AudioFlinger.h revision e33054eb968cbf8ccaee1b0ff0301403902deed6
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 size_t 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 size_t 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 uint32_t getPrimaryOutputSamplingRate(); 211 virtual size_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 size_t 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 uint32_t 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(); // mStepCount is an implicit input 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 mStepCount; // saves AudioBufferProvider::Buffer::frameCount as of 470 // time of releaseBuffer() for later use by step() 471 // we don't really need a lock for these 472 track_state mState; 473 const uint32_t mSampleRate; // initial sample rate only; for tracks which 474 // support dynamic rates, the current value is in control block 475 const audio_format_t mFormat; 476 const audio_channel_mask_t mChannelMask; 477 const uint8_t mChannelCount; 478 const size_t mFrameSize; // AudioFlinger's view of frame size in shared memory, 479 // where for AudioTrack (but not AudioRecord), 480 // 8-bit PCM samples are stored as 16-bit 481 bool mStepServerFailed; 482 const int mSessionId; 483 Vector < sp<SyncEvent> >mSyncEvents; 484 }; 485 486 enum { 487 CFG_EVENT_IO, 488 CFG_EVENT_PRIO 489 }; 490 491 class ConfigEvent { 492 public: 493 ConfigEvent(int type) : mType(type) {} 494 virtual ~ConfigEvent() {} 495 496 int type() const { return mType; } 497 498 virtual void dump(char *buffer, size_t size) = 0; 499 500 private: 501 const int mType; 502 }; 503 504 class IoConfigEvent : public ConfigEvent { 505 public: 506 IoConfigEvent(int event, int param) : 507 ConfigEvent(CFG_EVENT_IO), mEvent(event), mParam(event) {} 508 virtual ~IoConfigEvent() {} 509 510 int event() const { return mEvent; } 511 int param() const { return mParam; } 512 513 virtual void dump(char *buffer, size_t size) { 514 snprintf(buffer, size, "IO event: event %d, param %d\n", mEvent, mParam); 515 } 516 517 private: 518 const int mEvent; 519 const int mParam; 520 }; 521 522 class PrioConfigEvent : public ConfigEvent { 523 public: 524 PrioConfigEvent(pid_t pid, pid_t tid, int32_t prio) : 525 ConfigEvent(CFG_EVENT_PRIO), mPid(pid), mTid(tid), mPrio(prio) {} 526 virtual ~PrioConfigEvent() {} 527 528 pid_t pid() const { return mPid; } 529 pid_t tid() const { return mTid; } 530 int32_t prio() const { return mPrio; } 531 532 virtual void dump(char *buffer, size_t size) { 533 snprintf(buffer, size, "Prio event: pid %d, tid %d, prio %d\n", mPid, mTid, mPrio); 534 } 535 536 private: 537 const pid_t mPid; 538 const pid_t mTid; 539 const int32_t mPrio; 540 }; 541 542 543 class PMDeathRecipient : public IBinder::DeathRecipient { 544 public: 545 PMDeathRecipient(const wp<ThreadBase>& thread) : mThread(thread) {} 546 virtual ~PMDeathRecipient() {} 547 548 // IBinder::DeathRecipient 549 virtual void binderDied(const wp<IBinder>& who); 550 551 private: 552 PMDeathRecipient(const PMDeathRecipient&); 553 PMDeathRecipient& operator = (const PMDeathRecipient&); 554 555 wp<ThreadBase> mThread; 556 }; 557 558 virtual status_t initCheck() const = 0; 559 560 // static externally-visible 561 type_t type() const { return mType; } 562 audio_io_handle_t id() const { return mId;} 563 564 // dynamic externally-visible 565 uint32_t sampleRate() const { return mSampleRate; } 566 int channelCount() const { return mChannelCount; } 567 audio_channel_mask_t channelMask() const { return mChannelMask; } 568 audio_format_t format() const { return mFormat; } 569 // Called by AudioFlinger::frameCount(audio_io_handle_t output) and effects, 570 // and returns the normal mix buffer's frame count. 571 size_t frameCount() const { return mNormalFrameCount; } 572 // Return's the HAL's frame count i.e. fast mixer buffer size. 573 size_t frameCountHAL() const { return mFrameCount; } 574 575 // Should be "virtual status_t requestExitAndWait()" and override same 576 // method in Thread, but Thread::requestExitAndWait() is not yet virtual. 577 void exit(); 578 virtual bool checkForNewParameters_l() = 0; 579 virtual status_t setParameters(const String8& keyValuePairs); 580 virtual String8 getParameters(const String8& keys) = 0; 581 virtual void audioConfigChanged_l(int event, int param = 0) = 0; 582 void sendIoConfigEvent(int event, int param = 0); 583 void sendIoConfigEvent_l(int event, int param = 0); 584 void sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio); 585 void processConfigEvents(); 586 587 // see note at declaration of mStandby, mOutDevice and mInDevice 588 bool standby() const { return mStandby; } 589 audio_devices_t outDevice() const { return mOutDevice; } 590 audio_devices_t inDevice() const { return mInDevice; } 591 592 virtual audio_stream_t* stream() const = 0; 593 594 sp<EffectHandle> createEffect_l( 595 const sp<AudioFlinger::Client>& client, 596 const sp<IEffectClient>& effectClient, 597 int32_t priority, 598 int sessionId, 599 effect_descriptor_t *desc, 600 int *enabled, 601 status_t *status); 602 void disconnectEffect(const sp< EffectModule>& effect, 603 EffectHandle *handle, 604 bool unpinIfLast); 605 606 // return values for hasAudioSession (bit field) 607 enum effect_state { 608 EFFECT_SESSION = 0x1, // the audio session corresponds to at least one 609 // effect 610 TRACK_SESSION = 0x2 // the audio session corresponds to at least one 611 // track 612 }; 613 614 // get effect chain corresponding to session Id. 615 sp<EffectChain> getEffectChain(int sessionId); 616 // same as getEffectChain() but must be called with ThreadBase mutex locked 617 sp<EffectChain> getEffectChain_l(int sessionId) const; 618 // add an effect chain to the chain list (mEffectChains) 619 virtual status_t addEffectChain_l(const sp<EffectChain>& chain) = 0; 620 // remove an effect chain from the chain list (mEffectChains) 621 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain) = 0; 622 // lock all effect chains Mutexes. Must be called before releasing the 623 // ThreadBase mutex before processing the mixer and effects. This guarantees the 624 // integrity of the chains during the process. 625 // Also sets the parameter 'effectChains' to current value of mEffectChains. 626 void lockEffectChains_l(Vector< sp<EffectChain> >& effectChains); 627 // unlock effect chains after process 628 void unlockEffectChains(const Vector< sp<EffectChain> >& effectChains); 629 // set audio mode to all effect chains 630 void setMode(audio_mode_t mode); 631 // get effect module with corresponding ID on specified audio session 632 sp<AudioFlinger::EffectModule> getEffect(int sessionId, int effectId); 633 sp<AudioFlinger::EffectModule> getEffect_l(int sessionId, int effectId); 634 // add and effect module. Also creates the effect chain is none exists for 635 // the effects audio session 636 status_t addEffect_l(const sp< EffectModule>& effect); 637 // remove and effect module. Also removes the effect chain is this was the last 638 // effect 639 void removeEffect_l(const sp< EffectModule>& effect); 640 // detach all tracks connected to an auxiliary effect 641 virtual void detachAuxEffect_l(int effectId) {} 642 // returns either EFFECT_SESSION if effects on this audio session exist in one 643 // chain, or TRACK_SESSION if tracks on this audio session exist, or both 644 virtual uint32_t hasAudioSession(int sessionId) const = 0; 645 // the value returned by default implementation is not important as the 646 // strategy is only meaningful for PlaybackThread which implements this method 647 virtual uint32_t getStrategyForSession_l(int sessionId) { return 0; } 648 649 // suspend or restore effect according to the type of effect passed. a NULL 650 // type pointer means suspend all effects in the session 651 void setEffectSuspended(const effect_uuid_t *type, 652 bool suspend, 653 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 654 // check if some effects must be suspended/restored when an effect is enabled 655 // or disabled 656 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 657 bool enabled, 658 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 659 void checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 660 bool enabled, 661 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 662 663 virtual status_t setSyncEvent(const sp<SyncEvent>& event) = 0; 664 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const = 0; 665 666 667 mutable Mutex mLock; 668 669 protected: 670 671 // entry describing an effect being suspended in mSuspendedSessions keyed vector 672 class SuspendedSessionDesc : public RefBase { 673 public: 674 SuspendedSessionDesc() : mRefCount(0) {} 675 676 int mRefCount; // number of active suspend requests 677 effect_uuid_t mType; // effect type UUID 678 }; 679 680 void acquireWakeLock(); 681 void acquireWakeLock_l(); 682 void releaseWakeLock(); 683 void releaseWakeLock_l(); 684 void setEffectSuspended_l(const effect_uuid_t *type, 685 bool suspend, 686 int sessionId); 687 // updated mSuspendedSessions when an effect suspended or restored 688 void updateSuspendedSessions_l(const effect_uuid_t *type, 689 bool suspend, 690 int sessionId); 691 // check if some effects must be suspended when an effect chain is added 692 void checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain); 693 694 virtual void preExit() { } 695 696 friend class AudioFlinger; // for mEffectChains 697 698 const type_t mType; 699 700 // Used by parameters, config events, addTrack_l, exit 701 Condition mWaitWorkCV; 702 703 const sp<AudioFlinger> mAudioFlinger; 704 uint32_t mSampleRate; 705 size_t mFrameCount; // output HAL, direct output, record 706 size_t mNormalFrameCount; // normal mixer and effects 707 audio_channel_mask_t mChannelMask; 708 uint16_t mChannelCount; 709 size_t mFrameSize; 710 audio_format_t mFormat; 711 712 // Parameter sequence by client: binder thread calling setParameters(): 713 // 1. Lock mLock 714 // 2. Append to mNewParameters 715 // 3. mWaitWorkCV.signal 716 // 4. mParamCond.waitRelative with timeout 717 // 5. read mParamStatus 718 // 6. mWaitWorkCV.signal 719 // 7. Unlock 720 // 721 // Parameter sequence by server: threadLoop calling checkForNewParameters_l(): 722 // 1. Lock mLock 723 // 2. If there is an entry in mNewParameters proceed ... 724 // 2. Read first entry in mNewParameters 725 // 3. Process 726 // 4. Remove first entry from mNewParameters 727 // 5. Set mParamStatus 728 // 6. mParamCond.signal 729 // 7. mWaitWorkCV.wait with timeout (this is to avoid overwriting mParamStatus) 730 // 8. Unlock 731 Condition mParamCond; 732 Vector<String8> mNewParameters; 733 status_t mParamStatus; 734 735 Vector<ConfigEvent *> mConfigEvents; 736 737 // These fields are written and read by thread itself without lock or barrier, 738 // and read by other threads without lock or barrier via standby() , outDevice() 739 // and inDevice(). 740 // Because of the absence of a lock or barrier, any other thread that reads 741 // these fields must use the information in isolation, or be prepared to deal 742 // with possibility that it might be inconsistent with other information. 743 bool mStandby; // Whether thread is currently in standby. 744 audio_devices_t mOutDevice; // output device 745 audio_devices_t mInDevice; // input device 746 audio_source_t mAudioSource; // (see audio.h, audio_source_t) 747 748 const audio_io_handle_t mId; 749 Vector< sp<EffectChain> > mEffectChains; 750 751 static const int kNameLength = 16; // prctl(PR_SET_NAME) limit 752 char mName[kNameLength]; 753 sp<IPowerManager> mPowerManager; 754 sp<IBinder> mWakeLockToken; 755 const sp<PMDeathRecipient> mDeathRecipient; 756 // list of suspended effects per session and per type. The first vector is 757 // keyed by session ID, the second by type UUID timeLow field 758 KeyedVector< int, KeyedVector< int, sp<SuspendedSessionDesc> > > 759 mSuspendedSessions; 760 }; 761 762 struct stream_type_t { 763 stream_type_t() 764 : volume(1.0f), 765 mute(false) 766 { 767 } 768 float volume; 769 bool mute; 770 }; 771 772 // --- PlaybackThread --- 773 class PlaybackThread : public ThreadBase { 774 public: 775 776 enum mixer_state { 777 MIXER_IDLE, // no active tracks 778 MIXER_TRACKS_ENABLED, // at least one active track, but no track has any data ready 779 MIXER_TRACKS_READY // at least one active track, and at least one track has data 780 // standby mode does not have an enum value 781 // suspend by audio policy manager is orthogonal to mixer state 782 }; 783 784 // playback track 785 class Track : public TrackBase, public VolumeProvider { 786 public: 787 Track( PlaybackThread *thread, 788 const sp<Client>& client, 789 audio_stream_type_t streamType, 790 uint32_t sampleRate, 791 audio_format_t format, 792 audio_channel_mask_t channelMask, 793 size_t frameCount, 794 const sp<IMemory>& sharedBuffer, 795 int sessionId, 796 IAudioFlinger::track_flags_t flags); 797 virtual ~Track(); 798 799 static void appendDumpHeader(String8& result); 800 void dump(char* buffer, size_t size); 801 virtual status_t start(AudioSystem::sync_event_t event = 802 AudioSystem::SYNC_EVENT_NONE, 803 int triggerSession = 0); 804 virtual void stop(); 805 void pause(); 806 807 void flush(); 808 void destroy(); 809 void mute(bool); 810 int name() const { return mName; } 811 812 audio_stream_type_t streamType() const { 813 return mStreamType; 814 } 815 status_t attachAuxEffect(int EffectId); 816 void setAuxBuffer(int EffectId, int32_t *buffer); 817 int32_t *auxBuffer() const { return mAuxBuffer; } 818 void setMainBuffer(int16_t *buffer) { mMainBuffer = buffer; } 819 int16_t *mainBuffer() const { return mMainBuffer; } 820 int auxEffectId() const { return mAuxEffectId; } 821 822 // implement FastMixerState::VolumeProvider interface 823 virtual uint32_t getVolumeLR(); 824 825 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 826 827 protected: 828 // for numerous 829 friend class PlaybackThread; 830 friend class MixerThread; 831 friend class DirectOutputThread; 832 833 Track(const Track&); 834 Track& operator = (const Track&); 835 836 // AudioBufferProvider interface 837 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, 838 int64_t pts = kInvalidPTS); 839 // releaseBuffer() not overridden 840 841 virtual size_t framesReady() const; 842 843 bool isMuted() const { return mMute; } 844 bool isPausing() const { 845 return mState == PAUSING; 846 } 847 bool isPaused() const { 848 return mState == PAUSED; 849 } 850 bool isResuming() const { 851 return mState == RESUMING; 852 } 853 bool isReady() const; 854 void setPaused() { mState = PAUSED; } 855 void reset(); 856 857 bool isOutputTrack() const { 858 return (mStreamType == AUDIO_STREAM_CNT); 859 } 860 861 sp<IMemory> sharedBuffer() const { return mSharedBuffer; } 862 863 // framesWritten is cumulative, never reset, and is shared all tracks 864 // audioHalFrames is derived from output latency 865 // FIXME parameters not needed, could get them from the thread 866 bool presentationComplete(size_t framesWritten, size_t audioHalFrames); 867 868 public: 869 void triggerEvents(AudioSystem::sync_event_t type); 870 virtual bool isTimedTrack() const { return false; } 871 bool isFastTrack() const { return (mFlags & IAudioFlinger::TRACK_FAST) != 0; } 872 virtual bool isOut() const; 873 874 protected: 875 876 // written by Track::mute() called by binder thread(s), without a mutex or barrier. 877 // read by Track::isMuted() called by playback thread, also without a mutex or barrier. 878 // The lack of mutex or barrier is safe because the mute status is only used by itself. 879 bool mMute; 880 881 // FILLED state is used for suppressing volume ramp at begin of playing 882 enum {FS_INVALID, FS_FILLING, FS_FILLED, FS_ACTIVE}; 883 mutable uint8_t mFillingUpStatus; 884 int8_t mRetryCount; 885 const sp<IMemory> mSharedBuffer; 886 bool mResetDone; 887 const audio_stream_type_t mStreamType; 888 int mName; // track name on the normal mixer, 889 // allocated statically at track creation time, 890 // and is even allocated (though unused) for fast tracks 891 // FIXME don't allocate track name for fast tracks 892 int16_t *mMainBuffer; 893 int32_t *mAuxBuffer; 894 int mAuxEffectId; 895 bool mHasVolumeController; 896 size_t mPresentationCompleteFrames; // number of frames written to the 897 // audio HAL when this track will be fully rendered 898 // zero means not monitoring 899 private: 900 IAudioFlinger::track_flags_t mFlags; 901 902 // The following fields are only for fast tracks, and should be in a subclass 903 int mFastIndex; // index within FastMixerState::mFastTracks[]; 904 // either mFastIndex == -1 if not isFastTrack() 905 // or 0 < mFastIndex < FastMixerState::kMaxFast because 906 // index 0 is reserved for normal mixer's submix; 907 // index is allocated statically at track creation time 908 // but the slot is only used if track is active 909 FastTrackUnderruns mObservedUnderruns; // Most recently observed value of 910 // mFastMixerDumpState.mTracks[mFastIndex].mUnderruns 911 uint32_t mUnderrunCount; // Counter of total number of underruns, never reset 912 volatile float mCachedVolume; // combined master volume and stream type volume; 913 // 'volatile' means accessed without lock or 914 // barrier, but is read/written atomically 915 }; // end of Track 916 917 class TimedTrack : public Track { 918 public: 919 static sp<TimedTrack> create(PlaybackThread *thread, 920 const sp<Client>& client, 921 audio_stream_type_t streamType, 922 uint32_t sampleRate, 923 audio_format_t format, 924 audio_channel_mask_t channelMask, 925 size_t frameCount, 926 const sp<IMemory>& sharedBuffer, 927 int sessionId); 928 virtual ~TimedTrack(); 929 930 class TimedBuffer { 931 public: 932 TimedBuffer(); 933 TimedBuffer(const sp<IMemory>& buffer, int64_t pts); 934 const sp<IMemory>& buffer() const { return mBuffer; } 935 int64_t pts() const { return mPTS; } 936 uint32_t position() const { return mPosition; } 937 void setPosition(uint32_t pos) { mPosition = pos; } 938 private: 939 sp<IMemory> mBuffer; 940 int64_t mPTS; 941 uint32_t mPosition; 942 }; 943 944 // Mixer facing methods. 945 virtual bool isTimedTrack() const { return true; } 946 virtual size_t framesReady() const; 947 948 // AudioBufferProvider interface 949 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, 950 int64_t pts); 951 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 952 953 // Client/App facing methods. 954 status_t allocateTimedBuffer(size_t size, 955 sp<IMemory>* buffer); 956 status_t queueTimedBuffer(const sp<IMemory>& buffer, 957 int64_t pts); 958 status_t setMediaTimeTransform(const LinearTransform& xform, 959 TimedAudioTrack::TargetTimeline target); 960 961 private: 962 TimedTrack(PlaybackThread *thread, 963 const sp<Client>& client, 964 audio_stream_type_t streamType, 965 uint32_t sampleRate, 966 audio_format_t format, 967 audio_channel_mask_t channelMask, 968 size_t frameCount, 969 const sp<IMemory>& sharedBuffer, 970 int sessionId); 971 972 void timedYieldSamples_l(AudioBufferProvider::Buffer* buffer); 973 void timedYieldSilence_l(uint32_t numFrames, 974 AudioBufferProvider::Buffer* buffer); 975 void trimTimedBufferQueue_l(); 976 void trimTimedBufferQueueHead_l(const char* logTag); 977 void updateFramesPendingAfterTrim_l(const TimedBuffer& buf, 978 const char* logTag); 979 980 uint64_t mLocalTimeFreq; 981 LinearTransform mLocalTimeToSampleTransform; 982 LinearTransform mMediaTimeToSampleTransform; 983 sp<MemoryDealer> mTimedMemoryDealer; 984 985 Vector<TimedBuffer> mTimedBufferQueue; 986 bool mQueueHeadInFlight; 987 bool mTrimQueueHeadOnRelease; 988 uint32_t mFramesPendingInQueue; 989 990 uint8_t* mTimedSilenceBuffer; 991 uint32_t mTimedSilenceBufferSize; 992 mutable Mutex mTimedBufferQueueLock; 993 bool mTimedAudioOutputOnTime; 994 CCHelper mCCHelper; 995 996 Mutex mMediaTimeTransformLock; 997 LinearTransform mMediaTimeTransform; 998 bool mMediaTimeTransformValid; 999 TimedAudioTrack::TargetTimeline mMediaTimeTransformTarget; 1000 }; 1001 1002 1003 // playback track, used by DuplicatingThread 1004 class OutputTrack : public Track { 1005 public: 1006 1007 class Buffer : public AudioBufferProvider::Buffer { 1008 public: 1009 int16_t *mBuffer; 1010 }; 1011 1012 OutputTrack(PlaybackThread *thread, 1013 DuplicatingThread *sourceThread, 1014 uint32_t sampleRate, 1015 audio_format_t format, 1016 audio_channel_mask_t channelMask, 1017 size_t frameCount); 1018 virtual ~OutputTrack(); 1019 1020 virtual status_t start(AudioSystem::sync_event_t event = 1021 AudioSystem::SYNC_EVENT_NONE, 1022 int triggerSession = 0); 1023 virtual void stop(); 1024 bool write(int16_t* data, uint32_t frames); 1025 bool bufferQueueEmpty() const { return mBufferQueue.size() == 0; } 1026 bool isActive() const { return mActive; } 1027 const wp<ThreadBase>& thread() const { return mThread; } 1028 1029 private: 1030 1031 enum { 1032 NO_MORE_BUFFERS = 0x80000001, // same in AudioTrack.h, ok to be different value 1033 }; 1034 1035 status_t obtainBuffer(AudioBufferProvider::Buffer* buffer, 1036 uint32_t waitTimeMs); 1037 void clearBufferQueue(); 1038 1039 // Maximum number of pending buffers allocated by OutputTrack::write() 1040 static const uint8_t kMaxOverFlowBuffers = 10; 1041 1042 Vector < Buffer* > mBufferQueue; 1043 AudioBufferProvider::Buffer mOutBuffer; 1044 bool mActive; 1045 DuplicatingThread* const mSourceThread; // for waitTimeMs() in write() 1046 void* mBuffers; // starting address of buffers in plain memory 1047 }; // end of OutputTrack 1048 1049 PlaybackThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1050 audio_io_handle_t id, audio_devices_t device, type_t type); 1051 virtual ~PlaybackThread(); 1052 1053 void dump(int fd, const Vector<String16>& args); 1054 1055 // Thread virtuals 1056 virtual status_t readyToRun(); 1057 virtual bool threadLoop(); 1058 1059 // RefBase 1060 virtual void onFirstRef(); 1061 1062protected: 1063 // Code snippets that were lifted up out of threadLoop() 1064 virtual void threadLoop_mix() = 0; 1065 virtual void threadLoop_sleepTime() = 0; 1066 virtual void threadLoop_write(); 1067 virtual void threadLoop_standby(); 1068 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 1069 1070 // prepareTracks_l reads and writes mActiveTracks, and returns 1071 // the pending set of tracks to remove via Vector 'tracksToRemove'. The caller 1072 // is responsible for clearing or destroying this Vector later on, when it 1073 // is safe to do so. That will drop the final ref count and destroy the tracks. 1074 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove) = 0; 1075 1076 // ThreadBase virtuals 1077 virtual void preExit(); 1078 1079public: 1080 1081 virtual status_t initCheck() const { return (mOutput == NULL) ? NO_INIT : NO_ERROR; } 1082 1083 // return estimated latency in milliseconds, as reported by HAL 1084 uint32_t latency() const; 1085 // same, but lock must already be held 1086 uint32_t latency_l() const; 1087 1088 void setMasterVolume(float value); 1089 void setMasterMute(bool muted); 1090 1091 void setStreamVolume(audio_stream_type_t stream, float value); 1092 void setStreamMute(audio_stream_type_t stream, bool muted); 1093 1094 float streamVolume(audio_stream_type_t stream) const; 1095 1096 sp<Track> createTrack_l( 1097 const sp<AudioFlinger::Client>& client, 1098 audio_stream_type_t streamType, 1099 uint32_t sampleRate, 1100 audio_format_t format, 1101 audio_channel_mask_t channelMask, 1102 size_t frameCount, 1103 const sp<IMemory>& sharedBuffer, 1104 int sessionId, 1105 IAudioFlinger::track_flags_t *flags, 1106 pid_t tid, 1107 status_t *status); 1108 1109 AudioStreamOut* getOutput() const; 1110 AudioStreamOut* clearOutput(); 1111 virtual audio_stream_t* stream() const; 1112 1113 // a very large number of suspend() will eventually wraparound, but unlikely 1114 void suspend() { (void) android_atomic_inc(&mSuspended); } 1115 void restore() 1116 { 1117 // if restore() is done without suspend(), get back into 1118 // range so that the next suspend() will operate correctly 1119 if (android_atomic_dec(&mSuspended) <= 0) { 1120 android_atomic_release_store(0, &mSuspended); 1121 } 1122 } 1123 bool isSuspended() const 1124 { return android_atomic_acquire_load(&mSuspended) > 0; } 1125 1126 virtual String8 getParameters(const String8& keys); 1127 virtual void audioConfigChanged_l(int event, int param = 0); 1128 status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames); 1129 int16_t *mixBuffer() const { return mMixBuffer; }; 1130 1131 virtual void detachAuxEffect_l(int effectId); 1132 status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track> track, 1133 int EffectId); 1134 status_t attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track> track, 1135 int EffectId); 1136 1137 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1138 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1139 virtual uint32_t hasAudioSession(int sessionId) const; 1140 virtual uint32_t getStrategyForSession_l(int sessionId); 1141 1142 1143 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 1144 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 1145 void invalidateTracks(audio_stream_type_t streamType); 1146 1147 1148 protected: 1149 int16_t* mMixBuffer; 1150 1151 // suspend count, > 0 means suspended. While suspended, the thread continues to pull from 1152 // tracks and mix, but doesn't write to HAL. A2DP and SCO HAL implementations can't handle 1153 // concurrent use of both of them, so Audio Policy Service suspends one of the threads to 1154 // workaround that restriction. 1155 // 'volatile' means accessed via atomic operations and no lock. 1156 volatile int32_t mSuspended; 1157 1158 int mBytesWritten; 1159 private: 1160 // mMasterMute is in both PlaybackThread and in AudioFlinger. When a 1161 // PlaybackThread needs to find out if master-muted, it checks it's local 1162 // copy rather than the one in AudioFlinger. This optimization saves a lock. 1163 bool mMasterMute; 1164 void setMasterMute_l(bool muted) { mMasterMute = muted; } 1165 protected: 1166 SortedVector< wp<Track> > mActiveTracks; // FIXME check if this could be sp<> 1167 1168 // Allocate a track name for a given channel mask. 1169 // Returns name >= 0 if successful, -1 on failure. 1170 virtual int getTrackName_l(audio_channel_mask_t channelMask, int sessionId) = 0; 1171 virtual void deleteTrackName_l(int name) = 0; 1172 1173 // Time to sleep between cycles when: 1174 virtual uint32_t activeSleepTimeUs() const; // mixer state MIXER_TRACKS_ENABLED 1175 virtual uint32_t idleSleepTimeUs() const = 0; // mixer state MIXER_IDLE 1176 virtual uint32_t suspendSleepTimeUs() const = 0; // audio policy manager suspended us 1177 // No sleep when mixer state == MIXER_TRACKS_READY; relies on audio HAL stream->write() 1178 // No sleep in standby mode; waits on a condition 1179 1180 // Code snippets that are temporarily lifted up out of threadLoop() until the merge 1181 void checkSilentMode_l(); 1182 1183 // Non-trivial for DUPLICATING only 1184 virtual void saveOutputTracks() { } 1185 virtual void clearOutputTracks() { } 1186 1187 // Cache various calculated values, at threadLoop() entry and after a parameter change 1188 virtual void cacheParameters_l(); 1189 1190 virtual uint32_t correctLatency(uint32_t latency) const; 1191 1192 private: 1193 1194 friend class AudioFlinger; // for numerous 1195 1196 PlaybackThread(const Client&); 1197 PlaybackThread& operator = (const PlaybackThread&); 1198 1199 status_t addTrack_l(const sp<Track>& track); 1200 void destroyTrack_l(const sp<Track>& track); 1201 void removeTrack_l(const sp<Track>& track); 1202 1203 void readOutputParameters(); 1204 1205 virtual void dumpInternals(int fd, const Vector<String16>& args); 1206 void dumpTracks(int fd, const Vector<String16>& args); 1207 1208 SortedVector< sp<Track> > mTracks; 1209 // mStreamTypes[] uses 1 additional stream type internally for the OutputTrack used by 1210 // DuplicatingThread 1211 stream_type_t mStreamTypes[AUDIO_STREAM_CNT + 1]; 1212 AudioStreamOut *mOutput; 1213 1214 float mMasterVolume; 1215 nsecs_t mLastWriteTime; 1216 int mNumWrites; 1217 int mNumDelayedWrites; 1218 bool mInWrite; 1219 1220 // FIXME rename these former local variables of threadLoop to standard "m" names 1221 nsecs_t standbyTime; 1222 size_t mixBufferSize; 1223 1224 // cached copies of activeSleepTimeUs() and idleSleepTimeUs() made by cacheParameters_l() 1225 uint32_t activeSleepTime; 1226 uint32_t idleSleepTime; 1227 1228 uint32_t sleepTime; 1229 1230 // mixer status returned by prepareTracks_l() 1231 mixer_state mMixerStatus; // current cycle 1232 // previous cycle when in prepareTracks_l() 1233 mixer_state mMixerStatusIgnoringFastTracks; 1234 // FIXME or a separate ready state per track 1235 1236 // FIXME move these declarations into the specific sub-class that needs them 1237 // MIXER only 1238 uint32_t sleepTimeShift; 1239 1240 // same as AudioFlinger::mStandbyTimeInNsecs except for DIRECT which uses a shorter value 1241 nsecs_t standbyDelay; 1242 1243 // MIXER only 1244 nsecs_t maxPeriod; 1245 1246 // DUPLICATING only 1247 uint32_t writeFrames; 1248 1249 private: 1250 // The HAL output sink is treated as non-blocking, but current implementation is blocking 1251 sp<NBAIO_Sink> mOutputSink; 1252 // If a fast mixer is present, the blocking pipe sink, otherwise clear 1253 sp<NBAIO_Sink> mPipeSink; 1254 // The current sink for the normal mixer to write it's (sub)mix, mOutputSink or mPipeSink 1255 sp<NBAIO_Sink> mNormalSink; 1256 // For dumpsys 1257 sp<NBAIO_Sink> mTeeSink; 1258 sp<NBAIO_Source> mTeeSource; 1259 uint32_t mScreenState; // cached copy of gScreenState 1260 public: 1261 virtual bool hasFastMixer() const = 0; 1262 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const 1263 { FastTrackUnderruns dummy; return dummy; } 1264 1265 protected: 1266 // accessed by both binder threads and within threadLoop(), lock on mutex needed 1267 unsigned mFastTrackAvailMask; // bit i set if fast track [i] is available 1268 1269 }; 1270 1271 class MixerThread : public PlaybackThread { 1272 public: 1273 MixerThread(const sp<AudioFlinger>& audioFlinger, 1274 AudioStreamOut* output, 1275 audio_io_handle_t id, 1276 audio_devices_t device, 1277 type_t type = MIXER); 1278 virtual ~MixerThread(); 1279 1280 // Thread virtuals 1281 1282 virtual bool checkForNewParameters_l(); 1283 virtual void dumpInternals(int fd, const Vector<String16>& args); 1284 1285 protected: 1286 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1287 virtual int getTrackName_l(audio_channel_mask_t channelMask, int sessionId); 1288 virtual void deleteTrackName_l(int name); 1289 virtual uint32_t idleSleepTimeUs() const; 1290 virtual uint32_t suspendSleepTimeUs() const; 1291 virtual void cacheParameters_l(); 1292 1293 // threadLoop snippets 1294 virtual void threadLoop_write(); 1295 virtual void threadLoop_standby(); 1296 virtual void threadLoop_mix(); 1297 virtual void threadLoop_sleepTime(); 1298 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 1299 virtual uint32_t correctLatency(uint32_t latency) const; 1300 1301 AudioMixer* mAudioMixer; // normal mixer 1302 private: 1303 // one-time initialization, no locks required 1304 FastMixer* mFastMixer; // non-NULL if there is also a fast mixer 1305 sp<AudioWatchdog> mAudioWatchdog; // non-0 if there is an audio watchdog thread 1306 1307 // contents are not guaranteed to be consistent, no locks required 1308 FastMixerDumpState mFastMixerDumpState; 1309#ifdef STATE_QUEUE_DUMP 1310 StateQueueObserverDump mStateQueueObserverDump; 1311 StateQueueMutatorDump mStateQueueMutatorDump; 1312#endif 1313 AudioWatchdogDump mAudioWatchdogDump; 1314 1315 // accessible only within the threadLoop(), no locks required 1316 // mFastMixer->sq() // for mutating and pushing state 1317 int32_t mFastMixerFutex; // for cold idle 1318 1319 public: 1320 virtual bool hasFastMixer() const { return mFastMixer != NULL; } 1321 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const { 1322 ALOG_ASSERT(fastIndex < FastMixerState::kMaxFastTracks); 1323 return mFastMixerDumpState.mTracks[fastIndex].mUnderruns; 1324 } 1325 }; 1326 1327 class DirectOutputThread : public PlaybackThread { 1328 public: 1329 1330 DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1331 audio_io_handle_t id, audio_devices_t device); 1332 virtual ~DirectOutputThread(); 1333 1334 // Thread virtuals 1335 1336 virtual bool checkForNewParameters_l(); 1337 1338 protected: 1339 virtual int getTrackName_l(audio_channel_mask_t channelMask, int sessionId); 1340 virtual void deleteTrackName_l(int name); 1341 virtual uint32_t activeSleepTimeUs() const; 1342 virtual uint32_t idleSleepTimeUs() const; 1343 virtual uint32_t suspendSleepTimeUs() const; 1344 virtual void cacheParameters_l(); 1345 1346 // threadLoop snippets 1347 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1348 virtual void threadLoop_mix(); 1349 virtual void threadLoop_sleepTime(); 1350 1351 private: 1352 // volumes last sent to audio HAL with stream->set_volume() 1353 float mLeftVolFloat; 1354 float mRightVolFloat; 1355 1356 // prepareTracks_l() tells threadLoop_mix() the name of the single active track 1357 sp<Track> mActiveTrack; 1358 public: 1359 virtual bool hasFastMixer() const { return false; } 1360 }; 1361 1362 class DuplicatingThread : public MixerThread { 1363 public: 1364 DuplicatingThread(const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread, 1365 audio_io_handle_t id); 1366 virtual ~DuplicatingThread(); 1367 1368 // Thread virtuals 1369 void addOutputTrack(MixerThread* thread); 1370 void removeOutputTrack(MixerThread* thread); 1371 uint32_t waitTimeMs() const { return mWaitTimeMs; } 1372 protected: 1373 virtual uint32_t activeSleepTimeUs() const; 1374 1375 private: 1376 bool outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks); 1377 protected: 1378 // threadLoop snippets 1379 virtual void threadLoop_mix(); 1380 virtual void threadLoop_sleepTime(); 1381 virtual void threadLoop_write(); 1382 virtual void threadLoop_standby(); 1383 virtual void cacheParameters_l(); 1384 1385 private: 1386 // called from threadLoop, addOutputTrack, removeOutputTrack 1387 virtual void updateWaitTime_l(); 1388 protected: 1389 virtual void saveOutputTracks(); 1390 virtual void clearOutputTracks(); 1391 private: 1392 1393 uint32_t mWaitTimeMs; 1394 SortedVector < sp<OutputTrack> > outputTracks; 1395 SortedVector < sp<OutputTrack> > mOutputTracks; 1396 public: 1397 virtual bool hasFastMixer() const { return false; } 1398 }; 1399 1400 PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const; 1401 MixerThread *checkMixerThread_l(audio_io_handle_t output) const; 1402 RecordThread *checkRecordThread_l(audio_io_handle_t input) const; 1403 // no range check, AudioFlinger::mLock held 1404 bool streamMute_l(audio_stream_type_t stream) const 1405 { return mStreamTypes[stream].mute; } 1406 // no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held 1407 float streamVolume_l(audio_stream_type_t stream) const 1408 { return mStreamTypes[stream].volume; } 1409 void audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2); 1410 1411 // allocate an audio_io_handle_t, session ID, or effect ID 1412 uint32_t nextUniqueId(); 1413 1414 status_t moveEffectChain_l(int sessionId, 1415 PlaybackThread *srcThread, 1416 PlaybackThread *dstThread, 1417 bool reRegister); 1418 // return thread associated with primary hardware device, or NULL 1419 PlaybackThread *primaryPlaybackThread_l() const; 1420 audio_devices_t primaryOutputDevice_l() const; 1421 1422 sp<PlaybackThread> getEffectThread_l(int sessionId, int EffectId); 1423 1424 // server side of the client's IAudioTrack 1425 class TrackHandle : public android::BnAudioTrack { 1426 public: 1427 TrackHandle(const sp<PlaybackThread::Track>& track); 1428 virtual ~TrackHandle(); 1429 virtual sp<IMemory> getCblk() const; 1430 virtual status_t start(); 1431 virtual void stop(); 1432 virtual void flush(); 1433 virtual void mute(bool); 1434 virtual void pause(); 1435 virtual status_t attachAuxEffect(int effectId); 1436 virtual status_t allocateTimedBuffer(size_t size, 1437 sp<IMemory>* buffer); 1438 virtual status_t queueTimedBuffer(const sp<IMemory>& buffer, 1439 int64_t pts); 1440 virtual status_t setMediaTimeTransform(const LinearTransform& xform, 1441 int target); 1442 virtual status_t onTransact( 1443 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 1444 private: 1445 const sp<PlaybackThread::Track> mTrack; 1446 }; 1447 1448 void removeClient_l(pid_t pid); 1449 void removeNotificationClient(pid_t pid); 1450 1451 1452 // record thread 1453 class RecordThread : public ThreadBase, public AudioBufferProvider 1454 // derives from AudioBufferProvider interface for use by resampler 1455 { 1456 public: 1457 1458 // record track 1459 class RecordTrack : public TrackBase { 1460 public: 1461 RecordTrack(RecordThread *thread, 1462 const sp<Client>& client, 1463 uint32_t sampleRate, 1464 audio_format_t format, 1465 audio_channel_mask_t channelMask, 1466 size_t frameCount, 1467 int sessionId); 1468 virtual ~RecordTrack(); 1469 1470 virtual status_t start(AudioSystem::sync_event_t event, int triggerSession); 1471 virtual void stop(); 1472 1473 void destroy(); 1474 1475 // clear the buffer overflow flag 1476 void clearOverflow() { mOverflow = false; } 1477 // set the buffer overflow flag and return previous value 1478 bool setOverflow() { bool tmp = mOverflow; mOverflow = true; 1479 return tmp; } 1480 1481 static void appendDumpHeader(String8& result); 1482 void dump(char* buffer, size_t size); 1483 1484 virtual bool isOut() const; 1485 1486 private: 1487 friend class AudioFlinger; // for mState 1488 1489 RecordTrack(const RecordTrack&); 1490 RecordTrack& operator = (const RecordTrack&); 1491 1492 // AudioBufferProvider interface 1493 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, 1494 int64_t pts = kInvalidPTS); 1495 // releaseBuffer() not overridden 1496 1497 bool mOverflow; // overflow on most recent attempt to fill client buffer 1498 }; 1499 1500 RecordThread(const sp<AudioFlinger>& audioFlinger, 1501 AudioStreamIn *input, 1502 uint32_t sampleRate, 1503 audio_channel_mask_t channelMask, 1504 audio_io_handle_t id, 1505 audio_devices_t device, 1506 const sp<NBAIO_Sink>& teeSink); 1507 virtual ~RecordThread(); 1508 1509 // no addTrack_l ? 1510 void destroyTrack_l(const sp<RecordTrack>& track); 1511 void removeTrack_l(const sp<RecordTrack>& track); 1512 1513 void dumpInternals(int fd, const Vector<String16>& args); 1514 void dumpTracks(int fd, const Vector<String16>& args); 1515 1516 // Thread virtuals 1517 virtual bool threadLoop(); 1518 virtual status_t readyToRun(); 1519 1520 // RefBase 1521 virtual void onFirstRef(); 1522 1523 virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; } 1524 sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l( 1525 const sp<AudioFlinger::Client>& client, 1526 uint32_t sampleRate, 1527 audio_format_t format, 1528 audio_channel_mask_t channelMask, 1529 size_t frameCount, 1530 int sessionId, 1531 IAudioFlinger::track_flags_t flags, 1532 pid_t tid, 1533 status_t *status); 1534 1535 status_t start(RecordTrack* recordTrack, 1536 AudioSystem::sync_event_t event, 1537 int triggerSession); 1538 1539 // ask the thread to stop the specified track, and 1540 // return true if the caller should then do it's part of the stopping process 1541 bool stop_l(RecordTrack* recordTrack); 1542 1543 void dump(int fd, const Vector<String16>& args); 1544 AudioStreamIn* clearInput(); 1545 virtual audio_stream_t* stream() const; 1546 1547 // AudioBufferProvider interface 1548 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts); 1549 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 1550 1551 virtual bool checkForNewParameters_l(); 1552 virtual String8 getParameters(const String8& keys); 1553 virtual void audioConfigChanged_l(int event, int param = 0); 1554 void readInputParameters(); 1555 virtual unsigned int getInputFramesLost(); 1556 1557 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1558 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1559 virtual uint32_t hasAudioSession(int sessionId) const; 1560 1561 // Return the set of unique session IDs across all tracks. 1562 // The keys are the session IDs, and the associated values are meaningless. 1563 // FIXME replace by Set [and implement Bag/Multiset for other uses]. 1564 KeyedVector<int, bool> sessionIds() const; 1565 1566 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 1567 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 1568 1569 static void syncStartEventCallback(const wp<SyncEvent>& event); 1570 void handleSyncStartEvent(const sp<SyncEvent>& event); 1571 1572 private: 1573 void clearSyncStartEvent(); 1574 1575 // Enter standby if not already in standby, and set mStandby flag 1576 void standby(); 1577 1578 // Call the HAL standby method unconditionally, and don't change mStandby flag 1579 void inputStandBy(); 1580 1581 AudioStreamIn *mInput; 1582 SortedVector < sp<RecordTrack> > mTracks; 1583 // mActiveTrack has dual roles: it indicates the current active track, and 1584 // is used together with mStartStopCond to indicate start()/stop() progress 1585 sp<RecordTrack> mActiveTrack; 1586 Condition mStartStopCond; 1587 AudioResampler *mResampler; 1588 int32_t *mRsmpOutBuffer; 1589 int16_t *mRsmpInBuffer; 1590 size_t mRsmpInIndex; 1591 size_t mInputBytes; 1592 const int mReqChannelCount; 1593 const uint32_t mReqSampleRate; 1594 ssize_t mBytesRead; 1595 // sync event triggering actual audio capture. Frames read before this event will 1596 // be dropped and therefore not read by the application. 1597 sp<SyncEvent> mSyncStartEvent; 1598 // number of captured frames to drop after the start sync event has been received. 1599 // when < 0, maximum frames to drop before starting capture even if sync event is 1600 // not received 1601 ssize_t mFramestoDrop; 1602 1603 // For dumpsys 1604 const sp<NBAIO_Sink> mTeeSink; 1605 }; 1606 1607 // server side of the client's IAudioRecord 1608 class RecordHandle : public android::BnAudioRecord { 1609 public: 1610 RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack); 1611 virtual ~RecordHandle(); 1612 virtual sp<IMemory> getCblk() const; 1613 virtual status_t start(int /*AudioSystem::sync_event_t*/ event, int triggerSession); 1614 virtual void stop(); 1615 virtual status_t onTransact( 1616 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 1617 private: 1618 const sp<RecordThread::RecordTrack> mRecordTrack; 1619 1620 // for use from destructor 1621 void stop_nonvirtual(); 1622 }; 1623 1624 //--- Audio Effect Management 1625 1626 // EffectModule and EffectChain classes both have their own mutex to protect 1627 // state changes or resource modifications. Always respect the following order 1628 // if multiple mutexes must be acquired to avoid cross deadlock: 1629 // AudioFlinger -> ThreadBase -> EffectChain -> EffectModule 1630 1631 // The EffectModule class is a wrapper object controlling the effect engine implementation 1632 // in the effect library. It prevents concurrent calls to process() and command() functions 1633 // from different client threads. It keeps a list of EffectHandle objects corresponding 1634 // to all client applications using this effect and notifies applications of effect state, 1635 // control or parameter changes. It manages the activation state machine to send appropriate 1636 // reset, enable, disable commands to effect engine and provide volume 1637 // ramping when effects are activated/deactivated. 1638 // When controlling an auxiliary effect, the EffectModule also provides an input buffer used by 1639 // the attached track(s) to accumulate their auxiliary channel. 1640 class EffectModule : public RefBase { 1641 public: 1642 EffectModule(ThreadBase *thread, 1643 const wp<AudioFlinger::EffectChain>& chain, 1644 effect_descriptor_t *desc, 1645 int id, 1646 int sessionId); 1647 virtual ~EffectModule(); 1648 1649 enum effect_state { 1650 IDLE, 1651 RESTART, 1652 STARTING, 1653 ACTIVE, 1654 STOPPING, 1655 STOPPED, 1656 DESTROYED 1657 }; 1658 1659 int id() const { return mId; } 1660 void process(); 1661 void updateState(); 1662 status_t command(uint32_t cmdCode, 1663 uint32_t cmdSize, 1664 void *pCmdData, 1665 uint32_t *replySize, 1666 void *pReplyData); 1667 1668 void reset_l(); 1669 status_t configure(); 1670 status_t init(); 1671 effect_state state() const { 1672 return mState; 1673 } 1674 uint32_t status() { 1675 return mStatus; 1676 } 1677 int sessionId() const { 1678 return mSessionId; 1679 } 1680 status_t setEnabled(bool enabled); 1681 status_t setEnabled_l(bool enabled); 1682 bool isEnabled() const; 1683 bool isProcessEnabled() const; 1684 1685 void setInBuffer(int16_t *buffer) { mConfig.inputCfg.buffer.s16 = buffer; } 1686 int16_t *inBuffer() { return mConfig.inputCfg.buffer.s16; } 1687 void setOutBuffer(int16_t *buffer) { mConfig.outputCfg.buffer.s16 = buffer; } 1688 int16_t *outBuffer() { return mConfig.outputCfg.buffer.s16; } 1689 void setChain(const wp<EffectChain>& chain) { mChain = chain; } 1690 void setThread(const wp<ThreadBase>& thread) { mThread = thread; } 1691 const wp<ThreadBase>& thread() { return mThread; } 1692 1693 status_t addHandle(EffectHandle *handle); 1694 size_t disconnect(EffectHandle *handle, bool unpinIfLast); 1695 size_t removeHandle(EffectHandle *handle); 1696 1697 const effect_descriptor_t& desc() const { return mDescriptor; } 1698 wp<EffectChain>& chain() { return mChain; } 1699 1700 status_t setDevice(audio_devices_t device); 1701 status_t setVolume(uint32_t *left, uint32_t *right, bool controller); 1702 status_t setMode(audio_mode_t mode); 1703 status_t setAudioSource(audio_source_t source); 1704 status_t start(); 1705 status_t stop(); 1706 void setSuspended(bool suspended); 1707 bool suspended() const; 1708 1709 EffectHandle* controlHandle_l(); 1710 1711 bool isPinned() const { return mPinned; } 1712 void unPin() { mPinned = false; } 1713 bool purgeHandles(); 1714 void lock() { mLock.lock(); } 1715 void unlock() { mLock.unlock(); } 1716 1717 void dump(int fd, const Vector<String16>& args); 1718 1719 protected: 1720 friend class AudioFlinger; // for mHandles 1721 bool mPinned; 1722 1723 // Maximum time allocated to effect engines to complete the turn off sequence 1724 static const uint32_t MAX_DISABLE_TIME_MS = 10000; 1725 1726 EffectModule(const EffectModule&); 1727 EffectModule& operator = (const EffectModule&); 1728 1729 status_t start_l(); 1730 status_t stop_l(); 1731 1732mutable Mutex mLock; // mutex for process, commands and handles list protection 1733 wp<ThreadBase> mThread; // parent thread 1734 wp<EffectChain> mChain; // parent effect chain 1735 const int mId; // this instance unique ID 1736 const int mSessionId; // audio session ID 1737 const effect_descriptor_t mDescriptor;// effect descriptor received from effect engine 1738 effect_config_t mConfig; // input and output audio configuration 1739 effect_handle_t mEffectInterface; // Effect module C API 1740 status_t mStatus; // initialization status 1741 effect_state mState; // current activation state 1742 Vector<EffectHandle *> mHandles; // list of client handles 1743 // First handle in mHandles has highest priority and controls the effect module 1744 uint32_t mMaxDisableWaitCnt; // maximum grace period before forcing an effect off after 1745 // sending disable command. 1746 uint32_t mDisableWaitCnt; // current process() calls count during disable period. 1747 bool mSuspended; // effect is suspended: temporarily disabled by framework 1748 }; 1749 1750 // The EffectHandle class implements the IEffect interface. It provides resources 1751 // to receive parameter updates, keeps track of effect control 1752 // ownership and state and has a pointer to the EffectModule object it is controlling. 1753 // There is one EffectHandle object for each application controlling (or using) 1754 // an effect module. 1755 // The EffectHandle is obtained by calling AudioFlinger::createEffect(). 1756 class EffectHandle: public android::BnEffect { 1757 public: 1758 1759 EffectHandle(const sp<EffectModule>& effect, 1760 const sp<AudioFlinger::Client>& client, 1761 const sp<IEffectClient>& effectClient, 1762 int32_t priority); 1763 virtual ~EffectHandle(); 1764 1765 // IEffect 1766 virtual status_t enable(); 1767 virtual status_t disable(); 1768 virtual status_t command(uint32_t cmdCode, 1769 uint32_t cmdSize, 1770 void *pCmdData, 1771 uint32_t *replySize, 1772 void *pReplyData); 1773 virtual void disconnect(); 1774 private: 1775 void disconnect(bool unpinIfLast); 1776 public: 1777 virtual sp<IMemory> getCblk() const { return mCblkMemory; } 1778 virtual status_t onTransact(uint32_t code, const Parcel& data, 1779 Parcel* reply, uint32_t flags); 1780 1781 1782 // Give or take control of effect module 1783 // - hasControl: true if control is given, false if removed 1784 // - signal: true client app should be signaled of change, false otherwise 1785 // - enabled: state of the effect when control is passed 1786 void setControl(bool hasControl, bool signal, bool enabled); 1787 void commandExecuted(uint32_t cmdCode, 1788 uint32_t cmdSize, 1789 void *pCmdData, 1790 uint32_t replySize, 1791 void *pReplyData); 1792 void setEnabled(bool enabled); 1793 bool enabled() const { return mEnabled; } 1794 1795 // Getters 1796 int id() const { return mEffect->id(); } 1797 int priority() const { return mPriority; } 1798 bool hasControl() const { return mHasControl; } 1799 sp<EffectModule> effect() const { return mEffect; } 1800 // destroyed_l() must be called with the associated EffectModule mLock held 1801 bool destroyed_l() const { return mDestroyed; } 1802 1803 void dump(char* buffer, size_t size); 1804 1805 protected: 1806 friend class AudioFlinger; // for mEffect, mHasControl, mEnabled 1807 EffectHandle(const EffectHandle&); 1808 EffectHandle& operator =(const EffectHandle&); 1809 1810 sp<EffectModule> mEffect; // pointer to controlled EffectModule 1811 sp<IEffectClient> mEffectClient; // callback interface for client notifications 1812 /*const*/ sp<Client> mClient; // client for shared memory allocation, see disconnect() 1813 sp<IMemory> mCblkMemory; // shared memory for control block 1814 effect_param_cblk_t* mCblk; // control block for deferred parameter setting via 1815 // shared memory 1816 uint8_t* mBuffer; // pointer to parameter area in shared memory 1817 int mPriority; // client application priority to control the effect 1818 bool mHasControl; // true if this handle is controlling the effect 1819 bool mEnabled; // cached enable state: needed when the effect is 1820 // restored after being suspended 1821 bool mDestroyed; // Set to true by destructor. Access with EffectModule 1822 // mLock held 1823 }; 1824 1825 // the EffectChain class represents a group of effects associated to one audio session. 1826 // There can be any number of EffectChain objects per output mixer thread (PlaybackThread). 1827 // The EffecChain with session ID 0 contains global effects applied to the output mix. 1828 // Effects in this chain can be insert or auxiliary. Effects in other chains (attached to 1829 // tracks) are insert only. The EffectChain maintains an ordered list of effect module, the 1830 // order corresponding in the effect process order. When attached to a track (session ID != 0), 1831 // it also provide it's own input buffer used by the track as accumulation buffer. 1832 class EffectChain : public RefBase { 1833 public: 1834 EffectChain(const wp<ThreadBase>& wThread, int sessionId); 1835 EffectChain(ThreadBase *thread, int sessionId); 1836 virtual ~EffectChain(); 1837 1838 // special key used for an entry in mSuspendedEffects keyed vector 1839 // corresponding to a suspend all request. 1840 static const int kKeyForSuspendAll = 0; 1841 1842 // minimum duration during which we force calling effect process when last track on 1843 // a session is stopped or removed to allow effect tail to be rendered 1844 static const int kProcessTailDurationMs = 1000; 1845 1846 void process_l(); 1847 1848 void lock() { 1849 mLock.lock(); 1850 } 1851 void unlock() { 1852 mLock.unlock(); 1853 } 1854 1855 status_t addEffect_l(const sp<EffectModule>& handle); 1856 size_t removeEffect_l(const sp<EffectModule>& handle); 1857 1858 int sessionId() const { return mSessionId; } 1859 void setSessionId(int sessionId) { mSessionId = sessionId; } 1860 1861 sp<EffectModule> getEffectFromDesc_l(effect_descriptor_t *descriptor); 1862 sp<EffectModule> getEffectFromId_l(int id); 1863 sp<EffectModule> getEffectFromType_l(const effect_uuid_t *type); 1864 bool setVolume_l(uint32_t *left, uint32_t *right); 1865 void setDevice_l(audio_devices_t device); 1866 void setMode_l(audio_mode_t mode); 1867 void setAudioSource_l(audio_source_t source); 1868 1869 void setInBuffer(int16_t *buffer, bool ownsBuffer = false) { 1870 mInBuffer = buffer; 1871 mOwnInBuffer = ownsBuffer; 1872 } 1873 int16_t *inBuffer() const { 1874 return mInBuffer; 1875 } 1876 void setOutBuffer(int16_t *buffer) { 1877 mOutBuffer = buffer; 1878 } 1879 int16_t *outBuffer() const { 1880 return mOutBuffer; 1881 } 1882 1883 void incTrackCnt() { android_atomic_inc(&mTrackCnt); } 1884 void decTrackCnt() { android_atomic_dec(&mTrackCnt); } 1885 int32_t trackCnt() const { return android_atomic_acquire_load(&mTrackCnt); } 1886 1887 void incActiveTrackCnt() { android_atomic_inc(&mActiveTrackCnt); 1888 mTailBufferCount = mMaxTailBuffers; } 1889 void decActiveTrackCnt() { android_atomic_dec(&mActiveTrackCnt); } 1890 int32_t activeTrackCnt() const { return android_atomic_acquire_load(&mActiveTrackCnt); } 1891 1892 uint32_t strategy() const { return mStrategy; } 1893 void setStrategy(uint32_t strategy) 1894 { mStrategy = strategy; } 1895 1896 // suspend effect of the given type 1897 void setEffectSuspended_l(const effect_uuid_t *type, 1898 bool suspend); 1899 // suspend all eligible effects 1900 void setEffectSuspendedAll_l(bool suspend); 1901 // check if effects should be suspend or restored when a given effect is enable or disabled 1902 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1903 bool enabled); 1904 1905 void clearInputBuffer(); 1906 1907 void dump(int fd, const Vector<String16>& args); 1908 1909 protected: 1910 friend class AudioFlinger; // for mThread, mEffects 1911 EffectChain(const EffectChain&); 1912 EffectChain& operator =(const EffectChain&); 1913 1914 class SuspendedEffectDesc : public RefBase { 1915 public: 1916 SuspendedEffectDesc() : mRefCount(0) {} 1917 1918 int mRefCount; 1919 effect_uuid_t mType; 1920 wp<EffectModule> mEffect; 1921 }; 1922 1923 // get a list of effect modules to suspend when an effect of the type 1924 // passed is enabled. 1925 void getSuspendEligibleEffects(Vector< sp<EffectModule> > &effects); 1926 1927 // get an effect module if it is currently enable 1928 sp<EffectModule> getEffectIfEnabled(const effect_uuid_t *type); 1929 // true if the effect whose descriptor is passed can be suspended 1930 // OEMs can modify the rules implemented in this method to exclude specific effect 1931 // types or implementations from the suspend/restore mechanism. 1932 bool isEffectEligibleForSuspend(const effect_descriptor_t& desc); 1933 1934 void clearInputBuffer_l(sp<ThreadBase> thread); 1935 1936 wp<ThreadBase> mThread; // parent mixer thread 1937 Mutex mLock; // mutex protecting effect list 1938 Vector< sp<EffectModule> > mEffects; // list of effect modules 1939 int mSessionId; // audio session ID 1940 int16_t *mInBuffer; // chain input buffer 1941 int16_t *mOutBuffer; // chain output buffer 1942 1943 // 'volatile' here means these are accessed with atomic operations instead of mutex 1944 volatile int32_t mActiveTrackCnt; // number of active tracks connected 1945 volatile int32_t mTrackCnt; // number of tracks connected 1946 1947 int32_t mTailBufferCount; // current effect tail buffer count 1948 int32_t mMaxTailBuffers; // maximum effect tail buffers 1949 bool mOwnInBuffer; // true if the chain owns its input buffer 1950 int mVolumeCtrlIdx; // index of insert effect having control over volume 1951 uint32_t mLeftVolume; // previous volume on left channel 1952 uint32_t mRightVolume; // previous volume on right channel 1953 uint32_t mNewLeftVolume; // new volume on left channel 1954 uint32_t mNewRightVolume; // new volume on right channel 1955 uint32_t mStrategy; // strategy for this effect chain 1956 // mSuspendedEffects lists all effects currently suspended in the chain. 1957 // Use effect type UUID timelow field as key. There is no real risk of identical 1958 // timeLow fields among effect type UUIDs. 1959 // Updated by updateSuspendedSessions_l() only. 1960 KeyedVector< int, sp<SuspendedEffectDesc> > mSuspendedEffects; 1961 }; 1962 1963 class AudioHwDevice { 1964 public: 1965 enum Flags { 1966 AHWD_CAN_SET_MASTER_VOLUME = 0x1, 1967 AHWD_CAN_SET_MASTER_MUTE = 0x2, 1968 }; 1969 1970 AudioHwDevice(const char *moduleName, 1971 audio_hw_device_t *hwDevice, 1972 Flags flags) 1973 : mModuleName(strdup(moduleName)) 1974 , mHwDevice(hwDevice) 1975 , mFlags(flags) { } 1976 /*virtual*/ ~AudioHwDevice() { free((void *)mModuleName); } 1977 1978 bool canSetMasterVolume() const { 1979 return (0 != (mFlags & AHWD_CAN_SET_MASTER_VOLUME)); 1980 } 1981 1982 bool canSetMasterMute() const { 1983 return (0 != (mFlags & AHWD_CAN_SET_MASTER_MUTE)); 1984 } 1985 1986 const char *moduleName() const { return mModuleName; } 1987 audio_hw_device_t *hwDevice() const { return mHwDevice; } 1988 private: 1989 const char * const mModuleName; 1990 audio_hw_device_t * const mHwDevice; 1991 Flags mFlags; 1992 }; 1993 1994 // AudioStreamOut and AudioStreamIn are immutable, so their fields are const. 1995 // For emphasis, we could also make all pointers to them be "const *", 1996 // but that would clutter the code unnecessarily. 1997 1998 struct AudioStreamOut { 1999 AudioHwDevice* const audioHwDev; 2000 audio_stream_out_t* const stream; 2001 2002 audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); } 2003 2004 AudioStreamOut(AudioHwDevice *dev, audio_stream_out_t *out) : 2005 audioHwDev(dev), stream(out) {} 2006 }; 2007 2008 struct AudioStreamIn { 2009 AudioHwDevice* const audioHwDev; 2010 audio_stream_in_t* const stream; 2011 2012 audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); } 2013 2014 AudioStreamIn(AudioHwDevice *dev, audio_stream_in_t *in) : 2015 audioHwDev(dev), stream(in) {} 2016 }; 2017 2018 // for mAudioSessionRefs only 2019 struct AudioSessionRef { 2020 AudioSessionRef(int sessionid, pid_t pid) : 2021 mSessionid(sessionid), mPid(pid), mCnt(1) {} 2022 const int mSessionid; 2023 const pid_t mPid; 2024 int mCnt; 2025 }; 2026 2027 mutable Mutex mLock; 2028 2029 DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client() 2030 2031 mutable Mutex mHardwareLock; 2032 // NOTE: If both mLock and mHardwareLock mutexes must be held, 2033 // always take mLock before mHardwareLock 2034 2035 // These two fields are immutable after onFirstRef(), so no lock needed to access 2036 AudioHwDevice* mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL 2037 DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*> mAudioHwDevs; 2038 2039 // for dump, indicates which hardware operation is currently in progress (but not stream ops) 2040 enum hardware_call_state { 2041 AUDIO_HW_IDLE = 0, // no operation in progress 2042 AUDIO_HW_INIT, // init_check 2043 AUDIO_HW_OUTPUT_OPEN, // open_output_stream 2044 AUDIO_HW_OUTPUT_CLOSE, // unused 2045 AUDIO_HW_INPUT_OPEN, // unused 2046 AUDIO_HW_INPUT_CLOSE, // unused 2047 AUDIO_HW_STANDBY, // unused 2048 AUDIO_HW_SET_MASTER_VOLUME, // set_master_volume 2049 AUDIO_HW_GET_ROUTING, // unused 2050 AUDIO_HW_SET_ROUTING, // unused 2051 AUDIO_HW_GET_MODE, // unused 2052 AUDIO_HW_SET_MODE, // set_mode 2053 AUDIO_HW_GET_MIC_MUTE, // get_mic_mute 2054 AUDIO_HW_SET_MIC_MUTE, // set_mic_mute 2055 AUDIO_HW_SET_VOICE_VOLUME, // set_voice_volume 2056 AUDIO_HW_SET_PARAMETER, // set_parameters 2057 AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size 2058 AUDIO_HW_GET_MASTER_VOLUME, // get_master_volume 2059 AUDIO_HW_GET_PARAMETER, // get_parameters 2060 AUDIO_HW_SET_MASTER_MUTE, // set_master_mute 2061 AUDIO_HW_GET_MASTER_MUTE, // get_master_mute 2062 }; 2063 2064 mutable hardware_call_state mHardwareStatus; // for dump only 2065 2066 2067 DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads; 2068 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 2069 2070 // member variables below are protected by mLock 2071 float mMasterVolume; 2072 bool mMasterMute; 2073 // end of variables protected by mLock 2074 2075 DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads; 2076 2077 DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients; 2078 volatile int32_t mNextUniqueId; // updated by android_atomic_inc 2079 audio_mode_t mMode; 2080 bool mBtNrecIsOff; 2081 2082 // protected by mLock 2083 Vector<AudioSessionRef*> mAudioSessionRefs; 2084 2085 float masterVolume_l() const; 2086 bool masterMute_l() const; 2087 audio_module_handle_t loadHwModule_l(const char *name); 2088 2089 Vector < sp<SyncEvent> > mPendingSyncEvents; // sync events awaiting for a session 2090 // to be created 2091 2092private: 2093 sp<Client> registerPid_l(pid_t pid); // always returns non-0 2094 2095 // for use from destructor 2096 status_t closeOutput_nonvirtual(audio_io_handle_t output); 2097 status_t closeInput_nonvirtual(audio_io_handle_t input); 2098 2099 // all record threads serially share a common tee sink, which is re-created on format change 2100 sp<NBAIO_Sink> mRecordTeeSink; 2101 sp<NBAIO_Source> mRecordTeeSource; 2102 2103public: 2104 static void dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id = 0); 2105}; 2106 2107 2108// ---------------------------------------------------------------------------- 2109 2110}; // namespace android 2111 2112#endif // ANDROID_AUDIO_FLINGER_H 2113