AudioFlinger.h revision 4a3d5c23f79189eb7ab9f31c440c7da5b15947a2
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 "Configuration.h" 22#include <stdint.h> 23#include <sys/types.h> 24#include <limits.h> 25 26#include <cutils/compiler.h> 27 28#include <media/IAudioFlinger.h> 29#include <media/IAudioFlingerClient.h> 30#include <media/IAudioTrack.h> 31#include <media/IAudioRecord.h> 32#include <media/AudioSystem.h> 33#include <media/AudioTrack.h> 34 35#include <utils/Atomic.h> 36#include <utils/Errors.h> 37#include <utils/threads.h> 38#include <utils/SortedVector.h> 39#include <utils/TypeHelpers.h> 40#include <utils/Vector.h> 41 42#include <binder/BinderService.h> 43#include <binder/MemoryDealer.h> 44 45#include <system/audio.h> 46#include <hardware/audio.h> 47#include <hardware/audio_policy.h> 48 49#include <media/AudioBufferProvider.h> 50#include <media/ExtendedAudioBufferProvider.h> 51 52#include "FastCapture.h" 53#include "FastMixer.h" 54#include <media/nbaio/NBAIO.h> 55#include "AudioWatchdog.h" 56#include "AudioMixer.h" 57#include "AudioStreamOut.h" 58#include "SpdifStreamOut.h" 59#include "AudioHwDevice.h" 60#include "LinearMap.h" 61#include "LockWatch.h" 62 63#include <powermanager/IPowerManager.h> 64 65#include <media/nbaio/NBLog.h> 66#include <private/media/AudioTrackShared.h> 67 68namespace android { 69 70struct audio_track_cblk_t; 71struct effect_param_cblk_t; 72class AudioMixer; 73class AudioBuffer; 74class AudioResampler; 75class EffectsFactoryHalInterface; 76class FastMixer; 77class PassthruBufferProvider; 78class ServerProxy; 79 80// ---------------------------------------------------------------------------- 81 82static const nsecs_t kDefaultStandbyTimeInNsecs = seconds(3); 83 84 85// Max shared memory size for audio tracks and audio records per client process 86static const size_t kClientSharedHeapSizeBytes = 1024*1024; 87// Shared memory size multiplier for non low ram devices 88static const size_t kClientSharedHeapSizeMultiplier = 4; 89 90#define INCLUDING_FROM_AUDIOFLINGER_H 91 92class AudioFlinger : 93 public BinderService<AudioFlinger>, 94 public BnAudioFlinger 95{ 96 friend class BinderService<AudioFlinger>; // for AudioFlinger() 97public: 98 static const char* getServiceName() ANDROID_API { return "media.audio_flinger"; } 99 100 virtual status_t dump(int fd, const Vector<String16>& args); 101 102 // IAudioFlinger interface, in binder opcode order 103 virtual sp<IAudioTrack> createTrack( 104 audio_stream_type_t streamType, 105 uint32_t sampleRate, 106 audio_format_t format, 107 audio_channel_mask_t channelMask, 108 size_t *pFrameCount, 109 audio_output_flags_t *flags, 110 const sp<IMemory>& sharedBuffer, 111 audio_io_handle_t output, 112 pid_t pid, 113 pid_t tid, 114 audio_session_t *sessionId, 115 int clientUid, 116 status_t *status /*non-NULL*/); 117 118 virtual sp<IAudioRecord> openRecord( 119 audio_io_handle_t input, 120 uint32_t sampleRate, 121 audio_format_t format, 122 audio_channel_mask_t channelMask, 123 const String16& opPackageName, 124 size_t *pFrameCount, 125 audio_input_flags_t *flags, 126 pid_t pid, 127 pid_t tid, 128 int clientUid, 129 audio_session_t *sessionId, 130 size_t *notificationFrames, 131 sp<IMemory>& cblk, 132 sp<IMemory>& buffers, 133 status_t *status /*non-NULL*/); 134 135 virtual uint32_t sampleRate(audio_io_handle_t ioHandle) const; 136 virtual audio_format_t format(audio_io_handle_t output) const; 137 virtual size_t frameCount(audio_io_handle_t ioHandle) const; 138 virtual size_t frameCountHAL(audio_io_handle_t ioHandle) const; 139 virtual uint32_t latency(audio_io_handle_t output) const; 140 141 virtual status_t setMasterVolume(float value); 142 virtual status_t setMasterMute(bool muted); 143 144 virtual float masterVolume() const; 145 virtual bool masterMute() const; 146 147 virtual status_t setStreamVolume(audio_stream_type_t stream, float value, 148 audio_io_handle_t output); 149 virtual status_t setStreamMute(audio_stream_type_t stream, bool muted); 150 151 virtual float streamVolume(audio_stream_type_t stream, 152 audio_io_handle_t output) const; 153 virtual bool streamMute(audio_stream_type_t stream) const; 154 155 virtual status_t setMode(audio_mode_t mode); 156 157 virtual status_t setMicMute(bool state); 158 virtual bool getMicMute() const; 159 160 virtual status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs); 161 virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) const; 162 163 virtual void registerClient(const sp<IAudioFlingerClient>& client); 164 165 virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, 166 audio_channel_mask_t channelMask) const; 167 168 virtual status_t openOutput(audio_module_handle_t module, 169 audio_io_handle_t *output, 170 audio_config_t *config, 171 audio_devices_t *devices, 172 const String8& address, 173 uint32_t *latencyMs, 174 audio_output_flags_t flags); 175 176 virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1, 177 audio_io_handle_t output2); 178 179 virtual status_t closeOutput(audio_io_handle_t output); 180 181 virtual status_t suspendOutput(audio_io_handle_t output); 182 183 virtual status_t restoreOutput(audio_io_handle_t output); 184 185 virtual status_t openInput(audio_module_handle_t module, 186 audio_io_handle_t *input, 187 audio_config_t *config, 188 audio_devices_t *device, 189 const String8& address, 190 audio_source_t source, 191 audio_input_flags_t flags); 192 193 virtual status_t closeInput(audio_io_handle_t input); 194 195 virtual status_t invalidateStream(audio_stream_type_t stream); 196 197 virtual status_t setVoiceVolume(float volume); 198 199 virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 200 audio_io_handle_t output) const; 201 202 virtual uint32_t getInputFramesLost(audio_io_handle_t ioHandle) const; 203 204 // This is the binder API. For the internal API see nextUniqueId(). 205 virtual audio_unique_id_t newAudioUniqueId(audio_unique_id_use_t use); 206 207 virtual void acquireAudioSessionId(audio_session_t audioSession, pid_t pid); 208 209 virtual void releaseAudioSessionId(audio_session_t audioSession, pid_t pid); 210 211 virtual status_t queryNumberEffects(uint32_t *numEffects) const; 212 213 virtual status_t queryEffect(uint32_t index, effect_descriptor_t *descriptor) const; 214 215 virtual status_t getEffectDescriptor(const effect_uuid_t *pUuid, 216 effect_descriptor_t *descriptor) const; 217 218 virtual sp<IEffect> createEffect( 219 effect_descriptor_t *pDesc, 220 const sp<IEffectClient>& effectClient, 221 int32_t priority, 222 audio_io_handle_t io, 223 audio_session_t sessionId, 224 const String16& opPackageName, 225 status_t *status /*non-NULL*/, 226 int *id, 227 int *enabled); 228 229 virtual status_t moveEffects(audio_session_t sessionId, audio_io_handle_t srcOutput, 230 audio_io_handle_t dstOutput); 231 232 virtual audio_module_handle_t loadHwModule(const char *name); 233 234 virtual uint32_t getPrimaryOutputSamplingRate(); 235 virtual size_t getPrimaryOutputFrameCount(); 236 237 virtual status_t setLowRamDevice(bool isLowRamDevice); 238 239 /* List available audio ports and their attributes */ 240 virtual status_t listAudioPorts(unsigned int *num_ports, 241 struct audio_port *ports); 242 243 /* Get attributes for a given audio port */ 244 virtual status_t getAudioPort(struct audio_port *port); 245 246 /* Create an audio patch between several source and sink ports */ 247 virtual status_t createAudioPatch(const struct audio_patch *patch, 248 audio_patch_handle_t *handle); 249 250 /* Release an audio patch */ 251 virtual status_t releaseAudioPatch(audio_patch_handle_t handle); 252 253 /* List existing audio patches */ 254 virtual status_t listAudioPatches(unsigned int *num_patches, 255 struct audio_patch *patches); 256 257 /* Set audio port configuration */ 258 virtual status_t setAudioPortConfig(const struct audio_port_config *config); 259 260 /* Get the HW synchronization source used for an audio session */ 261 virtual audio_hw_sync_t getAudioHwSyncForSession(audio_session_t sessionId); 262 263 /* Indicate JAVA services are ready (scheduling, power management ...) */ 264 virtual status_t systemReady(); 265 266 virtual status_t onTransact( 267 uint32_t code, 268 const Parcel& data, 269 Parcel* reply, 270 uint32_t flags); 271 272 // end of IAudioFlinger interface 273 274 sp<NBLog::Writer> newWriter_l(size_t size, const char *name); 275 void unregisterWriter(const sp<NBLog::Writer>& writer); 276 sp<EffectsFactoryHalInterface> getEffectsFactory(); 277private: 278 static const size_t kLogMemorySize = 40 * 1024; 279 sp<MemoryDealer> mLogMemoryDealer; // == 0 when NBLog is disabled 280 // When a log writer is unregistered, it is done lazily so that media.log can continue to see it 281 // for as long as possible. The memory is only freed when it is needed for another log writer. 282 Vector< sp<NBLog::Writer> > mUnregisteredWriters; 283 Mutex mUnregisteredWritersLock; 284public: 285 286 class SyncEvent; 287 288 typedef void (*sync_event_callback_t)(const wp<SyncEvent>& event) ; 289 290 class SyncEvent : public RefBase { 291 public: 292 SyncEvent(AudioSystem::sync_event_t type, 293 audio_session_t triggerSession, 294 audio_session_t listenerSession, 295 sync_event_callback_t callBack, 296 wp<RefBase> cookie) 297 : mType(type), mTriggerSession(triggerSession), mListenerSession(listenerSession), 298 mCallback(callBack), mCookie(cookie) 299 {} 300 301 virtual ~SyncEvent() {} 302 303 void trigger() { Mutex::Autolock _l(mLock); if (mCallback) mCallback(this); } 304 bool isCancelled() const { Mutex::Autolock _l(mLock); return (mCallback == NULL); } 305 void cancel() { Mutex::Autolock _l(mLock); mCallback = NULL; } 306 AudioSystem::sync_event_t type() const { return mType; } 307 audio_session_t triggerSession() const { return mTriggerSession; } 308 audio_session_t listenerSession() const { return mListenerSession; } 309 wp<RefBase> cookie() const { return mCookie; } 310 311 private: 312 const AudioSystem::sync_event_t mType; 313 const audio_session_t mTriggerSession; 314 const audio_session_t mListenerSession; 315 sync_event_callback_t mCallback; 316 const wp<RefBase> mCookie; 317 mutable Mutex mLock; 318 }; 319 320 sp<SyncEvent> createSyncEvent(AudioSystem::sync_event_t type, 321 audio_session_t triggerSession, 322 audio_session_t listenerSession, 323 sync_event_callback_t callBack, 324 const wp<RefBase>& cookie); 325 326private: 327 328 audio_mode_t getMode() const { return mMode; } 329 330 bool btNrecIsOff() const { return mBtNrecIsOff; } 331 332 AudioFlinger() ANDROID_API; 333 virtual ~AudioFlinger(); 334 335 // call in any IAudioFlinger method that accesses mPrimaryHardwareDev 336 status_t initCheck() const { return mPrimaryHardwareDev == NULL ? 337 NO_INIT : NO_ERROR; } 338 339 // RefBase 340 virtual void onFirstRef(); 341 342 AudioHwDevice* findSuitableHwDev_l(audio_module_handle_t module, 343 audio_devices_t devices); 344 void purgeStaleEffects_l(); 345 346 // Set kEnableExtendedChannels to true to enable greater than stereo output 347 // for the MixerThread and device sink. Number of channels allowed is 348 // FCC_2 <= channels <= AudioMixer::MAX_NUM_CHANNELS. 349 static const bool kEnableExtendedChannels = true; 350 351 // Returns true if channel mask is permitted for the PCM sink in the MixerThread 352 static inline bool isValidPcmSinkChannelMask(audio_channel_mask_t channelMask) { 353 switch (audio_channel_mask_get_representation(channelMask)) { 354 case AUDIO_CHANNEL_REPRESENTATION_POSITION: { 355 uint32_t channelCount = FCC_2; // stereo is default 356 if (kEnableExtendedChannels) { 357 channelCount = audio_channel_count_from_out_mask(channelMask); 358 if (channelCount < FCC_2 // mono is not supported at this time 359 || channelCount > AudioMixer::MAX_NUM_CHANNELS) { 360 return false; 361 } 362 } 363 // check that channelMask is the "canonical" one we expect for the channelCount. 364 return channelMask == audio_channel_out_mask_from_count(channelCount); 365 } 366 case AUDIO_CHANNEL_REPRESENTATION_INDEX: 367 if (kEnableExtendedChannels) { 368 const uint32_t channelCount = audio_channel_count_from_out_mask(channelMask); 369 if (channelCount >= FCC_2 // mono is not supported at this time 370 && channelCount <= AudioMixer::MAX_NUM_CHANNELS) { 371 return true; 372 } 373 } 374 return false; 375 default: 376 return false; 377 } 378 } 379 380 // Set kEnableExtendedPrecision to true to use extended precision in MixerThread 381 static const bool kEnableExtendedPrecision = true; 382 383 // Returns true if format is permitted for the PCM sink in the MixerThread 384 static inline bool isValidPcmSinkFormat(audio_format_t format) { 385 switch (format) { 386 case AUDIO_FORMAT_PCM_16_BIT: 387 return true; 388 case AUDIO_FORMAT_PCM_FLOAT: 389 case AUDIO_FORMAT_PCM_24_BIT_PACKED: 390 case AUDIO_FORMAT_PCM_32_BIT: 391 case AUDIO_FORMAT_PCM_8_24_BIT: 392 return kEnableExtendedPrecision; 393 default: 394 return false; 395 } 396 } 397 398 // standby delay for MIXER and DUPLICATING playback threads is read from property 399 // ro.audio.flinger_standbytime_ms or defaults to kDefaultStandbyTimeInNsecs 400 static nsecs_t mStandbyTimeInNsecs; 401 402 // incremented by 2 when screen state changes, bit 0 == 1 means "off" 403 // AudioFlinger::setParameters() updates, other threads read w/o lock 404 static uint32_t mScreenState; 405 406 // Internal dump utilities. 407 static const int kDumpLockRetries = 50; 408 static const int kDumpLockSleepUs = 20000; 409 static bool dumpTryLock(Mutex& mutex); 410 void dumpPermissionDenial(int fd, const Vector<String16>& args); 411 void dumpClients(int fd, const Vector<String16>& args); 412 void dumpInternals(int fd, const Vector<String16>& args); 413 414 // --- Client --- 415 class Client : public RefBase { 416 public: 417 Client(const sp<AudioFlinger>& audioFlinger, pid_t pid); 418 virtual ~Client(); 419 sp<MemoryDealer> heap() const; 420 pid_t pid() const { return mPid; } 421 sp<AudioFlinger> audioFlinger() const { return mAudioFlinger; } 422 423 private: 424 Client(const Client&); 425 Client& operator = (const Client&); 426 const sp<AudioFlinger> mAudioFlinger; 427 sp<MemoryDealer> mMemoryDealer; 428 const pid_t mPid; 429 }; 430 431 // --- Notification Client --- 432 class NotificationClient : public IBinder::DeathRecipient { 433 public: 434 NotificationClient(const sp<AudioFlinger>& audioFlinger, 435 const sp<IAudioFlingerClient>& client, 436 pid_t pid); 437 virtual ~NotificationClient(); 438 439 sp<IAudioFlingerClient> audioFlingerClient() const { return mAudioFlingerClient; } 440 441 // IBinder::DeathRecipient 442 virtual void binderDied(const wp<IBinder>& who); 443 444 private: 445 NotificationClient(const NotificationClient&); 446 NotificationClient& operator = (const NotificationClient&); 447 448 const sp<AudioFlinger> mAudioFlinger; 449 const pid_t mPid; 450 const sp<IAudioFlingerClient> mAudioFlingerClient; 451 }; 452 453 class TrackHandle; 454 class RecordHandle; 455 class RecordThread; 456 class PlaybackThread; 457 class MixerThread; 458 class DirectOutputThread; 459 class OffloadThread; 460 class DuplicatingThread; 461 class AsyncCallbackThread; 462 class Track; 463 class RecordTrack; 464 class EffectModule; 465 class EffectHandle; 466 class EffectChain; 467 468 struct AudioStreamIn; 469 470 struct stream_type_t { 471 stream_type_t() 472 : volume(1.0f), 473 mute(false) 474 { 475 } 476 float volume; 477 bool mute; 478 }; 479 480 // --- PlaybackThread --- 481 482#include "Threads.h" 483 484#include "Effects.h" 485 486#include "PatchPanel.h" 487 488 // server side of the client's IAudioTrack 489 class TrackHandle : public android::BnAudioTrack { 490 public: 491 explicit TrackHandle(const sp<PlaybackThread::Track>& track); 492 virtual ~TrackHandle(); 493 virtual sp<IMemory> getCblk() const; 494 virtual status_t start(); 495 virtual void stop(); 496 virtual void flush(); 497 virtual void pause(); 498 virtual status_t attachAuxEffect(int effectId); 499 virtual status_t setParameters(const String8& keyValuePairs); 500 virtual status_t getTimestamp(AudioTimestamp& timestamp); 501 virtual void signal(); // signal playback thread for a change in control block 502 503 virtual status_t onTransact( 504 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 505 506 private: 507 const sp<PlaybackThread::Track> mTrack; 508 }; 509 510 // server side of the client's IAudioRecord 511 class RecordHandle : public android::BnAudioRecord { 512 public: 513 explicit RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack); 514 virtual ~RecordHandle(); 515 virtual status_t start(int /*AudioSystem::sync_event_t*/ event, 516 audio_session_t triggerSession); 517 virtual void stop(); 518 virtual status_t onTransact( 519 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 520 private: 521 const sp<RecordThread::RecordTrack> mRecordTrack; 522 523 // for use from destructor 524 void stop_nonvirtual(); 525 }; 526 527 528 ThreadBase *checkThread_l(audio_io_handle_t ioHandle) const; 529 PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const; 530 MixerThread *checkMixerThread_l(audio_io_handle_t output) const; 531 RecordThread *checkRecordThread_l(audio_io_handle_t input) const; 532 sp<RecordThread> openInput_l(audio_module_handle_t module, 533 audio_io_handle_t *input, 534 audio_config_t *config, 535 audio_devices_t device, 536 const String8& address, 537 audio_source_t source, 538 audio_input_flags_t flags); 539 sp<PlaybackThread> openOutput_l(audio_module_handle_t module, 540 audio_io_handle_t *output, 541 audio_config_t *config, 542 audio_devices_t devices, 543 const String8& address, 544 audio_output_flags_t flags); 545 546 void closeOutputFinish(const sp<PlaybackThread>& thread); 547 void closeInputFinish(const sp<RecordThread>& thread); 548 549 // no range check, AudioFlinger::mLock held 550 bool streamMute_l(audio_stream_type_t stream) const 551 { return mStreamTypes[stream].mute; } 552 // no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held 553 float streamVolume_l(audio_stream_type_t stream) const 554 { return mStreamTypes[stream].volume; } 555 void ioConfigChanged(audio_io_config_event event, 556 const sp<AudioIoDescriptor>& ioDesc, 557 pid_t pid = 0); 558 559 // Allocate an audio_unique_id_t. 560 // Specific types are audio_io_handle_t, audio_session_t, effect ID (int), 561 // audio_module_handle_t, and audio_patch_handle_t. 562 // They all share the same ID space, but the namespaces are actually independent 563 // because there are separate KeyedVectors for each kind of ID. 564 // The return value is cast to the specific type depending on how the ID will be used. 565 // FIXME This API does not handle rollover to zero (for unsigned IDs), 566 // or from positive to negative (for signed IDs). 567 // Thus it may fail by returning an ID of the wrong sign, 568 // or by returning a non-unique ID. 569 // This is the internal API. For the binder API see newAudioUniqueId(). 570 audio_unique_id_t nextUniqueId(audio_unique_id_use_t use); 571 572 status_t moveEffectChain_l(audio_session_t sessionId, 573 PlaybackThread *srcThread, 574 PlaybackThread *dstThread, 575 bool reRegister); 576 577 // return thread associated with primary hardware device, or NULL 578 PlaybackThread *primaryPlaybackThread_l() const; 579 audio_devices_t primaryOutputDevice_l() const; 580 581 // return the playback thread with smallest HAL buffer size, and prefer fast 582 PlaybackThread *fastPlaybackThread_l() const; 583 584 sp<PlaybackThread> getEffectThread_l(audio_session_t sessionId, int EffectId); 585 586 587 void removeClient_l(pid_t pid); 588 void removeNotificationClient(pid_t pid); 589 bool isNonOffloadableGlobalEffectEnabled_l(); 590 void onNonOffloadableGlobalEffectEnable(); 591 592 // Store an effect chain to mOrphanEffectChains keyed vector. 593 // Called when a thread exits and effects are still attached to it. 594 // If effects are later created on the same session, they will reuse the same 595 // effect chain and same instances in the effect library. 596 // return ALREADY_EXISTS if a chain with the same session already exists in 597 // mOrphanEffectChains. Note that this should never happen as there is only one 598 // chain for a given session and it is attached to only one thread at a time. 599 status_t putOrphanEffectChain_l(const sp<EffectChain>& chain); 600 // Get an effect chain for the specified session in mOrphanEffectChains and remove 601 // it if found. Returns 0 if not found (this is the most common case). 602 sp<EffectChain> getOrphanEffectChain_l(audio_session_t session); 603 // Called when the last effect handle on an effect instance is removed. If this 604 // effect belongs to an effect chain in mOrphanEffectChains, the chain is updated 605 // and removed from mOrphanEffectChains if it does not contain any effect. 606 // Return true if the effect was found in mOrphanEffectChains, false otherwise. 607 bool updateOrphanEffectChains(const sp<EffectModule>& effect); 608 609 void broacastParametersToRecordThreads_l(const String8& keyValuePairs); 610 611 // AudioStreamIn is immutable, so their fields are const. 612 // For emphasis, we could also make all pointers to them be "const *", 613 // but that would clutter the code unnecessarily. 614 615 struct AudioStreamIn { 616 AudioHwDevice* const audioHwDev; 617 audio_stream_in_t* const stream; 618 audio_input_flags_t flags; 619 620 audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); } 621 622 AudioStreamIn(AudioHwDevice *dev, audio_stream_in_t *in, audio_input_flags_t flags) : 623 audioHwDev(dev), stream(in), flags(flags) {} 624 }; 625 626 // for mAudioSessionRefs only 627 struct AudioSessionRef { 628 AudioSessionRef(audio_session_t sessionid, pid_t pid) : 629 mSessionid(sessionid), mPid(pid), mCnt(1) {} 630 const audio_session_t mSessionid; 631 const pid_t mPid; 632 int mCnt; 633 }; 634 635 mutable Mutex mLock; 636 sp<LockWatch> mLockWatch; 637 // protects mClients and mNotificationClients. 638 // must be locked after mLock and ThreadBase::mLock if both must be locked 639 // avoids acquiring AudioFlinger::mLock from inside thread loop. 640 mutable Mutex mClientLock; 641 // protected by mClientLock 642 DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client() 643 644 mutable Mutex mHardwareLock; 645 // NOTE: If both mLock and mHardwareLock mutexes must be held, 646 // always take mLock before mHardwareLock 647 648 // These two fields are immutable after onFirstRef(), so no lock needed to access 649 AudioHwDevice* mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL 650 DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*> mAudioHwDevs; 651 652 // for dump, indicates which hardware operation is currently in progress (but not stream ops) 653 enum hardware_call_state { 654 AUDIO_HW_IDLE = 0, // no operation in progress 655 AUDIO_HW_INIT, // init_check 656 AUDIO_HW_OUTPUT_OPEN, // open_output_stream 657 AUDIO_HW_OUTPUT_CLOSE, // unused 658 AUDIO_HW_INPUT_OPEN, // unused 659 AUDIO_HW_INPUT_CLOSE, // unused 660 AUDIO_HW_STANDBY, // unused 661 AUDIO_HW_SET_MASTER_VOLUME, // set_master_volume 662 AUDIO_HW_GET_ROUTING, // unused 663 AUDIO_HW_SET_ROUTING, // unused 664 AUDIO_HW_GET_MODE, // unused 665 AUDIO_HW_SET_MODE, // set_mode 666 AUDIO_HW_GET_MIC_MUTE, // get_mic_mute 667 AUDIO_HW_SET_MIC_MUTE, // set_mic_mute 668 AUDIO_HW_SET_VOICE_VOLUME, // set_voice_volume 669 AUDIO_HW_SET_PARAMETER, // set_parameters 670 AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size 671 AUDIO_HW_GET_MASTER_VOLUME, // get_master_volume 672 AUDIO_HW_GET_PARAMETER, // get_parameters 673 AUDIO_HW_SET_MASTER_MUTE, // set_master_mute 674 AUDIO_HW_GET_MASTER_MUTE, // get_master_mute 675 }; 676 677 mutable hardware_call_state mHardwareStatus; // for dump only 678 679 680 DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads; 681 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 682 683 // member variables below are protected by mLock 684 float mMasterVolume; 685 bool mMasterMute; 686 // end of variables protected by mLock 687 688 DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads; 689 690 // protected by mClientLock 691 DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients; 692 693 // updated by atomic_fetch_add_explicit 694 volatile atomic_uint_fast32_t mNextUniqueIds[AUDIO_UNIQUE_ID_USE_MAX]; 695 696 audio_mode_t mMode; 697 bool mBtNrecIsOff; 698 699 // protected by mLock 700 Vector<AudioSessionRef*> mAudioSessionRefs; 701 702 float masterVolume_l() const; 703 bool masterMute_l() const; 704 audio_module_handle_t loadHwModule_l(const char *name); 705 706 Vector < sp<SyncEvent> > mPendingSyncEvents; // sync events awaiting for a session 707 // to be created 708 709 // Effect chains without a valid thread 710 DefaultKeyedVector< audio_session_t , sp<EffectChain> > mOrphanEffectChains; 711 712 // list of sessions for which a valid HW A/V sync ID was retrieved from the HAL 713 DefaultKeyedVector< audio_session_t , audio_hw_sync_t >mHwAvSyncIds; 714private: 715 sp<Client> registerPid(pid_t pid); // always returns non-0 716 717 // for use from destructor 718 status_t closeOutput_nonvirtual(audio_io_handle_t output); 719 void closeOutputInternal_l(const sp<PlaybackThread>& thread); 720 status_t closeInput_nonvirtual(audio_io_handle_t input); 721 void closeInputInternal_l(const sp<RecordThread>& thread); 722 void setAudioHwSyncForSession_l(PlaybackThread *thread, audio_session_t sessionId); 723 724 status_t checkStreamType(audio_stream_type_t stream) const; 725 726#ifdef TEE_SINK 727 // all record threads serially share a common tee sink, which is re-created on format change 728 sp<NBAIO_Sink> mRecordTeeSink; 729 sp<NBAIO_Source> mRecordTeeSource; 730#endif 731 732public: 733 734#ifdef TEE_SINK 735 // tee sink, if enabled by property, allows dumpsys to write most recent audio to .wav file 736 static void dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id = 0); 737 738 // whether tee sink is enabled by property 739 static bool mTeeSinkInputEnabled; 740 static bool mTeeSinkOutputEnabled; 741 static bool mTeeSinkTrackEnabled; 742 743 // runtime configured size of each tee sink pipe, in frames 744 static size_t mTeeSinkInputFrames; 745 static size_t mTeeSinkOutputFrames; 746 static size_t mTeeSinkTrackFrames; 747 748 // compile-time default size of tee sink pipes, in frames 749 // 0x200000 stereo 16-bit PCM frames = 47.5 seconds at 44.1 kHz, 8 megabytes 750 static const size_t kTeeSinkInputFramesDefault = 0x200000; 751 static const size_t kTeeSinkOutputFramesDefault = 0x200000; 752 static const size_t kTeeSinkTrackFramesDefault = 0x200000; 753#endif 754 755 // This method reads from a variable without mLock, but the variable is updated under mLock. So 756 // we might read a stale value, or a value that's inconsistent with respect to other variables. 757 // In this case, it's safe because the return value isn't used for making an important decision. 758 // The reason we don't want to take mLock is because it could block the caller for a long time. 759 bool isLowRamDevice() const { return mIsLowRamDevice; } 760 761private: 762 bool mIsLowRamDevice; 763 bool mIsDeviceTypeKnown; 764 nsecs_t mGlobalEffectEnableTime; // when a global effect was last enabled 765 766 sp<PatchPanel> mPatchPanel; 767 sp<EffectsFactoryHalInterface> mEffectsFactoryHal; 768 769 bool mSystemReady; 770}; 771 772#undef INCLUDING_FROM_AUDIOFLINGER_H 773 774const char *formatToString(audio_format_t format); 775String8 inputFlagsToString(audio_input_flags_t flags); 776String8 outputFlagsToString(audio_output_flags_t flags); 777String8 devicesToString(audio_devices_t devices); 778const char *sourceToString(audio_source_t source); 779 780// ---------------------------------------------------------------------------- 781 782} // namespace android 783 784#endif // ANDROID_AUDIO_FLINGER_H 785