AudioFlinger.h revision 913d06c099bd689375483a839e11057ccf284d1c
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 <system/audio_policy.h> 47 48#include <media/audiohal/StreamHalInterface.h> 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 DeviceHalInterface; 75class DevicesFactoryHalInterface; 76class EffectsFactoryHalInterface; 77class FastMixer; 78class PassthruBufferProvider; 79class ServerProxy; 80 81// ---------------------------------------------------------------------------- 82 83static const nsecs_t kDefaultStandbyTimeInNsecs = seconds(3); 84 85 86// Max shared memory size for audio tracks and audio records per client process 87static const size_t kClientSharedHeapSizeBytes = 1024*1024; 88// Shared memory size multiplier for non low ram devices 89static const size_t kClientSharedHeapSizeMultiplier = 4; 90 91#define INCLUDING_FROM_AUDIOFLINGER_H 92 93class AudioFlinger : 94 public BinderService<AudioFlinger>, 95 public BnAudioFlinger 96{ 97 friend class BinderService<AudioFlinger>; // for AudioFlinger() 98public: 99 static const char* getServiceName() ANDROID_API { return "media.audio_flinger"; } 100 101 virtual status_t dump(int fd, const Vector<String16>& args); 102 103 // IAudioFlinger interface, in binder opcode order 104 virtual sp<IAudioTrack> createTrack( 105 audio_stream_type_t streamType, 106 uint32_t sampleRate, 107 audio_format_t format, 108 audio_channel_mask_t channelMask, 109 size_t *pFrameCount, 110 audio_output_flags_t *flags, 111 const sp<IMemory>& sharedBuffer, 112 audio_io_handle_t output, 113 pid_t pid, 114 pid_t tid, 115 audio_session_t *sessionId, 116 int clientUid, 117 status_t *status /*non-NULL*/); 118 119 virtual sp<IAudioRecord> openRecord( 120 audio_io_handle_t input, 121 uint32_t sampleRate, 122 audio_format_t format, 123 audio_channel_mask_t channelMask, 124 const String16& opPackageName, 125 size_t *pFrameCount, 126 audio_input_flags_t *flags, 127 pid_t pid, 128 pid_t tid, 129 int clientUid, 130 audio_session_t *sessionId, 131 size_t *notificationFrames, 132 sp<IMemory>& cblk, 133 sp<IMemory>& buffers, 134 status_t *status /*non-NULL*/); 135 136 virtual uint32_t sampleRate(audio_io_handle_t ioHandle) const; 137 virtual audio_format_t format(audio_io_handle_t output) const; 138 virtual size_t frameCount(audio_io_handle_t ioHandle) const; 139 virtual size_t frameCountHAL(audio_io_handle_t ioHandle) const; 140 virtual uint32_t latency(audio_io_handle_t output) const; 141 142 virtual status_t setMasterVolume(float value); 143 virtual status_t setMasterMute(bool muted); 144 145 virtual float masterVolume() const; 146 virtual bool masterMute() const; 147 148 virtual status_t setStreamVolume(audio_stream_type_t stream, float value, 149 audio_io_handle_t output); 150 virtual status_t setStreamMute(audio_stream_type_t stream, bool muted); 151 152 virtual float streamVolume(audio_stream_type_t stream, 153 audio_io_handle_t output) const; 154 virtual bool streamMute(audio_stream_type_t stream) const; 155 156 virtual status_t setMode(audio_mode_t mode); 157 158 virtual status_t setMicMute(bool state); 159 virtual bool getMicMute() const; 160 161 virtual status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs); 162 virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) const; 163 164 virtual void registerClient(const sp<IAudioFlingerClient>& client); 165 166 virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, 167 audio_channel_mask_t channelMask) const; 168 169 virtual status_t openOutput(audio_module_handle_t module, 170 audio_io_handle_t *output, 171 audio_config_t *config, 172 audio_devices_t *devices, 173 const String8& address, 174 uint32_t *latencyMs, 175 audio_output_flags_t flags); 176 177 virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1, 178 audio_io_handle_t output2); 179 180 virtual status_t closeOutput(audio_io_handle_t output); 181 182 virtual status_t suspendOutput(audio_io_handle_t output); 183 184 virtual status_t restoreOutput(audio_io_handle_t output); 185 186 virtual status_t openInput(audio_module_handle_t module, 187 audio_io_handle_t *input, 188 audio_config_t *config, 189 audio_devices_t *device, 190 const String8& address, 191 audio_source_t source, 192 audio_input_flags_t flags); 193 194 virtual status_t closeInput(audio_io_handle_t input); 195 196 virtual status_t invalidateStream(audio_stream_type_t stream); 197 198 virtual status_t setVoiceVolume(float volume); 199 200 virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 201 audio_io_handle_t output) const; 202 203 virtual uint32_t getInputFramesLost(audio_io_handle_t ioHandle) const; 204 205 // This is the binder API. For the internal API see nextUniqueId(). 206 virtual audio_unique_id_t newAudioUniqueId(audio_unique_id_use_t use); 207 208 virtual void acquireAudioSessionId(audio_session_t audioSession, pid_t pid); 209 210 virtual void releaseAudioSessionId(audio_session_t audioSession, pid_t pid); 211 212 virtual status_t queryNumberEffects(uint32_t *numEffects) const; 213 214 virtual status_t queryEffect(uint32_t index, effect_descriptor_t *descriptor) const; 215 216 virtual status_t getEffectDescriptor(const effect_uuid_t *pUuid, 217 effect_descriptor_t *descriptor) const; 218 219 virtual sp<IEffect> createEffect( 220 effect_descriptor_t *pDesc, 221 const sp<IEffectClient>& effectClient, 222 int32_t priority, 223 audio_io_handle_t io, 224 audio_session_t sessionId, 225 const String16& opPackageName, 226 status_t *status /*non-NULL*/, 227 int *id, 228 int *enabled); 229 230 virtual status_t moveEffects(audio_session_t sessionId, audio_io_handle_t srcOutput, 231 audio_io_handle_t dstOutput); 232 233 virtual audio_module_handle_t loadHwModule(const char *name); 234 235 virtual uint32_t getPrimaryOutputSamplingRate(); 236 virtual size_t getPrimaryOutputFrameCount(); 237 238 virtual status_t setLowRamDevice(bool isLowRamDevice); 239 240 /* List available audio ports and their attributes */ 241 virtual status_t listAudioPorts(unsigned int *num_ports, 242 struct audio_port *ports); 243 244 /* Get attributes for a given audio port */ 245 virtual status_t getAudioPort(struct audio_port *port); 246 247 /* Create an audio patch between several source and sink ports */ 248 virtual status_t createAudioPatch(const struct audio_patch *patch, 249 audio_patch_handle_t *handle); 250 251 /* Release an audio patch */ 252 virtual status_t releaseAudioPatch(audio_patch_handle_t handle); 253 254 /* List existing audio patches */ 255 virtual status_t listAudioPatches(unsigned int *num_patches, 256 struct audio_patch *patches); 257 258 /* Set audio port configuration */ 259 virtual status_t setAudioPortConfig(const struct audio_port_config *config); 260 261 /* Get the HW synchronization source used for an audio session */ 262 virtual audio_hw_sync_t getAudioHwSyncForSession(audio_session_t sessionId); 263 264 /* Indicate JAVA services are ready (scheduling, power management ...) */ 265 virtual status_t systemReady(); 266 267 virtual status_t onTransact( 268 uint32_t code, 269 const Parcel& data, 270 Parcel* reply, 271 uint32_t flags); 272 273 // end of IAudioFlinger interface 274 275 sp<NBLog::Writer> newWriter_l(size_t size, const char *name); 276 void unregisterWriter(const sp<NBLog::Writer>& writer); 277 sp<EffectsFactoryHalInterface> getEffectsFactory(); 278private: 279 static const size_t kLogMemorySize = 40 * 1024; 280 sp<MemoryDealer> mLogMemoryDealer; // == 0 when NBLog is disabled 281 // When a log writer is unregistered, it is done lazily so that media.log can continue to see it 282 // for as long as possible. The memory is only freed when it is needed for another log writer. 283 Vector< sp<NBLog::Writer> > mUnregisteredWriters; 284 Mutex mUnregisteredWritersLock; 285public: 286 287 class SyncEvent; 288 289 typedef void (*sync_event_callback_t)(const wp<SyncEvent>& event) ; 290 291 class SyncEvent : public RefBase { 292 public: 293 SyncEvent(AudioSystem::sync_event_t type, 294 audio_session_t triggerSession, 295 audio_session_t listenerSession, 296 sync_event_callback_t callBack, 297 wp<RefBase> cookie) 298 : mType(type), mTriggerSession(triggerSession), mListenerSession(listenerSession), 299 mCallback(callBack), mCookie(cookie) 300 {} 301 302 virtual ~SyncEvent() {} 303 304 void trigger() { Mutex::Autolock _l(mLock); if (mCallback) mCallback(this); } 305 bool isCancelled() const { Mutex::Autolock _l(mLock); return (mCallback == NULL); } 306 void cancel() { Mutex::Autolock _l(mLock); mCallback = NULL; } 307 AudioSystem::sync_event_t type() const { return mType; } 308 audio_session_t triggerSession() const { return mTriggerSession; } 309 audio_session_t listenerSession() const { return mListenerSession; } 310 wp<RefBase> cookie() const { return mCookie; } 311 312 private: 313 const AudioSystem::sync_event_t mType; 314 const audio_session_t mTriggerSession; 315 const audio_session_t mListenerSession; 316 sync_event_callback_t mCallback; 317 const wp<RefBase> mCookie; 318 mutable Mutex mLock; 319 }; 320 321 sp<SyncEvent> createSyncEvent(AudioSystem::sync_event_t type, 322 audio_session_t triggerSession, 323 audio_session_t listenerSession, 324 sync_event_callback_t callBack, 325 const wp<RefBase>& cookie); 326 327private: 328 329 audio_mode_t getMode() const { return mMode; } 330 331 bool btNrecIsOff() const { return mBtNrecIsOff; } 332 333 AudioFlinger() ANDROID_API; 334 virtual ~AudioFlinger(); 335 336 // call in any IAudioFlinger method that accesses mPrimaryHardwareDev 337 status_t initCheck() const { return mPrimaryHardwareDev == NULL ? 338 NO_INIT : NO_ERROR; } 339 340 // RefBase 341 virtual void onFirstRef(); 342 343 AudioHwDevice* findSuitableHwDev_l(audio_module_handle_t module, 344 audio_devices_t devices); 345 void purgeStaleEffects_l(); 346 347 // Set kEnableExtendedChannels to true to enable greater than stereo output 348 // for the MixerThread and device sink. Number of channels allowed is 349 // FCC_2 <= channels <= AudioMixer::MAX_NUM_CHANNELS. 350 static const bool kEnableExtendedChannels = true; 351 352 // Returns true if channel mask is permitted for the PCM sink in the MixerThread 353 static inline bool isValidPcmSinkChannelMask(audio_channel_mask_t channelMask) { 354 switch (audio_channel_mask_get_representation(channelMask)) { 355 case AUDIO_CHANNEL_REPRESENTATION_POSITION: { 356 uint32_t channelCount = FCC_2; // stereo is default 357 if (kEnableExtendedChannels) { 358 channelCount = audio_channel_count_from_out_mask(channelMask); 359 if (channelCount < FCC_2 // mono is not supported at this time 360 || channelCount > AudioMixer::MAX_NUM_CHANNELS) { 361 return false; 362 } 363 } 364 // check that channelMask is the "canonical" one we expect for the channelCount. 365 return channelMask == audio_channel_out_mask_from_count(channelCount); 366 } 367 case AUDIO_CHANNEL_REPRESENTATION_INDEX: 368 if (kEnableExtendedChannels) { 369 const uint32_t channelCount = audio_channel_count_from_out_mask(channelMask); 370 if (channelCount >= FCC_2 // mono is not supported at this time 371 && channelCount <= AudioMixer::MAX_NUM_CHANNELS) { 372 return true; 373 } 374 } 375 return false; 376 default: 377 return false; 378 } 379 } 380 381 // Set kEnableExtendedPrecision to true to use extended precision in MixerThread 382 static const bool kEnableExtendedPrecision = true; 383 384 // Returns true if format is permitted for the PCM sink in the MixerThread 385 static inline bool isValidPcmSinkFormat(audio_format_t format) { 386 switch (format) { 387 case AUDIO_FORMAT_PCM_16_BIT: 388 return true; 389 case AUDIO_FORMAT_PCM_FLOAT: 390 case AUDIO_FORMAT_PCM_24_BIT_PACKED: 391 case AUDIO_FORMAT_PCM_32_BIT: 392 case AUDIO_FORMAT_PCM_8_24_BIT: 393 return kEnableExtendedPrecision; 394 default: 395 return false; 396 } 397 } 398 399 // standby delay for MIXER and DUPLICATING playback threads is read from property 400 // ro.audio.flinger_standbytime_ms or defaults to kDefaultStandbyTimeInNsecs 401 static nsecs_t mStandbyTimeInNsecs; 402 403 // incremented by 2 when screen state changes, bit 0 == 1 means "off" 404 // AudioFlinger::setParameters() updates, other threads read w/o lock 405 static uint32_t mScreenState; 406 407 // Internal dump utilities. 408 static const int kDumpLockRetries = 50; 409 static const int kDumpLockSleepUs = 20000; 410 static bool dumpTryLock(Mutex& mutex); 411 void dumpPermissionDenial(int fd, const Vector<String16>& args); 412 void dumpClients(int fd, const Vector<String16>& args); 413 void dumpInternals(int fd, const Vector<String16>& args); 414 415 // --- Client --- 416 class Client : public RefBase { 417 public: 418 Client(const sp<AudioFlinger>& audioFlinger, pid_t pid); 419 virtual ~Client(); 420 sp<MemoryDealer> heap() const; 421 pid_t pid() const { return mPid; } 422 sp<AudioFlinger> audioFlinger() const { return mAudioFlinger; } 423 424 private: 425 Client(const Client&); 426 Client& operator = (const Client&); 427 const sp<AudioFlinger> mAudioFlinger; 428 sp<MemoryDealer> mMemoryDealer; 429 const pid_t mPid; 430 }; 431 432 // --- Notification Client --- 433 class NotificationClient : public IBinder::DeathRecipient { 434 public: 435 NotificationClient(const sp<AudioFlinger>& audioFlinger, 436 const sp<IAudioFlingerClient>& client, 437 pid_t pid); 438 virtual ~NotificationClient(); 439 440 sp<IAudioFlingerClient> audioFlingerClient() const { return mAudioFlingerClient; } 441 442 // IBinder::DeathRecipient 443 virtual void binderDied(const wp<IBinder>& who); 444 445 private: 446 NotificationClient(const NotificationClient&); 447 NotificationClient& operator = (const NotificationClient&); 448 449 const sp<AudioFlinger> mAudioFlinger; 450 const pid_t mPid; 451 const sp<IAudioFlingerClient> mAudioFlingerClient; 452 }; 453 454 class TrackHandle; 455 class RecordHandle; 456 class RecordThread; 457 class PlaybackThread; 458 class MixerThread; 459 class DirectOutputThread; 460 class OffloadThread; 461 class DuplicatingThread; 462 class AsyncCallbackThread; 463 class Track; 464 class RecordTrack; 465 class EffectModule; 466 class EffectHandle; 467 class EffectChain; 468 469 struct AudioStreamIn; 470 471 struct stream_type_t { 472 stream_type_t() 473 : volume(1.0f), 474 mute(false) 475 { 476 } 477 float volume; 478 bool mute; 479 }; 480 481 // --- PlaybackThread --- 482 483#include "Threads.h" 484 485#include "Effects.h" 486 487#include "PatchPanel.h" 488 489 // server side of the client's IAudioTrack 490 class TrackHandle : public android::BnAudioTrack { 491 public: 492 explicit TrackHandle(const sp<PlaybackThread::Track>& track); 493 virtual ~TrackHandle(); 494 virtual sp<IMemory> getCblk() const; 495 virtual status_t start(); 496 virtual void stop(); 497 virtual void flush(); 498 virtual void pause(); 499 virtual status_t attachAuxEffect(int effectId); 500 virtual status_t setParameters(const String8& keyValuePairs); 501 virtual status_t getTimestamp(AudioTimestamp& timestamp); 502 virtual void signal(); // signal playback thread for a change in control block 503 504 virtual status_t onTransact( 505 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 506 507 private: 508 const sp<PlaybackThread::Track> mTrack; 509 }; 510 511 // server side of the client's IAudioRecord 512 class RecordHandle : public android::BnAudioRecord { 513 public: 514 explicit RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack); 515 virtual ~RecordHandle(); 516 virtual status_t start(int /*AudioSystem::sync_event_t*/ event, 517 audio_session_t triggerSession); 518 virtual void stop(); 519 virtual status_t onTransact( 520 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 521 private: 522 const sp<RecordThread::RecordTrack> mRecordTrack; 523 524 // for use from destructor 525 void stop_nonvirtual(); 526 }; 527 528 529 ThreadBase *checkThread_l(audio_io_handle_t ioHandle) const; 530 PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const; 531 MixerThread *checkMixerThread_l(audio_io_handle_t output) const; 532 RecordThread *checkRecordThread_l(audio_io_handle_t input) const; 533 sp<RecordThread> openInput_l(audio_module_handle_t module, 534 audio_io_handle_t *input, 535 audio_config_t *config, 536 audio_devices_t device, 537 const String8& address, 538 audio_source_t source, 539 audio_input_flags_t flags); 540 sp<PlaybackThread> openOutput_l(audio_module_handle_t module, 541 audio_io_handle_t *output, 542 audio_config_t *config, 543 audio_devices_t devices, 544 const String8& address, 545 audio_output_flags_t flags); 546 547 void closeOutputFinish(const sp<PlaybackThread>& thread); 548 void closeInputFinish(const sp<RecordThread>& thread); 549 550 // no range check, AudioFlinger::mLock held 551 bool streamMute_l(audio_stream_type_t stream) const 552 { return mStreamTypes[stream].mute; } 553 // no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held 554 float streamVolume_l(audio_stream_type_t stream) const 555 { return mStreamTypes[stream].volume; } 556 void ioConfigChanged(audio_io_config_event event, 557 const sp<AudioIoDescriptor>& ioDesc, 558 pid_t pid = 0); 559 560 // Allocate an audio_unique_id_t. 561 // Specific types are audio_io_handle_t, audio_session_t, effect ID (int), 562 // audio_module_handle_t, and audio_patch_handle_t. 563 // They all share the same ID space, but the namespaces are actually independent 564 // because there are separate KeyedVectors for each kind of ID. 565 // The return value is cast to the specific type depending on how the ID will be used. 566 // FIXME This API does not handle rollover to zero (for unsigned IDs), 567 // or from positive to negative (for signed IDs). 568 // Thus it may fail by returning an ID of the wrong sign, 569 // or by returning a non-unique ID. 570 // This is the internal API. For the binder API see newAudioUniqueId(). 571 audio_unique_id_t nextUniqueId(audio_unique_id_use_t use); 572 573 status_t moveEffectChain_l(audio_session_t sessionId, 574 PlaybackThread *srcThread, 575 PlaybackThread *dstThread, 576 bool reRegister); 577 578 // return thread associated with primary hardware device, or NULL 579 PlaybackThread *primaryPlaybackThread_l() const; 580 audio_devices_t primaryOutputDevice_l() const; 581 582 // return the playback thread with smallest HAL buffer size, and prefer fast 583 PlaybackThread *fastPlaybackThread_l() const; 584 585 sp<PlaybackThread> getEffectThread_l(audio_session_t sessionId, int EffectId); 586 587 588 void removeClient_l(pid_t pid); 589 void removeNotificationClient(pid_t pid); 590 bool isNonOffloadableGlobalEffectEnabled_l(); 591 void onNonOffloadableGlobalEffectEnable(); 592 593 // Store an effect chain to mOrphanEffectChains keyed vector. 594 // Called when a thread exits and effects are still attached to it. 595 // If effects are later created on the same session, they will reuse the same 596 // effect chain and same instances in the effect library. 597 // return ALREADY_EXISTS if a chain with the same session already exists in 598 // mOrphanEffectChains. Note that this should never happen as there is only one 599 // chain for a given session and it is attached to only one thread at a time. 600 status_t putOrphanEffectChain_l(const sp<EffectChain>& chain); 601 // Get an effect chain for the specified session in mOrphanEffectChains and remove 602 // it if found. Returns 0 if not found (this is the most common case). 603 sp<EffectChain> getOrphanEffectChain_l(audio_session_t session); 604 // Called when the last effect handle on an effect instance is removed. If this 605 // effect belongs to an effect chain in mOrphanEffectChains, the chain is updated 606 // and removed from mOrphanEffectChains if it does not contain any effect. 607 // Return true if the effect was found in mOrphanEffectChains, false otherwise. 608 bool updateOrphanEffectChains(const sp<EffectModule>& effect); 609 610 void broacastParametersToRecordThreads_l(const String8& keyValuePairs); 611 612 // AudioStreamIn is immutable, so their fields are const. 613 // For emphasis, we could also make all pointers to them be "const *", 614 // but that would clutter the code unnecessarily. 615 616 struct AudioStreamIn { 617 AudioHwDevice* const audioHwDev; 618 sp<StreamInHalInterface> stream; 619 audio_input_flags_t flags; 620 621 sp<DeviceHalInterface> hwDev() const { return audioHwDev->hwDevice(); } 622 623 AudioStreamIn(AudioHwDevice *dev, sp<StreamInHalInterface> in, audio_input_flags_t flags) : 624 audioHwDev(dev), stream(in), flags(flags) {} 625 }; 626 627 // for mAudioSessionRefs only 628 struct AudioSessionRef { 629 AudioSessionRef(audio_session_t sessionid, pid_t pid) : 630 mSessionid(sessionid), mPid(pid), mCnt(1) {} 631 const audio_session_t mSessionid; 632 const pid_t mPid; 633 int mCnt; 634 }; 635 636 mutable Mutex mLock; 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 sp<DevicesFactoryHalInterface> mDevicesFactoryHal; 653 654 // for dump, indicates which hardware operation is currently in progress (but not stream ops) 655 enum hardware_call_state { 656 AUDIO_HW_IDLE = 0, // no operation in progress 657 AUDIO_HW_INIT, // init_check 658 AUDIO_HW_OUTPUT_OPEN, // open_output_stream 659 AUDIO_HW_OUTPUT_CLOSE, // unused 660 AUDIO_HW_INPUT_OPEN, // unused 661 AUDIO_HW_INPUT_CLOSE, // unused 662 AUDIO_HW_STANDBY, // unused 663 AUDIO_HW_SET_MASTER_VOLUME, // set_master_volume 664 AUDIO_HW_GET_ROUTING, // unused 665 AUDIO_HW_SET_ROUTING, // unused 666 AUDIO_HW_GET_MODE, // unused 667 AUDIO_HW_SET_MODE, // set_mode 668 AUDIO_HW_GET_MIC_MUTE, // get_mic_mute 669 AUDIO_HW_SET_MIC_MUTE, // set_mic_mute 670 AUDIO_HW_SET_VOICE_VOLUME, // set_voice_volume 671 AUDIO_HW_SET_PARAMETER, // set_parameters 672 AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size 673 AUDIO_HW_GET_MASTER_VOLUME, // get_master_volume 674 AUDIO_HW_GET_PARAMETER, // get_parameters 675 AUDIO_HW_SET_MASTER_MUTE, // set_master_mute 676 AUDIO_HW_GET_MASTER_MUTE, // get_master_mute 677 }; 678 679 mutable hardware_call_state mHardwareStatus; // for dump only 680 681 682 DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads; 683 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 684 685 // member variables below are protected by mLock 686 float mMasterVolume; 687 bool mMasterMute; 688 // end of variables protected by mLock 689 690 DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads; 691 692 // protected by mClientLock 693 DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients; 694 695 // updated by atomic_fetch_add_explicit 696 volatile atomic_uint_fast32_t mNextUniqueIds[AUDIO_UNIQUE_ID_USE_MAX]; 697 698 audio_mode_t mMode; 699 bool mBtNrecIsOff; 700 701 // protected by mLock 702 Vector<AudioSessionRef*> mAudioSessionRefs; 703 704 float masterVolume_l() const; 705 bool masterMute_l() const; 706 audio_module_handle_t loadHwModule_l(const char *name); 707 708 Vector < sp<SyncEvent> > mPendingSyncEvents; // sync events awaiting for a session 709 // to be created 710 711 // Effect chains without a valid thread 712 DefaultKeyedVector< audio_session_t , sp<EffectChain> > mOrphanEffectChains; 713 714 // list of sessions for which a valid HW A/V sync ID was retrieved from the HAL 715 DefaultKeyedVector< audio_session_t , audio_hw_sync_t >mHwAvSyncIds; 716private: 717 sp<Client> registerPid(pid_t pid); // always returns non-0 718 719 // for use from destructor 720 status_t closeOutput_nonvirtual(audio_io_handle_t output); 721 void closeOutputInternal_l(const sp<PlaybackThread>& thread); 722 status_t closeInput_nonvirtual(audio_io_handle_t input); 723 void closeInputInternal_l(const sp<RecordThread>& thread); 724 void setAudioHwSyncForSession_l(PlaybackThread *thread, audio_session_t sessionId); 725 726 status_t checkStreamType(audio_stream_type_t stream) const; 727 728#ifdef TEE_SINK 729 // all record threads serially share a common tee sink, which is re-created on format change 730 sp<NBAIO_Sink> mRecordTeeSink; 731 sp<NBAIO_Source> mRecordTeeSource; 732#endif 733 734public: 735 736#ifdef TEE_SINK 737 // tee sink, if enabled by property, allows dumpsys to write most recent audio to .wav file 738 static void dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id = 0); 739 740 // whether tee sink is enabled by property 741 static bool mTeeSinkInputEnabled; 742 static bool mTeeSinkOutputEnabled; 743 static bool mTeeSinkTrackEnabled; 744 745 // runtime configured size of each tee sink pipe, in frames 746 static size_t mTeeSinkInputFrames; 747 static size_t mTeeSinkOutputFrames; 748 static size_t mTeeSinkTrackFrames; 749 750 // compile-time default size of tee sink pipes, in frames 751 // 0x200000 stereo 16-bit PCM frames = 47.5 seconds at 44.1 kHz, 8 megabytes 752 static const size_t kTeeSinkInputFramesDefault = 0x200000; 753 static const size_t kTeeSinkOutputFramesDefault = 0x200000; 754 static const size_t kTeeSinkTrackFramesDefault = 0x200000; 755#endif 756 757 // This method reads from a variable without mLock, but the variable is updated under mLock. So 758 // we might read a stale value, or a value that's inconsistent with respect to other variables. 759 // In this case, it's safe because the return value isn't used for making an important decision. 760 // The reason we don't want to take mLock is because it could block the caller for a long time. 761 bool isLowRamDevice() const { return mIsLowRamDevice; } 762 763private: 764 bool mIsLowRamDevice; 765 bool mIsDeviceTypeKnown; 766 nsecs_t mGlobalEffectEnableTime; // when a global effect was last enabled 767 768 sp<PatchPanel> mPatchPanel; 769 sp<EffectsFactoryHalInterface> mEffectsFactoryHal; 770 771 bool mSystemReady; 772}; 773 774#undef INCLUDING_FROM_AUDIOFLINGER_H 775 776std::string formatToString(audio_format_t format); 777std::string inputFlagsToString(audio_input_flags_t flags); 778std::string outputFlagsToString(audio_output_flags_t flags); 779std::string devicesToString(audio_devices_t devices); 780const char *sourceToString(audio_source_t source); 781 782// ---------------------------------------------------------------------------- 783 784} // namespace android 785 786#endif // ANDROID_AUDIO_FLINGER_H 787