AudioFlinger.h revision 3b16c766d1ae2cfd8487e8ffb2b23936fc0a8e17
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 uint32_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 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 size_t mFrameSize; // AudioFlinger's view of frame size in shared memory, 477 // where for AudioTrack (but not AudioRecord), 478 // 8-bit PCM samples are stored as 16-bit 479 // FIXME should be const 480 bool mStepServerFailed; 481 const int mSessionId; 482 uint8_t mChannelCount; 483 audio_channel_mask_t mChannelMask; 484 Vector < sp<SyncEvent> >mSyncEvents; 485 }; 486 487 enum { 488 CFG_EVENT_IO, 489 CFG_EVENT_PRIO 490 }; 491 492 class ConfigEvent { 493 public: 494 ConfigEvent(int type) : mType(type) {} 495 virtual ~ConfigEvent() {} 496 497 int type() const { return mType; } 498 499 virtual void dump(char *buffer, size_t size) = 0; 500 501 private: 502 const int mType; 503 }; 504 505 class IoConfigEvent : public ConfigEvent { 506 public: 507 IoConfigEvent(int event, int param) : 508 ConfigEvent(CFG_EVENT_IO), mEvent(event), mParam(event) {} 509 virtual ~IoConfigEvent() {} 510 511 int event() const { return mEvent; } 512 int param() const { return mParam; } 513 514 virtual void dump(char *buffer, size_t size) { 515 snprintf(buffer, size, "IO event: event %d, param %d\n", mEvent, mParam); 516 } 517 518 private: 519 const int mEvent; 520 const int mParam; 521 }; 522 523 class PrioConfigEvent : public ConfigEvent { 524 public: 525 PrioConfigEvent(pid_t pid, pid_t tid, int32_t prio) : 526 ConfigEvent(CFG_EVENT_PRIO), mPid(pid), mTid(tid), mPrio(prio) {} 527 virtual ~PrioConfigEvent() {} 528 529 pid_t pid() const { return mPid; } 530 pid_t tid() const { return mTid; } 531 int32_t prio() const { return mPrio; } 532 533 virtual void dump(char *buffer, size_t size) { 534 snprintf(buffer, size, "Prio event: pid %d, tid %d, prio %d\n", mPid, mTid, mPrio); 535 } 536 537 private: 538 const pid_t mPid; 539 const pid_t mTid; 540 const int32_t mPrio; 541 }; 542 543 544 class PMDeathRecipient : public IBinder::DeathRecipient { 545 public: 546 PMDeathRecipient(const wp<ThreadBase>& thread) : mThread(thread) {} 547 virtual ~PMDeathRecipient() {} 548 549 // IBinder::DeathRecipient 550 virtual void binderDied(const wp<IBinder>& who); 551 552 private: 553 PMDeathRecipient(const PMDeathRecipient&); 554 PMDeathRecipient& operator = (const PMDeathRecipient&); 555 556 wp<ThreadBase> mThread; 557 }; 558 559 virtual status_t initCheck() const = 0; 560 561 // static externally-visible 562 type_t type() const { return mType; } 563 audio_io_handle_t id() const { return mId;} 564 565 // dynamic externally-visible 566 uint32_t sampleRate() const { return mSampleRate; } 567 int channelCount() const { return mChannelCount; } 568 audio_channel_mask_t channelMask() const { return mChannelMask; } 569 audio_format_t format() const { return mFormat; } 570 // Called by AudioFlinger::frameCount(audio_io_handle_t output) and effects, 571 // and returns the normal mix buffer's frame count. 572 size_t frameCount() const { return mNormalFrameCount; } 573 // Return's the HAL's frame count i.e. fast mixer buffer size. 574 size_t frameCountHAL() const { return mFrameCount; } 575 576 // Should be "virtual status_t requestExitAndWait()" and override same 577 // method in Thread, but Thread::requestExitAndWait() is not yet virtual. 578 void exit(); 579 virtual bool checkForNewParameters_l() = 0; 580 virtual status_t setParameters(const String8& keyValuePairs); 581 virtual String8 getParameters(const String8& keys) = 0; 582 virtual void audioConfigChanged_l(int event, int param = 0) = 0; 583 void sendIoConfigEvent(int event, int param = 0); 584 void sendIoConfigEvent_l(int event, int param = 0); 585 void sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio); 586 void processConfigEvents(); 587 588 // see note at declaration of mStandby, mOutDevice and mInDevice 589 bool standby() const { return mStandby; } 590 audio_devices_t outDevice() const { return mOutDevice; } 591 audio_devices_t inDevice() const { return mInDevice; } 592 593 virtual audio_stream_t* stream() const = 0; 594 595 sp<EffectHandle> createEffect_l( 596 const sp<AudioFlinger::Client>& client, 597 const sp<IEffectClient>& effectClient, 598 int32_t priority, 599 int sessionId, 600 effect_descriptor_t *desc, 601 int *enabled, 602 status_t *status); 603 void disconnectEffect(const sp< EffectModule>& effect, 604 EffectHandle *handle, 605 bool unpinIfLast); 606 607 // return values for hasAudioSession (bit field) 608 enum effect_state { 609 EFFECT_SESSION = 0x1, // the audio session corresponds to at least one 610 // effect 611 TRACK_SESSION = 0x2 // the audio session corresponds to at least one 612 // track 613 }; 614 615 // get effect chain corresponding to session Id. 616 sp<EffectChain> getEffectChain(int sessionId); 617 // same as getEffectChain() but must be called with ThreadBase mutex locked 618 sp<EffectChain> getEffectChain_l(int sessionId) const; 619 // add an effect chain to the chain list (mEffectChains) 620 virtual status_t addEffectChain_l(const sp<EffectChain>& chain) = 0; 621 // remove an effect chain from the chain list (mEffectChains) 622 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain) = 0; 623 // lock all effect chains Mutexes. Must be called before releasing the 624 // ThreadBase mutex before processing the mixer and effects. This guarantees the 625 // integrity of the chains during the process. 626 // Also sets the parameter 'effectChains' to current value of mEffectChains. 627 void lockEffectChains_l(Vector< sp<EffectChain> >& effectChains); 628 // unlock effect chains after process 629 void unlockEffectChains(const Vector< sp<EffectChain> >& effectChains); 630 // set audio mode to all effect chains 631 void setMode(audio_mode_t mode); 632 // get effect module with corresponding ID on specified audio session 633 sp<AudioFlinger::EffectModule> getEffect(int sessionId, int effectId); 634 sp<AudioFlinger::EffectModule> getEffect_l(int sessionId, int effectId); 635 // add and effect module. Also creates the effect chain is none exists for 636 // the effects audio session 637 status_t addEffect_l(const sp< EffectModule>& effect); 638 // remove and effect module. Also removes the effect chain is this was the last 639 // effect 640 void removeEffect_l(const sp< EffectModule>& effect); 641 // detach all tracks connected to an auxiliary effect 642 virtual void detachAuxEffect_l(int effectId) {} 643 // returns either EFFECT_SESSION if effects on this audio session exist in one 644 // chain, or TRACK_SESSION if tracks on this audio session exist, or both 645 virtual uint32_t hasAudioSession(int sessionId) const = 0; 646 // the value returned by default implementation is not important as the 647 // strategy is only meaningful for PlaybackThread which implements this method 648 virtual uint32_t getStrategyForSession_l(int sessionId) { return 0; } 649 650 // suspend or restore effect according to the type of effect passed. a NULL 651 // type pointer means suspend all effects in the session 652 void setEffectSuspended(const effect_uuid_t *type, 653 bool suspend, 654 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 655 // check if some effects must be suspended/restored when an effect is enabled 656 // or disabled 657 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 658 bool enabled, 659 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 660 void checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 661 bool enabled, 662 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 663 664 virtual status_t setSyncEvent(const sp<SyncEvent>& event) = 0; 665 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const = 0; 666 667 668 mutable Mutex mLock; 669 670 protected: 671 672 // entry describing an effect being suspended in mSuspendedSessions keyed vector 673 class SuspendedSessionDesc : public RefBase { 674 public: 675 SuspendedSessionDesc() : mRefCount(0) {} 676 677 int mRefCount; // number of active suspend requests 678 effect_uuid_t mType; // effect type UUID 679 }; 680 681 void acquireWakeLock(); 682 void acquireWakeLock_l(); 683 void releaseWakeLock(); 684 void releaseWakeLock_l(); 685 void setEffectSuspended_l(const effect_uuid_t *type, 686 bool suspend, 687 int sessionId); 688 // updated mSuspendedSessions when an effect suspended or restored 689 void updateSuspendedSessions_l(const effect_uuid_t *type, 690 bool suspend, 691 int sessionId); 692 // check if some effects must be suspended when an effect chain is added 693 void checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain); 694 695 virtual void preExit() { } 696 697 friend class AudioFlinger; // for mEffectChains 698 699 const type_t mType; 700 701 // Used by parameters, config events, addTrack_l, exit 702 Condition mWaitWorkCV; 703 704 const sp<AudioFlinger> mAudioFlinger; 705 uint32_t mSampleRate; 706 size_t mFrameCount; // output HAL, direct output, record 707 size_t mNormalFrameCount; // normal mixer and effects 708 audio_channel_mask_t mChannelMask; 709 uint16_t mChannelCount; 710 size_t mFrameSize; 711 audio_format_t mFormat; 712 713 // Parameter sequence by client: binder thread calling setParameters(): 714 // 1. Lock mLock 715 // 2. Append to mNewParameters 716 // 3. mWaitWorkCV.signal 717 // 4. mParamCond.waitRelative with timeout 718 // 5. read mParamStatus 719 // 6. mWaitWorkCV.signal 720 // 7. Unlock 721 // 722 // Parameter sequence by server: threadLoop calling checkForNewParameters_l(): 723 // 1. Lock mLock 724 // 2. If there is an entry in mNewParameters proceed ... 725 // 2. Read first entry in mNewParameters 726 // 3. Process 727 // 4. Remove first entry from mNewParameters 728 // 5. Set mParamStatus 729 // 6. mParamCond.signal 730 // 7. mWaitWorkCV.wait with timeout (this is to avoid overwriting mParamStatus) 731 // 8. Unlock 732 Condition mParamCond; 733 Vector<String8> mNewParameters; 734 status_t mParamStatus; 735 736 Vector<ConfigEvent *> mConfigEvents; 737 738 // These fields are written and read by thread itself without lock or barrier, 739 // and read by other threads without lock or barrier via standby() , outDevice() 740 // and inDevice(). 741 // Because of the absence of a lock or barrier, any other thread that reads 742 // these fields must use the information in isolation, or be prepared to deal 743 // with possibility that it might be inconsistent with other information. 744 bool mStandby; // Whether thread is currently in standby. 745 audio_devices_t mOutDevice; // output device 746 audio_devices_t mInDevice; // input device 747 audio_source_t mAudioSource; // (see audio.h, audio_source_t) 748 749 const audio_io_handle_t mId; 750 Vector< sp<EffectChain> > mEffectChains; 751 752 static const int kNameLength = 16; // prctl(PR_SET_NAME) limit 753 char mName[kNameLength]; 754 sp<IPowerManager> mPowerManager; 755 sp<IBinder> mWakeLockToken; 756 const sp<PMDeathRecipient> mDeathRecipient; 757 // list of suspended effects per session and per type. The first vector is 758 // keyed by session ID, the second by type UUID timeLow field 759 KeyedVector< int, KeyedVector< int, sp<SuspendedSessionDesc> > > 760 mSuspendedSessions; 761 }; 762 763 struct stream_type_t { 764 stream_type_t() 765 : volume(1.0f), 766 mute(false) 767 { 768 } 769 float volume; 770 bool mute; 771 }; 772 773 // --- PlaybackThread --- 774 class PlaybackThread : public ThreadBase { 775 public: 776 777 enum mixer_state { 778 MIXER_IDLE, // no active tracks 779 MIXER_TRACKS_ENABLED, // at least one active track, but no track has any data ready 780 MIXER_TRACKS_READY // at least one active track, and at least one track has data 781 // standby mode does not have an enum value 782 // suspend by audio policy manager is orthogonal to mixer state 783 }; 784 785 // playback track 786 class Track : public TrackBase, public VolumeProvider { 787 public: 788 Track( PlaybackThread *thread, 789 const sp<Client>& client, 790 audio_stream_type_t streamType, 791 uint32_t sampleRate, 792 audio_format_t format, 793 audio_channel_mask_t channelMask, 794 int frameCount, 795 const sp<IMemory>& sharedBuffer, 796 int sessionId, 797 IAudioFlinger::track_flags_t flags); 798 virtual ~Track(); 799 800 static void appendDumpHeader(String8& result); 801 void dump(char* buffer, size_t size); 802 virtual status_t start(AudioSystem::sync_event_t event = 803 AudioSystem::SYNC_EVENT_NONE, 804 int triggerSession = 0); 805 virtual void stop(); 806 void pause(); 807 808 void flush(); 809 void destroy(); 810 void mute(bool); 811 int name() const { return mName; } 812 813 audio_stream_type_t streamType() const { 814 return mStreamType; 815 } 816 status_t attachAuxEffect(int EffectId); 817 void setAuxBuffer(int EffectId, int32_t *buffer); 818 int32_t *auxBuffer() const { return mAuxBuffer; } 819 void setMainBuffer(int16_t *buffer) { mMainBuffer = buffer; } 820 int16_t *mainBuffer() const { return mMainBuffer; } 821 int auxEffectId() const { return mAuxEffectId; } 822 823 // implement FastMixerState::VolumeProvider interface 824 virtual uint32_t getVolumeLR(); 825 826 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 827 828 protected: 829 // for numerous 830 friend class PlaybackThread; 831 friend class MixerThread; 832 friend class DirectOutputThread; 833 834 Track(const Track&); 835 Track& operator = (const Track&); 836 837 // AudioBufferProvider interface 838 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, 839 int64_t pts = kInvalidPTS); 840 // releaseBuffer() not overridden 841 842 virtual size_t framesReady() const; 843 844 bool isMuted() const { return mMute; } 845 bool isPausing() const { 846 return mState == PAUSING; 847 } 848 bool isPaused() const { 849 return mState == PAUSED; 850 } 851 bool isResuming() const { 852 return mState == RESUMING; 853 } 854 bool isReady() const; 855 void setPaused() { mState = PAUSED; } 856 void reset(); 857 858 bool isOutputTrack() const { 859 return (mStreamType == AUDIO_STREAM_CNT); 860 } 861 862 sp<IMemory> sharedBuffer() const { return mSharedBuffer; } 863 864 // framesWritten is cumulative, never reset, and is shared all tracks 865 // audioHalFrames is derived from output latency 866 // FIXME parameters not needed, could get them from the thread 867 bool presentationComplete(size_t framesWritten, size_t audioHalFrames); 868 869 public: 870 void triggerEvents(AudioSystem::sync_event_t type); 871 virtual bool isTimedTrack() const { return false; } 872 bool isFastTrack() const { return (mFlags & IAudioFlinger::TRACK_FAST) != 0; } 873 virtual bool isOut() const; 874 875 protected: 876 877 // written by Track::mute() called by binder thread(s), without a mutex or barrier. 878 // read by Track::isMuted() called by playback thread, also without a mutex or barrier. 879 // The lack of mutex or barrier is safe because the mute status is only used by itself. 880 bool mMute; 881 882 // FILLED state is used for suppressing volume ramp at begin of playing 883 enum {FS_INVALID, FS_FILLING, FS_FILLED, FS_ACTIVE}; 884 mutable uint8_t mFillingUpStatus; 885 int8_t mRetryCount; 886 const sp<IMemory> mSharedBuffer; 887 bool mResetDone; 888 const audio_stream_type_t mStreamType; 889 int mName; // track name on the normal mixer, 890 // allocated statically at track creation time, 891 // and is even allocated (though unused) for fast tracks 892 // FIXME don't allocate track name for fast tracks 893 int16_t *mMainBuffer; 894 int32_t *mAuxBuffer; 895 int mAuxEffectId; 896 bool mHasVolumeController; 897 size_t mPresentationCompleteFrames; // number of frames written to the 898 // audio HAL when this track will be fully rendered 899 // zero means not monitoring 900 private: 901 IAudioFlinger::track_flags_t mFlags; 902 903 // The following fields are only for fast tracks, and should be in a subclass 904 int mFastIndex; // index within FastMixerState::mFastTracks[]; 905 // either mFastIndex == -1 if not isFastTrack() 906 // or 0 < mFastIndex < FastMixerState::kMaxFast because 907 // index 0 is reserved for normal mixer's submix; 908 // index is allocated statically at track creation time 909 // but the slot is only used if track is active 910 FastTrackUnderruns mObservedUnderruns; // Most recently observed value of 911 // mFastMixerDumpState.mTracks[mFastIndex].mUnderruns 912 uint32_t mUnderrunCount; // Counter of total number of underruns, never reset 913 volatile float mCachedVolume; // combined master volume and stream type volume; 914 // 'volatile' means accessed without lock or 915 // barrier, but is read/written atomically 916 }; // end of Track 917 918 class TimedTrack : public Track { 919 public: 920 static sp<TimedTrack> create(PlaybackThread *thread, 921 const sp<Client>& client, 922 audio_stream_type_t streamType, 923 uint32_t sampleRate, 924 audio_format_t format, 925 audio_channel_mask_t channelMask, 926 int frameCount, 927 const sp<IMemory>& sharedBuffer, 928 int sessionId); 929 virtual ~TimedTrack(); 930 931 class TimedBuffer { 932 public: 933 TimedBuffer(); 934 TimedBuffer(const sp<IMemory>& buffer, int64_t pts); 935 const sp<IMemory>& buffer() const { return mBuffer; } 936 int64_t pts() const { return mPTS; } 937 uint32_t position() const { return mPosition; } 938 void setPosition(uint32_t pos) { mPosition = pos; } 939 private: 940 sp<IMemory> mBuffer; 941 int64_t mPTS; 942 uint32_t mPosition; 943 }; 944 945 // Mixer facing methods. 946 virtual bool isTimedTrack() const { return true; } 947 virtual size_t framesReady() const; 948 949 // AudioBufferProvider interface 950 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, 951 int64_t pts); 952 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 953 954 // Client/App facing methods. 955 status_t allocateTimedBuffer(size_t size, 956 sp<IMemory>* buffer); 957 status_t queueTimedBuffer(const sp<IMemory>& buffer, 958 int64_t pts); 959 status_t setMediaTimeTransform(const LinearTransform& xform, 960 TimedAudioTrack::TargetTimeline target); 961 962 private: 963 TimedTrack(PlaybackThread *thread, 964 const sp<Client>& client, 965 audio_stream_type_t streamType, 966 uint32_t sampleRate, 967 audio_format_t format, 968 audio_channel_mask_t channelMask, 969 int frameCount, 970 const sp<IMemory>& sharedBuffer, 971 int sessionId); 972 973 void timedYieldSamples_l(AudioBufferProvider::Buffer* buffer); 974 void timedYieldSilence_l(uint32_t numFrames, 975 AudioBufferProvider::Buffer* buffer); 976 void trimTimedBufferQueue_l(); 977 void trimTimedBufferQueueHead_l(const char* logTag); 978 void updateFramesPendingAfterTrim_l(const TimedBuffer& buf, 979 const char* logTag); 980 981 uint64_t mLocalTimeFreq; 982 LinearTransform mLocalTimeToSampleTransform; 983 LinearTransform mMediaTimeToSampleTransform; 984 sp<MemoryDealer> mTimedMemoryDealer; 985 986 Vector<TimedBuffer> mTimedBufferQueue; 987 bool mQueueHeadInFlight; 988 bool mTrimQueueHeadOnRelease; 989 uint32_t mFramesPendingInQueue; 990 991 uint8_t* mTimedSilenceBuffer; 992 uint32_t mTimedSilenceBufferSize; 993 mutable Mutex mTimedBufferQueueLock; 994 bool mTimedAudioOutputOnTime; 995 CCHelper mCCHelper; 996 997 Mutex mMediaTimeTransformLock; 998 LinearTransform mMediaTimeTransform; 999 bool mMediaTimeTransformValid; 1000 TimedAudioTrack::TargetTimeline mMediaTimeTransformTarget; 1001 }; 1002 1003 1004 // playback track, used by DuplicatingThread 1005 class OutputTrack : public Track { 1006 public: 1007 1008 class Buffer : public AudioBufferProvider::Buffer { 1009 public: 1010 int16_t *mBuffer; 1011 }; 1012 1013 OutputTrack(PlaybackThread *thread, 1014 DuplicatingThread *sourceThread, 1015 uint32_t sampleRate, 1016 audio_format_t format, 1017 audio_channel_mask_t channelMask, 1018 int frameCount); 1019 virtual ~OutputTrack(); 1020 1021 virtual status_t start(AudioSystem::sync_event_t event = 1022 AudioSystem::SYNC_EVENT_NONE, 1023 int triggerSession = 0); 1024 virtual void stop(); 1025 bool write(int16_t* data, uint32_t frames); 1026 bool bufferQueueEmpty() const { return mBufferQueue.size() == 0; } 1027 bool isActive() const { return mActive; } 1028 const wp<ThreadBase>& thread() const { return mThread; } 1029 1030 private: 1031 1032 enum { 1033 NO_MORE_BUFFERS = 0x80000001, // same in AudioTrack.h, ok to be different value 1034 }; 1035 1036 status_t obtainBuffer(AudioBufferProvider::Buffer* buffer, 1037 uint32_t waitTimeMs); 1038 void clearBufferQueue(); 1039 1040 // Maximum number of pending buffers allocated by OutputTrack::write() 1041 static const uint8_t kMaxOverFlowBuffers = 10; 1042 1043 Vector < Buffer* > mBufferQueue; 1044 AudioBufferProvider::Buffer mOutBuffer; 1045 bool mActive; 1046 DuplicatingThread* const mSourceThread; // for waitTimeMs() in write() 1047 void* mBuffers; // starting address of buffers in plain memory 1048 }; // end of OutputTrack 1049 1050 PlaybackThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1051 audio_io_handle_t id, audio_devices_t device, type_t type); 1052 virtual ~PlaybackThread(); 1053 1054 void dump(int fd, const Vector<String16>& args); 1055 1056 // Thread virtuals 1057 virtual status_t readyToRun(); 1058 virtual bool threadLoop(); 1059 1060 // RefBase 1061 virtual void onFirstRef(); 1062 1063protected: 1064 // Code snippets that were lifted up out of threadLoop() 1065 virtual void threadLoop_mix() = 0; 1066 virtual void threadLoop_sleepTime() = 0; 1067 virtual void threadLoop_write(); 1068 virtual void threadLoop_standby(); 1069 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 1070 1071 // prepareTracks_l reads and writes mActiveTracks, and returns 1072 // the pending set of tracks to remove via Vector 'tracksToRemove'. The caller 1073 // is responsible for clearing or destroying this Vector later on, when it 1074 // is safe to do so. That will drop the final ref count and destroy the tracks. 1075 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove) = 0; 1076 1077 // ThreadBase virtuals 1078 virtual void preExit(); 1079 1080public: 1081 1082 virtual status_t initCheck() const { return (mOutput == NULL) ? NO_INIT : NO_ERROR; } 1083 1084 // return estimated latency in milliseconds, as reported by HAL 1085 uint32_t latency() const; 1086 // same, but lock must already be held 1087 uint32_t latency_l() const; 1088 1089 void setMasterVolume(float value); 1090 void setMasterMute(bool muted); 1091 1092 void setStreamVolume(audio_stream_type_t stream, float value); 1093 void setStreamMute(audio_stream_type_t stream, bool muted); 1094 1095 float streamVolume(audio_stream_type_t stream) const; 1096 1097 sp<Track> createTrack_l( 1098 const sp<AudioFlinger::Client>& client, 1099 audio_stream_type_t streamType, 1100 uint32_t sampleRate, 1101 audio_format_t format, 1102 audio_channel_mask_t channelMask, 1103 int frameCount, 1104 const sp<IMemory>& sharedBuffer, 1105 int sessionId, 1106 IAudioFlinger::track_flags_t *flags, 1107 pid_t tid, 1108 status_t *status); 1109 1110 AudioStreamOut* getOutput() const; 1111 AudioStreamOut* clearOutput(); 1112 virtual audio_stream_t* stream() const; 1113 1114 // a very large number of suspend() will eventually wraparound, but unlikely 1115 void suspend() { (void) android_atomic_inc(&mSuspended); } 1116 void restore() 1117 { 1118 // if restore() is done without suspend(), get back into 1119 // range so that the next suspend() will operate correctly 1120 if (android_atomic_dec(&mSuspended) <= 0) { 1121 android_atomic_release_store(0, &mSuspended); 1122 } 1123 } 1124 bool isSuspended() const 1125 { return android_atomic_acquire_load(&mSuspended) > 0; } 1126 1127 virtual String8 getParameters(const String8& keys); 1128 virtual void audioConfigChanged_l(int event, int param = 0); 1129 status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames); 1130 int16_t *mixBuffer() const { return mMixBuffer; }; 1131 1132 virtual void detachAuxEffect_l(int effectId); 1133 status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track> track, 1134 int EffectId); 1135 status_t attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track> track, 1136 int EffectId); 1137 1138 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1139 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1140 virtual uint32_t hasAudioSession(int sessionId) const; 1141 virtual uint32_t getStrategyForSession_l(int sessionId); 1142 1143 1144 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 1145 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 1146 void invalidateTracks(audio_stream_type_t streamType); 1147 1148 1149 protected: 1150 int16_t* mMixBuffer; 1151 1152 // suspend count, > 0 means suspended. While suspended, the thread continues to pull from 1153 // tracks and mix, but doesn't write to HAL. A2DP and SCO HAL implementations can't handle 1154 // concurrent use of both of them, so Audio Policy Service suspends one of the threads to 1155 // workaround that restriction. 1156 // 'volatile' means accessed via atomic operations and no lock. 1157 volatile int32_t mSuspended; 1158 1159 int mBytesWritten; 1160 private: 1161 // mMasterMute is in both PlaybackThread and in AudioFlinger. When a 1162 // PlaybackThread needs to find out if master-muted, it checks it's local 1163 // copy rather than the one in AudioFlinger. This optimization saves a lock. 1164 bool mMasterMute; 1165 void setMasterMute_l(bool muted) { mMasterMute = muted; } 1166 protected: 1167 SortedVector< wp<Track> > mActiveTracks; // FIXME check if this could be sp<> 1168 1169 // Allocate a track name for a given channel mask. 1170 // Returns name >= 0 if successful, -1 on failure. 1171 virtual int getTrackName_l(audio_channel_mask_t channelMask, int sessionId) = 0; 1172 virtual void deleteTrackName_l(int name) = 0; 1173 1174 // Time to sleep between cycles when: 1175 virtual uint32_t activeSleepTimeUs() const; // mixer state MIXER_TRACKS_ENABLED 1176 virtual uint32_t idleSleepTimeUs() const = 0; // mixer state MIXER_IDLE 1177 virtual uint32_t suspendSleepTimeUs() const = 0; // audio policy manager suspended us 1178 // No sleep when mixer state == MIXER_TRACKS_READY; relies on audio HAL stream->write() 1179 // No sleep in standby mode; waits on a condition 1180 1181 // Code snippets that are temporarily lifted up out of threadLoop() until the merge 1182 void checkSilentMode_l(); 1183 1184 // Non-trivial for DUPLICATING only 1185 virtual void saveOutputTracks() { } 1186 virtual void clearOutputTracks() { } 1187 1188 // Cache various calculated values, at threadLoop() entry and after a parameter change 1189 virtual void cacheParameters_l(); 1190 1191 virtual uint32_t correctLatency(uint32_t latency) const; 1192 1193 private: 1194 1195 friend class AudioFlinger; // for numerous 1196 1197 PlaybackThread(const Client&); 1198 PlaybackThread& operator = (const PlaybackThread&); 1199 1200 status_t addTrack_l(const sp<Track>& track); 1201 void destroyTrack_l(const sp<Track>& track); 1202 void removeTrack_l(const sp<Track>& track); 1203 1204 void readOutputParameters(); 1205 1206 virtual void dumpInternals(int fd, const Vector<String16>& args); 1207 void dumpTracks(int fd, const Vector<String16>& args); 1208 1209 SortedVector< sp<Track> > mTracks; 1210 // mStreamTypes[] uses 1 additional stream type internally for the OutputTrack used by 1211 // DuplicatingThread 1212 stream_type_t mStreamTypes[AUDIO_STREAM_CNT + 1]; 1213 AudioStreamOut *mOutput; 1214 1215 float mMasterVolume; 1216 nsecs_t mLastWriteTime; 1217 int mNumWrites; 1218 int mNumDelayedWrites; 1219 bool mInWrite; 1220 1221 // FIXME rename these former local variables of threadLoop to standard "m" names 1222 nsecs_t standbyTime; 1223 size_t mixBufferSize; 1224 1225 // cached copies of activeSleepTimeUs() and idleSleepTimeUs() made by cacheParameters_l() 1226 uint32_t activeSleepTime; 1227 uint32_t idleSleepTime; 1228 1229 uint32_t sleepTime; 1230 1231 // mixer status returned by prepareTracks_l() 1232 mixer_state mMixerStatus; // current cycle 1233 // previous cycle when in prepareTracks_l() 1234 mixer_state mMixerStatusIgnoringFastTracks; 1235 // FIXME or a separate ready state per track 1236 1237 // FIXME move these declarations into the specific sub-class that needs them 1238 // MIXER only 1239 uint32_t sleepTimeShift; 1240 1241 // same as AudioFlinger::mStandbyTimeInNsecs except for DIRECT which uses a shorter value 1242 nsecs_t standbyDelay; 1243 1244 // MIXER only 1245 nsecs_t maxPeriod; 1246 1247 // DUPLICATING only 1248 uint32_t writeFrames; 1249 1250 private: 1251 // The HAL output sink is treated as non-blocking, but current implementation is blocking 1252 sp<NBAIO_Sink> mOutputSink; 1253 // If a fast mixer is present, the blocking pipe sink, otherwise clear 1254 sp<NBAIO_Sink> mPipeSink; 1255 // The current sink for the normal mixer to write it's (sub)mix, mOutputSink or mPipeSink 1256 sp<NBAIO_Sink> mNormalSink; 1257 // For dumpsys 1258 sp<NBAIO_Sink> mTeeSink; 1259 sp<NBAIO_Source> mTeeSource; 1260 uint32_t mScreenState; // cached copy of gScreenState 1261 public: 1262 virtual bool hasFastMixer() const = 0; 1263 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const 1264 { FastTrackUnderruns dummy; return dummy; } 1265 1266 protected: 1267 // accessed by both binder threads and within threadLoop(), lock on mutex needed 1268 unsigned mFastTrackAvailMask; // bit i set if fast track [i] is available 1269 1270 }; 1271 1272 class MixerThread : public PlaybackThread { 1273 public: 1274 MixerThread(const sp<AudioFlinger>& audioFlinger, 1275 AudioStreamOut* output, 1276 audio_io_handle_t id, 1277 audio_devices_t device, 1278 type_t type = MIXER); 1279 virtual ~MixerThread(); 1280 1281 // Thread virtuals 1282 1283 virtual bool checkForNewParameters_l(); 1284 virtual void dumpInternals(int fd, const Vector<String16>& args); 1285 1286 protected: 1287 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1288 virtual int getTrackName_l(audio_channel_mask_t channelMask, int sessionId); 1289 virtual void deleteTrackName_l(int name); 1290 virtual uint32_t idleSleepTimeUs() const; 1291 virtual uint32_t suspendSleepTimeUs() const; 1292 virtual void cacheParameters_l(); 1293 1294 // threadLoop snippets 1295 virtual void threadLoop_write(); 1296 virtual void threadLoop_standby(); 1297 virtual void threadLoop_mix(); 1298 virtual void threadLoop_sleepTime(); 1299 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 1300 virtual uint32_t correctLatency(uint32_t latency) const; 1301 1302 AudioMixer* mAudioMixer; // normal mixer 1303 private: 1304 // one-time initialization, no locks required 1305 FastMixer* mFastMixer; // non-NULL if there is also a fast mixer 1306 sp<AudioWatchdog> mAudioWatchdog; // non-0 if there is an audio watchdog thread 1307 1308 // contents are not guaranteed to be consistent, no locks required 1309 FastMixerDumpState mFastMixerDumpState; 1310#ifdef STATE_QUEUE_DUMP 1311 StateQueueObserverDump mStateQueueObserverDump; 1312 StateQueueMutatorDump mStateQueueMutatorDump; 1313#endif 1314 AudioWatchdogDump mAudioWatchdogDump; 1315 1316 // accessible only within the threadLoop(), no locks required 1317 // mFastMixer->sq() // for mutating and pushing state 1318 int32_t mFastMixerFutex; // for cold idle 1319 1320 public: 1321 virtual bool hasFastMixer() const { return mFastMixer != NULL; } 1322 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const { 1323 ALOG_ASSERT(fastIndex < FastMixerState::kMaxFastTracks); 1324 return mFastMixerDumpState.mTracks[fastIndex].mUnderruns; 1325 } 1326 }; 1327 1328 class DirectOutputThread : public PlaybackThread { 1329 public: 1330 1331 DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1332 audio_io_handle_t id, audio_devices_t device); 1333 virtual ~DirectOutputThread(); 1334 1335 // Thread virtuals 1336 1337 virtual bool checkForNewParameters_l(); 1338 1339 protected: 1340 virtual int getTrackName_l(audio_channel_mask_t channelMask, int sessionId); 1341 virtual void deleteTrackName_l(int name); 1342 virtual uint32_t activeSleepTimeUs() const; 1343 virtual uint32_t idleSleepTimeUs() const; 1344 virtual uint32_t suspendSleepTimeUs() const; 1345 virtual void cacheParameters_l(); 1346 1347 // threadLoop snippets 1348 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1349 virtual void threadLoop_mix(); 1350 virtual void threadLoop_sleepTime(); 1351 1352 private: 1353 // volumes last sent to audio HAL with stream->set_volume() 1354 float mLeftVolFloat; 1355 float mRightVolFloat; 1356 1357 // prepareTracks_l() tells threadLoop_mix() the name of the single active track 1358 sp<Track> mActiveTrack; 1359 public: 1360 virtual bool hasFastMixer() const { return false; } 1361 }; 1362 1363 class DuplicatingThread : public MixerThread { 1364 public: 1365 DuplicatingThread(const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread, 1366 audio_io_handle_t id); 1367 virtual ~DuplicatingThread(); 1368 1369 // Thread virtuals 1370 void addOutputTrack(MixerThread* thread); 1371 void removeOutputTrack(MixerThread* thread); 1372 uint32_t waitTimeMs() const { return mWaitTimeMs; } 1373 protected: 1374 virtual uint32_t activeSleepTimeUs() const; 1375 1376 private: 1377 bool outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks); 1378 protected: 1379 // threadLoop snippets 1380 virtual void threadLoop_mix(); 1381 virtual void threadLoop_sleepTime(); 1382 virtual void threadLoop_write(); 1383 virtual void threadLoop_standby(); 1384 virtual void cacheParameters_l(); 1385 1386 private: 1387 // called from threadLoop, addOutputTrack, removeOutputTrack 1388 virtual void updateWaitTime_l(); 1389 protected: 1390 virtual void saveOutputTracks(); 1391 virtual void clearOutputTracks(); 1392 private: 1393 1394 uint32_t mWaitTimeMs; 1395 SortedVector < sp<OutputTrack> > outputTracks; 1396 SortedVector < sp<OutputTrack> > mOutputTracks; 1397 public: 1398 virtual bool hasFastMixer() const { return false; } 1399 }; 1400 1401 PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const; 1402 MixerThread *checkMixerThread_l(audio_io_handle_t output) const; 1403 RecordThread *checkRecordThread_l(audio_io_handle_t input) const; 1404 // no range check, AudioFlinger::mLock held 1405 bool streamMute_l(audio_stream_type_t stream) const 1406 { return mStreamTypes[stream].mute; } 1407 // no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held 1408 float streamVolume_l(audio_stream_type_t stream) const 1409 { return mStreamTypes[stream].volume; } 1410 void audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2); 1411 1412 // allocate an audio_io_handle_t, session ID, or effect ID 1413 uint32_t nextUniqueId(); 1414 1415 status_t moveEffectChain_l(int sessionId, 1416 PlaybackThread *srcThread, 1417 PlaybackThread *dstThread, 1418 bool reRegister); 1419 // return thread associated with primary hardware device, or NULL 1420 PlaybackThread *primaryPlaybackThread_l() const; 1421 audio_devices_t primaryOutputDevice_l() const; 1422 1423 sp<PlaybackThread> getEffectThread_l(int sessionId, int EffectId); 1424 1425 // server side of the client's IAudioTrack 1426 class TrackHandle : public android::BnAudioTrack { 1427 public: 1428 TrackHandle(const sp<PlaybackThread::Track>& track); 1429 virtual ~TrackHandle(); 1430 virtual sp<IMemory> getCblk() const; 1431 virtual status_t start(); 1432 virtual void stop(); 1433 virtual void flush(); 1434 virtual void mute(bool); 1435 virtual void pause(); 1436 virtual status_t attachAuxEffect(int effectId); 1437 virtual status_t allocateTimedBuffer(size_t size, 1438 sp<IMemory>* buffer); 1439 virtual status_t queueTimedBuffer(const sp<IMemory>& buffer, 1440 int64_t pts); 1441 virtual status_t setMediaTimeTransform(const LinearTransform& xform, 1442 int target); 1443 virtual status_t onTransact( 1444 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 1445 private: 1446 const sp<PlaybackThread::Track> mTrack; 1447 }; 1448 1449 void removeClient_l(pid_t pid); 1450 void removeNotificationClient(pid_t pid); 1451 1452 1453 // record thread 1454 class RecordThread : public ThreadBase, public AudioBufferProvider 1455 // derives from AudioBufferProvider interface for use by resampler 1456 { 1457 public: 1458 1459 // record track 1460 class RecordTrack : public TrackBase { 1461 public: 1462 RecordTrack(RecordThread *thread, 1463 const sp<Client>& client, 1464 uint32_t sampleRate, 1465 audio_format_t format, 1466 audio_channel_mask_t channelMask, 1467 int frameCount, 1468 int sessionId); 1469 virtual ~RecordTrack(); 1470 1471 virtual status_t start(AudioSystem::sync_event_t event, int triggerSession); 1472 virtual void stop(); 1473 1474 void destroy(); 1475 1476 // clear the buffer overflow flag 1477 void clearOverflow() { mOverflow = false; } 1478 // set the buffer overflow flag and return previous value 1479 bool setOverflow() { bool tmp = mOverflow; mOverflow = true; 1480 return tmp; } 1481 1482 static void appendDumpHeader(String8& result); 1483 void dump(char* buffer, size_t size); 1484 1485 virtual bool isOut() const; 1486 1487 private: 1488 friend class AudioFlinger; // for mState 1489 1490 RecordTrack(const RecordTrack&); 1491 RecordTrack& operator = (const RecordTrack&); 1492 1493 // AudioBufferProvider interface 1494 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, 1495 int64_t pts = kInvalidPTS); 1496 // releaseBuffer() not overridden 1497 1498 bool mOverflow; // overflow on most recent attempt to fill client buffer 1499 }; 1500 1501 RecordThread(const sp<AudioFlinger>& audioFlinger, 1502 AudioStreamIn *input, 1503 uint32_t sampleRate, 1504 audio_channel_mask_t channelMask, 1505 audio_io_handle_t id, 1506 audio_devices_t device, 1507 const sp<NBAIO_Sink>& teeSink); 1508 virtual ~RecordThread(); 1509 1510 // no addTrack_l ? 1511 void destroyTrack_l(const sp<RecordTrack>& track); 1512 void removeTrack_l(const sp<RecordTrack>& track); 1513 1514 void dumpInternals(int fd, const Vector<String16>& args); 1515 void dumpTracks(int fd, const Vector<String16>& args); 1516 1517 // Thread virtuals 1518 virtual bool threadLoop(); 1519 virtual status_t readyToRun(); 1520 1521 // RefBase 1522 virtual void onFirstRef(); 1523 1524 virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; } 1525 sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l( 1526 const sp<AudioFlinger::Client>& client, 1527 uint32_t sampleRate, 1528 audio_format_t format, 1529 audio_channel_mask_t channelMask, 1530 int frameCount, 1531 int sessionId, 1532 IAudioFlinger::track_flags_t flags, 1533 pid_t tid, 1534 status_t *status); 1535 1536 status_t start(RecordTrack* recordTrack, 1537 AudioSystem::sync_event_t event, 1538 int triggerSession); 1539 1540 // ask the thread to stop the specified track, and 1541 // return true if the caller should then do it's part of the stopping process 1542 bool stop_l(RecordTrack* recordTrack); 1543 1544 void dump(int fd, const Vector<String16>& args); 1545 AudioStreamIn* clearInput(); 1546 virtual audio_stream_t* stream() const; 1547 1548 // AudioBufferProvider interface 1549 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts); 1550 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 1551 1552 virtual bool checkForNewParameters_l(); 1553 virtual String8 getParameters(const String8& keys); 1554 virtual void audioConfigChanged_l(int event, int param = 0); 1555 void readInputParameters(); 1556 virtual unsigned int getInputFramesLost(); 1557 1558 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1559 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1560 virtual uint32_t hasAudioSession(int sessionId) const; 1561 1562 // Return the set of unique session IDs across all tracks. 1563 // The keys are the session IDs, and the associated values are meaningless. 1564 // FIXME replace by Set [and implement Bag/Multiset for other uses]. 1565 KeyedVector<int, bool> sessionIds() const; 1566 1567 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 1568 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 1569 1570 static void syncStartEventCallback(const wp<SyncEvent>& event); 1571 void handleSyncStartEvent(const sp<SyncEvent>& event); 1572 1573 private: 1574 void clearSyncStartEvent(); 1575 1576 // Enter standby if not already in standby, and set mStandby flag 1577 void standby(); 1578 1579 // Call the HAL standby method unconditionally, and don't change mStandby flag 1580 void inputStandBy(); 1581 1582 AudioStreamIn *mInput; 1583 SortedVector < sp<RecordTrack> > mTracks; 1584 // mActiveTrack has dual roles: it indicates the current active track, and 1585 // is used together with mStartStopCond to indicate start()/stop() progress 1586 sp<RecordTrack> mActiveTrack; 1587 Condition mStartStopCond; 1588 AudioResampler *mResampler; 1589 int32_t *mRsmpOutBuffer; 1590 int16_t *mRsmpInBuffer; 1591 size_t mRsmpInIndex; 1592 size_t mInputBytes; 1593 const int mReqChannelCount; 1594 const uint32_t mReqSampleRate; 1595 ssize_t mBytesRead; 1596 // sync event triggering actual audio capture. Frames read before this event will 1597 // be dropped and therefore not read by the application. 1598 sp<SyncEvent> mSyncStartEvent; 1599 // number of captured frames to drop after the start sync event has been received. 1600 // when < 0, maximum frames to drop before starting capture even if sync event is 1601 // not received 1602 ssize_t mFramestoDrop; 1603 1604 // For dumpsys 1605 const sp<NBAIO_Sink> mTeeSink; 1606 }; 1607 1608 // server side of the client's IAudioRecord 1609 class RecordHandle : public android::BnAudioRecord { 1610 public: 1611 RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack); 1612 virtual ~RecordHandle(); 1613 virtual sp<IMemory> getCblk() const; 1614 virtual status_t start(int /*AudioSystem::sync_event_t*/ event, int triggerSession); 1615 virtual void stop(); 1616 virtual status_t onTransact( 1617 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 1618 private: 1619 const sp<RecordThread::RecordTrack> mRecordTrack; 1620 1621 // for use from destructor 1622 void stop_nonvirtual(); 1623 }; 1624 1625 //--- Audio Effect Management 1626 1627 // EffectModule and EffectChain classes both have their own mutex to protect 1628 // state changes or resource modifications. Always respect the following order 1629 // if multiple mutexes must be acquired to avoid cross deadlock: 1630 // AudioFlinger -> ThreadBase -> EffectChain -> EffectModule 1631 1632 // The EffectModule class is a wrapper object controlling the effect engine implementation 1633 // in the effect library. It prevents concurrent calls to process() and command() functions 1634 // from different client threads. It keeps a list of EffectHandle objects corresponding 1635 // to all client applications using this effect and notifies applications of effect state, 1636 // control or parameter changes. It manages the activation state machine to send appropriate 1637 // reset, enable, disable commands to effect engine and provide volume 1638 // ramping when effects are activated/deactivated. 1639 // When controlling an auxiliary effect, the EffectModule also provides an input buffer used by 1640 // the attached track(s) to accumulate their auxiliary channel. 1641 class EffectModule : public RefBase { 1642 public: 1643 EffectModule(ThreadBase *thread, 1644 const wp<AudioFlinger::EffectChain>& chain, 1645 effect_descriptor_t *desc, 1646 int id, 1647 int sessionId); 1648 virtual ~EffectModule(); 1649 1650 enum effect_state { 1651 IDLE, 1652 RESTART, 1653 STARTING, 1654 ACTIVE, 1655 STOPPING, 1656 STOPPED, 1657 DESTROYED 1658 }; 1659 1660 int id() const { return mId; } 1661 void process(); 1662 void updateState(); 1663 status_t command(uint32_t cmdCode, 1664 uint32_t cmdSize, 1665 void *pCmdData, 1666 uint32_t *replySize, 1667 void *pReplyData); 1668 1669 void reset_l(); 1670 status_t configure(); 1671 status_t init(); 1672 effect_state state() const { 1673 return mState; 1674 } 1675 uint32_t status() { 1676 return mStatus; 1677 } 1678 int sessionId() const { 1679 return mSessionId; 1680 } 1681 status_t setEnabled(bool enabled); 1682 status_t setEnabled_l(bool enabled); 1683 bool isEnabled() const; 1684 bool isProcessEnabled() const; 1685 1686 void setInBuffer(int16_t *buffer) { mConfig.inputCfg.buffer.s16 = buffer; } 1687 int16_t *inBuffer() { return mConfig.inputCfg.buffer.s16; } 1688 void setOutBuffer(int16_t *buffer) { mConfig.outputCfg.buffer.s16 = buffer; } 1689 int16_t *outBuffer() { return mConfig.outputCfg.buffer.s16; } 1690 void setChain(const wp<EffectChain>& chain) { mChain = chain; } 1691 void setThread(const wp<ThreadBase>& thread) { mThread = thread; } 1692 const wp<ThreadBase>& thread() { return mThread; } 1693 1694 status_t addHandle(EffectHandle *handle); 1695 size_t disconnect(EffectHandle *handle, bool unpinIfLast); 1696 size_t removeHandle(EffectHandle *handle); 1697 1698 const effect_descriptor_t& desc() const { return mDescriptor; } 1699 wp<EffectChain>& chain() { return mChain; } 1700 1701 status_t setDevice(audio_devices_t device); 1702 status_t setVolume(uint32_t *left, uint32_t *right, bool controller); 1703 status_t setMode(audio_mode_t mode); 1704 status_t setAudioSource(audio_source_t source); 1705 status_t start(); 1706 status_t stop(); 1707 void setSuspended(bool suspended); 1708 bool suspended() const; 1709 1710 EffectHandle* controlHandle_l(); 1711 1712 bool isPinned() const { return mPinned; } 1713 void unPin() { mPinned = false; } 1714 bool purgeHandles(); 1715 void lock() { mLock.lock(); } 1716 void unlock() { mLock.unlock(); } 1717 1718 void dump(int fd, const Vector<String16>& args); 1719 1720 protected: 1721 friend class AudioFlinger; // for mHandles 1722 bool mPinned; 1723 1724 // Maximum time allocated to effect engines to complete the turn off sequence 1725 static const uint32_t MAX_DISABLE_TIME_MS = 10000; 1726 1727 EffectModule(const EffectModule&); 1728 EffectModule& operator = (const EffectModule&); 1729 1730 status_t start_l(); 1731 status_t stop_l(); 1732 1733mutable Mutex mLock; // mutex for process, commands and handles list protection 1734 wp<ThreadBase> mThread; // parent thread 1735 wp<EffectChain> mChain; // parent effect chain 1736 const int mId; // this instance unique ID 1737 const int mSessionId; // audio session ID 1738 const effect_descriptor_t mDescriptor;// effect descriptor received from effect engine 1739 effect_config_t mConfig; // input and output audio configuration 1740 effect_handle_t mEffectInterface; // Effect module C API 1741 status_t mStatus; // initialization status 1742 effect_state mState; // current activation state 1743 Vector<EffectHandle *> mHandles; // list of client handles 1744 // First handle in mHandles has highest priority and controls the effect module 1745 uint32_t mMaxDisableWaitCnt; // maximum grace period before forcing an effect off after 1746 // sending disable command. 1747 uint32_t mDisableWaitCnt; // current process() calls count during disable period. 1748 bool mSuspended; // effect is suspended: temporarily disabled by framework 1749 }; 1750 1751 // The EffectHandle class implements the IEffect interface. It provides resources 1752 // to receive parameter updates, keeps track of effect control 1753 // ownership and state and has a pointer to the EffectModule object it is controlling. 1754 // There is one EffectHandle object for each application controlling (or using) 1755 // an effect module. 1756 // The EffectHandle is obtained by calling AudioFlinger::createEffect(). 1757 class EffectHandle: public android::BnEffect { 1758 public: 1759 1760 EffectHandle(const sp<EffectModule>& effect, 1761 const sp<AudioFlinger::Client>& client, 1762 const sp<IEffectClient>& effectClient, 1763 int32_t priority); 1764 virtual ~EffectHandle(); 1765 1766 // IEffect 1767 virtual status_t enable(); 1768 virtual status_t disable(); 1769 virtual status_t command(uint32_t cmdCode, 1770 uint32_t cmdSize, 1771 void *pCmdData, 1772 uint32_t *replySize, 1773 void *pReplyData); 1774 virtual void disconnect(); 1775 private: 1776 void disconnect(bool unpinIfLast); 1777 public: 1778 virtual sp<IMemory> getCblk() const { return mCblkMemory; } 1779 virtual status_t onTransact(uint32_t code, const Parcel& data, 1780 Parcel* reply, uint32_t flags); 1781 1782 1783 // Give or take control of effect module 1784 // - hasControl: true if control is given, false if removed 1785 // - signal: true client app should be signaled of change, false otherwise 1786 // - enabled: state of the effect when control is passed 1787 void setControl(bool hasControl, bool signal, bool enabled); 1788 void commandExecuted(uint32_t cmdCode, 1789 uint32_t cmdSize, 1790 void *pCmdData, 1791 uint32_t replySize, 1792 void *pReplyData); 1793 void setEnabled(bool enabled); 1794 bool enabled() const { return mEnabled; } 1795 1796 // Getters 1797 int id() const { return mEffect->id(); } 1798 int priority() const { return mPriority; } 1799 bool hasControl() const { return mHasControl; } 1800 sp<EffectModule> effect() const { return mEffect; } 1801 // destroyed_l() must be called with the associated EffectModule mLock held 1802 bool destroyed_l() const { return mDestroyed; } 1803 1804 void dump(char* buffer, size_t size); 1805 1806 protected: 1807 friend class AudioFlinger; // for mEffect, mHasControl, mEnabled 1808 EffectHandle(const EffectHandle&); 1809 EffectHandle& operator =(const EffectHandle&); 1810 1811 sp<EffectModule> mEffect; // pointer to controlled EffectModule 1812 sp<IEffectClient> mEffectClient; // callback interface for client notifications 1813 /*const*/ sp<Client> mClient; // client for shared memory allocation, see disconnect() 1814 sp<IMemory> mCblkMemory; // shared memory for control block 1815 effect_param_cblk_t* mCblk; // control block for deferred parameter setting via 1816 // shared memory 1817 uint8_t* mBuffer; // pointer to parameter area in shared memory 1818 int mPriority; // client application priority to control the effect 1819 bool mHasControl; // true if this handle is controlling the effect 1820 bool mEnabled; // cached enable state: needed when the effect is 1821 // restored after being suspended 1822 bool mDestroyed; // Set to true by destructor. Access with EffectModule 1823 // mLock held 1824 }; 1825 1826 // the EffectChain class represents a group of effects associated to one audio session. 1827 // There can be any number of EffectChain objects per output mixer thread (PlaybackThread). 1828 // The EffecChain with session ID 0 contains global effects applied to the output mix. 1829 // Effects in this chain can be insert or auxiliary. Effects in other chains (attached to 1830 // tracks) are insert only. The EffectChain maintains an ordered list of effect module, the 1831 // order corresponding in the effect process order. When attached to a track (session ID != 0), 1832 // it also provide it's own input buffer used by the track as accumulation buffer. 1833 class EffectChain : public RefBase { 1834 public: 1835 EffectChain(const wp<ThreadBase>& wThread, int sessionId); 1836 EffectChain(ThreadBase *thread, int sessionId); 1837 virtual ~EffectChain(); 1838 1839 // special key used for an entry in mSuspendedEffects keyed vector 1840 // corresponding to a suspend all request. 1841 static const int kKeyForSuspendAll = 0; 1842 1843 // minimum duration during which we force calling effect process when last track on 1844 // a session is stopped or removed to allow effect tail to be rendered 1845 static const int kProcessTailDurationMs = 1000; 1846 1847 void process_l(); 1848 1849 void lock() { 1850 mLock.lock(); 1851 } 1852 void unlock() { 1853 mLock.unlock(); 1854 } 1855 1856 status_t addEffect_l(const sp<EffectModule>& handle); 1857 size_t removeEffect_l(const sp<EffectModule>& handle); 1858 1859 int sessionId() const { return mSessionId; } 1860 void setSessionId(int sessionId) { mSessionId = sessionId; } 1861 1862 sp<EffectModule> getEffectFromDesc_l(effect_descriptor_t *descriptor); 1863 sp<EffectModule> getEffectFromId_l(int id); 1864 sp<EffectModule> getEffectFromType_l(const effect_uuid_t *type); 1865 bool setVolume_l(uint32_t *left, uint32_t *right); 1866 void setDevice_l(audio_devices_t device); 1867 void setMode_l(audio_mode_t mode); 1868 void setAudioSource_l(audio_source_t source); 1869 1870 void setInBuffer(int16_t *buffer, bool ownsBuffer = false) { 1871 mInBuffer = buffer; 1872 mOwnInBuffer = ownsBuffer; 1873 } 1874 int16_t *inBuffer() const { 1875 return mInBuffer; 1876 } 1877 void setOutBuffer(int16_t *buffer) { 1878 mOutBuffer = buffer; 1879 } 1880 int16_t *outBuffer() const { 1881 return mOutBuffer; 1882 } 1883 1884 void incTrackCnt() { android_atomic_inc(&mTrackCnt); } 1885 void decTrackCnt() { android_atomic_dec(&mTrackCnt); } 1886 int32_t trackCnt() const { return android_atomic_acquire_load(&mTrackCnt); } 1887 1888 void incActiveTrackCnt() { android_atomic_inc(&mActiveTrackCnt); 1889 mTailBufferCount = mMaxTailBuffers; } 1890 void decActiveTrackCnt() { android_atomic_dec(&mActiveTrackCnt); } 1891 int32_t activeTrackCnt() const { return android_atomic_acquire_load(&mActiveTrackCnt); } 1892 1893 uint32_t strategy() const { return mStrategy; } 1894 void setStrategy(uint32_t strategy) 1895 { mStrategy = strategy; } 1896 1897 // suspend effect of the given type 1898 void setEffectSuspended_l(const effect_uuid_t *type, 1899 bool suspend); 1900 // suspend all eligible effects 1901 void setEffectSuspendedAll_l(bool suspend); 1902 // check if effects should be suspend or restored when a given effect is enable or disabled 1903 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1904 bool enabled); 1905 1906 void clearInputBuffer(); 1907 1908 void dump(int fd, const Vector<String16>& args); 1909 1910 protected: 1911 friend class AudioFlinger; // for mThread, mEffects 1912 EffectChain(const EffectChain&); 1913 EffectChain& operator =(const EffectChain&); 1914 1915 class SuspendedEffectDesc : public RefBase { 1916 public: 1917 SuspendedEffectDesc() : mRefCount(0) {} 1918 1919 int mRefCount; 1920 effect_uuid_t mType; 1921 wp<EffectModule> mEffect; 1922 }; 1923 1924 // get a list of effect modules to suspend when an effect of the type 1925 // passed is enabled. 1926 void getSuspendEligibleEffects(Vector< sp<EffectModule> > &effects); 1927 1928 // get an effect module if it is currently enable 1929 sp<EffectModule> getEffectIfEnabled(const effect_uuid_t *type); 1930 // true if the effect whose descriptor is passed can be suspended 1931 // OEMs can modify the rules implemented in this method to exclude specific effect 1932 // types or implementations from the suspend/restore mechanism. 1933 bool isEffectEligibleForSuspend(const effect_descriptor_t& desc); 1934 1935 void clearInputBuffer_l(sp<ThreadBase> thread); 1936 1937 wp<ThreadBase> mThread; // parent mixer thread 1938 Mutex mLock; // mutex protecting effect list 1939 Vector< sp<EffectModule> > mEffects; // list of effect modules 1940 int mSessionId; // audio session ID 1941 int16_t *mInBuffer; // chain input buffer 1942 int16_t *mOutBuffer; // chain output buffer 1943 1944 // 'volatile' here means these are accessed with atomic operations instead of mutex 1945 volatile int32_t mActiveTrackCnt; // number of active tracks connected 1946 volatile int32_t mTrackCnt; // number of tracks connected 1947 1948 int32_t mTailBufferCount; // current effect tail buffer count 1949 int32_t mMaxTailBuffers; // maximum effect tail buffers 1950 bool mOwnInBuffer; // true if the chain owns its input buffer 1951 int mVolumeCtrlIdx; // index of insert effect having control over volume 1952 uint32_t mLeftVolume; // previous volume on left channel 1953 uint32_t mRightVolume; // previous volume on right channel 1954 uint32_t mNewLeftVolume; // new volume on left channel 1955 uint32_t mNewRightVolume; // new volume on right channel 1956 uint32_t mStrategy; // strategy for this effect chain 1957 // mSuspendedEffects lists all effects currently suspended in the chain. 1958 // Use effect type UUID timelow field as key. There is no real risk of identical 1959 // timeLow fields among effect type UUIDs. 1960 // Updated by updateSuspendedSessions_l() only. 1961 KeyedVector< int, sp<SuspendedEffectDesc> > mSuspendedEffects; 1962 }; 1963 1964 class AudioHwDevice { 1965 public: 1966 enum Flags { 1967 AHWD_CAN_SET_MASTER_VOLUME = 0x1, 1968 AHWD_CAN_SET_MASTER_MUTE = 0x2, 1969 }; 1970 1971 AudioHwDevice(const char *moduleName, 1972 audio_hw_device_t *hwDevice, 1973 Flags flags) 1974 : mModuleName(strdup(moduleName)) 1975 , mHwDevice(hwDevice) 1976 , mFlags(flags) { } 1977 /*virtual*/ ~AudioHwDevice() { free((void *)mModuleName); } 1978 1979 bool canSetMasterVolume() const { 1980 return (0 != (mFlags & AHWD_CAN_SET_MASTER_VOLUME)); 1981 } 1982 1983 bool canSetMasterMute() const { 1984 return (0 != (mFlags & AHWD_CAN_SET_MASTER_MUTE)); 1985 } 1986 1987 const char *moduleName() const { return mModuleName; } 1988 audio_hw_device_t *hwDevice() const { return mHwDevice; } 1989 private: 1990 const char * const mModuleName; 1991 audio_hw_device_t * const mHwDevice; 1992 Flags mFlags; 1993 }; 1994 1995 // AudioStreamOut and AudioStreamIn are immutable, so their fields are const. 1996 // For emphasis, we could also make all pointers to them be "const *", 1997 // but that would clutter the code unnecessarily. 1998 1999 struct AudioStreamOut { 2000 AudioHwDevice* const audioHwDev; 2001 audio_stream_out_t* const stream; 2002 2003 audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); } 2004 2005 AudioStreamOut(AudioHwDevice *dev, audio_stream_out_t *out) : 2006 audioHwDev(dev), stream(out) {} 2007 }; 2008 2009 struct AudioStreamIn { 2010 AudioHwDevice* const audioHwDev; 2011 audio_stream_in_t* const stream; 2012 2013 audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); } 2014 2015 AudioStreamIn(AudioHwDevice *dev, audio_stream_in_t *in) : 2016 audioHwDev(dev), stream(in) {} 2017 }; 2018 2019 // for mAudioSessionRefs only 2020 struct AudioSessionRef { 2021 AudioSessionRef(int sessionid, pid_t pid) : 2022 mSessionid(sessionid), mPid(pid), mCnt(1) {} 2023 const int mSessionid; 2024 const pid_t mPid; 2025 int mCnt; 2026 }; 2027 2028 mutable Mutex mLock; 2029 2030 DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client() 2031 2032 mutable Mutex mHardwareLock; 2033 // NOTE: If both mLock and mHardwareLock mutexes must be held, 2034 // always take mLock before mHardwareLock 2035 2036 // These two fields are immutable after onFirstRef(), so no lock needed to access 2037 AudioHwDevice* mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL 2038 DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*> mAudioHwDevs; 2039 2040 // for dump, indicates which hardware operation is currently in progress (but not stream ops) 2041 enum hardware_call_state { 2042 AUDIO_HW_IDLE = 0, // no operation in progress 2043 AUDIO_HW_INIT, // init_check 2044 AUDIO_HW_OUTPUT_OPEN, // open_output_stream 2045 AUDIO_HW_OUTPUT_CLOSE, // unused 2046 AUDIO_HW_INPUT_OPEN, // unused 2047 AUDIO_HW_INPUT_CLOSE, // unused 2048 AUDIO_HW_STANDBY, // unused 2049 AUDIO_HW_SET_MASTER_VOLUME, // set_master_volume 2050 AUDIO_HW_GET_ROUTING, // unused 2051 AUDIO_HW_SET_ROUTING, // unused 2052 AUDIO_HW_GET_MODE, // unused 2053 AUDIO_HW_SET_MODE, // set_mode 2054 AUDIO_HW_GET_MIC_MUTE, // get_mic_mute 2055 AUDIO_HW_SET_MIC_MUTE, // set_mic_mute 2056 AUDIO_HW_SET_VOICE_VOLUME, // set_voice_volume 2057 AUDIO_HW_SET_PARAMETER, // set_parameters 2058 AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size 2059 AUDIO_HW_GET_MASTER_VOLUME, // get_master_volume 2060 AUDIO_HW_GET_PARAMETER, // get_parameters 2061 AUDIO_HW_SET_MASTER_MUTE, // set_master_mute 2062 AUDIO_HW_GET_MASTER_MUTE, // get_master_mute 2063 }; 2064 2065 mutable hardware_call_state mHardwareStatus; // for dump only 2066 2067 2068 DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads; 2069 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 2070 2071 // member variables below are protected by mLock 2072 float mMasterVolume; 2073 bool mMasterMute; 2074 // end of variables protected by mLock 2075 2076 DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads; 2077 2078 DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients; 2079 volatile int32_t mNextUniqueId; // updated by android_atomic_inc 2080 audio_mode_t mMode; 2081 bool mBtNrecIsOff; 2082 2083 // protected by mLock 2084 Vector<AudioSessionRef*> mAudioSessionRefs; 2085 2086 float masterVolume_l() const; 2087 bool masterMute_l() const; 2088 audio_module_handle_t loadHwModule_l(const char *name); 2089 2090 Vector < sp<SyncEvent> > mPendingSyncEvents; // sync events awaiting for a session 2091 // to be created 2092 2093private: 2094 sp<Client> registerPid_l(pid_t pid); // always returns non-0 2095 2096 // for use from destructor 2097 status_t closeOutput_nonvirtual(audio_io_handle_t output); 2098 status_t closeInput_nonvirtual(audio_io_handle_t input); 2099 2100 // all record threads serially share a common tee sink, which is re-created on format change 2101 sp<NBAIO_Sink> mRecordTeeSink; 2102 sp<NBAIO_Source> mRecordTeeSource; 2103 2104public: 2105 static void dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id = 0); 2106}; 2107 2108 2109// ---------------------------------------------------------------------------- 2110 2111}; // namespace android 2112 2113#endif // ANDROID_AUDIO_FLINGER_H 2114