); } 240 void cancel() { Mutex::Autolock _l(mLock); mCallback = NULL; } 241 AudioSystem::sync_event_t type() const { return mType; } 242 int triggerSession() const { return mTriggerSession; } 243 int listenerSession() const { return mListenerSession; } 244 void *cookie() const { return mCookie; } 245 246 private: 247 const AudioSystem::sync_event_t mType; 248 const int mTriggerSession; 249 const int mListenerSession; 250 sync_event_callback_t mCallback; 251 void * const mCookie; 252 mutable Mutex mLock; 253 }; 254 255 sp<SyncEvent> createSyncEvent(AudioSystem::sync_event_t type, 256 int triggerSession, 257 int listenerSession, 258 sync_event_callback_t callBack, 259 void *cookie); 260 261private: 262 class AudioHwDevice; // fwd declaration for findSuitableHwDev_l 263 264 audio_mode_t getMode() const { return mMode; } 265 266 bool btNrecIsOff() const { return mBtNrecIsOff; } 267 268 AudioFlinger(); 269 virtual ~AudioFlinger(); 270 271 // call in any IAudioFlinger method that accesses mPrimaryHardwareDev 272 status_t initCheck() const { return mPrimaryHardwareDev == NULL ? NO_INIT : NO_ERROR; } 273 274 // RefBase 275 virtual void onFirstRef(); 276 277 AudioHwDevice* findSuitableHwDev_l(audio_module_handle_t module, audio_devices_t devices); 278 void purgeStaleEffects_l(); 279 280 // standby delay for MIXER and DUPLICATING playback threads is read from property 281 // ro.audio.flinger_standbytime_ms or defaults to kDefaultStandbyTimeInNsecs 282 static nsecs_t mStandbyTimeInNsecs; 283 284 // Internal dump utilities. 285 void dumpPermissionDenial(int fd, const Vector<String16>& args); 286 void dumpClients(int fd, const Vector<String16>& args); 287 void dumpInternals(int fd, const Vector<String16>& args); 288 289 // --- Client --- 290 class Client : public RefBase { 291 public: 292 Client(const sp<AudioFlinger>& audioFlinger, pid_t pid); 293 virtual ~Client(); 294 sp<MemoryDealer> heap() const; 295 pid_t pid() const { return mPid; } 296 sp<AudioFlinger> audioFlinger() const { return mAudioFlinger; } 297 298 bool reserveTimedTrack(); 299 void releaseTimedTrack(); 300 301 private: 302 Client(const Client&); 303 Client& operator = (const Client&); 304 const sp<AudioFlinger> mAudioFlinger; 305 const sp<MemoryDealer> mMemoryDealer; 306 const pid_t mPid; 307 308 Mutex mTimedTrackLock; 309 int mTimedTrackCount; 310 }; 311 312 // --- Notification Client --- 313 class NotificationClient : public IBinder::DeathRecipient { 314 public: 315 NotificationClient(const sp<AudioFlinger>& audioFlinger, 316 const sp<IAudioFlingerClient>& client, 317 pid_t pid); 318 virtual ~NotificationClient(); 319 320 sp<IAudioFlingerClient> audioFlingerClient() const { return mAudioFlingerClient; } 321 322 // IBinder::DeathRecipient 323 virtual void binderDied(const wp<IBinder>& who); 324 325 private: 326 NotificationClient(const NotificationClient&); 327 NotificationClient& operator = (const NotificationClient&); 328 329 const sp<AudioFlinger> mAudioFlinger; 330 const pid_t mPid; 331 const sp<IAudioFlingerClient> mAudioFlingerClient; 332 }; 333 334 class TrackHandle; 335 class RecordHandle; 336 class RecordThread; 337 class PlaybackThread; 338 class MixerThread; 339 class DirectOutputThread; 340 class DuplicatingThread; 341 class Track; 342 class RecordTrack; 343 class EffectModule; 344 class EffectHandle; 345 class EffectChain; 346 struct AudioStreamOut; 347 struct AudioStreamIn; 348 349 class ThreadBase : public Thread { 350 public: 351 352 enum type_t { 353 MIXER, // Thread class is MixerThread 354 DIRECT, // Thread class is DirectOutputThread 355 DUPLICATING, // Thread class is DuplicatingThread 356 RECORD // Thread class is RecordThread 357 }; 358 359 ThreadBase (const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 360 audio_devices_t outDevice, audio_devices_t inDevice, type_t type); 361 virtual ~ThreadBase(); 362 363 void dumpBase(int fd, const Vector<String16>& args); 364 void dumpEffectChains(int fd, const Vector<String16>& args); 365 366 void clearPowerManager(); 367 368 // base for record and playback 369 class TrackBase : public ExtendedAudioBufferProvider, public RefBase { 370 371 public: 372 enum track_state { 373 IDLE, 374 TERMINATED, 375 FLUSHED, 376 STOPPED, 377 // next 2 states are currently used for fast tracks only 378 STOPPING_1, // waiting for first underrun 379 STOPPING_2, // waiting for presentation complete 380 RESUMING, 381 ACTIVE, 382 PAUSING, 383 PAUSED 384 }; 385 386 TrackBase(ThreadBase *thread, 387 const sp<Client>& client, 388 uint32_t sampleRate, 389 audio_format_t format, 390 audio_channel_mask_t channelMask, 391 int frameCount, 392 const sp<IMemory>& sharedBuffer, 393 int sessionId); 394 virtual ~TrackBase(); 395 396 virtual status_t start(AudioSystem::sync_event_t event, 397 int triggerSession) = 0; 398 virtual void stop() = 0; 399 sp<IMemory> getCblk() const { return mCblkMemory; } 400 audio_track_cblk_t* cblk() const { return mCblk; } 401 int sessionId() const { return mSessionId; } 402 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 403 404 protected: 405 TrackBase(const TrackBase&); 406 TrackBase& operator = (const TrackBase&); 407 408 // AudioBufferProvider interface 409 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) = 0; 410 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 411 412 // ExtendedAudioBufferProvider interface is only needed for Track, 413 // but putting it in TrackBase avoids the complexity of virtual inheritance 414 virtual size_t framesReady() const { return SIZE_MAX; } 415 416 audio_format_t format() const { 417 return mFormat; 418 } 419 420 int channelCount() const { return mChannelCount; } 421 422 audio_channel_mask_t channelMask() const { return mChannelMask; } 423 424 int sampleRate() const; // FIXME inline after cblk sr moved 425 426 // Return a pointer to the start of a contiguous slice of the track buffer. 427 // Parameter 'offset' is the requested start position, expressed in 428 // monotonically increasing frame units relative to the track epoch. 429 // Parameter 'frames' is the requested length, also in frame units. 430 // Always returns non-NULL. It is the caller's responsibility to 431 // verify that this will be successful; the result of calling this 432 // function with invalid 'offset' or 'frames' is undefined. 433 void* getBuffer(uint32_t offset, uint32_t frames) const; 434 435 bool isStopped() const { 436 return (mState == STOPPED || mState == FLUSHED); 437 } 438 439 // for fast tracks only 440 bool isStopping() const { 441 return mState == STOPPING_1 || mState == STOPPING_2; 442 } 443 bool isStopping_1() const { 444 return mState == STOPPING_1; 445 } 446 bool isStopping_2() const { 447 return mState == STOPPING_2; 448 } 449 450 bool isTerminated() const { 451 return mState == TERMINATED; 452 } 453 454 bool step(); 455 void reset(); 456 457 const wp<ThreadBase> mThread; 458 /*const*/ sp<Client> mClient; // see explanation at ~TrackBase() why not const 459 sp<IMemory> mCblkMemory; 460 audio_track_cblk_t* mCblk; 461 void* mBuffer; // start of track buffer, typically in shared memory 462 void* mBufferEnd; // &mBuffer[mFrameCount * frameSize], where frameSize 463 // is based on mChannelCount and 16-bit samples 464 uint32_t mFrameCount; 465 // we don't really need a lock for these 466 track_state mState; 467 const uint32_t mSampleRate; // initial sample rate only; for tracks which 468 // support dynamic rates, the current value is in control block 469 const audio_format_t mFormat; 470 bool mStepServerFailed; 471 const int mSessionId; 472 uint8_t mChannelCount; 473 audio_channel_mask_t mChannelMask; 474 Vector < sp<SyncEvent> >mSyncEvents; 475 }; 476 477 enum { 478 CFG_EVENT_IO, 479 CFG_EVENT_PRIO 480 }; 481 482 class ConfigEvent { 483 public: 484 ConfigEvent(int type) : mType(type) {} 485 virtual ~ConfigEvent() {} 486 487 int type() const { return mType; } 488 489 virtual void dump(char *buffer, size_t size) = 0; 490 491 private: 492 const int mType; 493 }; 494 495 class IoConfigEvent : public ConfigEvent { 496 public: 497 IoConfigEvent(int event, int param) : 498 ConfigEvent(CFG_EVENT_IO), mEvent(event), mParam(event) {} 499 virtual ~IoConfigEvent() {} 500 501 int event() const { return mEvent; } 502 int param() const { return mParam; } 503 504 virtual void dump(char *buffer, size_t size) { 505 snprintf(buffer, size, "IO event: event %d, param %d\n", mEvent, mParam); 506 } 507 508 private: 509 const int mEvent; 510 const int mParam; 511 }; 512 513 class PrioConfigEvent : public ConfigEvent { 514 public: 515 PrioConfigEvent(pid_t pid, pid_t tid, int32_t prio) : 516 ConfigEvent(CFG_EVENT_PRIO), mPid(pid), mTid(tid), mPrio(prio) {} 517 virtual ~PrioConfigEvent() {} 518 519 pid_t pid() const { return mPid; } 520 pid_t tid() const { return mTid; } 521 int32_t prio() const { return mPrio; } 522 523 virtual void dump(char *buffer, size_t size) { 524 snprintf(buffer, size, "Prio event: pid %d, tid %d, prio %d\n", mPid, mTid, mPrio); 525 } 526 527 private: 528 const pid_t mPid; 529 const pid_t mTid; 530 const int32_t mPrio; 531 }; 532 533 534 class PMDeathRecipient : public IBinder::DeathRecipient { 535 public: 536 PMDeathRecipient(const wp<ThreadBase>& thread) : mThread(thread) {} 537 virtual ~PMDeathRecipient() {} 538 539 // IBinder::DeathRecipient 540 virtual void binderDied(const wp<IBinder>& who); 541 542 private: 543 PMDeathRecipient(const PMDeathRecipient&); 544 PMDeathRecipient& operator = (const PMDeathRecipient&); 545 546 wp<ThreadBase> mThread; 547 }; 548 549 virtual status_t initCheck() const = 0; 550 551 // static externally-visible 552 type_t type() const { return mType; } 553 audio_io_handle_t id() const { return mId;} 554 555 // dynamic externally-visible 556 uint32_t sampleRate() const { return mSampleRate; } 557 int channelCount() const { return mChannelCount; } 558 audio_channel_mask_t channelMask() const { return mChannelMask; } 559 audio_format_t format() const { return mFormat; } 560 // Called by AudioFlinger::frameCount(audio_io_handle_t output) and effects, 561 // and returns the normal mix buffer's frame count. 562 size_t frameCount() const { return mNormalFrameCount; } 563 // Return's the HAL's frame count i.e. fast mixer buffer size. 564 size_t frameCountHAL() const { return mFrameCount; } 565 566 // Should be "virtual status_t requestExitAndWait()" and override same 567 // method in Thread, but Thread::requestExitAndWait() is not yet virtual. 568 void exit(); 569 virtual bool checkForNewParameters_l() = 0; 570 virtual status_t setParameters(const String8& keyValuePairs); 571 virtual String8 getParameters(const String8& keys) = 0; 572 virtual void audioConfigChanged_l(int event, int param = 0) = 0; 573 void sendIoConfigEvent(int event, int param = 0); 574 void sendIoConfigEvent_l(int event, int param = 0); 575 void sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio); 576 void processConfigEvents(); 577 578 // see note at declaration of mStandby, mOutDevice and mInDevice 579 bool standby() const { return mStandby; } 580 audio_devices_t outDevice() const { return mOutDevice; } 581 audio_devices_t inDevice() const { return mInDevice; } 582 583 virtual audio_stream_t* stream() const = 0; 584 585 sp<EffectHandle> createEffect_l( 586 const sp<AudioFlinger::Client>& client, 587 const sp<IEffectClient>& effectClient, 588 int32_t priority, 589 int sessionId, 590 effect_descriptor_t *desc, 591 int *enabled, 592 status_t *status); 593 void disconnectEffect(const sp< EffectModule>& effect, 594 EffectHandle *handle, 595 bool unpinIfLast); 596 597 // return values for hasAudioSession (bit field) 598 enum effect_state { 599 EFFECT_SESSION = 0x1, // the audio session corresponds to at least one 600 // effect 601 TRACK_SESSION = 0x2 // the audio session corresponds to at least one 602 // track 603 }; 604 605 // get effect chain corresponding to session Id. 606 sp<EffectChain> getEffectChain(int sessionId); 607 // same as getEffectChain() but must be called with ThreadBase mutex locked 608 sp<EffectChain> getEffectChain_l(int sessionId) const; 609 // add an effect chain to the chain list (mEffectChains) 610 virtual status_t addEffectChain_l(const sp<EffectChain>& chain) = 0; 611 // remove an effect chain from the chain list (mEffectChains) 612 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain) = 0; 613 // lock all effect chains Mutexes. Must be called before releasing the 614 // ThreadBase mutex before processing the mixer and effects. This guarantees the 615 // integrity of the chains during the process. 616 // Also sets the parameter 'effectChains' to current value of mEffectChains. 617 void lockEffectChains_l(Vector< sp<EffectChain> >& effectChains); 618 // unlock effect chains after process 619 void unlockEffectChains(const Vector< sp<EffectChain> >& effectChains); 620 // set audio mode to all effect chains 621 void setMode(audio_mode_t mode); 622 // get effect module with corresponding ID on specified audio session 623 sp<AudioFlinger::EffectModule> getEffect(int sessionId, int effectId); 624 sp<AudioFlinger::EffectModule> getEffect_l(int sessionId, int effectId); 625 // add and effect module. Also creates the effect chain is none exists for 626 // the effects audio session 627 status_t addEffect_l(const sp< EffectModule>& effect); 628 // remove and effect module. Also removes the effect chain is this was the last 629 // effect 630 void removeEffect_l(const sp< EffectModule>& effect); 631 // detach all tracks connected to an auxiliary effect 632 virtual void detachAuxEffect_l(int effectId) {} 633 // returns either EFFECT_SESSION if effects on this audio session exist in one 634 // chain, or TRACK_SESSION if tracks on this audio session exist, or both 635 virtual uint32_t hasAudioSession(int sessionId) const = 0; 636 // the value returned by default implementation is not important as the 637 // strategy is only meaningful for PlaybackThread which implements this method 638 virtual uint32_t getStrategyForSession_l(int sessionId) { return 0; } 639 640 // suspend or restore effect according to the type of effect passed. a NULL 641 // type pointer means suspend all effects in the session 642 void setEffectSuspended(const effect_uuid_t *type, 643 bool suspend, 644 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 645 // check if some effects must be suspended/restored when an effect is enabled 646 // or disabled 647 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 648 bool enabled, 649 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 650 void checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 651 bool enabled, 652 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 653 654 virtual status_t setSyncEvent(const sp<SyncEvent>& event) = 0; 655 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const = 0; 656 657 658 mutable Mutex mLock; 659 660 protected: 661 662 // entry describing an effect being suspended in mSuspendedSessions keyed vector 663 class SuspendedSessionDesc : public RefBase { 664 public: 665 SuspendedSessionDesc() : mRefCount(0) {} 666 667 int mRefCount; // number of active suspend requests 668 effect_uuid_t mType; // effect type UUID 669 }; 670 671 void acquireWakeLock(); 672 void acquireWakeLock_l(); 673 void releaseWakeLock(); 674 void releaseWakeLock_l(); 675 void setEffectSuspended_l(const effect_uuid_t *type, 676 bool suspend, 677 int sessionId); 678 // updated mSuspendedSessions when an effect suspended or restored 679 void updateSuspendedSessions_l(const effect_uuid_t *type, 680 bool suspend, 681 int sessionId); 682 // check if some effects must be suspended when an effect chain is added 683 void checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain); 684 685 friend class AudioFlinger; // for mEffectChains 686 687 const type_t mType; 688 689 // Used by parameters, config events, addTrack_l, exit 690 Condition mWaitWorkCV; 691 692 const sp<AudioFlinger> mAudioFlinger; 693 uint32_t mSampleRate; 694 size_t mFrameCount; // output HAL, direct output, record 695 size_t mNormalFrameCount; // normal mixer and effects 696 audio_channel_mask_t mChannelMask; 697 uint16_t mChannelCount; 698 size_t mFrameSize; 699 audio_format_t mFormat; 700 701 // Parameter sequence by client: binder thread calling setParameters(): 702 // 1. Lock mLock 703 // 2. Append to mNewParameters 704 // 3. mWaitWorkCV.signal 705 // 4. mParamCond.waitRelative with timeout 706 // 5. read mParamStatus 707 // 6. mWaitWorkCV.signal 708 // 7. Unlock 709 // 710 // Parameter sequence by server: threadLoop calling checkForNewParameters_l(): 711 // 1. Lock mLock 712 // 2. If there is an entry in mNewParameters proceed ... 713 // 2. Read first entry in mNewParameters 714 // 3. Process 715 // 4. Remove first entry from mNewParameters 716 // 5. Set mParamStatus 717 // 6. mParamCond.signal 718 // 7. mWaitWorkCV.wait with timeout (this is to avoid overwriting mParamStatus) 719 // 8. Unlock 720 Condition mParamCond; 721 Vector<String8> mNewParameters; 722 status_t mParamStatus; 723 724 Vector<ConfigEvent *> mConfigEvents; 725 726 // These fields are written and read by thread itself without lock or barrier, 727 // and read by other threads without lock or barrier via standby() , outDevice() 728 // and inDevice(). 729 // Because of the absence of a lock or barrier, any other thread that reads 730 // these fields must use the information in isolation, or be prepared to deal 731 // with possibility that it might be inconsistent with other information. 732 bool mStandby; // Whether thread is currently in standby. 733 audio_devices_t mOutDevice; // output device 734 audio_devices_t mInDevice; // input device 735 audio_source_t mAudioSource; // (see audio.h, audio_source_t) 736 737 const audio_io_handle_t mId; 738 Vector< sp<EffectChain> > mEffectChains; 739 740 static const int kNameLength = 16; // prctl(PR_SET_NAME) limit 741 char mName[kNameLength]; 742 sp<IPowerManager> mPowerManager; 743 sp<IBinder> mWakeLockToken; 744 const sp<PMDeathRecipient> mDeathRecipient; 745 // list of suspended effects per session and per type. The first vector is 746 // keyed by session ID, the second by type UUID timeLow field 747 KeyedVector< int, KeyedVector< int, sp<SuspendedSessionDesc> > > mSuspendedSessions; 748 }; 749 750 struct stream_type_t { 751 stream_type_t() 752 : volume(1.0f), 753 mute(false) 754 { 755 } 756 float volume; 757 bool mute; 758 }; 759 760 // --- PlaybackThread --- 761 class PlaybackThread : public ThreadBase { 762 public: 763 764 enum mixer_state { 765 MIXER_IDLE, // no active tracks 766 MIXER_TRACKS_ENABLED, // at least one active track, but no track has any data ready 767 MIXER_TRACKS_READY // at least one active track, and at least one track has data 768 // standby mode does not have an enum value 769 // suspend by audio policy manager is orthogonal to mixer state 770 }; 771 772 // playback track 773 class Track : public TrackBase, public VolumeProvider { 774 public: 775 Track( PlaybackThread *thread, 776 const sp<Client>& client, 777 audio_stream_type_t streamType, 778 uint32_t sampleRate, 779 audio_format_t format, 780 audio_channel_mask_t channelMask, 781 int frameCount, 782 const sp<IMemory>& sharedBuffer, 783 int sessionId, 784 IAudioFlinger::track_flags_t flags); 785 virtual ~Track(); 786 787 static void appendDumpHeader(String8& result); 788 void dump(char* buffer, size_t size); 789 virtual status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 790 int triggerSession = 0); 791 virtual void stop(); 792 void pause(); 793 794 void flush(); 795 void destroy(); 796 void mute(bool); 797 int name() const { return mName; } 798 799 audio_stream_type_t streamType() const { 800 return mStreamType; 801 } 802 status_t attachAuxEffect(int EffectId); 803 void setAuxBuffer(int EffectId, int32_t *buffer); 804 int32_t *auxBuffer() const { return mAuxBuffer; } 805 void setMainBuffer(int16_t *buffer) { mMainBuffer = buffer; } 806 int16_t *mainBuffer() const { return mMainBuffer; } 807 int auxEffectId() const { return mAuxEffectId; } 808 809 // implement FastMixerState::VolumeProvider interface 810 virtual uint32_t getVolumeLR(); 811 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 812 813 protected: 814 // for numerous 815 friend class PlaybackThread; 816 friend class MixerThread; 817 friend class DirectOutputThread; 818 819 Track(const Track&); 820 Track& operator = (const Track&); 821 822 // AudioBufferProvider interface 823 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts = kInvalidPTS); 824 // releaseBuffer() not overridden 825 826 virtual size_t framesReady() const; 827 828 bool isMuted() const { return mMute; } 829 bool isPausing() const { 830 return mState == PAUSING; 831 } 832 bool isPaused() const { 833 return mState == PAUSED; 834 } 835 bool isResuming() const { 836 return mState == RESUMING; 837 } 838 bool isReady() const; 839 void setPaused() { mState = PAUSED; } 840 void reset(); 841 842 bool isOutputTrack() const { 843 return (mStreamType == AUDIO_STREAM_CNT); 844 } 845 846 sp<IMemory> sharedBuffer() const { return mSharedBuffer; } 847 848 bool presentationComplete(size_t framesWritten, size_t audioHalFrames); 849 850 public: 851 void triggerEvents(AudioSystem::sync_event_t type); 852 virtual bool isTimedTrack() const { return false; } 853 bool isFastTrack() const { return (mFlags & IAudioFlinger::TRACK_FAST) != 0; } 854 855 protected: 856 857 // written by Track::mute() called by binder thread(s), without a mutex or barrier. 858 // read by Track::isMuted() called by playback thread, also without a mutex or barrier. 859 // The lack of mutex or barrier is safe because the mute status is only used by itself. 860 bool mMute; 861 862 // FILLED state is used for suppressing volume ramp at begin of playing 863 enum {FS_INVALID, FS_FILLING, FS_FILLED, FS_ACTIVE}; 864 mutable uint8_t mFillingUpStatus; 865 int8_t mRetryCount; 866 const sp<IMemory> mSharedBuffer; 867 bool mResetDone; 868 const audio_stream_type_t mStreamType; 869 int mName; // track name on the normal mixer, 870 // allocated statically at track creation time, 871 // and is even allocated (though unused) for fast tracks 872 // FIXME don't allocate track name for fast tracks 873 int16_t *mMainBuffer; 874 int32_t *mAuxBuffer; 875 int mAuxEffectId; 876 bool mHasVolumeController; 877 size_t mPresentationCompleteFrames; // number of frames written to the audio HAL 878 // when this track will be fully rendered 879 private: 880 IAudioFlinger::track_flags_t mFlags; 881 882 // The following fields are only for fast tracks, and should be in a subclass 883 int mFastIndex; // index within FastMixerState::mFastTracks[]; 884 // either mFastIndex == -1 if not isFastTrack() 885 // or 0 < mFastIndex < FastMixerState::kMaxFast because 886 // index 0 is reserved for normal mixer's submix; 887 // index is allocated statically at track creation time 888 // but the slot is only used if track is active 889 FastTrackUnderruns mObservedUnderruns; // Most recently observed value of 890 // mFastMixerDumpState.mTracks[mFastIndex].mUnderruns 891 uint32_t mUnderrunCount; // Counter of total number of underruns, never reset 892 volatile float mCachedVolume; // combined master volume and stream type volume; 893 // 'volatile' means accessed without lock or 894 // barrier, but is read/written atomically 895 }; // end of Track 896 897 class TimedTrack : public Track { 898 public: 899 static sp<TimedTrack> create(PlaybackThread *thread, 900 const sp<Client>& client, 901 audio_stream_type_t streamType, 902 uint32_t sampleRate, 903 audio_format_t format, 904 audio_channel_mask_t channelMask, 905 int frameCount, 906 const sp<IMemory>& sharedBuffer, 907 int sessionId); 908 virtual ~TimedTrack(); 909 910 class TimedBuffer { 911 public: 912 TimedBuffer(); 913 TimedBuffer(const sp<IMemory>& buffer, int64_t pts); 914 const sp<IMemory>& buffer() const { return mBuffer; } 915 int64_t pts() const { return mPTS; } 916 uint32_t position() const { return mPosition; } 917 void setPosition(uint32_t pos) { mPosition = pos; } 918 private: 919 sp<IMemory> mBuffer; 920 int64_t mPTS; 921 uint32_t mPosition; 922 }; 923 924 // Mixer facing methods. 925 virtual bool isTimedTrack() const { return true; } 926 virtual size_t framesReady() const; 927 928 // AudioBufferProvider interface 929 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, 930 int64_t pts); 931 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 932 933 // Client/App facing methods. 934 status_t allocateTimedBuffer(size_t size, 935 sp<IMemory>* buffer); 936 status_t queueTimedBuffer(const sp<IMemory>& buffer, 937 int64_t pts); 938 status_t setMediaTimeTransform(const LinearTransform& xform, 939 TimedAudioTrack::TargetTimeline target); 940 941 private: 942 TimedTrack(PlaybackThread *thread, 943 const sp<Client>& client, 944 audio_stream_type_t streamType, 945 uint32_t sampleRate, 946 audio_format_t format, 947 audio_channel_mask_t channelMask, 948 int frameCount, 949 const sp<IMemory>& sharedBuffer, 950 int sessionId); 951 952 void timedYieldSamples_l(AudioBufferProvider::Buffer* buffer); 953 void timedYieldSilence_l(uint32_t numFrames, 954 AudioBufferProvider::Buffer* buffer); 955 void trimTimedBufferQueue_l(); 956 void trimTimedBufferQueueHead_l(const char* logTag); 957 void updateFramesPendingAfterTrim_l(const TimedBuffer& buf, 958 const char* logTag); 959 960 uint64_t mLocalTimeFreq; 961 LinearTransform mLocalTimeToSampleTransform; 962 LinearTransform mMediaTimeToSampleTransform; 963 sp<MemoryDealer> mTimedMemoryDealer; 964 965 Vector<TimedBuffer> mTimedBufferQueue; 966 bool mQueueHeadInFlight; 967 bool mTrimQueueHeadOnRelease; 968 uint32_t mFramesPendingInQueue; 969 970 uint8_t* mTimedSilenceBuffer; 971 uint32_t mTimedSilenceBufferSize; 972 mutable Mutex mTimedBufferQueueLock; 973 bool mTimedAudioOutputOnTime; 974 CCHelper mCCHelper; 975 976 Mutex mMediaTimeTransformLock; 977 LinearTransform mMediaTimeTransform; 978 bool mMediaTimeTransformValid; 979 TimedAudioTrack::TargetTimeline mMediaTimeTransformTarget; 980 }; 981 982 983 // playback track 984 class OutputTrack : public Track { 985 public: 986 987 class Buffer: public AudioBufferProvider::Buffer { 988 public: 989 int16_t *mBuffer; 990 }; 991 992 OutputTrack(PlaybackThread *thread, 993 DuplicatingThread *sourceThread, 994 uint32_t sampleRate, 995 audio_format_t format, 996 audio_channel_mask_t channelMask, 997 int frameCount); 998 virtual ~OutputTrack(); 999 1000 virtual status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 1001 int triggerSession = 0); 1002 virtual void stop(); 1003 bool write(int16_t* data, uint32_t frames); 1004 bool bufferQueueEmpty() const { return mBufferQueue.size() == 0; } 1005 bool isActive() const { return mActive; } 1006 const wp<ThreadBase>& thread() const { return mThread; } 1007 1008 private: 1009 1010 enum { 1011 NO_MORE_BUFFERS = 0x80000001, // same in AudioTrack.h, ok to be different value 1012 }; 1013 1014 status_t obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs); 1015 void clearBufferQueue(); 1016 1017 // Maximum number of pending buffers allocated by OutputTrack::write() 1018 static const uint8_t kMaxOverFlowBuffers = 10; 1019 1020 Vector < Buffer* > mBufferQueue; 1021 AudioBufferProvider::Buffer mOutBuffer; 1022 bool mActive; 1023 DuplicatingThread* const mSourceThread; // for waitTimeMs() in write() 1024 }; // end of OutputTrack 1025 1026 PlaybackThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1027 audio_io_handle_t id, audio_devices_t device, type_t type); 1028 virtual ~PlaybackThread(); 1029 1030 void dump(int fd, const Vector<String16>& args); 1031 1032 // Thread virtuals 1033 virtual status_t readyToRun(); 1034 virtual bool threadLoop(); 1035 1036 // RefBase 1037 virtual void onFirstRef(); 1038 1039protected: 1040 // Code snippets that were lifted up out of threadLoop() 1041 virtual void threadLoop_mix() = 0; 1042 virtual void threadLoop_sleepTime() = 0; 1043 virtual void threadLoop_write(); 1044 virtual void threadLoop_standby(); 1045 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 1046 1047 // prepareTracks_l reads and writes mActiveTracks, and returns 1048 // the pending set of tracks to remove via Vector 'tracksToRemove'. The caller 1049 // is responsible for clearing or destroying this Vector later on, when it 1050 // is safe to do so. That will drop the final ref count and destroy the tracks. 1051 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove) = 0; 1052 1053public: 1054 1055 virtual status_t initCheck() const { return (mOutput == NULL) ? NO_INIT : NO_ERROR; } 1056 1057 // return estimated latency in milliseconds, as reported by HAL 1058 uint32_t latency() const; 1059 // same, but lock must already be held 1060 uint32_t latency_l() const; 1061 1062 void setMasterVolume(float value); 1063 void setMasterMute(bool muted); 1064 1065 void setStreamVolume(audio_stream_type_t stream, float value); 1066 void setStreamMute(audio_stream_type_t stream, bool muted); 1067 1068 float streamVolume(audio_stream_type_t stream) const; 1069 1070 sp<Track> createTrack_l( 1071 const sp<AudioFlinger::Client>& client, 1072 audio_stream_type_t streamType, 1073 uint32_t sampleRate, 1074 audio_format_t format, 1075 audio_channel_mask_t channelMask, 1076 int frameCount, 1077 const sp<IMemory>& sharedBuffer, 1078 int sessionId, 1079 IAudioFlinger::track_flags_t flags, 1080 pid_t tid, 1081 status_t *status); 1082 1083 AudioStreamOut* getOutput() const; 1084 AudioStreamOut* clearOutput(); 1085 virtual audio_stream_t* stream() const; 1086 1087 // a very large number of suspend() will eventually wraparound, but unlikely 1088 void suspend() { (void) android_atomic_inc(&mSuspended); } 1089 void restore() 1090 { 1091 // if restore() is done without suspend(), get back into 1092 // range so that the next suspend() will operate correctly 1093 if (android_atomic_dec(&mSuspended) <= 0) { 1094 android_atomic_release_store(0, &mSuspended); 1095 } 1096 } 1097 bool isSuspended() const 1098 { return android_atomic_acquire_load(&mSuspended) > 0; } 1099 1100 virtual String8 getParameters(const String8& keys); 1101 virtual void audioConfigChanged_l(int event, int param = 0); 1102 status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames); 1103 int16_t *mixBuffer() const { return mMixBuffer; }; 1104 1105 virtual void detachAuxEffect_l(int effectId); 1106 status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track> track, 1107 int EffectId); 1108 status_t attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track> track, 1109 int EffectId); 1110 1111 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1112 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1113 virtual uint32_t hasAudioSession(int sessionId) const; 1114 virtual uint32_t getStrategyForSession_l(int sessionId); 1115 1116 1117 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 1118 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 1119 void invalidateTracks(audio_stream_type_t streamType); 1120 1121 1122 protected: 1123 int16_t* mMixBuffer; 1124 1125 // suspend count, > 0 means suspended. While suspended, the thread continues to pull from 1126 // tracks and mix, but doesn't write to HAL. A2DP and SCO HAL implementations can't handle 1127 // concurrent use of both of them, so Audio Policy Service suspends one of the threads to 1128 // workaround that restriction. 1129 // 'volatile' means accessed via atomic operations and no lock. 1130 volatile int32_t mSuspended; 1131 1132 int mBytesWritten; 1133 private: 1134 // mMasterMute is in both PlaybackThread and in AudioFlinger. When a 1135 // PlaybackThread needs to find out if master-muted, it checks it's local 1136 // copy rather than the one in AudioFlinger. This optimization saves a lock. 1137 bool mMasterMute; 1138 void setMasterMute_l(bool muted) { mMasterMute = muted; } 1139 protected: 1140 SortedVector< wp<Track> > mActiveTracks; // FIXME check if this could be sp<> 1141 1142 // Allocate a track name for a given channel mask. 1143 // Returns name >= 0 if successful, -1 on failure. 1144 virtual int getTrackName_l(audio_channel_mask_t channelMask, int sessionId) = 0; 1145 virtual void deleteTrackName_l(int name) = 0; 1146 1147 // Time to sleep between cycles when: 1148 virtual uint32_t activeSleepTimeUs() const; // mixer state MIXER_TRACKS_ENABLED 1149 virtual uint32_t idleSleepTimeUs() const = 0; // mixer state MIXER_IDLE 1150 virtual uint32_t suspendSleepTimeUs() const = 0; // audio policy manager suspended us 1151 // No sleep when mixer state == MIXER_TRACKS_READY; relies on audio HAL stream->write() 1152 // No sleep in standby mode; waits on a condition 1153 1154 // Code snippets that are temporarily lifted up out of threadLoop() until the merge 1155 void checkSilentMode_l(); 1156 1157 // Non-trivial for DUPLICATING only 1158 virtual void saveOutputTracks() { } 1159 virtual void clearOutputTracks() { } 1160 1161 // Cache various calculated values, at threadLoop() entry and after a parameter change 1162 virtual void cacheParameters_l(); 1163 1164 virtual uint32_t correctLatency(uint32_t latency) const; 1165 1166 private: 1167 1168 friend class AudioFlinger; // for numerous 1169 1170 PlaybackThread(const Client&); 1171 PlaybackThread& operator = (const PlaybackThread&); 1172 1173 status_t addTrack_l(const sp<Track>& track); 1174 void destroyTrack_l(const sp<Track>& track); 1175 void removeTrack_l(const sp<Track>& track); 1176 1177 void readOutputParameters(); 1178 1179 virtual void dumpInternals(int fd, const Vector<String16>& args); 1180 void dumpTracks(int fd, const Vector<String16>& args); 1181 1182 SortedVector< sp<Track> > mTracks; 1183 // mStreamTypes[] uses 1 additional stream type internally for the OutputTrack used by DuplicatingThread 1184 stream_type_t mStreamTypes[AUDIO_STREAM_CNT + 1]; 1185 AudioStreamOut *mOutput; 1186 1187 float mMasterVolume; 1188 nsecs_t mLastWriteTime; 1189 int mNumWrites; 1190 int mNumDelayedWrites; 1191 bool mInWrite; 1192 1193 // FIXME rename these former local variables of threadLoop to standard "m" names 1194 nsecs_t standbyTime; 1195 size_t mixBufferSize; 1196 1197 // cached copies of activeSleepTimeUs() and idleSleepTimeUs() made by cacheParameters_l() 1198 uint32_t activeSleepTime; 1199 uint32_t idleSleepTime; 1200 1201 uint32_t sleepTime; 1202 1203 // mixer status returned by prepareTracks_l() 1204 mixer_state mMixerStatus; // current cycle 1205 // previous cycle when in prepareTracks_l() 1206 mixer_state mMixerStatusIgnoringFastTracks; 1207 // FIXME or a separate ready state per track 1208 1209 // FIXME move these declarations into the specific sub-class that needs them 1210 // MIXER only 1211 uint32_t sleepTimeShift; 1212 1213 // same as AudioFlinger::mStandbyTimeInNsecs except for DIRECT which uses a shorter value 1214 nsecs_t standbyDelay; 1215 1216 // MIXER only 1217 nsecs_t maxPeriod; 1218 1219 // DUPLICATING only 1220 uint32_t writeFrames; 1221 1222 private: 1223 // The HAL output sink is treated as non-blocking, but current implementation is blocking 1224 sp<NBAIO_Sink> mOutputSink; 1225 // If a fast mixer is present, the blocking pipe sink, otherwise clear 1226 sp<NBAIO_Sink> mPipeSink; 1227 // The current sink for the normal mixer to write it's (sub)mix, mOutputSink or mPipeSink 1228 sp<NBAIO_Sink> mNormalSink; 1229 // For dumpsys 1230 sp<NBAIO_Sink> mTeeSink; 1231 sp<NBAIO_Source> mTeeSource; 1232 uint32_t mScreenState; // cached copy of gScreenState 1233 public: 1234 virtual bool hasFastMixer() const = 0; 1235 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const 1236 { FastTrackUnderruns dummy; return dummy; } 1237 1238 protected: 1239 // accessed by both binder threads and within threadLoop(), lock on mutex needed 1240 unsigned mFastTrackAvailMask; // bit i set if fast track [i] is available 1241 1242 }; 1243 1244 class MixerThread : public PlaybackThread { 1245 public: 1246 MixerThread (const sp<AudioFlinger>& audioFlinger, 1247 AudioStreamOut* output, 1248 audio_io_handle_t id, 1249 audio_devices_t device, 1250 type_t type = MIXER); 1251 virtual ~MixerThread(); 1252 1253 // Thread virtuals 1254 1255 virtual bool checkForNewParameters_l(); 1256 virtual void dumpInternals(int fd, const Vector<String16>& args); 1257 1258 protected: 1259 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1260 virtual int getTrackName_l(audio_channel_mask_t channelMask, int sessionId); 1261 virtual void deleteTrackName_l(int name); 1262 virtual uint32_t idleSleepTimeUs() const; 1263 virtual uint32_t suspendSleepTimeUs() const; 1264 virtual void cacheParameters_l(); 1265 1266 // threadLoop snippets 1267 virtual void threadLoop_write(); 1268 virtual void threadLoop_standby(); 1269 virtual void threadLoop_mix(); 1270 virtual void threadLoop_sleepTime(); 1271 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 1272 virtual uint32_t correctLatency(uint32_t latency) const; 1273 1274 AudioMixer* mAudioMixer; // normal mixer 1275 private: 1276 // one-time initialization, no locks required 1277 FastMixer* mFastMixer; // non-NULL if there is also a fast mixer 1278 sp<AudioWatchdog> mAudioWatchdog; // non-0 if there is an audio watchdog thread 1279 1280 // contents are not guaranteed to be consistent, no locks required 1281 FastMixerDumpState mFastMixerDumpState; 1282#ifdef STATE_QUEUE_DUMP 1283 StateQueueObserverDump mStateQueueObserverDump; 1284 StateQueueMutatorDump mStateQueueMutatorDump; 1285#endif 1286 AudioWatchdogDump mAudioWatchdogDump; 1287 1288 // accessible only within the threadLoop(), no locks required 1289 // mFastMixer->sq() // for mutating and pushing state 1290 int32_t mFastMixerFutex; // for cold idle 1291 1292 public: 1293 virtual bool hasFastMixer() const { return mFastMixer != NULL; } 1294 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const { 1295 ALOG_ASSERT(fastIndex < FastMixerState::kMaxFastTracks); 1296 return mFastMixerDumpState.mTracks[fastIndex].mUnderruns; 1297 } 1298 }; 1299 1300 class DirectOutputThread : public PlaybackThread { 1301 public: 1302 1303 DirectOutputThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1304 audio_io_handle_t id, audio_devices_t device); 1305 virtual ~DirectOutputThread(); 1306 1307 // Thread virtuals 1308 1309 virtual bool checkForNewParameters_l(); 1310 1311 protected: 1312 virtual int getTrackName_l(audio_channel_mask_t channelMask, int sessionId); 1313 virtual void deleteTrackName_l(int name); 1314 virtual uint32_t activeSleepTimeUs() const; 1315 virtual uint32_t idleSleepTimeUs() const; 1316 virtual uint32_t suspendSleepTimeUs() const; 1317 virtual void cacheParameters_l(); 1318 1319 // threadLoop snippets 1320 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1321 virtual void threadLoop_mix(); 1322 virtual void threadLoop_sleepTime(); 1323 1324 // volumes last sent to audio HAL with stream->set_volume() 1325 float mLeftVolFloat; 1326 float mRightVolFloat; 1327 1328private: 1329 // prepareTracks_l() tells threadLoop_mix() the name of the single active track 1330 sp<Track> mActiveTrack; 1331 public: 1332 virtual bool hasFastMixer() const { return false; } 1333 }; 1334 1335 class DuplicatingThread : public MixerThread { 1336 public: 1337 DuplicatingThread (const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread, 1338 audio_io_handle_t id); 1339 virtual ~DuplicatingThread(); 1340 1341 // Thread virtuals 1342 void addOutputTrack(MixerThread* thread); 1343 void removeOutputTrack(MixerThread* thread); 1344 uint32_t waitTimeMs() const { return mWaitTimeMs; } 1345 protected: 1346 virtual uint32_t activeSleepTimeUs() const; 1347 1348 private: 1349 bool outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks); 1350 protected: 1351 // threadLoop snippets 1352 virtual void threadLoop_mix(); 1353 virtual void threadLoop_sleepTime(); 1354 virtual void threadLoop_write(); 1355 virtual void threadLoop_standby(); 1356 virtual void cacheParameters_l(); 1357 1358 private: 1359 // called from threadLoop, addOutputTrack, removeOutputTrack 1360 virtual void updateWaitTime_l(); 1361 protected: 1362 virtual void saveOutputTracks(); 1363 virtual void clearOutputTracks(); 1364 private: 1365 1366 uint32_t mWaitTimeMs; 1367 SortedVector < sp<OutputTrack> > outputTracks; 1368 SortedVector < sp<OutputTrack> > mOutputTracks; 1369 public: 1370 virtual bool hasFastMixer() const { return false; } 1371 }; 1372 1373 PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const; 1374 MixerThread *checkMixerThread_l(audio_io_handle_t output) const; 1375 RecordThread *checkRecordThread_l(audio_io_handle_t input) const; 1376 // no range check, AudioFlinger::mLock held 1377 bool streamMute_l(audio_stream_type_t stream) const 1378 { return mStreamTypes[stream].mute; } 1379 // no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held 1380 float streamVolume_l(audio_stream_type_t stream) const 1381 { return mStreamTypes[stream].volume; } 1382 void audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2); 1383 1384 // allocate an audio_io_handle_t, session ID, or effect ID 1385 uint32_t nextUniqueId(); 1386 1387 status_t moveEffectChain_l(int sessionId, 1388 PlaybackThread *srcThread, 1389 PlaybackThread *dstThread, 1390 bool reRegister); 1391 // return thread associated with primary hardware device, or NULL 1392 PlaybackThread *primaryPlaybackThread_l() const; 1393 audio_devices_t primaryOutputDevice_l() const; 1394 1395 sp<PlaybackThread> getEffectThread_l(int sessionId, int EffectId); 1396 1397 // server side of the client's IAudioTrack 1398 class TrackHandle : public android::BnAudioTrack { 1399 public: 1400 TrackHandle(const sp<PlaybackThread::Track>& track); 1401 virtual ~TrackHandle(); 1402 virtual sp<IMemory> getCblk() const; 1403 virtual status_t start(); 1404 virtual void stop(); 1405 virtual void flush(); 1406 virtual void mute(bool); 1407 virtual void pause(); 1408 virtual status_t attachAuxEffect(int effectId); 1409 virtual status_t allocateTimedBuffer(size_t size, 1410 sp<IMemory>* buffer); 1411 virtual status_t queueTimedBuffer(const sp<IMemory>& buffer, 1412 int64_t pts); 1413 virtual status_t setMediaTimeTransform(const LinearTransform& xform, 1414 int target); 1415 virtual status_t onTransact( 1416 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 1417 private: 1418 const sp<PlaybackThread::Track> mTrack; 1419 }; 1420 1421 void removeClient_l(pid_t pid); 1422 void removeNotificationClient(pid_t pid); 1423 1424 1425 // record thread 1426 class RecordThread : public ThreadBase, public AudioBufferProvider 1427 // derives from AudioBufferProvider interface for use by resampler 1428 { 1429 public: 1430 1431 // record track 1432 class RecordTrack : public TrackBase { 1433 public: 1434 RecordTrack(RecordThread *thread, 1435 const sp<Client>& client, 1436 uint32_t sampleRate, 1437 audio_format_t format, 1438 audio_channel_mask_t channelMask, 1439 int frameCount, 1440 int sessionId); 1441 virtual ~RecordTrack(); 1442 1443 virtual status_t start(AudioSystem::sync_event_t event, int triggerSession); 1444 virtual void stop(); 1445 1446 void destroy(); 1447 1448 // clear the buffer overflow flag 1449 void clearOverflow() { mOverflow = false; } 1450 // set the buffer overflow flag and return previous value 1451 bool setOverflow() { bool tmp = mOverflow; mOverflow = true; return tmp; } 1452 1453 static void appendDumpHeader(String8& result); 1454 void dump(char* buffer, size_t size); 1455 1456 private: 1457 friend class AudioFlinger; // for mState 1458 1459 RecordTrack(const RecordTrack&); 1460 RecordTrack& operator = (const RecordTrack&); 1461 1462 // AudioBufferProvider interface 1463 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts = kInvalidPTS); 1464 // releaseBuffer() not overridden 1465 1466 bool mOverflow; // overflow on most recent attempt to fill client buffer 1467 }; 1468 1469 RecordThread(const sp<AudioFlinger>& audioFlinger, 1470 AudioStreamIn *input, 1471 uint32_t sampleRate, 1472 audio_channel_mask_t channelMask, 1473 audio_io_handle_t id, 1474 audio_devices_t device); 1475 virtual ~RecordThread(); 1476 1477 // no addTrack_l ? 1478 void destroyTrack_l(const sp<RecordTrack>& track); 1479 void removeTrack_l(const sp<RecordTrack>& track); 1480 1481 void dumpInternals(int fd, const Vector<String16>& args); 1482 void dumpTracks(int fd, const Vector<String16>& args); 1483 1484 // Thread virtuals 1485 virtual bool threadLoop(); 1486 virtual status_t readyToRun(); 1487 1488 // RefBase 1489 virtual void onFirstRef(); 1490 1491 virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; } 1492 sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l( 1493 const sp<AudioFlinger::Client>& client, 1494 uint32_t sampleRate, 1495 audio_format_t format, 1496 audio_channel_mask_t channelMask, 1497 int frameCount, 1498 int sessionId, 1499 IAudioFlinger::track_flags_t flags, 1500 pid_t tid, 1501 status_t *status); 1502 1503 status_t start(RecordTrack* recordTrack, 1504 AudioSystem::sync_event_t event, 1505 int triggerSession); 1506 1507 // ask the thread to stop the specified track, and 1508 // return true if the caller should then do it's part of the stopping process 1509 bool stop_l(RecordTrack* recordTrack); 1510 1511 void dump(int fd, const Vector<String16>& args); 1512 AudioStreamIn* clearInput(); 1513 virtual audio_stream_t* stream() const; 1514 1515 // AudioBufferProvider interface 1516 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts); 1517 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 1518 1519 virtual bool checkForNewParameters_l(); 1520 virtual String8 getParameters(const String8& keys); 1521 virtual void audioConfigChanged_l(int event, int param = 0); 1522 void readInputParameters(); 1523 virtual unsigned int getInputFramesLost(); 1524 1525 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1526 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1527 virtual uint32_t hasAudioSession(int sessionId) const; 1528 1529 // Return the set of unique session IDs across all tracks. 1530 // The keys are the session IDs, and the associated values are meaningless. 1531 // FIXME replace by Set [and implement Bag/Multiset for other uses]. 1532 KeyedVector<int, bool> sessionIds() const; 1533 1534 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 1535 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 1536 1537 static void syncStartEventCallback(const wp<SyncEvent>& event); 1538 void handleSyncStartEvent(const sp<SyncEvent>& event); 1539 1540 private: 1541 void clearSyncStartEvent(); 1542 1543 // Enter standby if not already in standby, and set mStandby flag 1544 void standby(); 1545 1546 // Call the HAL standby method unconditionally, and don't change mStandby flag 1547 void inputStandBy(); 1548 1549 AudioStreamIn *mInput; 1550 SortedVector < sp<RecordTrack> > mTracks; 1551 // mActiveTrack has dual roles: it indicates the current active track, and 1552 // is used together with mStartStopCond to indicate start()/stop() progress 1553 sp<RecordTrack> mActiveTrack; 1554 Condition mStartStopCond; 1555 AudioResampler *mResampler; 1556 int32_t *mRsmpOutBuffer; 1557 int16_t *mRsmpInBuffer; 1558 size_t mRsmpInIndex; 1559 size_t mInputBytes; 1560 const int mReqChannelCount; 1561 const uint32_t mReqSampleRate; 1562 ssize_t mBytesRead; 1563 // sync event triggering actual audio capture. Frames read before this event will 1564 // be dropped and therefore not read by the application. 1565 sp<SyncEvent> mSyncStartEvent; 1566 // number of captured frames to drop after the start sync event has been received. 1567 // when < 0, maximum frames to drop before starting capture even if sync event is 1568 // not received 1569 ssize_t mFramestoDrop; 1570 }; 1571 1572 // server side of the client's IAudioRecord 1573 class RecordHandle : public android::BnAudioRecord { 1574 public: 1575 RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack); 1576 virtual ~RecordHandle(); 1577 virtual sp<IMemory> getCblk() const; 1578 virtual status_t start(int /*AudioSystem::sync_event_t*/ event, int triggerSession); 1579 virtual void stop(); 1580 virtual status_t onTransact( 1581 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 1582 private: 1583 const sp<RecordThread::RecordTrack> mRecordTrack; 1584 1585 // for use from destructor 1586 void stop_nonvirtual(); 1587 }; 1588 1589 //--- Audio Effect Management 1590 1591 // EffectModule and EffectChain classes both have their own mutex to protect 1592 // state changes or resource modifications. Always respect the following order 1593 // if multiple mutexes must be acquired to avoid cross deadlock: 1594 // AudioFlinger -> ThreadBase -> EffectChain -> EffectModule 1595 1596 // The EffectModule class is a wrapper object controlling the effect engine implementation 1597 // in the effect library. It prevents concurrent calls to process() and command() functions 1598 // from different client threads. It keeps a list of EffectHandle objects corresponding 1599 // to all client applications using this effect and notifies applications of effect state, 1600 // control or parameter changes. It manages the activation state machine to send appropriate 1601 // reset, enable, disable commands to effect engine and provide volume 1602 // ramping when effects are activated/deactivated. 1603 // When controlling an auxiliary effect, the EffectModule also provides an input buffer used by 1604 // the attached track(s) to accumulate their auxiliary channel. 1605 class EffectModule: public RefBase { 1606 public: 1607 EffectModule(ThreadBase *thread, 1608 const wp<AudioFlinger::EffectChain>& chain, 1609 effect_descriptor_t *desc, 1610 int id, 1611 int sessionId); 1612 virtual ~EffectModule(); 1613 1614 enum effect_state { 1615 IDLE, 1616 RESTART, 1617 STARTING, 1618 ACTIVE, 1619 STOPPING, 1620 STOPPED, 1621 DESTROYED 1622 }; 1623 1624 int id() const { return mId; } 1625 void process(); 1626 void updateState(); 1627 status_t command(uint32_t cmdCode, 1628 uint32_t cmdSize, 1629 void *pCmdData, 1630 uint32_t *replySize, 1631 void *pReplyData); 1632 1633 void reset_l(); 1634 status_t configure(); 1635 status_t init(); 1636 effect_state state() const { 1637 return mState; 1638 } 1639 uint32_t status() { 1640 return mStatus; 1641 } 1642 int sessionId() const { 1643 return mSessionId; 1644 } 1645 status_t setEnabled(bool enabled); 1646 status_t setEnabled_l(bool enabled); 1647 bool isEnabled() const; 1648 bool isProcessEnabled() const; 1649 1650 void setInBuffer(int16_t *buffer) { mConfig.inputCfg.buffer.s16 = buffer; } 1651 int16_t *inBuffer() { return mConfig.inputCfg.buffer.s16; } 1652 void setOutBuffer(int16_t *buffer) { mConfig.outputCfg.buffer.s16 = buffer; } 1653 int16_t *outBuffer() { return mConfig.outputCfg.buffer.s16; } 1654 void setChain(const wp<EffectChain>& chain) { mChain = chain; } 1655 void setThread(const wp<ThreadBase>& thread) { mThread = thread; } 1656 const wp<ThreadBase>& thread() { return mThread; } 1657 1658 status_t addHandle(EffectHandle *handle); 1659 size_t disconnect(EffectHandle *handle, bool unpinIfLast); 1660 size_t removeHandle(EffectHandle *handle); 1661 1662 const effect_descriptor_t& desc() const { return mDescriptor; } 1663 wp<EffectChain>& chain() { return mChain; } 1664 1665 status_t setDevice(audio_devices_t device); 1666 status_t setVolume(uint32_t *left, uint32_t *right, bool controller); 1667 status_t setMode(audio_mode_t mode); 1668 status_t setAudioSource(audio_source_t source); 1669 status_t start(); 1670 status_t stop(); 1671 void setSuspended(bool suspended); 1672 bool suspended() const; 1673 1674 EffectHandle* controlHandle_l(); 1675 1676 bool isPinned() const { return mPinned; } 1677 void unPin() { mPinned = false; } 1678 bool purgeHandles(); 1679 void lock() { mLock.lock(); } 1680 void unlock() { mLock.unlock(); } 1681 1682 void dump(int fd, const Vector<String16>& args); 1683 1684 protected: 1685 friend class AudioFlinger; // for mHandles 1686 bool mPinned; 1687 1688 // Maximum time allocated to effect engines to complete the turn off sequence 1689 static const uint32_t MAX_DISABLE_TIME_MS = 10000; 1690 1691 EffectModule(const EffectModule&); 1692 EffectModule& operator = (const EffectModule&); 1693 1694 status_t start_l(); 1695 status_t stop_l(); 1696 1697mutable Mutex mLock; // mutex for process, commands and handles list protection 1698 wp<ThreadBase> mThread; // parent thread 1699 wp<EffectChain> mChain; // parent effect chain 1700 const int mId; // this instance unique ID 1701 const int mSessionId; // audio session ID 1702 const effect_descriptor_t mDescriptor;// effect descriptor received from effect engine 1703 effect_config_t mConfig; // input and output audio configuration 1704 effect_handle_t mEffectInterface; // Effect module C API 1705 status_t mStatus; // initialization status 1706 effect_state mState; // current activation state 1707 Vector<EffectHandle *> mHandles; // list of client handles 1708 // First handle in mHandles has highest priority and controls the effect module 1709 uint32_t mMaxDisableWaitCnt; // maximum grace period before forcing an effect off after 1710 // sending disable command. 1711 uint32_t mDisableWaitCnt; // current process() calls count during disable period. 1712 bool mSuspended; // effect is suspended: temporarily disabled by framework 1713 }; 1714 1715 // The EffectHandle class implements the IEffect interface. It provides resources 1716 // to receive parameter updates, keeps track of effect control 1717 // ownership and state and has a pointer to the EffectModule object it is controlling. 1718 // There is one EffectHandle object for each application controlling (or using) 1719 // an effect module. 1720 // The EffectHandle is obtained by calling AudioFlinger::createEffect(). 1721 class EffectHandle: public android::BnEffect { 1722 public: 1723 1724 EffectHandle(const sp<EffectModule>& effect, 1725 const sp<AudioFlinger::Client>& client, 1726 const sp<IEffectClient>& effectClient, 1727 int32_t priority); 1728 virtual ~EffectHandle(); 1729 1730 // IEffect 1731 virtual status_t enable(); 1732 virtual status_t disable(); 1733 virtual status_t command(uint32_t cmdCode, 1734 uint32_t cmdSize, 1735 void *pCmdData, 1736 uint32_t *replySize, 1737 void *pReplyData); 1738 virtual void disconnect(); 1739 private: 1740 void disconnect(bool unpinIfLast); 1741 public: 1742 virtual sp<IMemory> getCblk() const { return mCblkMemory; } 1743 virtual status_t onTransact(uint32_t code, const Parcel& data, 1744 Parcel* reply, uint32_t flags); 1745 1746 1747 // Give or take control of effect module 1748 // - hasControl: true if control is given, false if removed 1749 // - signal: true client app should be signaled of change, false otherwise 1750 // - enabled: state of the effect when control is passed 1751 void setControl(bool hasControl, bool signal, bool enabled); 1752 void commandExecuted(uint32_t cmdCode, 1753 uint32_t cmdSize, 1754 void *pCmdData, 1755 uint32_t replySize, 1756 void *pReplyData); 1757 void setEnabled(bool enabled); 1758 bool enabled() const { return mEnabled; } 1759 1760 // Getters 1761 int id() const { return mEffect->id(); } 1762 int priority() const { return mPriority; } 1763 bool hasControl() const { return mHasControl; } 1764 sp<EffectModule> effect() const { return mEffect; } 1765 // destroyed_l() must be called with the associated EffectModule mLock held 1766 bool destroyed_l() const { return mDestroyed; } 1767 1768 void dump(char* buffer, size_t size); 1769 1770 protected: 1771 friend class AudioFlinger; // for mEffect, mHasControl, mEnabled 1772 EffectHandle(const EffectHandle&); 1773 EffectHandle& operator =(const EffectHandle&); 1774 1775 sp<EffectModule> mEffect; // pointer to controlled EffectModule 1776 sp<IEffectClient> mEffectClient; // callback interface for client notifications 1777 /*const*/ sp<Client> mClient; // client for shared memory allocation, see disconnect() 1778 sp<IMemory> mCblkMemory; // shared memory for control block 1779 effect_param_cblk_t* mCblk; // control block for deferred parameter setting via shared memory 1780 uint8_t* mBuffer; // pointer to parameter area in shared memory 1781 int mPriority; // client application priority to control the effect 1782 bool mHasControl; // true if this handle is controlling the effect 1783 bool mEnabled; // cached enable state: needed when the effect is 1784 // restored after being suspended 1785 bool mDestroyed; // Set to true by destructor. Access with EffectModule 1786 // mLock held 1787 }; 1788 1789 // the EffectChain class represents a group of effects associated to one audio session. 1790 // There can be any number of EffectChain objects per output mixer thread (PlaybackThread). 1791 // The EffecChain with session ID 0 contains global effects applied to the output mix. 1792 // Effects in this chain can be insert or auxiliary. Effects in other chains (attached to tracks) 1793 // are insert only. The EffectChain maintains an ordered list of effect module, the order corresponding 1794 // in the effect process order. When attached to a track (session ID != 0), it also provide it's own 1795 // input buffer used by the track as accumulation buffer. 1796 class EffectChain: public RefBase { 1797 public: 1798 EffectChain(const wp<ThreadBase>& wThread, int sessionId); 1799 EffectChain(ThreadBase *thread, int sessionId); 1800 virtual ~EffectChain(); 1801 1802 // special key used for an entry in mSuspendedEffects keyed vector 1803 // corresponding to a suspend all request. 1804 static const int kKeyForSuspendAll = 0; 1805 1806 // minimum duration during which we force calling effect process when last track on 1807 // a session is stopped or removed to allow effect tail to be rendered 1808 static const int kProcessTailDurationMs = 1000; 1809 1810 void process_l(); 1811 1812 void lock() { 1813 mLock.lock(); 1814 } 1815 void unlock() { 1816 mLock.unlock(); 1817 } 1818 1819 status_t addEffect_l(const sp<EffectModule>& handle); 1820 size_t removeEffect_l(const sp<EffectModule>& handle); 1821 1822 int sessionId() const { return mSessionId; } 1823 void setSessionId(int sessionId) { mSessionId = sessionId; } 1824 1825 sp<EffectModule> getEffectFromDesc_l(effect_descriptor_t *descriptor); 1826 sp<EffectModule> getEffectFromId_l(int id); 1827 sp<EffectModule> getEffectFromType_l(const effect_uuid_t *type); 1828 bool setVolume_l(uint32_t *left, uint32_t *right); 1829 void setDevice_l(audio_devices_t device); 1830 void setMode_l(audio_mode_t mode); 1831 void setAudioSource_l(audio_source_t source); 1832 1833 void setInBuffer(int16_t *buffer, bool ownsBuffer = false) { 1834 mInBuffer = buffer; 1835 mOwnInBuffer = ownsBuffer; 1836 } 1837 int16_t *inBuffer() const { 1838 return mInBuffer; 1839 } 1840 void setOutBuffer(int16_t *buffer) { 1841 mOutBuffer = buffer; 1842 } 1843 int16_t *outBuffer() const { 1844 return mOutBuffer; 1845 } 1846 1847 void incTrackCnt() { android_atomic_inc(&mTrackCnt); } 1848 void decTrackCnt() { android_atomic_dec(&mTrackCnt); } 1849 int32_t trackCnt() const { return android_atomic_acquire_load(&mTrackCnt); } 1850 1851 void incActiveTrackCnt() { android_atomic_inc(&mActiveTrackCnt); 1852 mTailBufferCount = mMaxTailBuffers; } 1853 void decActiveTrackCnt() { android_atomic_dec(&mActiveTrackCnt); } 1854 int32_t activeTrackCnt() const { return android_atomic_acquire_load(&mActiveTrackCnt); } 1855 1856 uint32_t strategy() const { return mStrategy; } 1857 void setStrategy(uint32_t strategy) 1858 { mStrategy = strategy; } 1859 1860 // suspend effect of the given type 1861 void setEffectSuspended_l(const effect_uuid_t *type, 1862 bool suspend); 1863 // suspend all eligible effects 1864 void setEffectSuspendedAll_l(bool suspend); 1865 // check if effects should be suspend or restored when a given effect is enable or disabled 1866 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1867 bool enabled); 1868 1869 void clearInputBuffer(); 1870 1871 void dump(int fd, const Vector<String16>& args); 1872 1873 protected: 1874 friend class AudioFlinger; // for mThread, mEffects 1875 EffectChain(const EffectChain&); 1876 EffectChain& operator =(const EffectChain&); 1877 1878 class SuspendedEffectDesc : public RefBase { 1879 public: 1880 SuspendedEffectDesc() : mRefCount(0) {} 1881 1882 int mRefCount; 1883 effect_uuid_t mType; 1884 wp<EffectModule> mEffect; 1885 }; 1886 1887 // get a list of effect modules to suspend when an effect of the type 1888 // passed is enabled. 1889 void getSuspendEligibleEffects(Vector< sp<EffectModule> > &effects); 1890 1891 // get an effect module if it is currently enable 1892 sp<EffectModule> getEffectIfEnabled(const effect_uuid_t *type); 1893 // true if the effect whose descriptor is passed can be suspended 1894 // OEMs can modify the rules implemented in this method to exclude specific effect 1895 // types or implementations from the suspend/restore mechanism. 1896 bool isEffectEligibleForSuspend(const effect_descriptor_t& desc); 1897 1898 void clearInputBuffer_l(sp<ThreadBase> thread); 1899 1900 wp<ThreadBase> mThread; // parent mixer thread 1901 Mutex mLock; // mutex protecting effect list 1902 Vector< sp<EffectModule> > mEffects; // list of effect modules 1903 int mSessionId; // audio session ID 1904 int16_t *mInBuffer; // chain input buffer 1905 int16_t *mOutBuffer; // chain output buffer 1906 1907 // 'volatile' here means these are accessed with atomic operations instead of mutex 1908 volatile int32_t mActiveTrackCnt; // number of active tracks connected 1909 volatile int32_t mTrackCnt; // number of tracks connected 1910 1911 int32_t mTailBufferCount; // current effect tail buffer count 1912 int32_t mMaxTailBuffers; // maximum effect tail buffers 1913 bool mOwnInBuffer; // true if the chain owns its input buffer 1914 int mVolumeCtrlIdx; // index of insert effect having control over volume 1915 uint32_t mLeftVolume; // previous volume on left channel 1916 uint32_t mRightVolume; // previous volume on right channel 1917 uint32_t mNewLeftVolume; // new volume on left channel 1918 uint32_t mNewRightVolume; // new volume on right channel 1919 uint32_t mStrategy; // strategy for this effect chain 1920 // mSuspendedEffects lists all effects currently suspended in the chain. 1921 // Use effect type UUID timelow field as key. There is no real risk of identical 1922 // timeLow fields among effect type UUIDs. 1923 // Updated by updateSuspendedSessions_l() only. 1924 KeyedVector< int, sp<SuspendedEffectDesc> > mSuspendedEffects; 1925 }; 1926 1927 class AudioHwDevice { 1928 public: 1929 enum Flags { 1930 AHWD_CAN_SET_MASTER_VOLUME = 0x1, 1931 AHWD_CAN_SET_MASTER_MUTE = 0x2, 1932 }; 1933 1934 AudioHwDevice(const char *moduleName, 1935 audio_hw_device_t *hwDevice, 1936 Flags flags) 1937 : mModuleName(strdup(moduleName)) 1938 , mHwDevice(hwDevice) 1939 , mFlags(flags) { } 1940 /*virtual*/ ~AudioHwDevice() { free((void *)mModuleName); } 1941 1942 bool canSetMasterVolume() const { 1943 return (0 != (mFlags & AHWD_CAN_SET_MASTER_VOLUME)); 1944 } 1945 1946 bool canSetMasterMute() const { 1947 return (0 != (mFlags & AHWD_CAN_SET_MASTER_MUTE)); 1948 } 1949 1950 const char *moduleName() const { return mModuleName; } 1951 audio_hw_device_t *hwDevice() const { return mHwDevice; } 1952 private: 1953 const char * const mModuleName; 1954 audio_hw_device_t * const mHwDevice; 1955 Flags mFlags; 1956 }; 1957 1958 // AudioStreamOut and AudioStreamIn are immutable, so their fields are const. 1959 // For emphasis, we could also make all pointers to them be "const *", 1960 // but that would clutter the code unnecessarily. 1961 1962 struct AudioStreamOut { 1963 AudioHwDevice* const audioHwDev; 1964 audio_stream_out_t* const stream; 1965 1966 audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); } 1967 1968 AudioStreamOut(AudioHwDevice *dev, audio_stream_out_t *out) : 1969 audioHwDev(dev), stream(out) {} 1970 }; 1971 1972 struct AudioStreamIn { 1973 AudioHwDevice* const audioHwDev; 1974 audio_stream_in_t* const stream; 1975 1976 audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); } 1977 1978 AudioStreamIn(AudioHwDevice *dev, audio_stream_in_t *in) : 1979 audioHwDev(dev), stream(in) {} 1980 }; 1981 1982 // for mAudioSessionRefs only 1983 struct AudioSessionRef { 1984 AudioSessionRef(int sessionid, pid_t pid) : 1985 mSessionid(sessionid), mPid(pid), mCnt(1) {} 1986 const int mSessionid; 1987 const pid_t mPid; 1988 int mCnt; 1989 }; 1990 1991 mutable Mutex mLock; 1992 1993 DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client() 1994 1995 mutable Mutex mHardwareLock; 1996 // NOTE: If both mLock and mHardwareLock mutexes must be held, 1997 // always take mLock before mHardwareLock 1998 1999 // These two fields are immutable after onFirstRef(), so no lock needed to access 2000 AudioHwDevice* mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL 2001 DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*> mAudioHwDevs; 2002 2003 // for dump, indicates which hardware operation is currently in progress (but not stream ops) 2004 enum hardware_call_state { 2005 AUDIO_HW_IDLE = 0, // no operation in progress 2006 AUDIO_HW_INIT, // init_check 2007 AUDIO_HW_OUTPUT_OPEN, // open_output_stream 2008 AUDIO_HW_OUTPUT_CLOSE, // unused 2009 AUDIO_HW_INPUT_OPEN, // unused 2010 AUDIO_HW_INPUT_CLOSE, // unused 2011 AUDIO_HW_STANDBY, // unused 2012 AUDIO_HW_SET_MASTER_VOLUME, // set_master_volume 2013 AUDIO_HW_GET_ROUTING, // unused 2014 AUDIO_HW_SET_ROUTING, // unused 2015 AUDIO_HW_GET_MODE, // unused 2016 AUDIO_HW_SET_MODE, // set_mode 2017 AUDIO_HW_GET_MIC_MUTE, // get_mic_mute 2018 AUDIO_HW_SET_MIC_MUTE, // set_mic_mute 2019 AUDIO_HW_SET_VOICE_VOLUME, // set_voice_volume 2020 AUDIO_HW_SET_PARAMETER, // set_parameters 2021 AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size 2022 AUDIO_HW_GET_MASTER_VOLUME, // get_master_volume 2023 AUDIO_HW_GET_PARAMETER, // get_parameters 2024 AUDIO_HW_SET_MASTER_MUTE, // set_master_mute 2025 AUDIO_HW_GET_MASTER_MUTE, // get_master_mute 2026 }; 2027 2028 mutable hardware_call_state mHardwareStatus; // for dump only 2029 2030 2031 DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads; 2032 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 2033 2034 // member variables below are protected by mLock 2035 float mMasterVolume; 2036 bool mMasterMute; 2037 // end of variables protected by mLock 2038 2039 DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads; 2040 2041 DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients; 2042 volatile int32_t mNextUniqueId; // updated by android_atomic_inc 2043 audio_mode_t mMode; 2044 bool mBtNrecIsOff; 2045 2046 // protected by mLock 2047 Vector<AudioSessionRef*> mAudioSessionRefs; 2048 2049 float masterVolume_l() const; 2050 bool masterMute_l() const; 2051 audio_module_handle_t loadHwModule_l(const char *name); 2052 2053 Vector < sp<SyncEvent> > mPendingSyncEvents; // sync events awaiting for a session 2054 // to be created 2055 2056private: 2057 sp<Client> registerPid_l(pid_t pid); // always returns non-0 2058 2059 // for use from destructor 2060 status_t closeOutput_nonvirtual(audio_io_handle_t output); 2061 status_t closeInput_nonvirtual(audio_io_handle_t input); 2062}; 2063 2064 2065// ---------------------------------------------------------------------------- 2066 2067}; // namespace android 2068 2069#endif // ANDROID_AUDIO_FLINGER_H 2070