AudioFlinger.h revision 8af901cdea0af7e536579dee6d56e69987035a01
1/* 2** 3** Copyright 2007, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18#ifndef ANDROID_AUDIO_FLINGER_H 19#define ANDROID_AUDIO_FLINGER_H 20 21#include <stdint.h> 22#include <sys/types.h> 23#include <limits.h> 24 25#include <common_time/cc_helper.h> 26 27#include <media/IAudioFlinger.h> 28#include <media/IAudioFlingerClient.h> 29#include <media/IAudioTrack.h> 30#include <media/IAudioRecord.h> 31#include <media/AudioSystem.h> 32#include <media/AudioTrack.h> 33 34#include <utils/Atomic.h> 35#include <utils/Errors.h> 36#include <utils/threads.h> 37#include <utils/SortedVector.h> 38#include <utils/TypeHelpers.h> 39#include <utils/Vector.h> 40 41#include <binder/BinderService.h> 42#include <binder/MemoryDealer.h> 43 44#include <system/audio.h> 45#include <hardware/audio.h> 46#include <hardware/audio_policy.h> 47 48#include <media/AudioBufferProvider.h> 49#include <media/ExtendedAudioBufferProvider.h> 50#include "FastMixer.h" 51#include <media/nbaio/NBAIO.h> 52#include "AudioWatchdog.h" 53 54#include <powermanager/IPowerManager.h> 55 56namespace android { 57 58class audio_track_cblk_t; 59class effect_param_cblk_t; 60class AudioMixer; 61class AudioBuffer; 62class AudioResampler; 63class FastMixer; 64 65// ---------------------------------------------------------------------------- 66 67// AudioFlinger has a hard-coded upper limit of 2 channels for capture and playback. 68// There is support for > 2 channel tracks down-mixed to 2 channel output via a down-mix effect. 69// Adding full support for > 2 channel capture or playback would require more than simply changing 70// this #define. There is an independent hard-coded upper limit in AudioMixer; 71// removing that AudioMixer limit would be necessary but insufficient to support > 2 channels. 72// The macro FCC_2 highlights some (but not all) places where there is are 2-channel assumptions. 73// Search also for "2", "left", "right", "[0]", "[1]", ">> 16", "<< 16", etc. 74#define FCC_2 2 // FCC_2 = Fixed Channel Count 2 75 76static const nsecs_t kDefaultStandbyTimeInNsecs = seconds(3); 77 78class AudioFlinger : 79 public BinderService<AudioFlinger>, 80 public BnAudioFlinger 81{ 82 friend class BinderService<AudioFlinger>; // for AudioFlinger() 83public: 84 static const char* getServiceName() { return "media.audio_flinger"; } 85 86 virtual status_t dump(int fd, const Vector<String16>& args); 87 88 // IAudioFlinger interface, in binder opcode order 89 virtual sp<IAudioTrack> createTrack( 90 pid_t pid, 91 audio_stream_type_t streamType, 92 uint32_t sampleRate, 93 audio_format_t format, 94 audio_channel_mask_t channelMask, 95 int frameCount, 96 IAudioFlinger::track_flags_t flags, 97 const sp<IMemory>& sharedBuffer, 98 audio_io_handle_t output, 99 pid_t tid, 100 int *sessionId, 101 status_t *status); 102 103 virtual sp<IAudioRecord> openRecord( 104 pid_t pid, 105 audio_io_handle_t input, 106 uint32_t sampleRate, 107 audio_format_t format, 108 audio_channel_mask_t channelMask, 109 int frameCount, 110 IAudioFlinger::track_flags_t flags, 111 pid_t tid, 112 int *sessionId, 113 status_t *status); 114 115 virtual uint32_t sampleRate(audio_io_handle_t output) const; 116 virtual int channelCount(audio_io_handle_t output) const; 117 virtual audio_format_t format(audio_io_handle_t output) const; 118 virtual size_t frameCount(audio_io_handle_t output) const; 119 virtual uint32_t latency(audio_io_handle_t output) const; 120 121 virtual status_t setMasterVolume(float value); 122 virtual status_t setMasterMute(bool muted); 123 124 virtual float masterVolume() const; 125 virtual bool masterMute() const; 126 127 virtual status_t setStreamVolume(audio_stream_type_t stream, float value, 128 audio_io_handle_t output); 129 virtual status_t setStreamMute(audio_stream_type_t stream, bool muted); 130 131 virtual float streamVolume(audio_stream_type_t stream, 132 audio_io_handle_t output) const; 133 virtual bool streamMute(audio_stream_type_t stream) const; 134 135 virtual status_t setMode(audio_mode_t mode); 136 137 virtual status_t setMicMute(bool state); 138 virtual bool getMicMute() const; 139 140 virtual status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs); 141 virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) const; 142 143 virtual void registerClient(const sp<IAudioFlingerClient>& client); 144 145 virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, 146 audio_channel_mask_t channelMask) const; 147 148 virtual audio_io_handle_t openOutput(audio_module_handle_t module, 149 audio_devices_t *pDevices, 150 uint32_t *pSamplingRate, 151 audio_format_t *pFormat, 152 audio_channel_mask_t *pChannelMask, 153 uint32_t *pLatencyMs, 154 audio_output_flags_t flags); 155 156 virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1, 157 audio_io_handle_t output2); 158 159 virtual status_t closeOutput(audio_io_handle_t output); 160 161 virtual status_t suspendOutput(audio_io_handle_t output); 162 163 virtual status_t restoreOutput(audio_io_handle_t output); 164 165 virtual audio_io_handle_t openInput(audio_module_handle_t module, 166 audio_devices_t *pDevices, 167 uint32_t *pSamplingRate, 168 audio_format_t *pFormat, 169 audio_channel_mask_t *pChannelMask); 170 171 virtual status_t closeInput(audio_io_handle_t input); 172 173 virtual status_t setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output); 174 175 virtual status_t setVoiceVolume(float volume); 176 177 virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 178 audio_io_handle_t output) const; 179 180 virtual unsigned int getInputFramesLost(audio_io_handle_t ioHandle) const; 181 182 virtual int newAudioSessionId(); 183 184 virtual void acquireAudioSessionId(int audioSession); 185 186 virtual void releaseAudioSessionId(int audioSession); 187 188 virtual status_t queryNumberEffects(uint32_t *numEffects) const; 189 190 virtual status_t queryEffect(uint32_t index, effect_descriptor_t *descriptor) const; 191 192 virtual status_t getEffectDescriptor(const effect_uuid_t *pUuid, 193 effect_descriptor_t *descriptor) const; 194 195 virtual sp<IEffect> createEffect(pid_t pid, 196 effect_descriptor_t *pDesc, 197 const sp<IEffectClient>& effectClient, 198 int32_t priority, 199 audio_io_handle_t io, 200 int sessionId, 201 status_t *status, 202 int *id, 203 int *enabled); 204 205 virtual status_t moveEffects(int sessionId, audio_io_handle_t srcOutput, 206 audio_io_handle_t dstOutput); 207 208 virtual audio_module_handle_t loadHwModule(const char *name); 209 210 virtual int32_t getPrimaryOutputSamplingRate(); 211 virtual int32_t getPrimaryOutputFrameCount(); 212 213 virtual status_t onTransact( 214 uint32_t code, 215 const Parcel& data, 216 Parcel* reply, 217 uint32_t flags); 218 219 // end of IAudioFlinger interface 220 221 class SyncEvent; 222 223 typedef void (*sync_event_callback_t)(const wp<SyncEvent>& event) ; 224 225 class SyncEvent : public RefBase { 226 public: 227 SyncEvent(AudioSystem::sync_event_t type, 228 int triggerSession, 229 int listenerSession, 230 sync_event_callback_t callBack, 231 void *cookie) 232 : mType(type), mTriggerSession(triggerSession), mListenerSession(listenerSession), 233 mCallback(callBack), mCookie(cookie) 234 {} 235 236 virtual ~SyncEvent() {} 237 238 void trigger() { Mutex::Autolock _l(mLock); if (mCallback) mCallback(this); } 239 bool isCancelled() const { Mutex::Autolock _l(mLock); return (mCallback == NULL); } 240 void cancel() { Mutex::Autolock _l(mLock); mCallback = NULL; } 241 AudioSystem::sync_event_t type() const { return mType; } 242 int triggerSession() const { return mTriggerSession; } 243 int listenerSession() const { return mListenerSession; } 244 void *cookie() const { return mCookie; } 245 246 private: 247 const AudioSystem::sync_event_t mType; 248 const int mTriggerSession; 249 const int mListenerSession; 250 sync_event_callback_t mCallback; 251 void * const mCookie; 252 mutable Mutex mLock; 253 }; 254 255 sp<SyncEvent> createSyncEvent(AudioSystem::sync_event_t type, 256 int triggerSession, 257 int listenerSession, 258 sync_event_callback_t callBack, 259 void *cookie); 260 261private: 262 class AudioHwDevice; // fwd declaration for findSuitableHwDev_l 263 264 audio_mode_t getMode() const { return mMode; } 265 266 bool btNrecIsOff() const { return mBtNrecIsOff; } 267 268 AudioFlinger(); 269 virtual ~AudioFlinger(); 270 271 // call in any IAudioFlinger method that accesses mPrimaryHardwareDev 272 status_t initCheck() const { return mPrimaryHardwareDev == NULL ? 273 NO_INIT : NO_ERROR; } 274 275 // RefBase 276 virtual void onFirstRef(); 277 278 AudioHwDevice* findSuitableHwDev_l(audio_module_handle_t module, 279 audio_devices_t devices); 280 void purgeStaleEffects_l(); 281 282 // standby delay for MIXER and DUPLICATING playback threads is read from property 283 // ro.audio.flinger_standbytime_ms or defaults to kDefaultStandbyTimeInNsecs 284 static nsecs_t mStandbyTimeInNsecs; 285 286 // Internal dump utilities. 287 void dumpPermissionDenial(int fd, const Vector<String16>& args); 288 void dumpClients(int fd, const Vector<String16>& args); 289 void dumpInternals(int fd, const Vector<String16>& args); 290 291 // --- Client --- 292 class Client : public RefBase { 293 public: 294 Client(const sp<AudioFlinger>& audioFlinger, pid_t pid); 295 virtual ~Client(); 296 sp<MemoryDealer> heap() const; 297 pid_t pid() const { return mPid; } 298 sp<AudioFlinger> audioFlinger() const { return mAudioFlinger; } 299 300 bool reserveTimedTrack(); 301 void releaseTimedTrack(); 302 303 private: 304 Client(const Client&); 305 Client& operator = (const Client&); 306 const sp<AudioFlinger> mAudioFlinger; 307 const sp<MemoryDealer> mMemoryDealer; 308 const pid_t mPid; 309 310 Mutex mTimedTrackLock; 311 int mTimedTrackCount; 312 }; 313 314 // --- Notification Client --- 315 class NotificationClient : public IBinder::DeathRecipient { 316 public: 317 NotificationClient(const sp<AudioFlinger>& audioFlinger, 318 const sp<IAudioFlingerClient>& client, 319 pid_t pid); 320 virtual ~NotificationClient(); 321 322 sp<IAudioFlingerClient> audioFlingerClient() const { return mAudioFlingerClient; } 323 324 // IBinder::DeathRecipient 325 virtual void binderDied(const wp<IBinder>& who); 326 327 private: 328 NotificationClient(const NotificationClient&); 329 NotificationClient& operator = (const NotificationClient&); 330 331 const sp<AudioFlinger> mAudioFlinger; 332 const pid_t mPid; 333 const sp<IAudioFlingerClient> mAudioFlingerClient; 334 }; 335 336 class TrackHandle; 337 class RecordHandle; 338 class RecordThread; 339 class PlaybackThread; 340 class MixerThread; 341 class DirectOutputThread; 342 class DuplicatingThread; 343 class Track; 344 class RecordTrack; 345 class EffectModule; 346 class EffectHandle; 347 class EffectChain; 348 struct AudioStreamOut; 349 struct AudioStreamIn; 350 351 class ThreadBase : public Thread { 352 public: 353 354 enum type_t { 355 MIXER, // Thread class is MixerThread 356 DIRECT, // Thread class is DirectOutputThread 357 DUPLICATING, // Thread class is DuplicatingThread 358 RECORD // Thread class is RecordThread 359 }; 360 361 ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 362 audio_devices_t outDevice, audio_devices_t inDevice, type_t type); 363 virtual ~ThreadBase(); 364 365 void dumpBase(int fd, const Vector<String16>& args); 366 void dumpEffectChains(int fd, const Vector<String16>& args); 367 368 void clearPowerManager(); 369 370 // base for record and playback 371 class TrackBase : public ExtendedAudioBufferProvider, public RefBase { 372 373 public: 374 enum track_state { 375 IDLE, 376 TERMINATED, 377 FLUSHED, 378 STOPPED, 379 // next 2 states are currently used for fast tracks only 380 STOPPING_1, // waiting for first underrun 381 STOPPING_2, // waiting for presentation complete 382 RESUMING, 383 ACTIVE, 384 PAUSING, 385 PAUSED 386 }; 387 388 TrackBase(ThreadBase *thread, 389 const sp<Client>& client, 390 uint32_t sampleRate, 391 audio_format_t format, 392 audio_channel_mask_t channelMask, 393 int frameCount, 394 const sp<IMemory>& sharedBuffer, 395 int sessionId); 396 virtual ~TrackBase(); 397 398 virtual status_t start(AudioSystem::sync_event_t event, 399 int triggerSession) = 0; 400 virtual void stop() = 0; 401 sp<IMemory> getCblk() const { return mCblkMemory; } 402 audio_track_cblk_t* cblk() const { return mCblk; } 403 int sessionId() const { return mSessionId; } 404 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 405 406 protected: 407 TrackBase(const TrackBase&); 408 TrackBase& operator = (const TrackBase&); 409 410 // AudioBufferProvider interface 411 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) = 0; 412 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 413 414 // ExtendedAudioBufferProvider interface is only needed for Track, 415 // but putting it in TrackBase avoids the complexity of virtual inheritance 416 virtual size_t framesReady() const { return SIZE_MAX; } 417 418 audio_format_t format() const { 419 return mFormat; 420 } 421 422 int channelCount() const { return mChannelCount; } 423 424 audio_channel_mask_t channelMask() const { return mChannelMask; } 425 426 int sampleRate() const; // FIXME inline after cblk sr moved 427 428 // Return a pointer to the start of a contiguous slice of the track buffer. 429 // Parameter 'offset' is the requested start position, expressed in 430 // monotonically increasing frame units relative to the track epoch. 431 // Parameter 'frames' is the requested length, also in frame units. 432 // Always returns non-NULL. It is the caller's responsibility to 433 // verify that this will be successful; the result of calling this 434 // function with invalid 'offset' or 'frames' is undefined. 435 void* getBuffer(uint32_t offset, uint32_t frames) const; 436 437 bool isStopped() const { 438 return (mState == STOPPED || mState == FLUSHED); 439 } 440 441 // for fast tracks only 442 bool isStopping() const { 443 return mState == STOPPING_1 || mState == STOPPING_2; 444 } 445 bool isStopping_1() const { 446 return mState == STOPPING_1; 447 } 448 bool isStopping_2() const { 449 return mState == STOPPING_2; 450 } 451 452 bool isTerminated() const { 453 return mState == TERMINATED; 454 } 455 456 bool step(); 457 void reset(); 458 459 const wp<ThreadBase> mThread; 460 /*const*/ sp<Client> mClient; // see explanation at ~TrackBase() why not const 461 sp<IMemory> mCblkMemory; 462 audio_track_cblk_t* mCblk; 463 void* mBuffer; // start of track buffer, typically in shared memory 464 void* mBufferEnd; // &mBuffer[mFrameCount * frameSize], where frameSize 465 // is based on mChannelCount and 16-bit samples 466 uint32_t mFrameCount; 467 // we don't really need a lock for these 468 track_state mState; 469 const uint32_t mSampleRate; // initial sample rate only; for tracks which 470 // support dynamic rates, the current value is in control block 471 const audio_format_t mFormat; 472 bool mStepServerFailed; 473 const int mSessionId; 474 uint8_t mChannelCount; 475 audio_channel_mask_t mChannelMask; 476 Vector < sp<SyncEvent> >mSyncEvents; 477 }; 478 479 enum { 480 CFG_EVENT_IO, 481 CFG_EVENT_PRIO 482 }; 483 484 class ConfigEvent { 485 public: 486 ConfigEvent(int type) : mType(type) {} 487 virtual ~ConfigEvent() {} 488 489 int type() const { return mType; } 490 491 virtual void dump(char *buffer, size_t size) = 0; 492 493 private: 494 const int mType; 495 }; 496 497 class IoConfigEvent : public ConfigEvent { 498 public: 499 IoConfigEvent(int event, int param) : 500 ConfigEvent(CFG_EVENT_IO), mEvent(event), mParam(event) {} 501 virtual ~IoConfigEvent() {} 502 503 int event() const { return mEvent; } 504 int param() const { return mParam; } 505 506 virtual void dump(char *buffer, size_t size) { 507 snprintf(buffer, size, "IO event: event %d, param %d\n", mEvent, mParam); 508 } 509 510 private: 511 const int mEvent; 512 const int mParam; 513 }; 514 515 class PrioConfigEvent : public ConfigEvent { 516 public: 517 PrioConfigEvent(pid_t pid, pid_t tid, int32_t prio) : 518 ConfigEvent(CFG_EVENT_PRIO), mPid(pid), mTid(tid), mPrio(prio) {} 519 virtual ~PrioConfigEvent() {} 520 521 pid_t pid() const { return mPid; } 522 pid_t tid() const { return mTid; } 523 int32_t prio() const { return mPrio; } 524 525 virtual void dump(char *buffer, size_t size) { 526 snprintf(buffer, size, "Prio event: pid %d, tid %d, prio %d\n", mPid, mTid, mPrio); 527 } 528 529 private: 530 const pid_t mPid; 531 const pid_t mTid; 532 const int32_t mPrio; 533 }; 534 535 536 class PMDeathRecipient : public IBinder::DeathRecipient { 537 public: 538 PMDeathRecipient(const wp<ThreadBase>& thread) : mThread(thread) {} 539 virtual ~PMDeathRecipient() {} 540 541 // IBinder::DeathRecipient 542 virtual void binderDied(const wp<IBinder>& who); 543 544 private: 545 PMDeathRecipient(const PMDeathRecipient&); 546 PMDeathRecipient& operator = (const PMDeathRecipient&); 547 548 wp<ThreadBase> mThread; 549 }; 550 551 virtual status_t initCheck() const = 0; 552 553 // static externally-visible 554 type_t type() const { return mType; } 555 audio_io_handle_t id() const { return mId;} 556 557 // dynamic externally-visible 558 uint32_t sampleRate() const { return mSampleRate; } 559 int channelCount() const { return mChannelCount; } 560 audio_channel_mask_t channelMask() const { return mChannelMask; } 561 audio_format_t format() const { return mFormat; } 562 // Called by AudioFlinger::frameCount(audio_io_handle_t output) and effects, 563 // and returns the normal mix buffer's frame count. 564 size_t frameCount() const { return mNormalFrameCount; } 565 // Return's the HAL's frame count i.e. fast mixer buffer size. 566 size_t frameCountHAL() const { return mFrameCount; } 567 568 // Should be "virtual status_t requestExitAndWait()" and override same 569 // method in Thread, but Thread::requestExitAndWait() is not yet virtual. 570 void exit(); 571 virtual bool checkForNewParameters_l() = 0; 572 virtual status_t setParameters(const String8& keyValuePairs); 573 virtual String8 getParameters(const String8& keys) = 0; 574 virtual void audioConfigChanged_l(int event, int param = 0) = 0; 575 void sendIoConfigEvent(int event, int param = 0); 576 void sendIoConfigEvent_l(int event, int param = 0); 577 void sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio); 578 void processConfigEvents(); 579 580 // see note at declaration of mStandby, mOutDevice and mInDevice 581 bool standby() const { return mStandby; } 582 audio_devices_t outDevice() const { return mOutDevice; } 583 audio_devices_t inDevice() const { return mInDevice; } 584 585 virtual audio_stream_t* stream() const = 0; 586 587 sp<EffectHandle> createEffect_l( 588 const sp<AudioFlinger::Client>& client, 589 const sp<IEffectClient>& effectClient, 590 int32_t priority, 591 int sessionId, 592 effect_descriptor_t *desc, 593 int *enabled, 594 status_t *status); 595 void disconnectEffect(const sp< EffectModule>& effect, 596 EffectHandle *handle, 597 bool unpinIfLast); 598 599 // return values for hasAudioSession (bit field) 600 enum effect_state { 601 EFFECT_SESSION = 0x1, // the audio session corresponds to at least one 602 // effect 603 TRACK_SESSION = 0x2 // the audio session corresponds to at least one 604 // track 605 }; 606 607 // get effect chain corresponding to session Id. 608 sp<EffectChain> getEffectChain(int sessionId); 609 // same as getEffectChain() but must be called with ThreadBase mutex locked 610 sp<EffectChain> getEffectChain_l(int sessionId) const; 611 // add an effect chain to the chain list (mEffectChains) 612 virtual status_t addEffectChain_l(const sp<EffectChain>& chain) = 0; 613 // remove an effect chain from the chain list (mEffectChains) 614 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain) = 0; 615 // lock all effect chains Mutexes. Must be called before releasing the 616 // ThreadBase mutex before processing the mixer and effects. This guarantees the 617 // integrity of the chains during the process. 618 // Also sets the parameter 'effectChains' to current value of mEffectChains. 619 void lockEffectChains_l(Vector< sp<EffectChain> >& effectChains); 620 // unlock effect chains after process 621 void unlockEffectChains(const Vector< sp<EffectChain> >& effectChains); 622 // set audio mode to all effect chains 623 void setMode(audio_mode_t mode); 624 // get effect module with corresponding ID on specified audio session 625 sp<AudioFlinger::EffectModule> getEffect(int sessionId, int effectId); 626 sp<AudioFlinger::EffectModule> getEffect_l(int sessionId, int effectId); 627 // add and effect module. Also creates the effect chain is none exists for 628 // the effects audio session 629 status_t addEffect_l(const sp< EffectModule>& effect); 630 // remove and effect module. Also removes the effect chain is this was the last 631 // effect 632 void removeEffect_l(const sp< EffectModule>& effect); 633 // detach all tracks connected to an auxiliary effect 634 virtual void detachAuxEffect_l(int effectId) {} 635 // returns either EFFECT_SESSION if effects on this audio session exist in one 636 // chain, or TRACK_SESSION if tracks on this audio session exist, or both 637 virtual uint32_t hasAudioSession(int sessionId) const = 0; 638 // the value returned by default implementation is not important as the 639 // strategy is only meaningful for PlaybackThread which implements this method 640 virtual uint32_t getStrategyForSession_l(int sessionId) { return 0; } 641 642 // suspend or restore effect according to the type of effect passed. a NULL 643 // type pointer means suspend all effects in the session 644 void setEffectSuspended(const effect_uuid_t *type, 645 bool suspend, 646 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 647 // check if some effects must be suspended/restored when an effect is enabled 648 // or disabled 649 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 650 bool enabled, 651 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 652 void checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 653 bool enabled, 654 int sessionId = AUDIO_SESSION_OUTPUT_MIX); 655 656 virtual status_t setSyncEvent(const sp<SyncEvent>& event) = 0; 657 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const = 0; 658 659 660 mutable Mutex mLock; 661 662 protected: 663 664 // entry describing an effect being suspended in mSuspendedSessions keyed vector 665 class SuspendedSessionDesc : public RefBase { 666 public: 667 SuspendedSessionDesc() : mRefCount(0) {} 668 669 int mRefCount; // number of active suspend requests 670 effect_uuid_t mType; // effect type UUID 671 }; 672 673 void acquireWakeLock(); 674 void acquireWakeLock_l(); 675 void releaseWakeLock(); 676 void releaseWakeLock_l(); 677 void setEffectSuspended_l(const effect_uuid_t *type, 678 bool suspend, 679 int sessionId); 680 // updated mSuspendedSessions when an effect suspended or restored 681 void updateSuspendedSessions_l(const effect_uuid_t *type, 682 bool suspend, 683 int sessionId); 684 // check if some effects must be suspended when an effect chain is added 685 void checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain); 686 687 virtual void preExit() { } 688 689 friend class AudioFlinger; // for mEffectChains 690 691 const type_t mType; 692 693 // Used by parameters, config events, addTrack_l, exit 694 Condition mWaitWorkCV; 695 696 const sp<AudioFlinger> mAudioFlinger; 697 uint32_t mSampleRate; 698 size_t mFrameCount; // output HAL, direct output, record 699 size_t mNormalFrameCount; // normal mixer and effects 700 audio_channel_mask_t mChannelMask; 701 uint16_t mChannelCount; 702 size_t mFrameSize; 703 audio_format_t mFormat; 704 705 // Parameter sequence by client: binder thread calling setParameters(): 706 // 1. Lock mLock 707 // 2. Append to mNewParameters 708 // 3. mWaitWorkCV.signal 709 // 4. mParamCond.waitRelative with timeout 710 // 5. read mParamStatus 711 // 6. mWaitWorkCV.signal 712 // 7. Unlock 713 // 714 // Parameter sequence by server: threadLoop calling checkForNewParameters_l(): 715 // 1. Lock mLock 716 // 2. If there is an entry in mNewParameters proceed ... 717 // 2. Read first entry in mNewParameters 718 // 3. Process 719 // 4. Remove first entry from mNewParameters 720 // 5. Set mParamStatus 721 // 6. mParamCond.signal 722 // 7. mWaitWorkCV.wait with timeout (this is to avoid overwriting mParamStatus) 723 // 8. Unlock 724 Condition mParamCond; 725 Vector<String8> mNewParameters; 726 status_t mParamStatus; 727 728 Vector<ConfigEvent *> mConfigEvents; 729 730 // These fields are written and read by thread itself without lock or barrier, 731 // and read by other threads without lock or barrier via standby() , outDevice() 732 // and inDevice(). 733 // Because of the absence of a lock or barrier, any other thread that reads 734 // these fields must use the information in isolation, or be prepared to deal 735 // with possibility that it might be inconsistent with other information. 736 bool mStandby; // Whether thread is currently in standby. 737 audio_devices_t mOutDevice; // output device 738 audio_devices_t mInDevice; // input device 739 audio_source_t mAudioSource; // (see audio.h, audio_source_t) 740 741 const audio_io_handle_t mId; 742 Vector< sp<EffectChain> > mEffectChains; 743 744 static const int kNameLength = 16; // prctl(PR_SET_NAME) limit 745 char mName[kNameLength]; 746 sp<IPowerManager> mPowerManager; 747 sp<IBinder> mWakeLockToken; 748 const sp<PMDeathRecipient> mDeathRecipient; 749 // list of suspended effects per session and per type. The first vector is 750 // keyed by session ID, the second by type UUID timeLow field 751 KeyedVector< int, KeyedVector< int, sp<SuspendedSessionDesc> > > 752 mSuspendedSessions; 753 }; 754 755 struct stream_type_t { 756 stream_type_t() 757 : volume(1.0f), 758 mute(false) 759 { 760 } 761 float volume; 762 bool mute; 763 }; 764 765 // --- PlaybackThread --- 766 class PlaybackThread : public ThreadBase { 767 public: 768 769 enum mixer_state { 770 MIXER_IDLE, // no active tracks 771 MIXER_TRACKS_ENABLED, // at least one active track, but no track has any data ready 772 MIXER_TRACKS_READY // at least one active track, and at least one track has data 773 // standby mode does not have an enum value 774 // suspend by audio policy manager is orthogonal to mixer state 775 }; 776 777 // playback track 778 class Track : public TrackBase, public VolumeProvider { 779 public: 780 Track( PlaybackThread *thread, 781 const sp<Client>& client, 782 audio_stream_type_t streamType, 783 uint32_t sampleRate, 784 audio_format_t format, 785 audio_channel_mask_t channelMask, 786 int frameCount, 787 const sp<IMemory>& sharedBuffer, 788 int sessionId, 789 IAudioFlinger::track_flags_t flags); 790 virtual ~Track(); 791 792 static void appendDumpHeader(String8& result); 793 void dump(char* buffer, size_t size); 794 virtual status_t start(AudioSystem::sync_event_t event = 795 AudioSystem::SYNC_EVENT_NONE, 796 int triggerSession = 0); 797 virtual void stop(); 798 void pause(); 799 800 void flush(); 801 void destroy(); 802 void mute(bool); 803 int name() const { return mName; } 804 805 audio_stream_type_t streamType() const { 806 return mStreamType; 807 } 808 status_t attachAuxEffect(int EffectId); 809 void setAuxBuffer(int EffectId, int32_t *buffer); 810 int32_t *auxBuffer() const { return mAuxBuffer; } 811 void setMainBuffer(int16_t *buffer) { mMainBuffer = buffer; } 812 int16_t *mainBuffer() const { return mMainBuffer; } 813 int auxEffectId() const { return mAuxEffectId; } 814 815 // implement FastMixerState::VolumeProvider interface 816 virtual uint32_t getVolumeLR(); 817 818 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 819 820 protected: 821 // for numerous 822 friend class PlaybackThread; 823 friend class MixerThread; 824 friend class DirectOutputThread; 825 826 Track(const Track&); 827 Track& operator = (const Track&); 828 829 // AudioBufferProvider interface 830 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, 831 int64_t pts = kInvalidPTS); 832 // releaseBuffer() not overridden 833 834 virtual size_t framesReady() const; 835 836 bool isMuted() const { return mMute; } 837 bool isPausing() const { 838 return mState == PAUSING; 839 } 840 bool isPaused() const { 841 return mState == PAUSED; 842 } 843 bool isResuming() const { 844 return mState == RESUMING; 845 } 846 bool isReady() const; 847 void setPaused() { mState = PAUSED; } 848 void reset(); 849 850 bool isOutputTrack() const { 851 return (mStreamType == AUDIO_STREAM_CNT); 852 } 853 854 sp<IMemory> sharedBuffer() const { return mSharedBuffer; } 855 856 bool presentationComplete(size_t framesWritten, size_t audioHalFrames); 857 858 public: 859 void triggerEvents(AudioSystem::sync_event_t type); 860 virtual bool isTimedTrack() const { return false; } 861 bool isFastTrack() const { return (mFlags & IAudioFlinger::TRACK_FAST) != 0; } 862 863 protected: 864 865 // written by Track::mute() called by binder thread(s), without a mutex or barrier. 866 // read by Track::isMuted() called by playback thread, also without a mutex or barrier. 867 // The lack of mutex or barrier is safe because the mute status is only used by itself. 868 bool mMute; 869 870 // FILLED state is used for suppressing volume ramp at begin of playing 871 enum {FS_INVALID, FS_FILLING, FS_FILLED, FS_ACTIVE}; 872 mutable uint8_t mFillingUpStatus; 873 int8_t mRetryCount; 874 const sp<IMemory> mSharedBuffer; 875 bool mResetDone; 876 const audio_stream_type_t mStreamType; 877 int mName; // track name on the normal mixer, 878 // allocated statically at track creation time, 879 // and is even allocated (though unused) for fast tracks 880 // FIXME don't allocate track name for fast tracks 881 int16_t *mMainBuffer; 882 int32_t *mAuxBuffer; 883 int mAuxEffectId; 884 bool mHasVolumeController; 885 size_t mPresentationCompleteFrames; // number of frames written to the 886 // audio HAL when this track will be fully rendered 887 private: 888 IAudioFlinger::track_flags_t mFlags; 889 890 // The following fields are only for fast tracks, and should be in a subclass 891 int mFastIndex; // index within FastMixerState::mFastTracks[]; 892 // either mFastIndex == -1 if not isFastTrack() 893 // or 0 < mFastIndex < FastMixerState::kMaxFast because 894 // index 0 is reserved for normal mixer's submix; 895 // index is allocated statically at track creation time 896 // but the slot is only used if track is active 897 FastTrackUnderruns mObservedUnderruns; // Most recently observed value of 898 // mFastMixerDumpState.mTracks[mFastIndex].mUnderruns 899 uint32_t mUnderrunCount; // Counter of total number of underruns, never reset 900 volatile float mCachedVolume; // combined master volume and stream type volume; 901 // 'volatile' means accessed without lock or 902 // barrier, but is read/written atomically 903 }; // end of Track 904 905 class TimedTrack : public Track { 906 public: 907 static sp<TimedTrack> create(PlaybackThread *thread, 908 const sp<Client>& client, 909 audio_stream_type_t streamType, 910 uint32_t sampleRate, 911 audio_format_t format, 912 audio_channel_mask_t channelMask, 913 int frameCount, 914 const sp<IMemory>& sharedBuffer, 915 int sessionId); 916 virtual ~TimedTrack(); 917 918 class TimedBuffer { 919 public: 920 TimedBuffer(); 921 TimedBuffer(const sp<IMemory>& buffer, int64_t pts); 922 const sp<IMemory>& buffer() const { return mBuffer; } 923 int64_t pts() const { return mPTS; } 924 uint32_t position() const { return mPosition; } 925 void setPosition(uint32_t pos) { mPosition = pos; } 926 private: 927 sp<IMemory> mBuffer; 928 int64_t mPTS; 929 uint32_t mPosition; 930 }; 931 932 // Mixer facing methods. 933 virtual bool isTimedTrack() const { return true; } 934 virtual size_t framesReady() const; 935 936 // AudioBufferProvider interface 937 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, 938 int64_t pts); 939 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 940 941 // Client/App facing methods. 942 status_t allocateTimedBuffer(size_t size, 943 sp<IMemory>* buffer); 944 status_t queueTimedBuffer(const sp<IMemory>& buffer, 945 int64_t pts); 946 status_t setMediaTimeTransform(const LinearTransform& xform, 947 TimedAudioTrack::TargetTimeline target); 948 949 private: 950 TimedTrack(PlaybackThread *thread, 951 const sp<Client>& client, 952 audio_stream_type_t streamType, 953 uint32_t sampleRate, 954 audio_format_t format, 955 audio_channel_mask_t channelMask, 956 int frameCount, 957 const sp<IMemory>& sharedBuffer, 958 int sessionId); 959 960 void timedYieldSamples_l(AudioBufferProvider::Buffer* buffer); 961 void timedYieldSilence_l(uint32_t numFrames, 962 AudioBufferProvider::Buffer* buffer); 963 void trimTimedBufferQueue_l(); 964 void trimTimedBufferQueueHead_l(const char* logTag); 965 void updateFramesPendingAfterTrim_l(const TimedBuffer& buf, 966 const char* logTag); 967 968 uint64_t mLocalTimeFreq; 969 LinearTransform mLocalTimeToSampleTransform; 970 LinearTransform mMediaTimeToSampleTransform; 971 sp<MemoryDealer> mTimedMemoryDealer; 972 973 Vector<TimedBuffer> mTimedBufferQueue; 974 bool mQueueHeadInFlight; 975 bool mTrimQueueHeadOnRelease; 976 uint32_t mFramesPendingInQueue; 977 978 uint8_t* mTimedSilenceBuffer; 979 uint32_t mTimedSilenceBufferSize; 980 mutable Mutex mTimedBufferQueueLock; 981 bool mTimedAudioOutputOnTime; 982 CCHelper mCCHelper; 983 984 Mutex mMediaTimeTransformLock; 985 LinearTransform mMediaTimeTransform; 986 bool mMediaTimeTransformValid; 987 TimedAudioTrack::TargetTimeline mMediaTimeTransformTarget; 988 }; 989 990 991 // playback track 992 class OutputTrack : public Track { 993 public: 994 995 class Buffer : public AudioBufferProvider::Buffer { 996 public: 997 int16_t *mBuffer; 998 }; 999 1000 OutputTrack(PlaybackThread *thread, 1001 DuplicatingThread *sourceThread, 1002 uint32_t sampleRate, 1003 audio_format_t format, 1004 audio_channel_mask_t channelMask, 1005 int frameCount); 1006 virtual ~OutputTrack(); 1007 1008 virtual status_t start(AudioSystem::sync_event_t event = 1009 AudioSystem::SYNC_EVENT_NONE, 1010 int triggerSession = 0); 1011 virtual void stop(); 1012 bool write(int16_t* data, uint32_t frames); 1013 bool bufferQueueEmpty() const { return mBufferQueue.size() == 0; } 1014 bool isActive() const { return mActive; } 1015 const wp<ThreadBase>& thread() const { return mThread; } 1016 1017 private: 1018 1019 enum { 1020 NO_MORE_BUFFERS = 0x80000001, // same in AudioTrack.h, ok to be different value 1021 }; 1022 1023 status_t obtainBuffer(AudioBufferProvider::Buffer* buffer, 1024 uint32_t waitTimeMs); 1025 void clearBufferQueue(); 1026 1027 // Maximum number of pending buffers allocated by OutputTrack::write() 1028 static const uint8_t kMaxOverFlowBuffers = 10; 1029 1030 Vector < Buffer* > mBufferQueue; 1031 AudioBufferProvider::Buffer mOutBuffer; 1032 bool mActive; 1033 DuplicatingThread* const mSourceThread; // for waitTimeMs() in write() 1034 }; // end of OutputTrack 1035 1036 PlaybackThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1037 audio_io_handle_t id, audio_devices_t device, type_t type); 1038 virtual ~PlaybackThread(); 1039 1040 void dump(int fd, const Vector<String16>& args); 1041 1042 // Thread virtuals 1043 virtual status_t readyToRun(); 1044 virtual bool threadLoop(); 1045 1046 // RefBase 1047 virtual void onFirstRef(); 1048 1049protected: 1050 // Code snippets that were lifted up out of threadLoop() 1051 virtual void threadLoop_mix() = 0; 1052 virtual void threadLoop_sleepTime() = 0; 1053 virtual void threadLoop_write(); 1054 virtual void threadLoop_standby(); 1055 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 1056 1057 // prepareTracks_l reads and writes mActiveTracks, and returns 1058 // the pending set of tracks to remove via Vector 'tracksToRemove'. The caller 1059 // is responsible for clearing or destroying this Vector later on, when it 1060 // is safe to do so. That will drop the final ref count and destroy the tracks. 1061 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove) = 0; 1062 1063 // ThreadBase virtuals 1064 virtual void preExit(); 1065 1066public: 1067 1068 virtual status_t initCheck() const { return (mOutput == NULL) ? NO_INIT : NO_ERROR; } 1069 1070 // return estimated latency in milliseconds, as reported by HAL 1071 uint32_t latency() const; 1072 // same, but lock must already be held 1073 uint32_t latency_l() const; 1074 1075 void setMasterVolume(float value); 1076 void setMasterMute(bool muted); 1077 1078 void setStreamVolume(audio_stream_type_t stream, float value); 1079 void setStreamMute(audio_stream_type_t stream, bool muted); 1080 1081 float streamVolume(audio_stream_type_t stream) const; 1082 1083 sp<Track> createTrack_l( 1084 const sp<AudioFlinger::Client>& client, 1085 audio_stream_type_t streamType, 1086 uint32_t sampleRate, 1087 audio_format_t format, 1088 audio_channel_mask_t channelMask, 1089 int frameCount, 1090 const sp<IMemory>& sharedBuffer, 1091 int sessionId, 1092 IAudioFlinger::track_flags_t flags, 1093 pid_t tid, 1094 status_t *status); 1095 1096 AudioStreamOut* getOutput() const; 1097 AudioStreamOut* clearOutput(); 1098 virtual audio_stream_t* stream() const; 1099 1100 // a very large number of suspend() will eventually wraparound, but unlikely 1101 void suspend() { (void) android_atomic_inc(&mSuspended); } 1102 void restore() 1103 { 1104 // if restore() is done without suspend(), get back into 1105 // range so that the next suspend() will operate correctly 1106 if (android_atomic_dec(&mSuspended) <= 0) { 1107 android_atomic_release_store(0, &mSuspended); 1108 } 1109 } 1110 bool isSuspended() const 1111 { return android_atomic_acquire_load(&mSuspended) > 0; } 1112 1113 virtual String8 getParameters(const String8& keys); 1114 virtual void audioConfigChanged_l(int event, int param = 0); 1115 status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames); 1116 int16_t *mixBuffer() const { return mMixBuffer; }; 1117 1118 virtual void detachAuxEffect_l(int effectId); 1119 status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track> track, 1120 int EffectId); 1121 status_t attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track> track, 1122 int EffectId); 1123 1124 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1125 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1126 virtual uint32_t hasAudioSession(int sessionId) const; 1127 virtual uint32_t getStrategyForSession_l(int sessionId); 1128 1129 1130 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 1131 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 1132 void invalidateTracks(audio_stream_type_t streamType); 1133 1134 1135 protected: 1136 int16_t* mMixBuffer; 1137 1138 // suspend count, > 0 means suspended. While suspended, the thread continues to pull from 1139 // tracks and mix, but doesn't write to HAL. A2DP and SCO HAL implementations can't handle 1140 // concurrent use of both of them, so Audio Policy Service suspends one of the threads to 1141 // workaround that restriction. 1142 // 'volatile' means accessed via atomic operations and no lock. 1143 volatile int32_t mSuspended; 1144 1145 int mBytesWritten; 1146 private: 1147 // mMasterMute is in both PlaybackThread and in AudioFlinger. When a 1148 // PlaybackThread needs to find out if master-muted, it checks it's local 1149 // copy rather than the one in AudioFlinger. This optimization saves a lock. 1150 bool mMasterMute; 1151 void setMasterMute_l(bool muted) { mMasterMute = muted; } 1152 protected: 1153 SortedVector< wp<Track> > mActiveTracks; // FIXME check if this could be sp<> 1154 1155 // Allocate a track name for a given channel mask. 1156 // Returns name >= 0 if successful, -1 on failure. 1157 virtual int getTrackName_l(audio_channel_mask_t channelMask, int sessionId) = 0; 1158 virtual void deleteTrackName_l(int name) = 0; 1159 1160 // Time to sleep between cycles when: 1161 virtual uint32_t activeSleepTimeUs() const; // mixer state MIXER_TRACKS_ENABLED 1162 virtual uint32_t idleSleepTimeUs() const = 0; // mixer state MIXER_IDLE 1163 virtual uint32_t suspendSleepTimeUs() const = 0; // audio policy manager suspended us 1164 // No sleep when mixer state == MIXER_TRACKS_READY; relies on audio HAL stream->write() 1165 // No sleep in standby mode; waits on a condition 1166 1167 // Code snippets that are temporarily lifted up out of threadLoop() until the merge 1168 void checkSilentMode_l(); 1169 1170 // Non-trivial for DUPLICATING only 1171 virtual void saveOutputTracks() { } 1172 virtual void clearOutputTracks() { } 1173 1174 // Cache various calculated values, at threadLoop() entry and after a parameter change 1175 virtual void cacheParameters_l(); 1176 1177 virtual uint32_t correctLatency(uint32_t latency) const; 1178 1179 private: 1180 1181 friend class AudioFlinger; // for numerous 1182 1183 PlaybackThread(const Client&); 1184 PlaybackThread& operator = (const PlaybackThread&); 1185 1186 status_t addTrack_l(const sp<Track>& track); 1187 void destroyTrack_l(const sp<Track>& track); 1188 void removeTrack_l(const sp<Track>& track); 1189 1190 void readOutputParameters(); 1191 1192 virtual void dumpInternals(int fd, const Vector<String16>& args); 1193 void dumpTracks(int fd, const Vector<String16>& args); 1194 1195 SortedVector< sp<Track> > mTracks; 1196 // mStreamTypes[] uses 1 additional stream type internally for the OutputTrack used by 1197 // DuplicatingThread 1198 stream_type_t mStreamTypes[AUDIO_STREAM_CNT + 1]; 1199 AudioStreamOut *mOutput; 1200 1201 float mMasterVolume; 1202 nsecs_t mLastWriteTime; 1203 int mNumWrites; 1204 int mNumDelayedWrites; 1205 bool mInWrite; 1206 1207 // FIXME rename these former local variables of threadLoop to standard "m" names 1208 nsecs_t standbyTime; 1209 size_t mixBufferSize; 1210 1211 // cached copies of activeSleepTimeUs() and idleSleepTimeUs() made by cacheParameters_l() 1212 uint32_t activeSleepTime; 1213 uint32_t idleSleepTime; 1214 1215 uint32_t sleepTime; 1216 1217 // mixer status returned by prepareTracks_l() 1218 mixer_state mMixerStatus; // current cycle 1219 // previous cycle when in prepareTracks_l() 1220 mixer_state mMixerStatusIgnoringFastTracks; 1221 // FIXME or a separate ready state per track 1222 1223 // FIXME move these declarations into the specific sub-class that needs them 1224 // MIXER only 1225 uint32_t sleepTimeShift; 1226 1227 // same as AudioFlinger::mStandbyTimeInNsecs except for DIRECT which uses a shorter value 1228 nsecs_t standbyDelay; 1229 1230 // MIXER only 1231 nsecs_t maxPeriod; 1232 1233 // DUPLICATING only 1234 uint32_t writeFrames; 1235 1236 private: 1237 // The HAL output sink is treated as non-blocking, but current implementation is blocking 1238 sp<NBAIO_Sink> mOutputSink; 1239 // If a fast mixer is present, the blocking pipe sink, otherwise clear 1240 sp<NBAIO_Sink> mPipeSink; 1241 // The current sink for the normal mixer to write it's (sub)mix, mOutputSink or mPipeSink 1242 sp<NBAIO_Sink> mNormalSink; 1243 // For dumpsys 1244 sp<NBAIO_Sink> mTeeSink; 1245 sp<NBAIO_Source> mTeeSource; 1246 uint32_t mScreenState; // cached copy of gScreenState 1247 public: 1248 virtual bool hasFastMixer() const = 0; 1249 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const 1250 { FastTrackUnderruns dummy; return dummy; } 1251 1252 protected: 1253 // accessed by both binder threads and within threadLoop(), lock on mutex needed 1254 unsigned mFastTrackAvailMask; // bit i set if fast track [i] is available 1255 1256 }; 1257 1258 class MixerThread : public PlaybackThread { 1259 public: 1260 MixerThread(const sp<AudioFlinger>& audioFlinger, 1261 AudioStreamOut* output, 1262 audio_io_handle_t id, 1263 audio_devices_t device, 1264 type_t type = MIXER); 1265 virtual ~MixerThread(); 1266 1267 // Thread virtuals 1268 1269 virtual bool checkForNewParameters_l(); 1270 virtual void dumpInternals(int fd, const Vector<String16>& args); 1271 1272 protected: 1273 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1274 virtual int getTrackName_l(audio_channel_mask_t channelMask, int sessionId); 1275 virtual void deleteTrackName_l(int name); 1276 virtual uint32_t idleSleepTimeUs() const; 1277 virtual uint32_t suspendSleepTimeUs() const; 1278 virtual void cacheParameters_l(); 1279 1280 // threadLoop snippets 1281 virtual void threadLoop_write(); 1282 virtual void threadLoop_standby(); 1283 virtual void threadLoop_mix(); 1284 virtual void threadLoop_sleepTime(); 1285 virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); 1286 virtual uint32_t correctLatency(uint32_t latency) const; 1287 1288 AudioMixer* mAudioMixer; // normal mixer 1289 private: 1290 // one-time initialization, no locks required 1291 FastMixer* mFastMixer; // non-NULL if there is also a fast mixer 1292 sp<AudioWatchdog> mAudioWatchdog; // non-0 if there is an audio watchdog thread 1293 1294 // contents are not guaranteed to be consistent, no locks required 1295 FastMixerDumpState mFastMixerDumpState; 1296#ifdef STATE_QUEUE_DUMP 1297 StateQueueObserverDump mStateQueueObserverDump; 1298 StateQueueMutatorDump mStateQueueMutatorDump; 1299#endif 1300 AudioWatchdogDump mAudioWatchdogDump; 1301 1302 // accessible only within the threadLoop(), no locks required 1303 // mFastMixer->sq() // for mutating and pushing state 1304 int32_t mFastMixerFutex; // for cold idle 1305 1306 public: 1307 virtual bool hasFastMixer() const { return mFastMixer != NULL; } 1308 virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const { 1309 ALOG_ASSERT(fastIndex < FastMixerState::kMaxFastTracks); 1310 return mFastMixerDumpState.mTracks[fastIndex].mUnderruns; 1311 } 1312 }; 1313 1314 class DirectOutputThread : public PlaybackThread { 1315 public: 1316 1317 DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1318 audio_io_handle_t id, audio_devices_t device); 1319 virtual ~DirectOutputThread(); 1320 1321 // Thread virtuals 1322 1323 virtual bool checkForNewParameters_l(); 1324 1325 protected: 1326 virtual int getTrackName_l(audio_channel_mask_t channelMask, int sessionId); 1327 virtual void deleteTrackName_l(int name); 1328 virtual uint32_t activeSleepTimeUs() const; 1329 virtual uint32_t idleSleepTimeUs() const; 1330 virtual uint32_t suspendSleepTimeUs() const; 1331 virtual void cacheParameters_l(); 1332 1333 // threadLoop snippets 1334 virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); 1335 virtual void threadLoop_mix(); 1336 virtual void threadLoop_sleepTime(); 1337 1338 private: 1339 // volumes last sent to audio HAL with stream->set_volume() 1340 float mLeftVolFloat; 1341 float mRightVolFloat; 1342 1343 // prepareTracks_l() tells threadLoop_mix() the name of the single active track 1344 sp<Track> mActiveTrack; 1345 public: 1346 virtual bool hasFastMixer() const { return false; } 1347 }; 1348 1349 class DuplicatingThread : public MixerThread { 1350 public: 1351 DuplicatingThread(const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread, 1352 audio_io_handle_t id); 1353 virtual ~DuplicatingThread(); 1354 1355 // Thread virtuals 1356 void addOutputTrack(MixerThread* thread); 1357 void removeOutputTrack(MixerThread* thread); 1358 uint32_t waitTimeMs() const { return mWaitTimeMs; } 1359 protected: 1360 virtual uint32_t activeSleepTimeUs() const; 1361 1362 private: 1363 bool outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks); 1364 protected: 1365 // threadLoop snippets 1366 virtual void threadLoop_mix(); 1367 virtual void threadLoop_sleepTime(); 1368 virtual void threadLoop_write(); 1369 virtual void threadLoop_standby(); 1370 virtual void cacheParameters_l(); 1371 1372 private: 1373 // called from threadLoop, addOutputTrack, removeOutputTrack 1374 virtual void updateWaitTime_l(); 1375 protected: 1376 virtual void saveOutputTracks(); 1377 virtual void clearOutputTracks(); 1378 private: 1379 1380 uint32_t mWaitTimeMs; 1381 SortedVector < sp<OutputTrack> > outputTracks; 1382 SortedVector < sp<OutputTrack> > mOutputTracks; 1383 public: 1384 virtual bool hasFastMixer() const { return false; } 1385 }; 1386 1387 PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const; 1388 MixerThread *checkMixerThread_l(audio_io_handle_t output) const; 1389 RecordThread *checkRecordThread_l(audio_io_handle_t input) const; 1390 // no range check, AudioFlinger::mLock held 1391 bool streamMute_l(audio_stream_type_t stream) const 1392 { return mStreamTypes[stream].mute; } 1393 // no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held 1394 float streamVolume_l(audio_stream_type_t stream) const 1395 { return mStreamTypes[stream].volume; } 1396 void audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2); 1397 1398 // allocate an audio_io_handle_t, session ID, or effect ID 1399 uint32_t nextUniqueId(); 1400 1401 status_t moveEffectChain_l(int sessionId, 1402 PlaybackThread *srcThread, 1403 PlaybackThread *dstThread, 1404 bool reRegister); 1405 // return thread associated with primary hardware device, or NULL 1406 PlaybackThread *primaryPlaybackThread_l() const; 1407 audio_devices_t primaryOutputDevice_l() const; 1408 1409 sp<PlaybackThread> getEffectThread_l(int sessionId, int EffectId); 1410 1411 // server side of the client's IAudioTrack 1412 class TrackHandle : public android::BnAudioTrack { 1413 public: 1414 TrackHandle(const sp<PlaybackThread::Track>& track); 1415 virtual ~TrackHandle(); 1416 virtual sp<IMemory> getCblk() const; 1417 virtual status_t start(); 1418 virtual void stop(); 1419 virtual void flush(); 1420 virtual void mute(bool); 1421 virtual void pause(); 1422 virtual status_t attachAuxEffect(int effectId); 1423 virtual status_t allocateTimedBuffer(size_t size, 1424 sp<IMemory>* buffer); 1425 virtual status_t queueTimedBuffer(const sp<IMemory>& buffer, 1426 int64_t pts); 1427 virtual status_t setMediaTimeTransform(const LinearTransform& xform, 1428 int target); 1429 virtual status_t onTransact( 1430 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 1431 private: 1432 const sp<PlaybackThread::Track> mTrack; 1433 }; 1434 1435 void removeClient_l(pid_t pid); 1436 void removeNotificationClient(pid_t pid); 1437 1438 1439 // record thread 1440 class RecordThread : public ThreadBase, public AudioBufferProvider 1441 // derives from AudioBufferProvider interface for use by resampler 1442 { 1443 public: 1444 1445 // record track 1446 class RecordTrack : public TrackBase { 1447 public: 1448 RecordTrack(RecordThread *thread, 1449 const sp<Client>& client, 1450 uint32_t sampleRate, 1451 audio_format_t format, 1452 audio_channel_mask_t channelMask, 1453 int frameCount, 1454 int sessionId); 1455 virtual ~RecordTrack(); 1456 1457 virtual status_t start(AudioSystem::sync_event_t event, int triggerSession); 1458 virtual void stop(); 1459 1460 void destroy(); 1461 1462 // clear the buffer overflow flag 1463 void clearOverflow() { mOverflow = false; } 1464 // set the buffer overflow flag and return previous value 1465 bool setOverflow() { bool tmp = mOverflow; mOverflow = true; 1466 return tmp; } 1467 1468 static void appendDumpHeader(String8& result); 1469 void dump(char* buffer, size_t size); 1470 1471 private: 1472 friend class AudioFlinger; // for mState 1473 1474 RecordTrack(const RecordTrack&); 1475 RecordTrack& operator = (const RecordTrack&); 1476 1477 // AudioBufferProvider interface 1478 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, 1479 int64_t pts = kInvalidPTS); 1480 // releaseBuffer() not overridden 1481 1482 bool mOverflow; // overflow on most recent attempt to fill client buffer 1483 }; 1484 1485 RecordThread(const sp<AudioFlinger>& audioFlinger, 1486 AudioStreamIn *input, 1487 uint32_t sampleRate, 1488 audio_channel_mask_t channelMask, 1489 audio_io_handle_t id, 1490 audio_devices_t device, 1491 const sp<NBAIO_Sink>& teeSink); 1492 virtual ~RecordThread(); 1493 1494 // no addTrack_l ? 1495 void destroyTrack_l(const sp<RecordTrack>& track); 1496 void removeTrack_l(const sp<RecordTrack>& track); 1497 1498 void dumpInternals(int fd, const Vector<String16>& args); 1499 void dumpTracks(int fd, const Vector<String16>& args); 1500 1501 // Thread virtuals 1502 virtual bool threadLoop(); 1503 virtual status_t readyToRun(); 1504 1505 // RefBase 1506 virtual void onFirstRef(); 1507 1508 virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; } 1509 sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l( 1510 const sp<AudioFlinger::Client>& client, 1511 uint32_t sampleRate, 1512 audio_format_t format, 1513 audio_channel_mask_t channelMask, 1514 int frameCount, 1515 int sessionId, 1516 IAudioFlinger::track_flags_t flags, 1517 pid_t tid, 1518 status_t *status); 1519 1520 status_t start(RecordTrack* recordTrack, 1521 AudioSystem::sync_event_t event, 1522 int triggerSession); 1523 1524 // ask the thread to stop the specified track, and 1525 // return true if the caller should then do it's part of the stopping process 1526 bool stop_l(RecordTrack* recordTrack); 1527 1528 void dump(int fd, const Vector<String16>& args); 1529 AudioStreamIn* clearInput(); 1530 virtual audio_stream_t* stream() const; 1531 1532 // AudioBufferProvider interface 1533 virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts); 1534 virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); 1535 1536 virtual bool checkForNewParameters_l(); 1537 virtual String8 getParameters(const String8& keys); 1538 virtual void audioConfigChanged_l(int event, int param = 0); 1539 void readInputParameters(); 1540 virtual unsigned int getInputFramesLost(); 1541 1542 virtual status_t addEffectChain_l(const sp<EffectChain>& chain); 1543 virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); 1544 virtual uint32_t hasAudioSession(int sessionId) const; 1545 1546 // Return the set of unique session IDs across all tracks. 1547 // The keys are the session IDs, and the associated values are meaningless. 1548 // FIXME replace by Set [and implement Bag/Multiset for other uses]. 1549 KeyedVector<int, bool> sessionIds() const; 1550 1551 virtual status_t setSyncEvent(const sp<SyncEvent>& event); 1552 virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; 1553 1554 static void syncStartEventCallback(const wp<SyncEvent>& event); 1555 void handleSyncStartEvent(const sp<SyncEvent>& event); 1556 1557 private: 1558 void clearSyncStartEvent(); 1559 1560 // Enter standby if not already in standby, and set mStandby flag 1561 void standby(); 1562 1563 // Call the HAL standby method unconditionally, and don't change mStandby flag 1564 void inputStandBy(); 1565 1566 AudioStreamIn *mInput; 1567 SortedVector < sp<RecordTrack> > mTracks; 1568 // mActiveTrack has dual roles: it indicates the current active track, and 1569 // is used together with mStartStopCond to indicate start()/stop() progress 1570 sp<RecordTrack> mActiveTrack; 1571 Condition mStartStopCond; 1572 AudioResampler *mResampler; 1573 int32_t *mRsmpOutBuffer; 1574 int16_t *mRsmpInBuffer; 1575 size_t mRsmpInIndex; 1576 size_t mInputBytes; 1577 const int mReqChannelCount; 1578 const uint32_t mReqSampleRate; 1579 ssize_t mBytesRead; 1580 // sync event triggering actual audio capture. Frames read before this event will 1581 // be dropped and therefore not read by the application. 1582 sp<SyncEvent> mSyncStartEvent; 1583 // number of captured frames to drop after the start sync event has been received. 1584 // when < 0, maximum frames to drop before starting capture even if sync event is 1585 // not received 1586 ssize_t mFramestoDrop; 1587 1588 // For dumpsys 1589 const sp<NBAIO_Sink> mTeeSink; 1590 }; 1591 1592 // server side of the client's IAudioRecord 1593 class RecordHandle : public android::BnAudioRecord { 1594 public: 1595 RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack); 1596 virtual ~RecordHandle(); 1597 virtual sp<IMemory> getCblk() const; 1598 virtual status_t start(int /*AudioSystem::sync_event_t*/ event, int triggerSession); 1599 virtual void stop(); 1600 virtual status_t onTransact( 1601 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 1602 private: 1603 const sp<RecordThread::RecordTrack> mRecordTrack; 1604 1605 // for use from destructor 1606 void stop_nonvirtual(); 1607 }; 1608 1609 //--- Audio Effect Management 1610 1611 // EffectModule and EffectChain classes both have their own mutex to protect 1612 // state changes or resource modifications. Always respect the following order 1613 // if multiple mutexes must be acquired to avoid cross deadlock: 1614 // AudioFlinger -> ThreadBase -> EffectChain -> EffectModule 1615 1616 // The EffectModule class is a wrapper object controlling the effect engine implementation 1617 // in the effect library. It prevents concurrent calls to process() and command() functions 1618 // from different client threads. It keeps a list of EffectHandle objects corresponding 1619 // to all client applications using this effect and notifies applications of effect state, 1620 // control or parameter changes. It manages the activation state machine to send appropriate 1621 // reset, enable, disable commands to effect engine and provide volume 1622 // ramping when effects are activated/deactivated. 1623 // When controlling an auxiliary effect, the EffectModule also provides an input buffer used by 1624 // the attached track(s) to accumulate their auxiliary channel. 1625 class EffectModule : public RefBase { 1626 public: 1627 EffectModule(ThreadBase *thread, 1628 const wp<AudioFlinger::EffectChain>& chain, 1629 effect_descriptor_t *desc, 1630 int id, 1631 int sessionId); 1632 virtual ~EffectModule(); 1633 1634 enum effect_state { 1635 IDLE, 1636 RESTART, 1637 STARTING, 1638 ACTIVE, 1639 STOPPING, 1640 STOPPED, 1641 DESTROYED 1642 }; 1643 1644 int id() const { return mId; } 1645 void process(); 1646 void updateState(); 1647 status_t command(uint32_t cmdCode, 1648 uint32_t cmdSize, 1649 void *pCmdData, 1650 uint32_t *replySize, 1651 void *pReplyData); 1652 1653 void reset_l(); 1654 status_t configure(); 1655 status_t init(); 1656 effect_state state() const { 1657 return mState; 1658 } 1659 uint32_t status() { 1660 return mStatus; 1661 } 1662 int sessionId() const { 1663 return mSessionId; 1664 } 1665 status_t setEnabled(bool enabled); 1666 status_t setEnabled_l(bool enabled); 1667 bool isEnabled() const; 1668 bool isProcessEnabled() const; 1669 1670 void setInBuffer(int16_t *buffer) { mConfig.inputCfg.buffer.s16 = buffer; } 1671 int16_t *inBuffer() { return mConfig.inputCfg.buffer.s16; } 1672 void setOutBuffer(int16_t *buffer) { mConfig.outputCfg.buffer.s16 = buffer; } 1673 int16_t *outBuffer() { return mConfig.outputCfg.buffer.s16; } 1674 void setChain(const wp<EffectChain>& chain) { mChain = chain; } 1675 void setThread(const wp<ThreadBase>& thread) { mThread = thread; } 1676 const wp<ThreadBase>& thread() { return mThread; } 1677 1678 status_t addHandle(EffectHandle *handle); 1679 size_t disconnect(EffectHandle *handle, bool unpinIfLast); 1680 size_t removeHandle(EffectHandle *handle); 1681 1682 const effect_descriptor_t& desc() const { return mDescriptor; } 1683 wp<EffectChain>& chain() { return mChain; } 1684 1685 status_t setDevice(audio_devices_t device); 1686 status_t setVolume(uint32_t *left, uint32_t *right, bool controller); 1687 status_t setMode(audio_mode_t mode); 1688 status_t setAudioSource(audio_source_t source); 1689 status_t start(); 1690 status_t stop(); 1691 void setSuspended(bool suspended); 1692 bool suspended() const; 1693 1694 EffectHandle* controlHandle_l(); 1695 1696 bool isPinned() const { return mPinned; } 1697 void unPin() { mPinned = false; } 1698 bool purgeHandles(); 1699 void lock() { mLock.lock(); } 1700 void unlock() { mLock.unlock(); } 1701 1702 void dump(int fd, const Vector<String16>& args); 1703 1704 protected: 1705 friend class AudioFlinger; // for mHandles 1706 bool mPinned; 1707 1708 // Maximum time allocated to effect engines to complete the turn off sequence 1709 static const uint32_t MAX_DISABLE_TIME_MS = 10000; 1710 1711 EffectModule(const EffectModule&); 1712 EffectModule& operator = (const EffectModule&); 1713 1714 status_t start_l(); 1715 status_t stop_l(); 1716 1717mutable Mutex mLock; // mutex for process, commands and handles list protection 1718 wp<ThreadBase> mThread; // parent thread 1719 wp<EffectChain> mChain; // parent effect chain 1720 const int mId; // this instance unique ID 1721 const int mSessionId; // audio session ID 1722 const effect_descriptor_t mDescriptor;// effect descriptor received from effect engine 1723 effect_config_t mConfig; // input and output audio configuration 1724 effect_handle_t mEffectInterface; // Effect module C API 1725 status_t mStatus; // initialization status 1726 effect_state mState; // current activation state 1727 Vector<EffectHandle *> mHandles; // list of client handles 1728 // First handle in mHandles has highest priority and controls the effect module 1729 uint32_t mMaxDisableWaitCnt; // maximum grace period before forcing an effect off after 1730 // sending disable command. 1731 uint32_t mDisableWaitCnt; // current process() calls count during disable period. 1732 bool mSuspended; // effect is suspended: temporarily disabled by framework 1733 }; 1734 1735 // The EffectHandle class implements the IEffect interface. It provides resources 1736 // to receive parameter updates, keeps track of effect control 1737 // ownership and state and has a pointer to the EffectModule object it is controlling. 1738 // There is one EffectHandle object for each application controlling (or using) 1739 // an effect module. 1740 // The EffectHandle is obtained by calling AudioFlinger::createEffect(). 1741 class EffectHandle: public android::BnEffect { 1742 public: 1743 1744 EffectHandle(const sp<EffectModule>& effect, 1745 const sp<AudioFlinger::Client>& client, 1746 const sp<IEffectClient>& effectClient, 1747 int32_t priority); 1748 virtual ~EffectHandle(); 1749 1750 // IEffect 1751 virtual status_t enable(); 1752 virtual status_t disable(); 1753 virtual status_t command(uint32_t cmdCode, 1754 uint32_t cmdSize, 1755 void *pCmdData, 1756 uint32_t *replySize, 1757 void *pReplyData); 1758 virtual void disconnect(); 1759 private: 1760 void disconnect(bool unpinIfLast); 1761 public: 1762 virtual sp<IMemory> getCblk() const { return mCblkMemory; } 1763 virtual status_t onTransact(uint32_t code, const Parcel& data, 1764 Parcel* reply, uint32_t flags); 1765 1766 1767 // Give or take control of effect module 1768 // - hasControl: true if control is given, false if removed 1769 // - signal: true client app should be signaled of change, false otherwise 1770 // - enabled: state of the effect when control is passed 1771 void setControl(bool hasControl, bool signal, bool enabled); 1772 void commandExecuted(uint32_t cmdCode, 1773 uint32_t cmdSize, 1774 void *pCmdData, 1775 uint32_t replySize, 1776 void *pReplyData); 1777 void setEnabled(bool enabled); 1778 bool enabled() const { return mEnabled; } 1779 1780 // Getters 1781 int id() const { return mEffect->id(); } 1782 int priority() const { return mPriority; } 1783 bool hasControl() const { return mHasControl; } 1784 sp<EffectModule> effect() const { return mEffect; } 1785 // destroyed_l() must be called with the associated EffectModule mLock held 1786 bool destroyed_l() const { return mDestroyed; } 1787 1788 void dump(char* buffer, size_t size); 1789 1790 protected: 1791 friend class AudioFlinger; // for mEffect, mHasControl, mEnabled 1792 EffectHandle(const EffectHandle&); 1793 EffectHandle& operator =(const EffectHandle&); 1794 1795 sp<EffectModule> mEffect; // pointer to controlled EffectModule 1796 sp<IEffectClient> mEffectClient; // callback interface for client notifications 1797 /*const*/ sp<Client> mClient; // client for shared memory allocation, see disconnect() 1798 sp<IMemory> mCblkMemory; // shared memory for control block 1799 effect_param_cblk_t* mCblk; // control block for deferred parameter setting via 1800 // shared memory 1801 uint8_t* mBuffer; // pointer to parameter area in shared memory 1802 int mPriority; // client application priority to control the effect 1803 bool mHasControl; // true if this handle is controlling the effect 1804 bool mEnabled; // cached enable state: needed when the effect is 1805 // restored after being suspended 1806 bool mDestroyed; // Set to true by destructor. Access with EffectModule 1807 // mLock held 1808 }; 1809 1810 // the EffectChain class represents a group of effects associated to one audio session. 1811 // There can be any number of EffectChain objects per output mixer thread (PlaybackThread). 1812 // The EffecChain with session ID 0 contains global effects applied to the output mix. 1813 // Effects in this chain can be insert or auxiliary. Effects in other chains (attached to 1814 // tracks) are insert only. The EffectChain maintains an ordered list of effect module, the 1815 // order corresponding in the effect process order. When attached to a track (session ID != 0), 1816 // it also provide it's own input buffer used by the track as accumulation buffer. 1817 class EffectChain : public RefBase { 1818 public: 1819 EffectChain(const wp<ThreadBase>& wThread, int sessionId); 1820 EffectChain(ThreadBase *thread, int sessionId); 1821 virtual ~EffectChain(); 1822 1823 // special key used for an entry in mSuspendedEffects keyed vector 1824 // corresponding to a suspend all request. 1825 static const int kKeyForSuspendAll = 0; 1826 1827 // minimum duration during which we force calling effect process when last track on 1828 // a session is stopped or removed to allow effect tail to be rendered 1829 static const int kProcessTailDurationMs = 1000; 1830 1831 void process_l(); 1832 1833 void lock() { 1834 mLock.lock(); 1835 } 1836 void unlock() { 1837 mLock.unlock(); 1838 } 1839 1840 status_t addEffect_l(const sp<EffectModule>& handle); 1841 size_t removeEffect_l(const sp<EffectModule>& handle); 1842 1843 int sessionId() const { return mSessionId; } 1844 void setSessionId(int sessionId) { mSessionId = sessionId; } 1845 1846 sp<EffectModule> getEffectFromDesc_l(effect_descriptor_t *descriptor); 1847 sp<EffectModule> getEffectFromId_l(int id); 1848 sp<EffectModule> getEffectFromType_l(const effect_uuid_t *type); 1849 bool setVolume_l(uint32_t *left, uint32_t *right); 1850 void setDevice_l(audio_devices_t device); 1851 void setMode_l(audio_mode_t mode); 1852 void setAudioSource_l(audio_source_t source); 1853 1854 void setInBuffer(int16_t *buffer, bool ownsBuffer = false) { 1855 mInBuffer = buffer; 1856 mOwnInBuffer = ownsBuffer; 1857 } 1858 int16_t *inBuffer() const { 1859 return mInBuffer; 1860 } 1861 void setOutBuffer(int16_t *buffer) { 1862 mOutBuffer = buffer; 1863 } 1864 int16_t *outBuffer() const { 1865 return mOutBuffer; 1866 } 1867 1868 void incTrackCnt() { android_atomic_inc(&mTrackCnt); } 1869 void decTrackCnt() { android_atomic_dec(&mTrackCnt); } 1870 int32_t trackCnt() const { return android_atomic_acquire_load(&mTrackCnt); } 1871 1872 void incActiveTrackCnt() { android_atomic_inc(&mActiveTrackCnt); 1873 mTailBufferCount = mMaxTailBuffers; } 1874 void decActiveTrackCnt() { android_atomic_dec(&mActiveTrackCnt); } 1875 int32_t activeTrackCnt() const { return android_atomic_acquire_load(&mActiveTrackCnt); } 1876 1877 uint32_t strategy() const { return mStrategy; } 1878 void setStrategy(uint32_t strategy) 1879 { mStrategy = strategy; } 1880 1881 // suspend effect of the given type 1882 void setEffectSuspended_l(const effect_uuid_t *type, 1883 bool suspend); 1884 // suspend all eligible effects 1885 void setEffectSuspendedAll_l(bool suspend); 1886 // check if effects should be suspend or restored when a given effect is enable or disabled 1887 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1888 bool enabled); 1889 1890 void clearInputBuffer(); 1891 1892 void dump(int fd, const Vector<String16>& args); 1893 1894 protected: 1895 friend class AudioFlinger; // for mThread, mEffects 1896 EffectChain(const EffectChain&); 1897 EffectChain& operator =(const EffectChain&); 1898 1899 class SuspendedEffectDesc : public RefBase { 1900 public: 1901 SuspendedEffectDesc() : mRefCount(0) {} 1902 1903 int mRefCount; 1904 effect_uuid_t mType; 1905 wp<EffectModule> mEffect; 1906 }; 1907 1908 // get a list of effect modules to suspend when an effect of the type 1909 // passed is enabled. 1910 void getSuspendEligibleEffects(Vector< sp<EffectModule> > &effects); 1911 1912 // get an effect module if it is currently enable 1913 sp<EffectModule> getEffectIfEnabled(const effect_uuid_t *type); 1914 // true if the effect whose descriptor is passed can be suspended 1915 // OEMs can modify the rules implemented in this method to exclude specific effect 1916 // types or implementations from the suspend/restore mechanism. 1917 bool isEffectEligibleForSuspend(const effect_descriptor_t& desc); 1918 1919 void clearInputBuffer_l(sp<ThreadBase> thread); 1920 1921 wp<ThreadBase> mThread; // parent mixer thread 1922 Mutex mLock; // mutex protecting effect list 1923 Vector< sp<EffectModule> > mEffects; // list of effect modules 1924 int mSessionId; // audio session ID 1925 int16_t *mInBuffer; // chain input buffer 1926 int16_t *mOutBuffer; // chain output buffer 1927 1928 // 'volatile' here means these are accessed with atomic operations instead of mutex 1929 volatile int32_t mActiveTrackCnt; // number of active tracks connected 1930 volatile int32_t mTrackCnt; // number of tracks connected 1931 1932 int32_t mTailBufferCount; // current effect tail buffer count 1933 int32_t mMaxTailBuffers; // maximum effect tail buffers 1934 bool mOwnInBuffer; // true if the chain owns its input buffer 1935 int mVolumeCtrlIdx; // index of insert effect having control over volume 1936 uint32_t mLeftVolume; // previous volume on left channel 1937 uint32_t mRightVolume; // previous volume on right channel 1938 uint32_t mNewLeftVolume; // new volume on left channel 1939 uint32_t mNewRightVolume; // new volume on right channel 1940 uint32_t mStrategy; // strategy for this effect chain 1941 // mSuspendedEffects lists all effects currently suspended in the chain. 1942 // Use effect type UUID timelow field as key. There is no real risk of identical 1943 // timeLow fields among effect type UUIDs. 1944 // Updated by updateSuspendedSessions_l() only. 1945 KeyedVector< int, sp<SuspendedEffectDesc> > mSuspendedEffects; 1946 }; 1947 1948 class AudioHwDevice { 1949 public: 1950 enum Flags { 1951 AHWD_CAN_SET_MASTER_VOLUME = 0x1, 1952 AHWD_CAN_SET_MASTER_MUTE = 0x2, 1953 }; 1954 1955 AudioHwDevice(const char *moduleName, 1956 audio_hw_device_t *hwDevice, 1957 Flags flags) 1958 : mModuleName(strdup(moduleName)) 1959 , mHwDevice(hwDevice) 1960 , mFlags(flags) { } 1961 /*virtual*/ ~AudioHwDevice() { free((void *)mModuleName); } 1962 1963 bool canSetMasterVolume() const { 1964 return (0 != (mFlags & AHWD_CAN_SET_MASTER_VOLUME)); 1965 } 1966 1967 bool canSetMasterMute() const { 1968 return (0 != (mFlags & AHWD_CAN_SET_MASTER_MUTE)); 1969 } 1970 1971 const char *moduleName() const { return mModuleName; } 1972 audio_hw_device_t *hwDevice() const { return mHwDevice; } 1973 private: 1974 const char * const mModuleName; 1975 audio_hw_device_t * const mHwDevice; 1976 Flags mFlags; 1977 }; 1978 1979 // AudioStreamOut and AudioStreamIn are immutable, so their fields are const. 1980 // For emphasis, we could also make all pointers to them be "const *", 1981 // but that would clutter the code unnecessarily. 1982 1983 struct AudioStreamOut { 1984 AudioHwDevice* const audioHwDev; 1985 audio_stream_out_t* const stream; 1986 1987 audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); } 1988 1989 AudioStreamOut(AudioHwDevice *dev, audio_stream_out_t *out) : 1990 audioHwDev(dev), stream(out) {} 1991 }; 1992 1993 struct AudioStreamIn { 1994 AudioHwDevice* const audioHwDev; 1995 audio_stream_in_t* const stream; 1996 1997 audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); } 1998 1999 AudioStreamIn(AudioHwDevice *dev, audio_stream_in_t *in) : 2000 audioHwDev(dev), stream(in) {} 2001 }; 2002 2003 // for mAudioSessionRefs only 2004 struct AudioSessionRef { 2005 AudioSessionRef(int sessionid, pid_t pid) : 2006 mSessionid(sessionid), mPid(pid), mCnt(1) {} 2007 const int mSessionid; 2008 const pid_t mPid; 2009 int mCnt; 2010 }; 2011 2012 mutable Mutex mLock; 2013 2014 DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client() 2015 2016 mutable Mutex mHardwareLock; 2017 // NOTE: If both mLock and mHardwareLock mutexes must be held, 2018 // always take mLock before mHardwareLock 2019 2020 // These two fields are immutable after onFirstRef(), so no lock needed to access 2021 AudioHwDevice* mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL 2022 DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*> mAudioHwDevs; 2023 2024 // for dump, indicates which hardware operation is currently in progress (but not stream ops) 2025 enum hardware_call_state { 2026 AUDIO_HW_IDLE = 0, // no operation in progress 2027 AUDIO_HW_INIT, // init_check 2028 AUDIO_HW_OUTPUT_OPEN, // open_output_stream 2029 AUDIO_HW_OUTPUT_CLOSE, // unused 2030 AUDIO_HW_INPUT_OPEN, // unused 2031 AUDIO_HW_INPUT_CLOSE, // unused 2032 AUDIO_HW_STANDBY, // unused 2033 AUDIO_HW_SET_MASTER_VOLUME, // set_master_volume 2034 AUDIO_HW_GET_ROUTING, // unused 2035 AUDIO_HW_SET_ROUTING, // unused 2036 AUDIO_HW_GET_MODE, // unused 2037 AUDIO_HW_SET_MODE, // set_mode 2038 AUDIO_HW_GET_MIC_MUTE, // get_mic_mute 2039 AUDIO_HW_SET_MIC_MUTE, // set_mic_mute 2040 AUDIO_HW_SET_VOICE_VOLUME, // set_voice_volume 2041 AUDIO_HW_SET_PARAMETER, // set_parameters 2042 AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size 2043 AUDIO_HW_GET_MASTER_VOLUME, // get_master_volume 2044 AUDIO_HW_GET_PARAMETER, // get_parameters 2045 AUDIO_HW_SET_MASTER_MUTE, // set_master_mute 2046 AUDIO_HW_GET_MASTER_MUTE, // get_master_mute 2047 }; 2048 2049 mutable hardware_call_state mHardwareStatus; // for dump only 2050 2051 2052 DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads; 2053 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 2054 2055 // member variables below are protected by mLock 2056 float mMasterVolume; 2057 bool mMasterMute; 2058 // end of variables protected by mLock 2059 2060 DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads; 2061 2062 DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients; 2063 volatile int32_t mNextUniqueId; // updated by android_atomic_inc 2064 audio_mode_t mMode; 2065 bool mBtNrecIsOff; 2066 2067 // protected by mLock 2068 Vector<AudioSessionRef*> mAudioSessionRefs; 2069 2070 float masterVolume_l() const; 2071 bool masterMute_l() const; 2072 audio_module_handle_t loadHwModule_l(const char *name); 2073 2074 Vector < sp<SyncEvent> > mPendingSyncEvents; // sync events awaiting for a session 2075 // to be created 2076 2077private: 2078 sp<Client> registerPid_l(pid_t pid); // always returns non-0 2079 2080 // for use from destructor 2081 status_t closeOutput_nonvirtual(audio_io_handle_t output); 2082 status_t closeInput_nonvirtual(audio_io_handle_t input); 2083 2084 // all record threads serially share a common tee sink, which is re-created on format change 2085 sp<NBAIO_Sink> mRecordTeeSink; 2086 sp<NBAIO_Source> mRecordTeeSource; 2087 2088public: 2089 static void dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id = 0); 2090}; 2091 2092 2093// ---------------------------------------------------------------------------- 2094 2095}; // namespace android 2096 2097#endif // ANDROID_AUDIO_FLINGER_H 2098