AudioFlinger.h revision b929e417853694e37aba1ef4399f188987b709d9
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 ? 273 NO_INIT : NO_ERROR; } 274 275 // RefBase 276 virtual void onFirstRef(); 277 278 AudioHwDevice* findSuitableHwDev_l(audio_module_handle_t module, 279 audio_devices_t devices); 280 void purgeStaleEffects_l(); 281 282 // standby delay for MIXER and DUPLICATING playback threads is read from property 283 // ro.audio.flinger_standbytime_ms or defaults to kDefaultStandbyTimeInNsecs 284 static nsecs_t mStandbyTimeInNsecs; 285 286 // Internal dump utilities. 287 void dumpPermissionDenial(int fd, const Vector<String16>& args); 288 void dumpClients(int fd, const Vector<String16>& args); 289 void dumpInternals(int fd, const Vector<String16>& args); 290 291 // --- Client --- 292 class Client : public RefBase { 293 public: 294 Client(const sp<AudioFlinger>& audioFlinger, pid_t pid); 295 virtual ~Client(); 296 sp<MemoryDealer> heap() const; 297 pid_t pid() const { return mPid; } 298 sp<AudioFlinger> audioFlinger() const { return mAudioFlinger; } 299 300 bool reserveTimedTrack(); 301 void releaseTimedTrack(); 302 303 private: 304 Client(const Client&); 305 Client& operator = (const Client&); 306 const sp<AudioFlinger> mAudioFlinger; 307 const sp<MemoryDealer> mMemoryDealer; 308 const pid_t mPid; 309 310 Mutex mTimedTrackLock; 311 int mTimedTrackCount; 312 }; 313 314 // --- Notification Client --- 315 class NotificationClient : public IBinder::DeathRecipient { 316 public: 317 NotificationClient(const sp<AudioFlinger>& audioFlinger, 318 const sp<IAudioFlingerClient>& client, 319 pid_t pid); 320 virtual ~NotificationClient(); 321 322 sp<IAudioFlingerClient> audioFlingerClient() const { return mAudioFlingerClient; } 323 324 // IBinder::DeathRecipient 325 virtual void binderDied(const wp<IBinder>& who); 326 327 private: 328 NotificationClient(const NotificationClient&); 329 NotificationClient& operator = (const NotificationClient&); 330 331 const sp<AudioFlinger> mAudioFlinger; 332 const pid_t mPid; 333 const sp<IAudioFlingerClient> mAudioFlingerClient; 334 }; 335 336 class TrackHandle; 337 class RecordHandle; 338 class RecordThread; 339 class PlaybackThread; 340 class MixerThread; 341 class DirectOutputThread; 342 class DuplicatingThread; 343 class Track; 344 class RecordTrack; 345 class EffectModule; 346 class EffectHandle; 347 class EffectChain; 348 struct AudioStreamOut; 349 struct AudioStreamIn; 350 351 class ThreadBase : public Thread { 352 public: 353 354 enum type_t { 355 MIXER, // Thread class is MixerThread 356 DIRECT, // Thread class is DirectOutputThread 357 DUPLICATING, // Thread class is DuplicatingThread 358 RECORD // Thread class is RecordThread 359 }; 360 361 ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 362 audio_devices_t outDevice, audio_devices_t inDevice, type_t type); 363 virtual ~ThreadBase(); 364 365 void dumpBase(int fd, const Vector<String16>& args); 366 void dumpEffectChains(int fd, const Vector<String16>& args); 367 368 void clearPowerManager(); 369 370 // base for record and playback 371 class TrackBase : public ExtendedAudioBufferProvider, public RefBase { 372 373 public: 374 enum track_state { 375 IDLE, 376 TERMINATED, 377 FLUSHED, 378 STOPPED, 379 // next 2 states are currently used for fast tracks only 380 STOPPING_1, // waiting for first underrun 381 STOPPING_2, // waiting for presentation complete 382 RESUMING, 383 ACTIVE, 384 PAUSING, 385 PAUSED 386 }; 387 388 TrackBase(ThreadBase *thread, 389 const sp<Client>& client, 390 uint32_t sampleRate, 391 audio_format_t format, 392 audio_channel_mask_t channelMask, 393 int frameCount, 394 const sp<IMemory>& sharedBuffer, 395 int sessionId); 396 virtual ~TrackBase(); 397 398 virtual status_t start(AudioSystem::sync_event_t event, 399 int triggerSession) = 0; 400 virtual void stop() = 0; 401 sp<IMemory> getCblk() const { return mCblkMemory; } 402 audio_track_cblk_t* cblk() const { return mCblk; } 403 int sessionId() const { return mSessionId; } 404 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 405 406 protected: 407 TrackBase(const TrackBase&); 408 TrackBase& operator = (const TrackBase&); 409 410 // AudioBufferProvider interface 411 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) = 0; 412 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 413 414 // ExtendedAudioBufferProvider interface is only needed for Track, 415 // but putting it in TrackBase avoids the complexity of virtual inheritance 416 virtual size_t framesReady() const { return SIZE_MAX; } 417 418 audio_format_t format() const { 419 return mFormat; 420 } 421 422 int channelCount() const { return mChannelCount; } 423 424 audio_channel_mask_t channelMask() const { return mChannelMask; } 425 426 int sampleRate() const; // FIXME inline after cblk sr moved 427 428 // Return a pointer to the start of a contiguous slice of the track buffer. 429 // Parameter 'offset' is the requested start position, expressed in 430 // monotonically increasing frame units relative to the track epoch. 431 // Parameter 'frames' is the requested length, also in frame units. 432 // Always returns non-NULL. It is the caller's responsibility to 433 // verify that this will be successful; the result of calling this 434 // function with invalid 'offset' or 'frames' is undefined. 435 void* getBuffer(uint32_t offset, uint32_t frames) const; 436 437 bool isStopped() const { 438 return (mState == STOPPED || mState == FLUSHED); 439 } 440 441 // for fast tracks only 442 bool isStopping() const { 443 return mState == STOPPING_1 || mState == STOPPING_2; 444 } 445 bool isStopping_1() const { 446 return mState == STOPPING_1; 447 } 448 bool isStopping_2() const { 449 return mState == STOPPING_2; 450 } 451 452 bool isTerminated() const { 453 return mState == TERMINATED; 454 } 455 456 bool step(); 457 void reset(); 458 459 virtual bool isOut() const = 0; // true for Track and TimedTrack, false for RecordTrack, 460 // this could be a track type if needed later 461 462 const wp<ThreadBase> mThread; 463 /*const*/ sp<Client> mClient; // see explanation at ~TrackBase() why not const 464 sp<IMemory> mCblkMemory; 465 audio_track_cblk_t* mCblk; 466 void* mBuffer; // start of track buffer, typically in shared memory 467 void* mBufferEnd; // &mBuffer[mFrameCount * frameSize], where frameSize 468 // is based on mChannelCount and 16-bit samples 469 uint32_t mFrameCount; 470 // we don't really need a lock for these 471 track_state mState; 472 const uint32_t mSampleRate; // initial sample rate only; for tracks which 473 // support dynamic rates, the current value is in control block 474 const audio_format_t mFormat; 475 bool mStepServerFailed; 476 const int mSessionId; 477 uint8_t mChannelCount; 478 audio_channel_mask_t mChannelMask; 479 Vector < sp<SyncEvent> >mSyncEvents; 480 }; 481 482 enum { 483 CFG_EVENT_IO, 484 CFG_EVENT_PRIO 485 }; 486 487 class ConfigEvent { 488 public: 489 ConfigEvent(int type) : mType(type) {} 490 virtual ~ConfigEvent() {} 491 492 int type() const { return mType; } 493 494 virtual void dump(char *buffer, size_t size) = 0; 495 496 private: 497 const int mType; 498 }; 499 500 class IoConfigEvent : public ConfigEvent { 501 public: 502 IoConfigEvent(int event, int param) : 503 ConfigEvent(CFG_EVENT_IO), mEvent(event), mParam(event) {} 504 virtual ~IoConfigEvent() {} 505 506 int event() const { return mEvent; } 507 int param() const { return mParam; } 508 509 virtual void dump(char *buffer, size_t size) { 510 snprintf(buffer, size, "IO event: event %d, param %d\n", mEvent, mParam); 511 } 512 513 private: 514 const int mEvent; 515 const int mParam; 516 }; 517 518 class PrioConfigEvent : public ConfigEvent { 519 public: 520 PrioConfigEvent(pid_t pid, pid_t tid, int32_t prio) : 521 ConfigEvent(CFG_EVENT_PRIO), mPid(pid), mTid(tid), mPrio(prio) {} 522 virtual ~PrioConfigEvent() {} 523 524 pid_t pid() const { return mPid; } 525 pid_t tid() const { return mTid; } 526 int32_t prio() const { return mPrio; } 527 528 virtual void dump(char *buffer, size_t size) { 529 snprintf(buffer, size, "Prio event: pid %d, tid %d, prio %d\n", mPid, mTid, mPrio); 530 } 531 532 private: 533 const pid_t mPid; 534 const pid_t mTid; 535 const int32_t mPrio; 536 }; 537 538 539 class PMDeathRecipient : public IBinder::DeathRecipient { 540 public: 541 PMDeathRecipient(const wp<ThreadBase>& thread) : mThread(thread) {} 542 virtual ~PMDeathRecipient() {} 543 544 // IBinder::DeathRecipient 545 virtual void binderDied(const wp<IBinder>& who); 546 547 private: 548 PMDeathRecipient(const PMDeathRecipient&); 549 PMDeathRecipient& operator = (const PMDeathRecipient&); 550 551 wp<ThreadBase> mThread; 552 }; 553 554 virtual status_t initCheck() const = 0; 555 556 // static externally-visible 557 type_t type() const { return mType; } 558 audio_io_handle_t id() const { return mId;} 559 560 // dynamic externally-visible 561 uint32_t sampleRate() const { return mSampleRate; } 562 int channelCount() const { return mChannelCount; } 563 audio_channel_mask_t channelMask() const { return mChannelMask; } 564 audio_format_t format() const { return mFormat; } 565 // Called by AudioFlinger::frameCount(audio_io_handle_t output) and effects, 566 // and returns the normal mix buffer's frame count. 567 size_t frameCount() const { return mNormalFrameCount; } 568 // Return's the HAL's frame count i.e. fast mixer buffer size. 569 size_t frameCountHAL() const { return mFrameCount; } 570 571 // Should be "virtual status_t requestExitAndWait()" and override same 572 // method in Thread, but Thread::requestExitAndWait() is not yet virtual. 573 void exit(); 574 virtual bool checkForNewParameters_l() = 0; 575 virtual status_t setParameters(const String8& keyValuePairs); 576 virtual String8 getParameters(const String8& keys) = 0; 577 virtual void audioConfigChanged_l(int event, int param = 0) = 0; 578 void sendIoConfigEvent(int event, int param = 0); 579 void sendIoConfigEvent_l(int event, int param = 0); 580 void sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio); 581 void processConfigEvents(); 582 583 // see note at declaration of mStandby, mOutDevice and mInDevice 584 bool standby() const { return mStandby; } 585 audio_devices_t outDevice() const { return mOutDevice; } 586 audio_devices_t inDevice() const { return mInDevice; } 587 588 virtual audio_stream_t* stream() const = 0; 589 590 sp<EffectHandle> createEffect_l( 591 const sp<AudioFlinger::Client>& client, 592 const sp<IEffectClient>& effectClient, 593 int32_t priority, 594 int sessionId, 595 effect_descriptor_t *desc, 596 int *enabled, 597 status_t *status); 598 void disconnectEffect(const sp< EffectModule>& effect, 599 EffectHandle *handle, 600 bool unpinIfLast); 601 602 // return values for hasAudioSession (bit field) 603 enum effect_state { 604 EFFECT_SESSION = 0x1, // the audio session corresponds to at least one 605 // effect 606 TRACK_SESSION = 0x2 // the audio session corresponds to at least one 607 // track 608 }; 609 610 // get effect chain corresponding to session Id. 611 sp<EffectChain> getEffectChain(int sessionId); 612 // same as getEffectChain() but must be called with ThreadBase mutex locked 613 sp<EffectChain> getEffectChain_l(int sessionId) const; 614 // add an effect chain to the chain list (mEffectChains) 615 virtual status_t addEffectChain_l(const sp<EffectChain>& chain) = 0; 616 // remove an effect chain from the chain list (mEffectChains) 617 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain) = 0; 618 // lock all effect chains Mutexes. Must be called before releasing the 619 // ThreadBase mutex before processing the mixer and effects. This guarantees the 620 // integrity of the chains during the process. 621 // Also sets the parameter 'effectChains' to current value of mEffectChains. 622 void lockEffectChains_l(Vector< sp<EffectChain> >& effectChains); 623 // unlock effect chains after process 624 void unlockEffectChains(const Vector< sp<EffectChain> >& effectChains); 625 // set audio mode to all effect chains 626 void setMode(audio_mode_t mode); 627 // get effect module with corresponding ID on specified audio session 628 sp<AudioFlinger::EffectModule> getEffect(int sessionId, int effectId); 629 sp<AudioFlinger::EffectModule> getEffect_l(int sessionId, int effectId); 630 // add and effect module. Also creates the effect chain is none exists for 631 // the effects audio session 632 status_t addEffect_l(const sp< EffectModule>& effect); 633 // remove and effect module. Also removes the effect chain is this was the last 634 // effect 635 void removeEffect_l(const sp< EffectModule>& effect); 636 // detach all tracks connected to an auxiliary effect 637 virtual void detachAuxEffect_l(int effectId) {} 638 // returns either EFFECT_SESSION if effects on this audio session exist in one 639 // chain, or TRACK_SESSION if tracks on this audio session exist, or both 640 virtual uint32_t hasAudioSession(int sessionId) const = 0; 641 // the value returned by default implementation is not important as the 642 // strategy is only meaningful for PlaybackThread which implements this method 643 virtual uint32_t getStrategyForSession_l(int sessionId) { return 0; } 644 645 // suspend or restore effect according to the type of effect passed. a NULL 646 // type pointer means suspend all effects in the session 647 void setEffectSuspended(const effect_uuid_t *type, 648 bool suspend, 649 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 650 // check if some effects must be suspended/restored when an effect is enabled 651 // or disabled 652 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 653 bool enabled, 654 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 655 void checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 656 bool enabled, 657 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 658 659 virtual status_t setSyncEvent(const sp<SyncEvent>& event) = 0; 660 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const = 0; 661 662 663 mutable Mutex mLock; 664 665 protected: 666 667 // entry describing an effect being suspended in mSuspendedSessions keyed vector 668 class SuspendedSessionDesc : public RefBase { 669 public: 670 SuspendedSessionDesc() : mRefCount(0) {} 671 672 int mRefCount; // number of active suspend requests 673 effect_uuid_t mType; // effect type UUID 674 }; 675 676 void acquireWakeLock(); 677 void acquireWakeLock_l(); 678 void releaseWakeLock(); 679 void releaseWakeLock_l(); 680 void setEffectSuspended_l(const effect_uuid_t *type, 681 bool suspend, 682 int sessionId); 683 // updated mSuspendedSessions when an effect suspended or restored 684 void updateSuspendedSessions_l(const effect_uuid_t *type, 685 bool suspend, 686 int sessionId); 687 // check if some effects must be suspended when an effect chain is added 688 void checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain); 689 690 virtual void preExit() { } 691 692 friend class AudioFlinger; // for mEffectChains 693 694 const type_t mType; 695 696 // Used by parameters, config events, addTrack_l, exit 697 Condition mWaitWorkCV; 698 699 const sp<AudioFlinger> mAudioFlinger; 700 uint32_t mSampleRate; 701 size_t mFrameCount; // output HAL, direct output, record 702 size_t mNormalFrameCount; // normal mixer and effects 703 audio_channel_mask_t mChannelMask; 704 uint16_t mChannelCount; 705 size_t mFrameSize; 706 audio_format_t mFormat; 707 708 // Parameter sequence by client: binder thread calling setParameters(): 709 // 1. Lock mLock 710 // 2. Append to mNewParameters 711 // 3. mWaitWorkCV.signal 712 // 4. mParamCond.waitRelative with timeout 713 // 5. read mParamStatus 714 // 6. mWaitWorkCV.signal 715 // 7. Unlock 716 // 717 // Parameter sequence by server: threadLoop calling checkForNewParameters_l(): 718 // 1. Lock mLock 719 // 2. If there is an entry in mNewParameters proceed ... 720 // 2. Read first entry in mNewParameters 721 // 3. Process 722 // 4. Remove first entry from mNewParameters 723 // 5. Set mParamStatus 724 // 6. mParamCond.signal 725 // 7. mWaitWorkCV.wait with timeout (this is to avoid overwriting mParamStatus) 726 // 8. Unlock 727 Condition mParamCond; 728 Vector<String8> mNewParameters; 729 status_t mParamStatus; 730 731 Vector<ConfigEvent *> mConfigEvents; 732 733 // These fields are written and read by thread itself without lock or barrier, 734 // and read by other threads without lock or barrier via standby() , outDevice() 735 // and inDevice(). 736 // Because of the absence of a lock or barrier, any other thread that reads 737 // these fields must use the information in isolation, or be prepared to deal 738 // with possibility that it might be inconsistent with other information. 739 bool mStandby; // Whether thread is currently in standby. 740 audio_devices_t mOutDevice; // output device 741 audio_devices_t mInDevice; // input device 742 audio_source_t mAudioSource; // (see audio.h, audio_source_t) 743 744 const audio_io_handle_t mId; 745 Vector< sp<EffectChain> > mEffectChains; 746 747 static const int kNameLength = 16; // prctl(PR_SET_NAME) limit 748 char mName[kNameLength]; 749 sp<IPowerManager> mPowerManager; 750 sp<IBinder> mWakeLockToken; 751 const sp<PMDeathRecipient> mDeathRecipient; 752 // list of suspended effects per session and per type. The first vector is 753 // keyed by session ID, the second by type UUID timeLow field 754 KeyedVector< int, KeyedVector< int, sp<SuspendedSessionDesc> > > 755 mSuspendedSessions; 756 }; 757 758 struct stream_type_t { 759 stream_type_t() 760 : volume(1.0f), 761 mute(false) 762 { 763 } 764 float volume; 765 bool mute; 766 }; 767 768 // --- PlaybackThread --- 769 class PlaybackThread : public ThreadBase { 770 public: 771 772 enum mixer_state { 773 MIXER_IDLE, // no active tracks 774 MIXER_TRACKS_ENABLED, // at least one active track, but no track has any data ready 775 MIXER_TRACKS_READY // at least one active track, and at least one track has data 776 // standby mode does not have an enum value 777 // suspend by audio policy manager is orthogonal to mixer state 778 }; 779 780 // playback track 781 class Track : public TrackBase, public VolumeProvider { 782 public: 783 Track( PlaybackThread *thread, 784 const sp<Client>& client, 785 audio_stream_type_t streamType, 786 uint32_t sampleRate, 787 audio_format_t format, 788 audio_channel_mask_t channelMask, 789 int frameCount, 790 const sp<IMemory>& sharedBuffer, 791 int sessionId, 792 IAudioFlinger::track_flags_t flags); 793 virtual ~Track(); 794 795 static void appendDumpHeader(String8& result); 796 void dump(char* buffer, size_t size); 797 virtual status_t start(AudioSystem::sync_event_t event = 798 AudioSystem::SYNC_EVENT_NONE, 799 int triggerSession = 0); 800 virtual void stop(); 801 void pause(); 802 803 void flush(); 804 void destroy(); 805 void mute(bool); 806 int name() const { return mName; } 807 808 audio_stream_type_t streamType() const { 809 return mStreamType; 810 } 811 status_t attachAuxEffect(int EffectId); 812 void setAuxBuffer(int EffectId, int32_t *buffer); 813 int32_t *auxBuffer() const { return mAuxBuffer; } 814 void setMainBuffer(int16_t *buffer) { mMainBuffer = buffer; } 815 int16_t *mainBuffer() const { return mMainBuffer; } 816 int auxEffectId() const { return mAuxEffectId; } 817 818 // implement FastMixerState::VolumeProvider interface 819 virtual uint32_t getVolumeLR(); 820 821 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 822 823 protected: 824 // for numerous 825 friend class PlaybackThread; 826 friend class MixerThread; 827 friend class DirectOutputThread; 828 829 Track(const Track&); 830 Track& operator = (const Track&); 831 832 // AudioBufferProvider interface 833 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, 834 int64_t pts = kInvalidPTS); 835 // releaseBuffer() not overridden 836 837 virtual size_t framesReady() const; 838 839 bool isMuted() const { return mMute; } 840 bool isPausing() const { 841 return mState == PAUSING; 842 } 843 bool isPaused() const { 844 return mState == PAUSED; 845 } 846 bool isResuming() const { 847 return mState == RESUMING; 848 } 849 bool isReady() const; 850 void setPaused() { mState = PAUSED; } 851 void reset(); 852 853 bool isOutputTrack() const { 854 return (mStreamType == AUDIO_STREAM_CNT); 855 } 856 857 sp<IMemory> sharedBuffer() const { return mSharedBuffer; } 858 859 bool presentationComplete(size_t framesWritten, size_t audioHalFrames); 860 861 public: 862 void triggerEvents(AudioSystem::sync_event_t type); 863 virtual bool isTimedTrack() const { return false; } 864 bool isFastTrack() const { return (mFlags & IAudioFlinger::TRACK_FAST) != 0; } 865 virtual bool isOut() const; 866 867 protected: 868 869 // written by Track::mute() called by binder thread(s), without a mutex or barrier. 870 // read by Track::isMuted() called by playback thread, also without a mutex or barrier. 871 // The lack of mutex or barrier is safe because the mute status is only used by itself. 872 bool mMute; 873 874 // FILLED state is used for suppressing volume ramp at begin of playing 875 enum {FS_INVALID, FS_FILLING, FS_FILLED, FS_ACTIVE}; 876 mutable uint8_t mFillingUpStatus; 877 int8_t mRetryCount; 878 const sp<IMemory> mSharedBuffer; 879 bool mResetDone; 880 const audio_stream_type_t mStreamType; 881 int mName; // track name on the normal mixer, 882 // allocated statically at track creation time, 883 // and is even allocated (though unused) for fast tracks 884 // FIXME don't allocate track name for fast tracks 885 int16_t *mMainBuffer; 886 int32_t *mAuxBuffer; 887 int mAuxEffectId; 888 bool mHasVolumeController; 889 size_t mPresentationCompleteFrames; // number of frames written to the 890 // audio HAL when this track will be fully rendered 891 private: 892 IAudioFlinger::track_flags_t mFlags; 893 894 // The following fields are only for fast tracks, and should be in a subclass 895 int mFastIndex; // index within FastMixerState::mFastTracks[]; 896 // either mFastIndex == -1 if not isFastTrack() 897 // or 0 < mFastIndex < FastMixerState::kMaxFast because 898 // index 0 is reserved for normal mixer's submix; 899 // index is allocated statically at track creation time 900 // but the slot is only used if track is active 901 FastTrackUnderruns mObservedUnderruns; // Most recently observed value of 902 // mFastMixerDumpState.mTracks[mFastIndex].mUnderruns 903 uint32_t mUnderrunCount; // Counter of total number of underruns, never reset 904 volatile float mCachedVolume; // combined master volume and stream type volume; 905 // 'volatile' means accessed without lock or 906 // barrier, but is read/written atomically 907 }; // end of Track 908 909 class TimedTrack : public Track { 910 public: 911 static sp<TimedTrack> create(PlaybackThread *thread, 912 const sp<Client>& client, 913 audio_stream_type_t streamType, 914 uint32_t sampleRate, 915 audio_format_t format, 916 audio_channel_mask_t channelMask, 917 int frameCount, 918 const sp<IMemory>& sharedBuffer, 919 int sessionId); 920 virtual ~TimedTrack(); 921 922 class TimedBuffer { 923 public: 924 TimedBuffer(); 925 TimedBuffer(const sp<IMemory>& buffer, int64_t pts); 926 const sp<IMemory>& buffer() const { return mBuffer; } 927 int64_t pts() const { return mPTS; } 928 uint32_t position() const { return mPosition; } 929 void setPosition(uint32_t pos) { mPosition = pos; } 930 private: 931 sp<IMemory> mBuffer; 932 int64_t mPTS; 933 uint32_t mPosition; 934 }; 935 936 // Mixer facing methods. 937 virtual bool isTimedTrack() const { return true; } 938 virtual size_t framesReady() const; 939 940 // AudioBufferProvider interface 941 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, 942 int64_t pts); 943 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 944 945 // Client/App facing methods. 946 status_t allocateTimedBuffer(size_t size, 947 sp<IMemory>* buffer); 948 status_t queueTimedBuffer(const sp<IMemory>& buffer, 949 int64_t pts); 950 status_t setMediaTimeTransform(const LinearTransform& xform, 951 TimedAudioTrack::TargetTimeline target); 952 953 private: 954 TimedTrack(PlaybackThread *thread, 955 const sp<Client>& client, 956 audio_stream_type_t streamType, 957 uint32_t sampleRate, 958 audio_format_t format, 959 audio_channel_mask_t channelMask, 960 int frameCount, 961 const sp<IMemory>& sharedBuffer, 962 int sessionId); 963 964 void timedYieldSamples_l(AudioBufferProvider::Buffer* buffer); 965 void timedYieldSilence_l(uint32_t numFrames, 966 AudioBufferProvider::Buffer* buffer); 967 void trimTimedBufferQueue_l(); 968 void trimTimedBufferQueueHead_l(const char* logTag); 969 void updateFramesPendingAfterTrim_l(const TimedBuffer& buf, 970 const char* logTag); 971 972 uint64_t mLocalTimeFreq; 973 LinearTransform mLocalTimeToSampleTransform; 974 LinearTransform mMediaTimeToSampleTransform; 975 sp<MemoryDealer> mTimedMemoryDealer; 976 977 Vector<TimedBuffer> mTimedBufferQueue; 978 bool mQueueHeadInFlight; 979 bool mTrimQueueHeadOnRelease; 980 uint32_t mFramesPendingInQueue; 981 982 uint8_t* mTimedSilenceBuffer; 983 uint32_t mTimedSilenceBufferSize; 984 mutable Mutex mTimedBufferQueueLock; 985 bool mTimedAudioOutputOnTime; 986 CCHelper mCCHelper; 987 988 Mutex mMediaTimeTransformLock; 989 LinearTransform mMediaTimeTransform; 990 bool mMediaTimeTransformValid; 991 TimedAudioTrack::TargetTimeline mMediaTimeTransformTarget; 992 }; 993 994 995 // playback track 996 class OutputTrack : public Track { 997 public: 998 999 class Buffer : public AudioBufferProvider::Buffer { 1000 public: 1001 int16_t *mBuffer; 1002 }; 1003 1004 OutputTrack(PlaybackThread *thread, 1005 DuplicatingThread *sourceThread, 1006 uint32_t sampleRate, 1007 audio_format_t format, 1008 audio_channel_mask_t channelMask, 1009 int frameCount); 1010 virtual ~OutputTrack(); 1011 1012 virtual status_t start(AudioSystem::sync_event_t event = 1013 AudioSystem::SYNC_EVENT_NONE, 1014 int triggerSession = 0); 1015 virtual void stop(); 1016 bool write(int16_t* data, uint32_t frames); 1017 bool bufferQueueEmpty() const { return mBufferQueue.size() == 0; } 1018 bool isActive() const { return mActive; } 1019 const wp<ThreadBase>& thread() const { return mThread; } 1020 1021 private: 1022 1023 enum { 1024 NO_MORE_BUFFERS = 0x80000001, // same in AudioTrack.h, ok to be different value 1025 }; 1026 1027 status_t obtainBuffer(AudioBufferProvider::Buffer* buffer, 1028 uint32_t waitTimeMs); 1029 void clearBufferQueue(); 1030 1031 // Maximum number of pending buffers allocated by OutputTrack::write() 1032 static const uint8_t kMaxOverFlowBuffers = 10; 1033 1034 Vector < Buffer* > mBufferQueue; 1035 AudioBufferProvider::Buffer mOutBuffer; 1036 bool mActive; 1037 DuplicatingThread* const mSourceThread; // for waitTimeMs() in write() 1038 void* mBuffers; // starting address of buffers in plain memory 1039 }; // end of OutputTrack 1040 1041 PlaybackThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1042 audio_io_handle_t id, audio_devices_t device, type_t type); 1043 virtual ~PlaybackThread(); 1044 1045 void dump(int fd, const Vector<String16>& args); 1046 1047 // Thread virtuals 1048 virtual status_t readyToRun(); 1049 virtual bool threadLoop(); 1050 1051 // RefBase 1052 virtual void onFirstRef(); 1053 1054protected: 1055 // Code snippets that were lifted up out of threadLoop() 1056 virtual void threadLoop_mix() = 0; 1057 virtual void threadLoop_sleepTime() = 0; 1058 virtual void threadLoop_write(); 1059 virtual void threadLoop_standby(); 1060 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 1061 1062 // prepareTracks_l reads and writes mActiveTracks, and returns 1063 // the pending set of tracks to remove via Vector 'tracksToRemove'. The caller 1064 // is responsible for clearing or destroying this Vector later on, when it 1065 // is safe to do so. That will drop the final ref count and destroy the tracks. 1066 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove) = 0; 1067 1068 // ThreadBase virtuals 1069 virtual void preExit(); 1070 1071public: 1072 1073 virtual status_t initCheck() const { return (mOutput == NULL) ? NO_INIT : NO_ERROR; } 1074 1075 // return estimated latency in milliseconds, as reported by HAL 1076 uint32_t latency() const; 1077 // same, but lock must already be held 1078 uint32_t latency_l() const; 1079 1080 void setMasterVolume(float value); 1081 void setMasterMute(bool muted); 1082 1083 void setStreamVolume(audio_stream_type_t stream, float value); 1084 void setStreamMute(audio_stream_type_t stream, bool muted); 1085 1086 float streamVolume(audio_stream_type_t stream) const; 1087 1088 sp<Track> createTrack_l( 1089 const sp<AudioFlinger::Client>& client, 1090 audio_stream_type_t streamType, 1091 uint32_t sampleRate, 1092 audio_format_t format, 1093 audio_channel_mask_t channelMask, 1094 int frameCount, 1095 const sp<IMemory>& sharedBuffer, 1096 int sessionId, 1097 IAudioFlinger::track_flags_t *flags, 1098 pid_t tid, 1099 status_t *status); 1100 1101 AudioStreamOut* getOutput() const; 1102 AudioStreamOut* clearOutput(); 1103 virtual audio_stream_t* stream() const; 1104 1105 // a very large number of suspend() will eventually wraparound, but unlikely 1106 void suspend() { (void) android_atomic_inc(&mSuspended); } 1107 void restore() 1108 { 1109 // if restore() is done without suspend(), get back into 1110 // range so that the next suspend() will operate correctly 1111 if (android_atomic_dec(&mSuspended) <= 0) { 1112 android_atomic_release_store(0, &mSuspended); 1113 } 1114 } 1115 bool isSuspended() const 1116 { return android_atomic_acquire_load(&mSuspended) > 0; } 1117 1118 virtual String8 getParameters(const String8& keys); 1119 virtual void audioConfigChanged_l(int event, int param = 0); 1120 status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames); 1121 int16_t *mixBuffer() const { return mMixBuffer; }; 1122 1123 virtual void detachAuxEffect_l(int effectId); 1124 status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track> track, 1125 int EffectId); 1126 status_t attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track> track, 1127 int EffectId); 1128 1129 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1130 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1131 virtual uint32_t hasAudioSession(int sessionId) const; 1132 virtual uint32_t getStrategyForSession_l(int sessionId); 1133 1134 1135 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 1136 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 1137 void invalidateTracks(audio_stream_type_t streamType); 1138 1139 1140 protected: 1141 int16_t* mMixBuffer; 1142 1143 // suspend count, > 0 means suspended. While suspended, the thread continues to pull from 1144 // tracks and mix, but doesn't write to HAL. A2DP and SCO HAL implementations can't handle 1145 // concurrent use of both of them, so Audio Policy Service suspends one of the threads to 1146 // workaround that restriction. 1147 // 'volatile' means accessed via atomic operations and no lock. 1148 volatile int32_t mSuspended; 1149 1150 int mBytesWritten; 1151 private: 1152 // mMasterMute is in both PlaybackThread and in AudioFlinger. When a 1153 // PlaybackThread needs to find out if master-muted, it checks it's local 1154 // copy rather than the one in AudioFlinger. This optimization saves a lock. 1155 bool mMasterMute; 1156 void setMasterMute_l(bool muted) { mMasterMute = muted; } 1157 protected: 1158 SortedVector< wp<Track> > mActiveTracks; // FIXME check if this could be sp<> 1159 1160 // Allocate a track name for a given channel mask. 1161 // Returns name >= 0 if successful, -1 on failure. 1162 virtual int getTrackName_l(audio_channel_mask_t channelMask, int sessionId) = 0; 1163 virtual void deleteTrackName_l(int name) = 0; 1164 1165 // Time to sleep between cycles when: 1166 virtual uint32_t activeSleepTimeUs() const; // mixer state MIXER_TRACKS_ENABLED 1167 virtual uint32_t idleSleepTimeUs() const = 0; // mixer state MIXER_IDLE 1168 virtual uint32_t suspendSleepTimeUs() const = 0; // audio policy manager suspended us 1169 // No sleep when mixer state == MIXER_TRACKS_READY; relies on audio HAL stream->write() 1170 // No sleep in standby mode; waits on a condition 1171 1172 // Code snippets that are temporarily lifted up out of threadLoop() until the merge 1173 void checkSilentMode_l(); 1174 1175 // Non-trivial for DUPLICATING only 1176 virtual void saveOutputTracks() { } 1177 virtual void clearOutputTracks() { } 1178 1179 // Cache various calculated values, at threadLoop() entry and after a parameter change 1180 virtual void cacheParameters_l(); 1181 1182 virtual uint32_t correctLatency(uint32_t latency) const; 1183 1184 private: 1185 1186 friend class AudioFlinger; // for numerous 1187 1188 PlaybackThread(const Client&); 1189 PlaybackThread& operator = (const PlaybackThread&); 1190 1191 status_t addTrack_l(const sp<Track>& track); 1192 void destroyTrack_l(const sp<Track>& track); 1193 void removeTrack_l(const sp<Track>& track); 1194 1195 void readOutputParameters(); 1196 1197 virtual void dumpInternals(int fd, const Vector<String16>& args); 1198 void dumpTracks(int fd, const Vector<String16>& args); 1199 1200 SortedVector< sp<Track> > mTracks; 1201 // mStreamTypes[] uses 1 additional stream type internally for the OutputTrack used by 1202 // DuplicatingThread 1203 stream_type_t mStreamTypes[AUDIO_STREAM_CNT + 1]; 1204 AudioStreamOut *mOutput; 1205 1206 float mMasterVolume; 1207 nsecs_t mLastWriteTime; 1208 int mNumWrites; 1209 int mNumDelayedWrites; 1210 bool mInWrite; 1211 1212 // FIXME rename these former local variables of threadLoop to standard "m" names 1213 nsecs_t standbyTime; 1214 size_t mixBufferSize; 1215 1216 // cached copies of activeSleepTimeUs() and idleSleepTimeUs() made by cacheParameters_l() 1217 uint32_t activeSleepTime; 1218 uint32_t idleSleepTime; 1219 1220 uint32_t sleepTime; 1221 1222 // mixer status returned by prepareTracks_l() 1223 mixer_state mMixerStatus; // current cycle 1224 // previous cycle when in prepareTracks_l() 1225 mixer_state mMixerStatusIgnoringFastTracks; 1226 // FIXME or a separate ready state per track 1227 1228 // FIXME move these declarations into the specific sub-class that needs them 1229 // MIXER only 1230 uint32_t sleepTimeShift; 1231 1232 // same as AudioFlinger::mStandbyTimeInNsecs except for DIRECT which uses a shorter value 1233 nsecs_t standbyDelay; 1234 1235 // MIXER only 1236 nsecs_t maxPeriod; 1237 1238 // DUPLICATING only 1239 uint32_t writeFrames; 1240 1241 private: 1242 // The HAL output sink is treated as non-blocking, but current implementation is blocking 1243 sp<NBAIO_Sink> mOutputSink; 1244 // If a fast mixer is present, the blocking pipe sink, otherwise clear 1245 sp<NBAIO_Sink> mPipeSink; 1246 // The current sink for the normal mixer to write it's (sub)mix, mOutputSink or mPipeSink 1247 sp<NBAIO_Sink> mNormalSink; 1248 // For dumpsys 1249 sp<NBAIO_Sink> mTeeSink; 1250 sp<NBAIO_Source> mTeeSource; 1251 uint32_t mScreenState; // cached copy of gScreenState 1252 public: 1253 virtual bool hasFastMixer() const = 0; 1254 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const 1255 { FastTrackUnderruns dummy; return dummy; } 1256 1257 protected: 1258 // accessed by both binder threads and within threadLoop(), lock on mutex needed 1259 unsigned mFastTrackAvailMask; // bit i set if fast track [i] is available 1260 1261 }; 1262 1263 class MixerThread : public PlaybackThread { 1264 public: 1265 MixerThread(const sp<AudioFlinger>& audioFlinger, 1266 AudioStreamOut* output, 1267 audio_io_handle_t id, 1268 audio_devices_t device, 1269 type_t type = MIXER); 1270 virtual ~MixerThread(); 1271 1272 // Thread virtuals 1273 1274 virtual bool checkForNewParameters_l(); 1275 virtual void dumpInternals(int fd, const Vector<String16>& args); 1276 1277 protected: 1278 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1279 virtual int getTrackName_l(audio_channel_mask_t channelMask, int sessionId); 1280 virtual void deleteTrackName_l(int name); 1281 virtual uint32_t idleSleepTimeUs() const; 1282 virtual uint32_t suspendSleepTimeUs() const; 1283 virtual void cacheParameters_l(); 1284 1285 // threadLoop snippets 1286 virtual void threadLoop_write(); 1287 virtual void threadLoop_standby(); 1288 virtual void threadLoop_mix(); 1289 virtual void threadLoop_sleepTime(); 1290 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 1291 virtual uint32_t correctLatency(uint32_t latency) const; 1292 1293 AudioMixer* mAudioMixer; // normal mixer 1294 private: 1295 // one-time initialization, no locks required 1296 FastMixer* mFastMixer; // non-NULL if there is also a fast mixer 1297 sp<AudioWatchdog> mAudioWatchdog; // non-0 if there is an audio watchdog thread 1298 1299 // contents are not guaranteed to be consistent, no locks required 1300 FastMixerDumpState mFastMixerDumpState; 1301#ifdef STATE_QUEUE_DUMP 1302 StateQueueObserverDump mStateQueueObserverDump; 1303 StateQueueMutatorDump mStateQueueMutatorDump; 1304#endif 1305 AudioWatchdogDump mAudioWatchdogDump; 1306 1307 // accessible only within the threadLoop(), no locks required 1308 // mFastMixer->sq() // for mutating and pushing state 1309 int32_t mFastMixerFutex; // for cold idle 1310 1311 public: 1312 virtual bool hasFastMixer() const { return mFastMixer != NULL; } 1313 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const { 1314 ALOG_ASSERT(fastIndex < FastMixerState::kMaxFastTracks); 1315 return mFastMixerDumpState.mTracks[fastIndex].mUnderruns; 1316 } 1317 }; 1318 1319 class DirectOutputThread : public PlaybackThread { 1320 public: 1321 1322 DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1323 audio_io_handle_t id, audio_devices_t device); 1324 virtual ~DirectOutputThread(); 1325 1326 // Thread virtuals 1327 1328 virtual bool checkForNewParameters_l(); 1329 1330 protected: 1331 virtual int getTrackName_l(audio_channel_mask_t channelMask, int sessionId); 1332 virtual void deleteTrackName_l(int name); 1333 virtual uint32_t activeSleepTimeUs() const; 1334 virtual uint32_t idleSleepTimeUs() const; 1335 virtual uint32_t suspendSleepTimeUs() const; 1336 virtual void cacheParameters_l(); 1337 1338 // threadLoop snippets 1339 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1340 virtual void threadLoop_mix(); 1341 virtual void threadLoop_sleepTime(); 1342 1343 private: 1344 // volumes last sent to audio HAL with stream->set_volume() 1345 float mLeftVolFloat; 1346 float mRightVolFloat; 1347 1348 // prepareTracks_l() tells threadLoop_mix() the name of the single active track 1349 sp<Track> mActiveTrack; 1350 public: 1351 virtual bool hasFastMixer() const { return false; } 1352 }; 1353 1354 class DuplicatingThread : public MixerThread { 1355 public: 1356 DuplicatingThread(const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread, 1357 audio_io_handle_t id); 1358 virtual ~DuplicatingThread(); 1359 1360 // Thread virtuals 1361 void addOutputTrack(MixerThread* thread); 1362 void removeOutputTrack(MixerThread* thread); 1363 uint32_t waitTimeMs() const { return mWaitTimeMs; } 1364 protected: 1365 virtual uint32_t activeSleepTimeUs() const; 1366 1367 private: 1368 bool outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks); 1369 protected: 1370 // threadLoop snippets 1371 virtual void threadLoop_mix(); 1372 virtual void threadLoop_sleepTime(); 1373 virtual void threadLoop_write(); 1374 virtual void threadLoop_standby(); 1375 virtual void cacheParameters_l(); 1376 1377 private: 1378 // called from threadLoop, addOutputTrack, removeOutputTrack 1379 virtual void updateWaitTime_l(); 1380 protected: 1381 virtual void saveOutputTracks(); 1382 virtual void clearOutputTracks(); 1383 private: 1384 1385 uint32_t mWaitTimeMs; 1386 SortedVector < sp<OutputTrack> > outputTracks; 1387 SortedVector < sp<OutputTrack> > mOutputTracks; 1388 public: 1389 virtual bool hasFastMixer() const { return false; } 1390 }; 1391 1392 PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const; 1393 MixerThread *checkMixerThread_l(audio_io_handle_t output) const; 1394 RecordThread *checkRecordThread_l(audio_io_handle_t input) const; 1395 // no range check, AudioFlinger::mLock held 1396 bool streamMute_l(audio_stream_type_t stream) const 1397 { return mStreamTypes[stream].mute; } 1398 // no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held 1399 float streamVolume_l(audio_stream_type_t stream) const 1400 { return mStreamTypes[stream].volume; } 1401 void audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2); 1402 1403 // allocate an audio_io_handle_t, session ID, or effect ID 1404 uint32_t nextUniqueId(); 1405 1406 status_t moveEffectChain_l(int sessionId, 1407 PlaybackThread *srcThread, 1408 PlaybackThread *dstThread, 1409 bool reRegister); 1410 // return thread associated with primary hardware device, or NULL 1411 PlaybackThread *primaryPlaybackThread_l() const; 1412 audio_devices_t primaryOutputDevice_l() const; 1413 1414 sp<PlaybackThread> getEffectThread_l(int sessionId, int EffectId); 1415 1416 // server side of the client's IAudioTrack 1417 class TrackHandle : public android::BnAudioTrack { 1418 public: 1419 TrackHandle(const sp<PlaybackThread::Track>& track); 1420 virtual ~TrackHandle(); 1421 virtual sp<IMemory> getCblk() const; 1422 virtual status_t start(); 1423 virtual void stop(); 1424 virtual void flush(); 1425 virtual void mute(bool); 1426 virtual void pause(); 1427 virtual status_t attachAuxEffect(int effectId); 1428 virtual status_t allocateTimedBuffer(size_t size, 1429 sp<IMemory>* buffer); 1430 virtual status_t queueTimedBuffer(const sp<IMemory>& buffer, 1431 int64_t pts); 1432 virtual status_t setMediaTimeTransform(const LinearTransform& xform, 1433 int target); 1434 virtual status_t onTransact( 1435 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 1436 private: 1437 const sp<PlaybackThread::Track> mTrack; 1438 }; 1439 1440 void removeClient_l(pid_t pid); 1441 void removeNotificationClient(pid_t pid); 1442 1443 1444 // record thread 1445 class RecordThread : public ThreadBase, public AudioBufferProvider 1446 // derives from AudioBufferProvider interface for use by resampler 1447 { 1448 public: 1449 1450 // record track 1451 class RecordTrack : public TrackBase { 1452 public: 1453 RecordTrack(RecordThread *thread, 1454 const sp<Client>& client, 1455 uint32_t sampleRate, 1456 audio_format_t format, 1457 audio_channel_mask_t channelMask, 1458 int frameCount, 1459 int sessionId); 1460 virtual ~RecordTrack(); 1461 1462 virtual status_t start(AudioSystem::sync_event_t event, int triggerSession); 1463 virtual void stop(); 1464 1465 void destroy(); 1466 1467 // clear the buffer overflow flag 1468 void clearOverflow() { mOverflow = false; } 1469 // set the buffer overflow flag and return previous value 1470 bool setOverflow() { bool tmp = mOverflow; mOverflow = true; 1471 return tmp; } 1472 1473 static void appendDumpHeader(String8& result); 1474 void dump(char* buffer, size_t size); 1475 1476 virtual bool isOut() const; 1477 1478 private: 1479 friend class AudioFlinger; // for mState 1480 1481 RecordTrack(const RecordTrack&); 1482 RecordTrack& operator = (const RecordTrack&); 1483 1484 // AudioBufferProvider interface 1485 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, 1486 int64_t pts = kInvalidPTS); 1487 // releaseBuffer() not overridden 1488 1489 bool mOverflow; // overflow on most recent attempt to fill client buffer 1490 }; 1491 1492 RecordThread(const sp<AudioFlinger>& audioFlinger, 1493 AudioStreamIn *input, 1494 uint32_t sampleRate, 1495 audio_channel_mask_t channelMask, 1496 audio_io_handle_t id, 1497 audio_devices_t device, 1498 const sp<NBAIO_Sink>& teeSink); 1499 virtual ~RecordThread(); 1500 1501 // no addTrack_l ? 1502 void destroyTrack_l(const sp<RecordTrack>& track); 1503 void removeTrack_l(const sp<RecordTrack>& track); 1504 1505 void dumpInternals(int fd, const Vector<String16>& args); 1506 void dumpTracks(int fd, const Vector<String16>& args); 1507 1508 // Thread virtuals 1509 virtual bool threadLoop(); 1510 virtual status_t readyToRun(); 1511 1512 // RefBase 1513 virtual void onFirstRef(); 1514 1515 virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; } 1516 sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l( 1517 const sp<AudioFlinger::Client>& client, 1518 uint32_t sampleRate, 1519 audio_format_t format, 1520 audio_channel_mask_t channelMask, 1521 int frameCount, 1522 int sessionId, 1523 IAudioFlinger::track_flags_t flags, 1524 pid_t tid, 1525 status_t *status); 1526 1527 status_t start(RecordTrack* recordTrack, 1528 AudioSystem::sync_event_t event, 1529 int triggerSession); 1530 1531 // ask the thread to stop the specified track, and 1532 // return true if the caller should then do it's part of the stopping process 1533 bool stop_l(RecordTrack* recordTrack); 1534 1535 void dump(int fd, const Vector<String16>& args); 1536 AudioStreamIn* clearInput(); 1537 virtual audio_stream_t* stream() const; 1538 1539 // AudioBufferProvider interface 1540 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts); 1541 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 1542 1543 virtual bool checkForNewParameters_l(); 1544 virtual String8 getParameters(const String8& keys); 1545 virtual void audioConfigChanged_l(int event, int param = 0); 1546 void readInputParameters(); 1547 virtual unsigned int getInputFramesLost(); 1548 1549 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1550 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1551 virtual uint32_t hasAudioSession(int sessionId) const; 1552 1553 // Return the set of unique session IDs across all tracks. 1554 // The keys are the session IDs, and the associated values are meaningless. 1555 // FIXME replace by Set [and implement Bag/Multiset for other uses]. 1556 KeyedVector<int, bool> sessionIds() const; 1557 1558 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 1559 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 1560 1561 static void syncStartEventCallback(const wp<SyncEvent>& event); 1562 void handleSyncStartEvent(const sp<SyncEvent>& event); 1563 1564 private: 1565 void clearSyncStartEvent(); 1566 1567 // Enter standby if not already in standby, and set mStandby flag 1568 void standby(); 1569 1570 // Call the HAL standby method unconditionally, and don't change mStandby flag 1571 void inputStandBy(); 1572 1573 AudioStreamIn *mInput; 1574 SortedVector < sp<RecordTrack> > mTracks; 1575 // mActiveTrack has dual roles: it indicates the current active track, and 1576 // is used together with mStartStopCond to indicate start()/stop() progress 1577 sp<RecordTrack> mActiveTrack; 1578 Condition mStartStopCond; 1579 AudioResampler *mResampler; 1580 int32_t *mRsmpOutBuffer; 1581 int16_t *mRsmpInBuffer; 1582 size_t mRsmpInIndex; 1583 size_t mInputBytes; 1584 const int mReqChannelCount; 1585 const uint32_t mReqSampleRate; 1586 ssize_t mBytesRead; 1587 // sync event triggering actual audio capture. Frames read before this event will 1588 // be dropped and therefore not read by the application. 1589 sp<SyncEvent> mSyncStartEvent; 1590 // number of captured frames to drop after the start sync event has been received. 1591 // when < 0, maximum frames to drop before starting capture even if sync event is 1592 // not received 1593 ssize_t mFramestoDrop; 1594 1595 // For dumpsys 1596 const sp<NBAIO_Sink> mTeeSink; 1597 }; 1598 1599 // server side of the client's IAudioRecord 1600 class RecordHandle : public android::BnAudioRecord { 1601 public: 1602 RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack); 1603 virtual ~RecordHandle(); 1604 virtual sp<IMemory> getCblk() const; 1605 virtual status_t start(int /*AudioSystem::sync_event_t*/ event, int triggerSession); 1606 virtual void stop(); 1607 virtual status_t onTransact( 1608 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 1609 private: 1610 const sp<RecordThread::RecordTrack> mRecordTrack; 1611 1612 // for use from destructor 1613 void stop_nonvirtual(); 1614 }; 1615 1616 //--- Audio Effect Management 1617 1618 // EffectModule and EffectChain classes both have their own mutex to protect 1619 // state changes or resource modifications. Always respect the following order 1620 // if multiple mutexes must be acquired to avoid cross deadlock: 1621 // AudioFlinger -> ThreadBase -> EffectChain -> EffectModule 1622 1623 // The EffectModule class is a wrapper object controlling the effect engine implementation 1624 // in the effect library. It prevents concurrent calls to process() and command() functions 1625 // from different client threads. It keeps a list of EffectHandle objects corresponding 1626 // to all client applications using this effect and notifies applications of effect state, 1627 // control or parameter changes. It manages the activation state machine to send appropriate 1628 // reset, enable, disable commands to effect engine and provide volume 1629 // ramping when effects are activated/deactivated. 1630 // When controlling an auxiliary effect, the EffectModule also provides an input buffer used by 1631 // the attached track(s) to accumulate their auxiliary channel. 1632 class EffectModule : public RefBase { 1633 public: 1634 EffectModule(ThreadBase *thread, 1635 const wp<AudioFlinger::EffectChain>& chain, 1636 effect_descriptor_t *desc, 1637 int id, 1638 int sessionId); 1639 virtual ~EffectModule(); 1640 1641 enum effect_state { 1642 IDLE, 1643 RESTART, 1644 STARTING, 1645 ACTIVE, 1646 STOPPING, 1647 STOPPED, 1648 DESTROYED 1649 }; 1650 1651 int id() const { return mId; } 1652 void process(); 1653 void updateState(); 1654 status_t command(uint32_t cmdCode, 1655 uint32_t cmdSize, 1656 void *pCmdData, 1657 uint32_t *replySize, 1658 void *pReplyData); 1659 1660 void reset_l(); 1661 status_t configure(); 1662 status_t init(); 1663 effect_state state() const { 1664 return mState; 1665 } 1666 uint32_t status() { 1667 return mStatus; 1668 } 1669 int sessionId() const { 1670 return mSessionId; 1671 } 1672 status_t setEnabled(bool enabled); 1673 status_t setEnabled_l(bool enabled); 1674 bool isEnabled() const; 1675 bool isProcessEnabled() const; 1676 1677 void setInBuffer(int16_t *buffer) { mConfig.inputCfg.buffer.s16 = buffer; } 1678 int16_t *inBuffer() { return mConfig.inputCfg.buffer.s16; } 1679 void setOutBuffer(int16_t *buffer) { mConfig.outputCfg.buffer.s16 = buffer; } 1680 int16_t *outBuffer() { return mConfig.outputCfg.buffer.s16; } 1681 void setChain(const wp<EffectChain>& chain) { mChain = chain; } 1682 void setThread(const wp<ThreadBase>& thread) { mThread = thread; } 1683 const wp<ThreadBase>& thread() { return mThread; } 1684 1685 status_t addHandle(EffectHandle *handle); 1686 size_t disconnect(EffectHandle *handle, bool unpinIfLast); 1687 size_t removeHandle(EffectHandle *handle); 1688 1689 const effect_descriptor_t& desc() const { return mDescriptor; } 1690 wp<EffectChain>& chain() { return mChain; } 1691 1692 status_t setDevice(audio_devices_t device); 1693 status_t setVolume(uint32_t *left, uint32_t *right, bool controller); 1694 status_t setMode(audio_mode_t mode); 1695 status_t setAudioSource(audio_source_t source); 1696 status_t start(); 1697 status_t stop(); 1698 void setSuspended(bool suspended); 1699 bool suspended() const; 1700 1701 EffectHandle* controlHandle_l(); 1702 1703 bool isPinned() const { return mPinned; } 1704 void unPin() { mPinned = false; } 1705 bool purgeHandles(); 1706 void lock() { mLock.lock(); } 1707 void unlock() { mLock.unlock(); } 1708 1709 void dump(int fd, const Vector<String16>& args); 1710 1711 protected: 1712 friend class AudioFlinger; // for mHandles 1713 bool mPinned; 1714 1715 // Maximum time allocated to effect engines to complete the turn off sequence 1716 static const uint32_t MAX_DISABLE_TIME_MS = 10000; 1717 1718 EffectModule(const EffectModule&); 1719 EffectModule& operator = (const EffectModule&); 1720 1721 status_t start_l(); 1722 status_t stop_l(); 1723 1724mutable Mutex mLock; // mutex for process, commands and handles list protection 1725 wp<ThreadBase> mThread; // parent thread 1726 wp<EffectChain> mChain; // parent effect chain 1727 const int mId; // this instance unique ID 1728 const int mSessionId; // audio session ID 1729 const effect_descriptor_t mDescriptor;// effect descriptor received from effect engine 1730 effect_config_t mConfig; // input and output audio configuration 1731 effect_handle_t mEffectInterface; // Effect module C API 1732 status_t mStatus; // initialization status 1733 effect_state mState; // current activation state 1734 Vector<EffectHandle *> mHandles; // list of client handles 1735 // First handle in mHandles has highest priority and controls the effect module 1736 uint32_t mMaxDisableWaitCnt; // maximum grace period before forcing an effect off after 1737 // sending disable command. 1738 uint32_t mDisableWaitCnt; // current process() calls count during disable period. 1739 bool mSuspended; // effect is suspended: temporarily disabled by framework 1740 }; 1741 1742 // The EffectHandle class implements the IEffect interface. It provides resources 1743 // to receive parameter updates, keeps track of effect control 1744 // ownership and state and has a pointer to the EffectModule object it is controlling. 1745 // There is one EffectHandle object for each application controlling (or using) 1746 // an effect module. 1747 // The EffectHandle is obtained by calling AudioFlinger::createEffect(). 1748 class EffectHandle: public android::BnEffect { 1749 public: 1750 1751 EffectHandle(const sp<EffectModule>& effect, 1752 const sp<AudioFlinger::Client>& client, 1753 const sp<IEffectClient>& effectClient, 1754 int32_t priority); 1755 virtual ~EffectHandle(); 1756 1757 // IEffect 1758 virtual status_t enable(); 1759 virtual status_t disable(); 1760 virtual status_t command(uint32_t cmdCode, 1761 uint32_t cmdSize, 1762 void *pCmdData, 1763 uint32_t *replySize, 1764 void *pReplyData); 1765 virtual void disconnect(); 1766 private: 1767 void disconnect(bool unpinIfLast); 1768 public: 1769 virtual sp<IMemory> getCblk() const { return mCblkMemory; } 1770 virtual status_t onTransact(uint32_t code, const Parcel& data, 1771 Parcel* reply, uint32_t flags); 1772 1773 1774 // Give or take control of effect module 1775 // - hasControl: true if control is given, false if removed 1776 // - signal: true client app should be signaled of change, false otherwise 1777 // - enabled: state of the effect when control is passed 1778 void setControl(bool hasControl, bool signal, bool enabled); 1779 void commandExecuted(uint32_t cmdCode, 1780 uint32_t cmdSize, 1781 void *pCmdData, 1782 uint32_t replySize, 1783 void *pReplyData); 1784 void setEnabled(bool enabled); 1785 bool enabled() const { return mEnabled; } 1786 1787 // Getters 1788 int id() const { return mEffect->id(); } 1789 int priority() const { return mPriority; } 1790 bool hasControl() const { return mHasControl; } 1791 sp<EffectModule> effect() const { return mEffect; } 1792 // destroyed_l() must be called with the associated EffectModule mLock held 1793 bool destroyed_l() const { return mDestroyed; } 1794 1795 void dump(char* buffer, size_t size); 1796 1797 protected: 1798 friend class AudioFlinger; // for mEffect, mHasControl, mEnabled 1799 EffectHandle(const EffectHandle&); 1800 EffectHandle& operator =(const EffectHandle&); 1801 1802 sp<EffectModule> mEffect; // pointer to controlled EffectModule 1803 sp<IEffectClient> mEffectClient; // callback interface for client notifications 1804 /*const*/ sp<Client> mClient; // client for shared memory allocation, see disconnect() 1805 sp<IMemory> mCblkMemory; // shared memory for control block 1806 effect_param_cblk_t* mCblk; // control block for deferred parameter setting via 1807 // shared memory 1808 uint8_t* mBuffer; // pointer to parameter area in shared memory 1809 int mPriority; // client application priority to control the effect 1810 bool mHasControl; // true if this handle is controlling the effect 1811 bool mEnabled; // cached enable state: needed when the effect is 1812 // restored after being suspended 1813 bool mDestroyed; // Set to true by destructor. Access with EffectModule 1814 // mLock held 1815 }; 1816 1817 // the EffectChain class represents a group of effects associated to one audio session. 1818 // There can be any number of EffectChain objects per output mixer thread (PlaybackThread). 1819 // The EffecChain with session ID 0 contains global effects applied to the output mix. 1820 // Effects in this chain can be insert or auxiliary. Effects in other chains (attached to 1821 // tracks) are insert only. The EffectChain maintains an ordered list of effect module, the 1822 // order corresponding in the effect process order. When attached to a track (session ID != 0), 1823 // it also provide it's own input buffer used by the track as accumulation buffer. 1824 class EffectChain : public RefBase { 1825 public: 1826 EffectChain(const wp<ThreadBase>& wThread, int sessionId); 1827 EffectChain(ThreadBase *thread, int sessionId); 1828 virtual ~EffectChain(); 1829 1830 // special key used for an entry in mSuspendedEffects keyed vector 1831 // corresponding to a suspend all request. 1832 static const int kKeyForSuspendAll = 0; 1833 1834 // minimum duration during which we force calling effect process when last track on 1835 // a session is stopped or removed to allow effect tail to be rendered 1836 static const int kProcessTailDurationMs = 1000; 1837 1838 void process_l(); 1839 1840 void lock() { 1841 mLock.lock(); 1842 } 1843 void unlock() { 1844 mLock.unlock(); 1845 } 1846 1847 status_t addEffect_l(const sp<EffectModule>& handle); 1848 size_t removeEffect_l(const sp<EffectModule>& handle); 1849 1850 int sessionId() const { return mSessionId; } 1851 void setSessionId(int sessionId) { mSessionId = sessionId; } 1852 1853 sp<EffectModule> getEffectFromDesc_l(effect_descriptor_t *descriptor); 1854 sp<EffectModule> getEffectFromId_l(int id); 1855 sp<EffectModule> getEffectFromType_l(const effect_uuid_t *type); 1856 bool setVolume_l(uint32_t *left, uint32_t *right); 1857 void setDevice_l(audio_devices_t device); 1858 void setMode_l(audio_mode_t mode); 1859 void setAudioSource_l(audio_source_t source); 1860 1861 void setInBuffer(int16_t *buffer, bool ownsBuffer = false) { 1862 mInBuffer = buffer; 1863 mOwnInBuffer = ownsBuffer; 1864 } 1865 int16_t *inBuffer() const { 1866 return mInBuffer; 1867 } 1868 void setOutBuffer(int16_t *buffer) { 1869 mOutBuffer = buffer; 1870 } 1871 int16_t *outBuffer() const { 1872 return mOutBuffer; 1873 } 1874 1875 void incTrackCnt() { android_atomic_inc(&mTrackCnt); } 1876 void decTrackCnt() { android_atomic_dec(&mTrackCnt); } 1877 int32_t trackCnt() const { return android_atomic_acquire_load(&mTrackCnt); } 1878 1879 void incActiveTrackCnt() { android_atomic_inc(&mActiveTrackCnt); 1880 mTailBufferCount = mMaxTailBuffers; } 1881 void decActiveTrackCnt() { android_atomic_dec(&mActiveTrackCnt); } 1882 int32_t activeTrackCnt() const { return android_atomic_acquire_load(&mActiveTrackCnt); } 1883 1884 uint32_t strategy() const { return mStrategy; } 1885 void setStrategy(uint32_t strategy) 1886 { mStrategy = strategy; } 1887 1888 // suspend effect of the given type 1889 void setEffectSuspended_l(const effect_uuid_t *type, 1890 bool suspend); 1891 // suspend all eligible effects 1892 void setEffectSuspendedAll_l(bool suspend); 1893 // check if effects should be suspend or restored when a given effect is enable or disabled 1894 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1895 bool enabled); 1896 1897 void clearInputBuffer(); 1898 1899 void dump(int fd, const Vector<String16>& args); 1900 1901 protected: 1902 friend class AudioFlinger; // for mThread, mEffects 1903 EffectChain(const EffectChain&); 1904 EffectChain& operator =(const EffectChain&); 1905 1906 class SuspendedEffectDesc : public RefBase { 1907 public: 1908 SuspendedEffectDesc() : mRefCount(0) {} 1909 1910 int mRefCount; 1911 effect_uuid_t mType; 1912 wp<EffectModule> mEffect; 1913 }; 1914 1915 // get a list of effect modules to suspend when an effect of the type 1916 // passed is enabled. 1917 void getSuspendEligibleEffects(Vector< sp<EffectModule> > &effects); 1918 1919 // get an effect module if it is currently enable 1920 sp<EffectModule> getEffectIfEnabled(const effect_uuid_t *type); 1921 // true if the effect whose descriptor is passed can be suspended 1922 // OEMs can modify the rules implemented in this method to exclude specific effect 1923 // types or implementations from the suspend/restore mechanism. 1924 bool isEffectEligibleForSuspend(const effect_descriptor_t& desc); 1925 1926 void clearInputBuffer_l(sp<ThreadBase> thread); 1927 1928 wp<ThreadBase> mThread; // parent mixer thread 1929 Mutex mLock; // mutex protecting effect list 1930 Vector< sp<EffectModule> > mEffects; // list of effect modules 1931 int mSessionId; // audio session ID 1932 int16_t *mInBuffer; // chain input buffer 1933 int16_t *mOutBuffer; // chain output buffer 1934 1935 // 'volatile' here means these are accessed with atomic operations instead of mutex 1936 volatile int32_t mActiveTrackCnt; // number of active tracks connected 1937 volatile int32_t mTrackCnt; // number of tracks connected 1938 1939 int32_t mTailBufferCount; // current effect tail buffer count 1940 int32_t mMaxTailBuffers; // maximum effect tail buffers 1941 bool mOwnInBuffer; // true if the chain owns its input buffer 1942 int mVolumeCtrlIdx; // index of insert effect having control over volume 1943 uint32_t mLeftVolume; // previous volume on left channel 1944 uint32_t mRightVolume; // previous volume on right channel 1945 uint32_t mNewLeftVolume; // new volume on left channel 1946 uint32_t mNewRightVolume; // new volume on right channel 1947 uint32_t mStrategy; // strategy for this effect chain 1948 // mSuspendedEffects lists all effects currently suspended in the chain. 1949 // Use effect type UUID timelow field as key. There is no real risk of identical 1950 // timeLow fields among effect type UUIDs. 1951 // Updated by updateSuspendedSessions_l() only. 1952 KeyedVector< int, sp<SuspendedEffectDesc> > mSuspendedEffects; 1953 }; 1954 1955 class AudioHwDevice { 1956 public: 1957 enum Flags { 1958 AHWD_CAN_SET_MASTER_VOLUME = 0x1, 1959 AHWD_CAN_SET_MASTER_MUTE = 0x2, 1960 }; 1961 1962 AudioHwDevice(const char *moduleName, 1963 audio_hw_device_t *hwDevice, 1964 Flags flags) 1965 : mModuleName(strdup(moduleName)) 1966 , mHwDevice(hwDevice) 1967 , mFlags(flags) { } 1968 /*virtual*/ ~AudioHwDevice() { free((void *)mModuleName); } 1969 1970 bool canSetMasterVolume() const { 1971 return (0 != (mFlags & AHWD_CAN_SET_MASTER_VOLUME)); 1972 } 1973 1974 bool canSetMasterMute() const { 1975 return (0 != (mFlags & AHWD_CAN_SET_MASTER_MUTE)); 1976 } 1977 1978 const char *moduleName() const { return mModuleName; } 1979 audio_hw_device_t *hwDevice() const { return mHwDevice; } 1980 private: 1981 const char * const mModuleName; 1982 audio_hw_device_t * const mHwDevice; 1983 Flags mFlags; 1984 }; 1985 1986 // AudioStreamOut and AudioStreamIn are immutable, so their fields are const. 1987 // For emphasis, we could also make all pointers to them be "const *", 1988 // but that would clutter the code unnecessarily. 1989 1990 struct AudioStreamOut { 1991 AudioHwDevice* const audioHwDev; 1992 audio_stream_out_t* const stream; 1993 1994 audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); } 1995 1996 AudioStreamOut(AudioHwDevice *dev, audio_stream_out_t *out) : 1997 audioHwDev(dev), stream(out) {} 1998 }; 1999 2000 struct AudioStreamIn { 2001 AudioHwDevice* const audioHwDev; 2002 audio_stream_in_t* const stream; 2003 2004 audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); } 2005 2006 AudioStreamIn(AudioHwDevice *dev, audio_stream_in_t *in) : 2007 audioHwDev(dev), stream(in) {} 2008 }; 2009 2010 // for mAudioSessionRefs only 2011 struct AudioSessionRef { 2012 AudioSessionRef(int sessionid, pid_t pid) : 2013 mSessionid(sessionid), mPid(pid), mCnt(1) {} 2014 const int mSessionid; 2015 const pid_t mPid; 2016 int mCnt; 2017 }; 2018 2019 mutable Mutex mLock; 2020 2021 DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client() 2022 2023 mutable Mutex mHardwareLock; 2024 // NOTE: If both mLock and mHardwareLock mutexes must be held, 2025 // always take mLock before mHardwareLock 2026 2027 // These two fields are immutable after onFirstRef(), so no lock needed to access 2028 AudioHwDevice* mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL 2029 DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*> mAudioHwDevs; 2030 2031 // for dump, indicates which hardware operation is currently in progress (but not stream ops) 2032 enum hardware_call_state { 2033 AUDIO_HW_IDLE = 0, // no operation in progress 2034 AUDIO_HW_INIT, // init_check 2035 AUDIO_HW_OUTPUT_OPEN, // open_output_stream 2036 AUDIO_HW_OUTPUT_CLOSE, // unused 2037 AUDIO_HW_INPUT_OPEN, // unused 2038 AUDIO_HW_INPUT_CLOSE, // unused 2039 AUDIO_HW_STANDBY, // unused 2040 AUDIO_HW_SET_MASTER_VOLUME, // set_master_volume 2041 AUDIO_HW_GET_ROUTING, // unused 2042 AUDIO_HW_SET_ROUTING, // unused 2043 AUDIO_HW_GET_MODE, // unused 2044 AUDIO_HW_SET_MODE, // set_mode 2045 AUDIO_HW_GET_MIC_MUTE, // get_mic_mute 2046 AUDIO_HW_SET_MIC_MUTE, // set_mic_mute 2047 AUDIO_HW_SET_VOICE_VOLUME, // set_voice_volume 2048 AUDIO_HW_SET_PARAMETER, // set_parameters 2049 AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size 2050 AUDIO_HW_GET_MASTER_VOLUME, // get_master_volume 2051 AUDIO_HW_GET_PARAMETER, // get_parameters 2052 AUDIO_HW_SET_MASTER_MUTE, // set_master_mute 2053 AUDIO_HW_GET_MASTER_MUTE, // get_master_mute 2054 }; 2055 2056 mutable hardware_call_state mHardwareStatus; // for dump only 2057 2058 2059 DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads; 2060 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 2061 2062 // member variables below are protected by mLock 2063 float mMasterVolume; 2064 bool mMasterMute; 2065 // end of variables protected by mLock 2066 2067 DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads; 2068 2069 DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients; 2070 volatile int32_t mNextUniqueId; // updated by android_atomic_inc 2071 audio_mode_t mMode; 2072 bool mBtNrecIsOff; 2073 2074 // protected by mLock 2075 Vector<AudioSessionRef*> mAudioSessionRefs; 2076 2077 float masterVolume_l() const; 2078 bool masterMute_l() const; 2079 audio_module_handle_t loadHwModule_l(const char *name); 2080 2081 Vector < sp<SyncEvent> > mPendingSyncEvents; // sync events awaiting for a session 2082 // to be created 2083 2084private: 2085 sp<Client> registerPid_l(pid_t pid); // always returns non-0 2086 2087 // for use from destructor 2088 status_t closeOutput_nonvirtual(audio_io_handle_t output); 2089 status_t closeInput_nonvirtual(audio_io_handle_t input); 2090 2091 // all record threads serially share a common tee sink, which is re-created on format change 2092 sp<NBAIO_Sink> mRecordTeeSink; 2093 sp<NBAIO_Source> mRecordTeeSource; 2094 2095public: 2096 static void dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id = 0); 2097}; 2098 2099 2100// ---------------------------------------------------------------------------- 2101 2102}; // namespace android 2103 2104#endif // ANDROID_AUDIO_FLINGER_H 2105