AudioFlinger.h revision 4a8308b11b92e608cdaf29f73f7919e75706f9a2
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 62#include <powermanager/IPowerManager.h> 63 64#include <media/nbaio/NBLog.h> 65#include <private/media/AudioTrackShared.h> 66 67namespace android { 68 69struct audio_track_cblk_t; 70struct effect_param_cblk_t; 71class AudioMixer; 72class AudioBuffer; 73class AudioResampler; 74class FastMixer; 75class PassthruBufferProvider; 76class ServerProxy; 77 78// ---------------------------------------------------------------------------- 79 80static const nsecs_t kDefaultStandbyTimeInNsecs = seconds(3); 81 82 83// Max shared memory size for audio tracks and audio records per client process 84static const size_t kClientSharedHeapSizeBytes = 1024*1024; 85// Shared memory size multiplier for non low ram devices 86static const size_t kClientSharedHeapSizeMultiplier = 4; 87 88#define INCLUDING_FROM_AUDIOFLINGER_H 89 90class AudioFlinger : 91 public BinderService<AudioFlinger>, 92 public BnAudioFlinger 93{ 94 friend class BinderService<AudioFlinger>; // for AudioFlinger() 95public: 96 static const char* getServiceName() ANDROID_API { return "media.audio_flinger"; } 97 98 virtual status_t dump(int fd, const Vector<String16>& args); 99 100 // IAudioFlinger interface, in binder opcode order 101 virtual sp<IAudioTrack> createTrack( 102 audio_stream_type_t streamType, 103 uint32_t sampleRate, 104 audio_format_t format, 105 audio_channel_mask_t channelMask, 106 size_t *pFrameCount, 107 IAudioFlinger::track_flags_t *flags, 108 const sp<IMemory>& sharedBuffer, 109 audio_io_handle_t output, 110 pid_t tid, 111 audio_session_t *sessionId, 112 int clientUid, 113 status_t *status /*non-NULL*/); 114 115 virtual sp<IAudioRecord> openRecord( 116 audio_io_handle_t input, 117 uint32_t sampleRate, 118 audio_format_t format, 119 audio_channel_mask_t channelMask, 120 const String16& opPackageName, 121 size_t *pFrameCount, 122 IAudioFlinger::track_flags_t *flags, 123 pid_t tid, 124 int clientUid, 125 audio_session_t *sessionId, 126 size_t *notificationFrames, 127 sp<IMemory>& cblk, 128 sp<IMemory>& buffers, 129 status_t *status /*non-NULL*/); 130 131 virtual uint32_t sampleRate(audio_io_handle_t ioHandle) const; 132 virtual audio_format_t format(audio_io_handle_t output) const; 133 virtual size_t frameCount(audio_io_handle_t ioHandle) const; 134 virtual size_t frameCountHAL(audio_io_handle_t ioHandle) const; 135 virtual uint32_t latency(audio_io_handle_t output) const; 136 137 virtual status_t setMasterVolume(float value); 138 virtual status_t setMasterMute(bool muted); 139 140 virtual float masterVolume() const; 141 virtual bool masterMute() const; 142 143 virtual status_t setStreamVolume(audio_stream_type_t stream, float value, 144 audio_io_handle_t output); 145 virtual status_t setStreamMute(audio_stream_type_t stream, bool muted); 146 147 virtual float streamVolume(audio_stream_type_t stream, 148 audio_io_handle_t output) const; 149 virtual bool streamMute(audio_stream_type_t stream) const; 150 151 virtual status_t setMode(audio_mode_t mode); 152 153 virtual status_t setMicMute(bool state); 154 virtual bool getMicMute() const; 155 156 virtual status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs); 157 virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) const; 158 159 virtual void registerClient(const sp<IAudioFlingerClient>& client); 160 161 virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, 162 audio_channel_mask_t channelMask) const; 163 164 virtual status_t openOutput(audio_module_handle_t module, 165 audio_io_handle_t *output, 166 audio_config_t *config, 167 audio_devices_t *devices, 168 const String8& address, 169 uint32_t *latencyMs, 170 audio_output_flags_t flags); 171 172 virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1, 173 audio_io_handle_t output2); 174 175 virtual status_t closeOutput(audio_io_handle_t output); 176 177 virtual status_t suspendOutput(audio_io_handle_t output); 178 179 virtual status_t restoreOutput(audio_io_handle_t output); 180 181 virtual status_t openInput(audio_module_handle_t module, 182 audio_io_handle_t *input, 183 audio_config_t *config, 184 audio_devices_t *device, 185 const String8& address, 186 audio_source_t source, 187 audio_input_flags_t flags); 188 189 virtual status_t closeInput(audio_io_handle_t input); 190 191 virtual status_t invalidateStream(audio_stream_type_t stream); 192 193 virtual status_t setVoiceVolume(float volume); 194 195 virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 196 audio_io_handle_t output) const; 197 198 virtual uint32_t getInputFramesLost(audio_io_handle_t ioHandle) const; 199 200 // This is the binder API. For the internal API see nextUniqueId(). 201 virtual audio_unique_id_t newAudioUniqueId(audio_unique_id_use_t use); 202 203 virtual void acquireAudioSessionId(audio_session_t audioSession, pid_t pid); 204 205 virtual void releaseAudioSessionId(audio_session_t audioSession, pid_t pid); 206 207 virtual status_t queryNumberEffects(uint32_t *numEffects) const; 208 209 virtual status_t queryEffect(uint32_t index, effect_descriptor_t *descriptor) const; 210 211 virtual status_t getEffectDescriptor(const effect_uuid_t *pUuid, 212 effect_descriptor_t *descriptor) const; 213 214 virtual sp<IEffect> createEffect( 215 effect_descriptor_t *pDesc, 216 const sp<IEffectClient>& effectClient, 217 int32_t priority, 218 audio_io_handle_t io, 219 audio_session_t sessionId, 220 const String16& opPackageName, 221 status_t *status /*non-NULL*/, 222 int *id, 223 int *enabled); 224 225 virtual status_t moveEffects(audio_session_t sessionId, audio_io_handle_t srcOutput, 226 audio_io_handle_t dstOutput); 227 228 virtual audio_module_handle_t loadHwModule(const char *name); 229 230 virtual uint32_t getPrimaryOutputSamplingRate(); 231 virtual size_t getPrimaryOutputFrameCount(); 232 233 virtual status_t setLowRamDevice(bool isLowRamDevice); 234 235 /* List available audio ports and their attributes */ 236 virtual status_t listAudioPorts(unsigned int *num_ports, 237 struct audio_port *ports); 238 239 /* Get attributes for a given audio port */ 240 virtual status_t getAudioPort(struct audio_port *port); 241 242 /* Create an audio patch between several source and sink ports */ 243 virtual status_t createAudioPatch(const struct audio_patch *patch, 244 audio_patch_handle_t *handle); 245 246 /* Release an audio patch */ 247 virtual status_t releaseAudioPatch(audio_patch_handle_t handle); 248 249 /* List existing audio patches */ 250 virtual status_t listAudioPatches(unsigned int *num_patches, 251 struct audio_patch *patches); 252 253 /* Set audio port configuration */ 254 virtual status_t setAudioPortConfig(const struct audio_port_config *config); 255 256 /* Get the HW synchronization source used for an audio session */ 257 virtual audio_hw_sync_t getAudioHwSyncForSession(audio_session_t sessionId); 258 259 /* Indicate JAVA services are ready (scheduling, power management ...) */ 260 virtual status_t systemReady(); 261 262 virtual status_t onTransact( 263 uint32_t code, 264 const Parcel& data, 265 Parcel* reply, 266 uint32_t flags); 267 268 // end of IAudioFlinger interface 269 270 sp<NBLog::Writer> newWriter_l(size_t size, const char *name); 271 void unregisterWriter(const sp<NBLog::Writer>& writer); 272private: 273 static const size_t kLogMemorySize = 40 * 1024; 274 sp<MemoryDealer> mLogMemoryDealer; // == 0 when NBLog is disabled 275 // When a log writer is unregistered, it is done lazily so that media.log can continue to see it 276 // for as long as possible. The memory is only freed when it is needed for another log writer. 277 Vector< sp<NBLog::Writer> > mUnregisteredWriters; 278 Mutex mUnregisteredWritersLock; 279public: 280 281 class SyncEvent; 282 283 typedef void (*sync_event_callback_t)(const wp<SyncEvent>& event) ; 284 285 class SyncEvent : public RefBase { 286 public: 287 SyncEvent(AudioSystem::sync_event_t type, 288 audio_session_t triggerSession, 289 audio_session_t listenerSession, 290 sync_event_callback_t callBack, 291 wp<RefBase> cookie) 292 : mType(type), mTriggerSession(triggerSession), mListenerSession(listenerSession), 293 mCallback(callBack), mCookie(cookie) 294 {} 295 296 virtual ~SyncEvent() {} 297 298 void trigger() { Mutex::Autolock _l(mLock); if (mCallback) mCallback(this); } 299 bool isCancelled() const { Mutex::Autolock _l(mLock); return (mCallback == NULL); } 300 void cancel() { Mutex::Autolock _l(mLock); mCallback = NULL; } 301 AudioSystem::sync_event_t type() const { return mType; } 302 audio_session_t triggerSession() const { return mTriggerSession; } 303 audio_session_t listenerSession() const { return mListenerSession; } 304 wp<RefBase> cookie() const { return mCookie; } 305 306 private: 307 const AudioSystem::sync_event_t mType; 308 const audio_session_t mTriggerSession; 309 const audio_session_t mListenerSession; 310 sync_event_callback_t mCallback; 311 const wp<RefBase> mCookie; 312 mutable Mutex mLock; 313 }; 314 315 sp<SyncEvent> createSyncEvent(AudioSystem::sync_event_t type, 316 audio_session_t triggerSession, 317 audio_session_t listenerSession, 318 sync_event_callback_t callBack, 319 wp<RefBase> cookie); 320 321private: 322 323 audio_mode_t getMode() const { return mMode; } 324 325 bool btNrecIsOff() const { return mBtNrecIsOff; } 326 327 AudioFlinger() ANDROID_API; 328 virtual ~AudioFlinger(); 329 330 // call in any IAudioFlinger method that accesses mPrimaryHardwareDev 331 status_t initCheck() const { return mPrimaryHardwareDev == NULL ? 332 NO_INIT : NO_ERROR; } 333 334 // RefBase 335 virtual void onFirstRef(); 336 337 AudioHwDevice* findSuitableHwDev_l(audio_module_handle_t module, 338 audio_devices_t devices); 339 void purgeStaleEffects_l(); 340 341 // Set kEnableExtendedChannels to true to enable greater than stereo output 342 // for the MixerThread and device sink. Number of channels allowed is 343 // FCC_2 <= channels <= AudioMixer::MAX_NUM_CHANNELS. 344 static const bool kEnableExtendedChannels = true; 345 346 // Returns true if channel mask is permitted for the PCM sink in the MixerThread 347 static inline bool isValidPcmSinkChannelMask(audio_channel_mask_t channelMask) { 348 switch (audio_channel_mask_get_representation(channelMask)) { 349 case AUDIO_CHANNEL_REPRESENTATION_POSITION: { 350 uint32_t channelCount = FCC_2; // stereo is default 351 if (kEnableExtendedChannels) { 352 channelCount = audio_channel_count_from_out_mask(channelMask); 353 if (channelCount < FCC_2 // mono is not supported at this time 354 || channelCount > AudioMixer::MAX_NUM_CHANNELS) { 355 return false; 356 } 357 } 358 // check that channelMask is the "canonical" one we expect for the channelCount. 359 return channelMask == audio_channel_out_mask_from_count(channelCount); 360 } 361 case AUDIO_CHANNEL_REPRESENTATION_INDEX: 362 if (kEnableExtendedChannels) { 363 const uint32_t channelCount = audio_channel_count_from_out_mask(channelMask); 364 if (channelCount >= FCC_2 // mono is not supported at this time 365 && channelCount <= AudioMixer::MAX_NUM_CHANNELS) { 366 return true; 367 } 368 } 369 return false; 370 default: 371 return false; 372 } 373 } 374 375 // Set kEnableExtendedPrecision to true to use extended precision in MixerThread 376 static const bool kEnableExtendedPrecision = true; 377 378 // Returns true if format is permitted for the PCM sink in the MixerThread 379 static inline bool isValidPcmSinkFormat(audio_format_t format) { 380 switch (format) { 381 case AUDIO_FORMAT_PCM_16_BIT: 382 return true; 383 case AUDIO_FORMAT_PCM_FLOAT: 384 case AUDIO_FORMAT_PCM_24_BIT_PACKED: 385 case AUDIO_FORMAT_PCM_32_BIT: 386 case AUDIO_FORMAT_PCM_8_24_BIT: 387 return kEnableExtendedPrecision; 388 default: 389 return false; 390 } 391 } 392 393 // standby delay for MIXER and DUPLICATING playback threads is read from property 394 // ro.audio.flinger_standbytime_ms or defaults to kDefaultStandbyTimeInNsecs 395 static nsecs_t mStandbyTimeInNsecs; 396 397 // incremented by 2 when screen state changes, bit 0 == 1 means "off" 398 // AudioFlinger::setParameters() updates, other threads read w/o lock 399 static uint32_t mScreenState; 400 401 // Internal dump utilities. 402 static const int kDumpLockRetries = 50; 403 static const int kDumpLockSleepUs = 20000; 404 static bool dumpTryLock(Mutex& mutex); 405 void dumpPermissionDenial(int fd, const Vector<String16>& args); 406 void dumpClients(int fd, const Vector<String16>& args); 407 void dumpInternals(int fd, const Vector<String16>& args); 408 409 // --- Client --- 410 class Client : public RefBase { 411 public: 412 Client(const sp<AudioFlinger>& audioFlinger, pid_t pid); 413 virtual ~Client(); 414 sp<MemoryDealer> heap() const; 415 pid_t pid() const { return mPid; } 416 sp<AudioFlinger> audioFlinger() const { return mAudioFlinger; } 417 418 private: 419 Client(const Client&); 420 Client& operator = (const Client&); 421 const sp<AudioFlinger> mAudioFlinger; 422 sp<MemoryDealer> mMemoryDealer; 423 const pid_t mPid; 424 }; 425 426 // --- Notification Client --- 427 class NotificationClient : public IBinder::DeathRecipient { 428 public: 429 NotificationClient(const sp<AudioFlinger>& audioFlinger, 430 const sp<IAudioFlingerClient>& client, 431 pid_t pid); 432 virtual ~NotificationClient(); 433 434 sp<IAudioFlingerClient> audioFlingerClient() const { return mAudioFlingerClient; } 435 436 // IBinder::DeathRecipient 437 virtual void binderDied(const wp<IBinder>& who); 438 439 private: 440 NotificationClient(const NotificationClient&); 441 NotificationClient& operator = (const NotificationClient&); 442 443 const sp<AudioFlinger> mAudioFlinger; 444 const pid_t mPid; 445 const sp<IAudioFlingerClient> mAudioFlingerClient; 446 }; 447 448 class TrackHandle; 449 class RecordHandle; 450 class RecordThread; 451 class PlaybackThread; 452 class MixerThread; 453 class DirectOutputThread; 454 class OffloadThread; 455 class DuplicatingThread; 456 class AsyncCallbackThread; 457 class Track; 458 class RecordTrack; 459 class EffectModule; 460 class EffectHandle; 461 class EffectChain; 462 463 struct AudioStreamIn; 464 465 struct stream_type_t { 466 stream_type_t() 467 : volume(1.0f), 468 mute(false) 469 { 470 } 471 float volume; 472 bool mute; 473 }; 474 475 // --- PlaybackThread --- 476 477#include "Threads.h" 478 479#include "Effects.h" 480 481#include "PatchPanel.h" 482 483 // server side of the client's IAudioTrack 484 class TrackHandle : public android::BnAudioTrack { 485 public: 486 TrackHandle(const sp<PlaybackThread::Track>& track); 487 virtual ~TrackHandle(); 488 virtual sp<IMemory> getCblk() const; 489 virtual status_t start(); 490 virtual void stop(); 491 virtual void flush(); 492 virtual void pause(); 493 virtual status_t attachAuxEffect(int effectId); 494 virtual status_t setParameters(const String8& keyValuePairs); 495 virtual status_t getTimestamp(AudioTimestamp& timestamp); 496 virtual void signal(); // signal playback thread for a change in control block 497 498 virtual status_t onTransact( 499 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 500 501 private: 502 const sp<PlaybackThread::Track> mTrack; 503 }; 504 505 // server side of the client's IAudioRecord 506 class RecordHandle : public android::BnAudioRecord { 507 public: 508 RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack); 509 virtual ~RecordHandle(); 510 virtual status_t start(int /*AudioSystem::sync_event_t*/ event, 511 audio_session_t triggerSession); 512 virtual void stop(); 513 virtual status_t onTransact( 514 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 515 private: 516 const sp<RecordThread::RecordTrack> mRecordTrack; 517 518 // for use from destructor 519 void stop_nonvirtual(); 520 }; 521 522 523 ThreadBase *checkThread_l(audio_io_handle_t ioHandle) const; 524 PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const; 525 MixerThread *checkMixerThread_l(audio_io_handle_t output) const; 526 RecordThread *checkRecordThread_l(audio_io_handle_t input) const; 527 sp<RecordThread> openInput_l(audio_module_handle_t module, 528 audio_io_handle_t *input, 529 audio_config_t *config, 530 audio_devices_t device, 531 const String8& address, 532 audio_source_t source, 533 audio_input_flags_t flags); 534 sp<PlaybackThread> openOutput_l(audio_module_handle_t module, 535 audio_io_handle_t *output, 536 audio_config_t *config, 537 audio_devices_t devices, 538 const String8& address, 539 audio_output_flags_t flags); 540 541 void closeOutputFinish(sp<PlaybackThread> thread); 542 void closeInputFinish(sp<RecordThread> thread); 543 544 // no range check, AudioFlinger::mLock held 545 bool streamMute_l(audio_stream_type_t stream) const 546 { return mStreamTypes[stream].mute; } 547 // no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held 548 float streamVolume_l(audio_stream_type_t stream) const 549 { return mStreamTypes[stream].volume; } 550 void ioConfigChanged(audio_io_config_event event, 551 const sp<AudioIoDescriptor>& ioDesc, 552 pid_t pid = 0); 553 554 // Allocate an audio_unique_id_t. 555 // Specific types are audio_io_handle_t, audio_session_t, effect ID (int), 556 // audio_module_handle_t, and audio_patch_handle_t. 557 // They all share the same ID space, but the namespaces are actually independent 558 // because there are separate KeyedVectors for each kind of ID. 559 // The return value is cast to the specific type depending on how the ID will be used. 560 // FIXME This API does not handle rollover to zero (for unsigned IDs), 561 // or from positive to negative (for signed IDs). 562 // Thus it may fail by returning an ID of the wrong sign, 563 // or by returning a non-unique ID. 564 // This is the internal API. For the binder API see newAudioUniqueId(). 565 audio_unique_id_t nextUniqueId(audio_unique_id_use_t use); 566 567 status_t moveEffectChain_l(audio_session_t sessionId, 568 PlaybackThread *srcThread, 569 PlaybackThread *dstThread, 570 bool reRegister); 571 572 // return thread associated with primary hardware device, or NULL 573 PlaybackThread *primaryPlaybackThread_l() const; 574 audio_devices_t primaryOutputDevice_l() const; 575 576 sp<PlaybackThread> getEffectThread_l(audio_session_t sessionId, int EffectId); 577 578 579 void removeClient_l(pid_t pid); 580 void removeNotificationClient(pid_t pid); 581 bool isNonOffloadableGlobalEffectEnabled_l(); 582 void onNonOffloadableGlobalEffectEnable(); 583 584 // Store an effect chain to mOrphanEffectChains keyed vector. 585 // Called when a thread exits and effects are still attached to it. 586 // If effects are later created on the same session, they will reuse the same 587 // effect chain and same instances in the effect library. 588 // return ALREADY_EXISTS if a chain with the same session already exists in 589 // mOrphanEffectChains. Note that this should never happen as there is only one 590 // chain for a given session and it is attached to only one thread at a time. 591 status_t putOrphanEffectChain_l(const sp<EffectChain>& chain); 592 // Get an effect chain for the specified session in mOrphanEffectChains and remove 593 // it if found. Returns 0 if not found (this is the most common case). 594 sp<EffectChain> getOrphanEffectChain_l(audio_session_t session); 595 // Called when the last effect handle on an effect instance is removed. If this 596 // effect belongs to an effect chain in mOrphanEffectChains, the chain is updated 597 // and removed from mOrphanEffectChains if it does not contain any effect. 598 // Return true if the effect was found in mOrphanEffectChains, false otherwise. 599 bool updateOrphanEffectChains(const sp<EffectModule>& effect); 600 601 void broacastParametersToRecordThreads_l(const String8& keyValuePairs); 602 603 // AudioStreamIn is immutable, so their fields are const. 604 // For emphasis, we could also make all pointers to them be "const *", 605 // but that would clutter the code unnecessarily. 606 607 struct AudioStreamIn { 608 AudioHwDevice* const audioHwDev; 609 audio_stream_in_t* const stream; 610 611 audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); } 612 613 AudioStreamIn(AudioHwDevice *dev, audio_stream_in_t *in) : 614 audioHwDev(dev), stream(in) {} 615 }; 616 617 // for mAudioSessionRefs only 618 struct AudioSessionRef { 619 AudioSessionRef(audio_session_t sessionid, pid_t pid) : 620 mSessionid(sessionid), mPid(pid), mCnt(1) {} 621 const audio_session_t mSessionid; 622 const pid_t mPid; 623 int mCnt; 624 }; 625 626 mutable Mutex mLock; 627 // protects mClients and mNotificationClients. 628 // must be locked after mLock and ThreadBase::mLock if both must be locked 629 // avoids acquiring AudioFlinger::mLock from inside thread loop. 630 mutable Mutex mClientLock; 631 // protected by mClientLock 632 DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client() 633 634 mutable Mutex mHardwareLock; 635 // NOTE: If both mLock and mHardwareLock mutexes must be held, 636 // always take mLock before mHardwareLock 637 638 // These two fields are immutable after onFirstRef(), so no lock needed to access 639 AudioHwDevice* mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL 640 DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*> mAudioHwDevs; 641 642 // for dump, indicates which hardware operation is currently in progress (but not stream ops) 643 enum hardware_call_state { 644 AUDIO_HW_IDLE = 0, // no operation in progress 645 AUDIO_HW_INIT, // init_check 646 AUDIO_HW_OUTPUT_OPEN, // open_output_stream 647 AUDIO_HW_OUTPUT_CLOSE, // unused 648 AUDIO_HW_INPUT_OPEN, // unused 649 AUDIO_HW_INPUT_CLOSE, // unused 650 AUDIO_HW_STANDBY, // unused 651 AUDIO_HW_SET_MASTER_VOLUME, // set_master_volume 652 AUDIO_HW_GET_ROUTING, // unused 653 AUDIO_HW_SET_ROUTING, // unused 654 AUDIO_HW_GET_MODE, // unused 655 AUDIO_HW_SET_MODE, // set_mode 656 AUDIO_HW_GET_MIC_MUTE, // get_mic_mute 657 AUDIO_HW_SET_MIC_MUTE, // set_mic_mute 658 AUDIO_HW_SET_VOICE_VOLUME, // set_voice_volume 659 AUDIO_HW_SET_PARAMETER, // set_parameters 660 AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size 661 AUDIO_HW_GET_MASTER_VOLUME, // get_master_volume 662 AUDIO_HW_GET_PARAMETER, // get_parameters 663 AUDIO_HW_SET_MASTER_MUTE, // set_master_mute 664 AUDIO_HW_GET_MASTER_MUTE, // get_master_mute 665 }; 666 667 mutable hardware_call_state mHardwareStatus; // for dump only 668 669 670 DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads; 671 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 672 673 // member variables below are protected by mLock 674 float mMasterVolume; 675 bool mMasterMute; 676 // end of variables protected by mLock 677 678 DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads; 679 680 // protected by mClientLock 681 DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients; 682 683 // updated by atomic_fetch_add_explicit 684 volatile atomic_uint_fast32_t mNextUniqueIds[AUDIO_UNIQUE_ID_USE_MAX]; 685 686 audio_mode_t mMode; 687 bool mBtNrecIsOff; 688 689 // protected by mLock 690 Vector<AudioSessionRef*> mAudioSessionRefs; 691 692 float masterVolume_l() const; 693 bool masterMute_l() const; 694 audio_module_handle_t loadHwModule_l(const char *name); 695 696 Vector < sp<SyncEvent> > mPendingSyncEvents; // sync events awaiting for a session 697 // to be created 698 699 // Effect chains without a valid thread 700 DefaultKeyedVector< audio_session_t , sp<EffectChain> > mOrphanEffectChains; 701 702 // list of sessions for which a valid HW A/V sync ID was retrieved from the HAL 703 DefaultKeyedVector< audio_session_t , audio_hw_sync_t >mHwAvSyncIds; 704private: 705 sp<Client> registerPid(pid_t pid); // always returns non-0 706 707 // for use from destructor 708 status_t closeOutput_nonvirtual(audio_io_handle_t output); 709 void closeOutputInternal_l(sp<PlaybackThread> thread); 710 status_t closeInput_nonvirtual(audio_io_handle_t input); 711 void closeInputInternal_l(sp<RecordThread> thread); 712 void setAudioHwSyncForSession_l(PlaybackThread *thread, audio_session_t sessionId); 713 714 status_t checkStreamType(audio_stream_type_t stream) const; 715 716#ifdef TEE_SINK 717 // all record threads serially share a common tee sink, which is re-created on format change 718 sp<NBAIO_Sink> mRecordTeeSink; 719 sp<NBAIO_Source> mRecordTeeSource; 720#endif 721 722public: 723 724#ifdef TEE_SINK 725 // tee sink, if enabled by property, allows dumpsys to write most recent audio to .wav file 726 static void dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id = 0); 727 728 // whether tee sink is enabled by property 729 static bool mTeeSinkInputEnabled; 730 static bool mTeeSinkOutputEnabled; 731 static bool mTeeSinkTrackEnabled; 732 733 // runtime configured size of each tee sink pipe, in frames 734 static size_t mTeeSinkInputFrames; 735 static size_t mTeeSinkOutputFrames; 736 static size_t mTeeSinkTrackFrames; 737 738 // compile-time default size of tee sink pipes, in frames 739 // 0x200000 stereo 16-bit PCM frames = 47.5 seconds at 44.1 kHz, 8 megabytes 740 static const size_t kTeeSinkInputFramesDefault = 0x200000; 741 static const size_t kTeeSinkOutputFramesDefault = 0x200000; 742 static const size_t kTeeSinkTrackFramesDefault = 0x200000; 743#endif 744 745 // This method reads from a variable without mLock, but the variable is updated under mLock. So 746 // we might read a stale value, or a value that's inconsistent with respect to other variables. 747 // In this case, it's safe because the return value isn't used for making an important decision. 748 // The reason we don't want to take mLock is because it could block the caller for a long time. 749 bool isLowRamDevice() const { return mIsLowRamDevice; } 750 751private: 752 bool mIsLowRamDevice; 753 bool mIsDeviceTypeKnown; 754 nsecs_t mGlobalEffectEnableTime; // when a global effect was last enabled 755 756 sp<PatchPanel> mPatchPanel; 757 758 bool mSystemReady; 759}; 760 761#undef INCLUDING_FROM_AUDIOFLINGER_H 762 763const char *formatToString(audio_format_t format); 764String8 inputFlagsToString(audio_input_flags_t flags); 765String8 outputFlagsToString(audio_output_flags_t flags); 766String8 devicesToString(audio_devices_t devices); 767const char *sourceToString(audio_source_t source); 768 769// ---------------------------------------------------------------------------- 770 771} // namespace android 772 773#endif // ANDROID_AUDIO_FLINGER_H 774