AudioFlinger.h revision cb4432af8dc3abc8dc4cc8e5d6080cc68a862ea1
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 <common_time/cc_helper.h> 27 28#include <cutils/compiler.h> 29 30#include <media/IAudioFlinger.h> 31#include <media/IAudioFlingerClient.h> 32#include <media/IAudioTrack.h> 33#include <media/IAudioRecord.h> 34#include <media/AudioSystem.h> 35#include <media/AudioTrack.h> 36 37#include <utils/Atomic.h> 38#include <utils/Errors.h> 39#include <utils/threads.h> 40#include <utils/SortedVector.h> 41#include <utils/TypeHelpers.h> 42#include <utils/Vector.h> 43 44#include <binder/BinderService.h> 45#include <binder/MemoryDealer.h> 46 47#include <system/audio.h> 48#include <hardware/audio.h> 49#include <hardware/audio_policy.h> 50 51#include <media/AudioBufferProvider.h> 52#include <media/ExtendedAudioBufferProvider.h> 53 54#include "FastCapture.h" 55#include "FastMixer.h" 56#include <media/nbaio/NBAIO.h> 57#include "AudioWatchdog.h" 58#include "AudioMixer.h" 59#include "AudioStreamOut.h" 60#include "SpdifStreamOut.h" 61#include "AudioHwDevice.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 FastMixer; 76class PassthruBufferProvider; 77class ServerProxy; 78 79// ---------------------------------------------------------------------------- 80 81static const nsecs_t kDefaultStandbyTimeInNsecs = seconds(3); 82 83 84// Max shared memory size for audio tracks and audio records per client process 85static const size_t kClientSharedHeapSizeBytes = 1024*1024; 86// Shared memory size multiplier for non low ram devices 87static const size_t kClientSharedHeapSizeMultiplier = 4; 88 89#define INCLUDING_FROM_AUDIOFLINGER_H 90 91class AudioFlinger : 92 public BinderService<AudioFlinger>, 93 public BnAudioFlinger 94{ 95 friend class BinderService<AudioFlinger>; // for AudioFlinger() 96public: 97 static const char* getServiceName() ANDROID_API { return "media.audio_flinger"; } 98 99 virtual status_t dump(int fd, const Vector<String16>& args); 100 101 // IAudioFlinger interface, in binder opcode order 102 virtual sp<IAudioTrack> createTrack( 103 audio_stream_type_t streamType, 104 uint32_t sampleRate, 105 audio_format_t format, 106 audio_channel_mask_t channelMask, 107 size_t *pFrameCount, 108 IAudioFlinger::track_flags_t *flags, 109 const sp<IMemory>& sharedBuffer, 110 audio_io_handle_t output, 111 pid_t tid, 112 int *sessionId, 113 int clientUid, 114 status_t *status /*non-NULL*/); 115 116 virtual sp<IAudioRecord> openRecord( 117 audio_io_handle_t input, 118 uint32_t sampleRate, 119 audio_format_t format, 120 audio_channel_mask_t channelMask, 121 const String16& opPackageName, 122 size_t *pFrameCount, 123 IAudioFlinger::track_flags_t *flags, 124 pid_t tid, 125 int clientUid, 126 int *sessionId, 127 size_t *notificationFrames, 128 sp<IMemory>& cblk, 129 sp<IMemory>& buffers, 130 status_t *status /*non-NULL*/); 131 132 virtual uint32_t sampleRate(audio_io_handle_t output) const; 133 virtual audio_format_t format(audio_io_handle_t output) const; 134 virtual size_t frameCount(audio_io_handle_t output) 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 virtual audio_unique_id_t newAudioUniqueId(); 201 202 virtual void acquireAudioSessionId(int audioSession, pid_t pid); 203 204 virtual void releaseAudioSessionId(int audioSession, pid_t pid); 205 206 virtual status_t queryNumberEffects(uint32_t *numEffects) const; 207 208 virtual status_t queryEffect(uint32_t index, effect_descriptor_t *descriptor) const; 209 210 virtual status_t getEffectDescriptor(const effect_uuid_t *pUuid, 211 effect_descriptor_t *descriptor) const; 212 213 virtual sp<IEffect> createEffect( 214 effect_descriptor_t *pDesc, 215 const sp<IEffectClient>& effectClient, 216 int32_t priority, 217 audio_io_handle_t io, 218 int sessionId, 219 const String16& opPackageName, 220 status_t *status /*non-NULL*/, 221 int *id, 222 int *enabled); 223 224 virtual status_t moveEffects(int sessionId, audio_io_handle_t srcOutput, 225 audio_io_handle_t dstOutput); 226 227 virtual audio_module_handle_t loadHwModule(const char *name); 228 229 virtual uint32_t getPrimaryOutputSamplingRate(); 230 virtual size_t getPrimaryOutputFrameCount(); 231 232 virtual status_t setLowRamDevice(bool isLowRamDevice); 233 234 /* List available audio ports and their attributes */ 235 virtual status_t listAudioPorts(unsigned int *num_ports, 236 struct audio_port *ports); 237 238 /* Get attributes for a given audio port */ 239 virtual status_t getAudioPort(struct audio_port *port); 240 241 /* Create an audio patch between several source and sink ports */ 242 virtual status_t createAudioPatch(const struct audio_patch *patch, 243 audio_patch_handle_t *handle); 244 245 /* Release an audio patch */ 246 virtual status_t releaseAudioPatch(audio_patch_handle_t handle); 247 248 /* List existing audio patches */ 249 virtual status_t listAudioPatches(unsigned int *num_patches, 250 struct audio_patch *patches); 251 252 /* Set audio port configuration */ 253 virtual status_t setAudioPortConfig(const struct audio_port_config *config); 254 255 /* Get the HW synchronization source used for an audio session */ 256 virtual audio_hw_sync_t getAudioHwSyncForSession(audio_session_t sessionId); 257 258 /* Indicate JAVA services are ready (scheduling, power management ...) */ 259 virtual status_t systemReady(); 260 261 virtual status_t onTransact( 262 uint32_t code, 263 const Parcel& data, 264 Parcel* reply, 265 uint32_t flags); 266 267 // end of IAudioFlinger interface 268 269 sp<NBLog::Writer> newWriter_l(size_t size, const char *name); 270 void unregisterWriter(const sp<NBLog::Writer>& writer); 271private: 272 static const size_t kLogMemorySize = 40 * 1024; 273 sp<MemoryDealer> mLogMemoryDealer; // == 0 when NBLog is disabled 274 // When a log writer is unregistered, it is done lazily so that media.log can continue to see it 275 // for as long as possible. The memory is only freed when it is needed for another log writer. 276 Vector< sp<NBLog::Writer> > mUnregisteredWriters; 277 Mutex mUnregisteredWritersLock; 278public: 279 280 class SyncEvent; 281 282 typedef void (*sync_event_callback_t)(const wp<SyncEvent>& event) ; 283 284 class SyncEvent : public RefBase { 285 public: 286 SyncEvent(AudioSystem::sync_event_t type, 287 int triggerSession, 288 int listenerSession, 289 sync_event_callback_t callBack, 290 wp<RefBase> cookie) 291 : mType(type), mTriggerSession(triggerSession), mListenerSession(listenerSession), 292 mCallback(callBack), mCookie(cookie) 293 {} 294 295 virtual ~SyncEvent() {} 296 297 void trigger() { Mutex::Autolock _l(mLock); if (mCallback) mCallback(this); } 298 bool isCancelled() const { Mutex::Autolock _l(mLock); return (mCallback == NULL); } 299 void cancel() { Mutex::Autolock _l(mLock); mCallback = NULL; } 300 AudioSystem::sync_event_t type() const { return mType; } 301 int triggerSession() const { return mTriggerSession; } 302 int listenerSession() const { return mListenerSession; } 303 wp<RefBase> cookie() const { return mCookie; } 304 305 private: 306 const AudioSystem::sync_event_t mType; 307 const int mTriggerSession; 308 const int mListenerSession; 309 sync_event_callback_t mCallback; 310 const wp<RefBase> mCookie; 311 mutable Mutex mLock; 312 }; 313 314 sp<SyncEvent> createSyncEvent(AudioSystem::sync_event_t type, 315 int triggerSession, 316 int listenerSession, 317 sync_event_callback_t callBack, 318 wp<RefBase> cookie); 319 320private: 321 322 audio_mode_t getMode() const { return mMode; } 323 324 bool btNrecIsOff() const { return mBtNrecIsOff; } 325 326 AudioFlinger() ANDROID_API; 327 virtual ~AudioFlinger(); 328 329 // call in any IAudioFlinger method that accesses mPrimaryHardwareDev 330 status_t initCheck() const { return mPrimaryHardwareDev == NULL ? 331 NO_INIT : NO_ERROR; } 332 333 // RefBase 334 virtual void onFirstRef(); 335 336 AudioHwDevice* findSuitableHwDev_l(audio_module_handle_t module, 337 audio_devices_t devices); 338 void purgeStaleEffects_l(); 339 340 // Set kEnableExtendedChannels to true to enable greater than stereo output 341 // for the MixerThread and device sink. Number of channels allowed is 342 // FCC_2 <= channels <= AudioMixer::MAX_NUM_CHANNELS. 343 static const bool kEnableExtendedChannels = true; 344 345 // Returns true if channel mask is permitted for the PCM sink in the MixerThread 346 static inline bool isValidPcmSinkChannelMask(audio_channel_mask_t channelMask) { 347 switch (audio_channel_mask_get_representation(channelMask)) { 348 case AUDIO_CHANNEL_REPRESENTATION_POSITION: { 349 uint32_t channelCount = FCC_2; // stereo is default 350 if (kEnableExtendedChannels) { 351 channelCount = audio_channel_count_from_out_mask(channelMask); 352 if (channelCount < FCC_2 // mono is not supported at this time 353 || channelCount > AudioMixer::MAX_NUM_CHANNELS) { 354 return false; 355 } 356 } 357 // check that channelMask is the "canonical" one we expect for the channelCount. 358 return channelMask == audio_channel_out_mask_from_count(channelCount); 359 } 360 case AUDIO_CHANNEL_REPRESENTATION_INDEX: 361 if (kEnableExtendedChannels) { 362 const uint32_t channelCount = audio_channel_count_from_out_mask(channelMask); 363 if (channelCount >= FCC_2 // mono is not supported at this time 364 && channelCount <= AudioMixer::MAX_NUM_CHANNELS) { 365 return true; 366 } 367 } 368 return false; 369 default: 370 return false; 371 } 372 } 373 374 // Set kEnableExtendedPrecision to true to use extended precision in MixerThread 375 static const bool kEnableExtendedPrecision = true; 376 377 // Returns true if format is permitted for the PCM sink in the MixerThread 378 static inline bool isValidPcmSinkFormat(audio_format_t format) { 379 switch (format) { 380 case AUDIO_FORMAT_PCM_16_BIT: 381 return true; 382 case AUDIO_FORMAT_PCM_FLOAT: 383 case AUDIO_FORMAT_PCM_24_BIT_PACKED: 384 case AUDIO_FORMAT_PCM_32_BIT: 385 case AUDIO_FORMAT_PCM_8_24_BIT: 386 return kEnableExtendedPrecision; 387 default: 388 return false; 389 } 390 } 391 392 // standby delay for MIXER and DUPLICATING playback threads is read from property 393 // ro.audio.flinger_standbytime_ms or defaults to kDefaultStandbyTimeInNsecs 394 static nsecs_t mStandbyTimeInNsecs; 395 396 // incremented by 2 when screen state changes, bit 0 == 1 means "off" 397 // AudioFlinger::setParameters() updates, other threads read w/o lock 398 static uint32_t mScreenState; 399 400 // Internal dump utilities. 401 static const int kDumpLockRetries = 50; 402 static const int kDumpLockSleepUs = 20000; 403 static bool dumpTryLock(Mutex& mutex); 404 void dumpPermissionDenial(int fd, const Vector<String16>& args); 405 void dumpClients(int fd, const Vector<String16>& args); 406 void dumpInternals(int fd, const Vector<String16>& args); 407 408 // --- Client --- 409 class Client : public RefBase { 410 public: 411 Client(const sp<AudioFlinger>& audioFlinger, pid_t pid); 412 virtual ~Client(); 413 sp<MemoryDealer> heap() const; 414 pid_t pid() const { return mPid; } 415 sp<AudioFlinger> audioFlinger() const { return mAudioFlinger; } 416 417 bool reserveTimedTrack(); 418 void releaseTimedTrack(); 419 420 private: 421 Client(const Client&); 422 Client& operator = (const Client&); 423 const sp<AudioFlinger> mAudioFlinger; 424 sp<MemoryDealer> mMemoryDealer; 425 const pid_t mPid; 426 427 Mutex mTimedTrackLock; 428 int mTimedTrackCount; 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 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 allocateTimedBuffer(size_t size, 500 sp<IMemory>* buffer); 501 virtual status_t queueTimedBuffer(const sp<IMemory>& buffer, 502 int64_t pts); 503 virtual status_t setMediaTimeTransform(const LinearTransform& xform, 504 int target); 505 virtual status_t setParameters(const String8& keyValuePairs); 506 virtual status_t getTimestamp(AudioTimestamp& timestamp); 507 virtual void signal(); // signal playback thread for a change in control block 508 509 virtual status_t onTransact( 510 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 511 512 private: 513 const sp<PlaybackThread::Track> mTrack; 514 }; 515 516 // server side of the client's IAudioRecord 517 class RecordHandle : public android::BnAudioRecord { 518 public: 519 RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack); 520 virtual ~RecordHandle(); 521 virtual status_t start(int /*AudioSystem::sync_event_t*/ event, int triggerSession); 522 virtual void stop(); 523 virtual status_t onTransact( 524 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 525 private: 526 const sp<RecordThread::RecordTrack> mRecordTrack; 527 528 // for use from destructor 529 void stop_nonvirtual(); 530 }; 531 532 533 PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const; 534 MixerThread *checkMixerThread_l(audio_io_handle_t output) const; 535 RecordThread *checkRecordThread_l(audio_io_handle_t input) const; 536 sp<RecordThread> openInput_l(audio_module_handle_t module, 537 audio_io_handle_t *input, 538 audio_config_t *config, 539 audio_devices_t device, 540 const String8& address, 541 audio_source_t source, 542 audio_input_flags_t flags); 543 sp<PlaybackThread> openOutput_l(audio_module_handle_t module, 544 audio_io_handle_t *output, 545 audio_config_t *config, 546 audio_devices_t devices, 547 const String8& address, 548 audio_output_flags_t flags); 549 550 void closeOutputFinish(sp<PlaybackThread> thread); 551 void closeInputFinish(sp<RecordThread> thread); 552 553 // no range check, AudioFlinger::mLock held 554 bool streamMute_l(audio_stream_type_t stream) const 555 { return mStreamTypes[stream].mute; } 556 // no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held 557 float streamVolume_l(audio_stream_type_t stream) const 558 { return mStreamTypes[stream].volume; } 559 void ioConfigChanged(audio_io_config_event event, 560 const sp<AudioIoDescriptor>& ioDesc, 561 pid_t pid = 0); 562 563 // Allocate an audio_io_handle_t, session ID, effect ID, or audio_module_handle_t. 564 // They all share the same ID space, but the namespaces are actually independent 565 // because there are separate KeyedVectors for each kind of ID. 566 // The return value is uint32_t, but is cast to signed for some IDs. 567 // FIXME This API does not handle rollover to zero (for unsigned IDs), 568 // or from positive to negative (for signed IDs). 569 // Thus it may fail by returning an ID of the wrong sign, 570 // or by returning a non-unique ID. 571 uint32_t nextUniqueId(); 572 573 status_t moveEffectChain_l(int sessionId, 574 PlaybackThread *srcThread, 575 PlaybackThread *dstThread, 576 bool reRegister); 577 // return thread associated with primary hardware device, or NULL 578 PlaybackThread *primaryPlaybackThread_l() const; 579 audio_devices_t primaryOutputDevice_l() const; 580 581 sp<PlaybackThread> getEffectThread_l(int sessionId, int EffectId); 582 583 584 void removeClient_l(pid_t pid); 585 void removeNotificationClient(pid_t pid); 586 bool isNonOffloadableGlobalEffectEnabled_l(); 587 void onNonOffloadableGlobalEffectEnable(); 588 589 // Store an effect chain to mOrphanEffectChains keyed vector. 590 // Called when a thread exits and effects are still attached to it. 591 // If effects are later created on the same session, they will reuse the same 592 // effect chain and same instances in the effect library. 593 // return ALREADY_EXISTS if a chain with the same session already exists in 594 // mOrphanEffectChains. Note that this should never happen as there is only one 595 // chain for a given session and it is attached to only one thread at a time. 596 status_t putOrphanEffectChain_l(const sp<EffectChain>& chain); 597 // Get an effect chain for the specified session in mOrphanEffectChains and remove 598 // it if found. Returns 0 if not found (this is the most common case). 599 sp<EffectChain> getOrphanEffectChain_l(audio_session_t session); 600 // Called when the last effect handle on an effect instance is removed. If this 601 // effect belongs to an effect chain in mOrphanEffectChains, the chain is updated 602 // and removed from mOrphanEffectChains if it does not contain any effect. 603 // Return true if the effect was found in mOrphanEffectChains, false otherwise. 604 bool updateOrphanEffectChains(const sp<EffectModule>& effect); 605 606 void broacastParametersToRecordThreads_l(const String8& keyValuePairs); 607 608 // AudioStreamIn is immutable, so their fields are const. 609 // For emphasis, we could also make all pointers to them be "const *", 610 // but that would clutter the code unnecessarily. 611 612 struct AudioStreamIn { 613 AudioHwDevice* const audioHwDev; 614 audio_stream_in_t* const stream; 615 616 audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); } 617 618 AudioStreamIn(AudioHwDevice *dev, audio_stream_in_t *in) : 619 audioHwDev(dev), stream(in) {} 620 }; 621 622 // for mAudioSessionRefs only 623 struct AudioSessionRef { 624 AudioSessionRef(int sessionid, pid_t pid) : 625 mSessionid(sessionid), mPid(pid), mCnt(1) {} 626 const int mSessionid; 627 const pid_t mPid; 628 int mCnt; 629 }; 630 631 mutable Mutex mLock; 632 // protects mClients and mNotificationClients. 633 // must be locked after mLock and ThreadBase::mLock if both must be locked 634 // avoids acquiring AudioFlinger::mLock from inside thread loop. 635 mutable Mutex mClientLock; 636 // protected by mClientLock 637 DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client() 638 639 mutable Mutex mHardwareLock; 640 // NOTE: If both mLock and mHardwareLock mutexes must be held, 641 // always take mLock before mHardwareLock 642 643 // These two fields are immutable after onFirstRef(), so no lock needed to access 644 AudioHwDevice* mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL 645 DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*> mAudioHwDevs; 646 647 // for dump, indicates which hardware operation is currently in progress (but not stream ops) 648 enum hardware_call_state { 649 AUDIO_HW_IDLE = 0, // no operation in progress 650 AUDIO_HW_INIT, // init_check 651 AUDIO_HW_OUTPUT_OPEN, // open_output_stream 652 AUDIO_HW_OUTPUT_CLOSE, // unused 653 AUDIO_HW_INPUT_OPEN, // unused 654 AUDIO_HW_INPUT_CLOSE, // unused 655 AUDIO_HW_STANDBY, // unused 656 AUDIO_HW_SET_MASTER_VOLUME, // set_master_volume 657 AUDIO_HW_GET_ROUTING, // unused 658 AUDIO_HW_SET_ROUTING, // unused 659 AUDIO_HW_GET_MODE, // unused 660 AUDIO_HW_SET_MODE, // set_mode 661 AUDIO_HW_GET_MIC_MUTE, // get_mic_mute 662 AUDIO_HW_SET_MIC_MUTE, // set_mic_mute 663 AUDIO_HW_SET_VOICE_VOLUME, // set_voice_volume 664 AUDIO_HW_SET_PARAMETER, // set_parameters 665 AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size 666 AUDIO_HW_GET_MASTER_VOLUME, // get_master_volume 667 AUDIO_HW_GET_PARAMETER, // get_parameters 668 AUDIO_HW_SET_MASTER_MUTE, // set_master_mute 669 AUDIO_HW_GET_MASTER_MUTE, // get_master_mute 670 }; 671 672 mutable hardware_call_state mHardwareStatus; // for dump only 673 674 675 DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads; 676 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 677 678 // member variables below are protected by mLock 679 float mMasterVolume; 680 bool mMasterMute; 681 // end of variables protected by mLock 682 683 DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads; 684 685 // protected by mClientLock 686 DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients; 687 688 volatile int32_t mNextUniqueId; // updated by android_atomic_inc 689 // nextUniqueId() returns uint32_t, but this is declared int32_t 690 // because the atomic operations require an int32_t 691 692 audio_mode_t mMode; 693 bool mBtNrecIsOff; 694 695 // protected by mLock 696 Vector<AudioSessionRef*> mAudioSessionRefs; 697 698 float masterVolume_l() const; 699 bool masterMute_l() const; 700 audio_module_handle_t loadHwModule_l(const char *name); 701 702 Vector < sp<SyncEvent> > mPendingSyncEvents; // sync events awaiting for a session 703 // to be created 704 705 // Effect chains without a valid thread 706 DefaultKeyedVector< audio_session_t , sp<EffectChain> > mOrphanEffectChains; 707 708 // list of sessions for which a valid HW A/V sync ID was retrieved from the HAL 709 DefaultKeyedVector< audio_session_t , audio_hw_sync_t >mHwAvSyncIds; 710private: 711 sp<Client> registerPid(pid_t pid); // always returns non-0 712 713 // for use from destructor 714 status_t closeOutput_nonvirtual(audio_io_handle_t output); 715 void closeOutputInternal_l(sp<PlaybackThread> thread); 716 status_t closeInput_nonvirtual(audio_io_handle_t input); 717 void closeInputInternal_l(sp<RecordThread> thread); 718 void setAudioHwSyncForSession_l(PlaybackThread *thread, audio_session_t sessionId); 719 720 status_t checkStreamType(audio_stream_type_t stream) const; 721 722#ifdef TEE_SINK 723 // all record threads serially share a common tee sink, which is re-created on format change 724 sp<NBAIO_Sink> mRecordTeeSink; 725 sp<NBAIO_Source> mRecordTeeSource; 726#endif 727 728public: 729 730#ifdef TEE_SINK 731 // tee sink, if enabled by property, allows dumpsys to write most recent audio to .wav file 732 static void dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id = 0); 733 734 // whether tee sink is enabled by property 735 static bool mTeeSinkInputEnabled; 736 static bool mTeeSinkOutputEnabled; 737 static bool mTeeSinkTrackEnabled; 738 739 // runtime configured size of each tee sink pipe, in frames 740 static size_t mTeeSinkInputFrames; 741 static size_t mTeeSinkOutputFrames; 742 static size_t mTeeSinkTrackFrames; 743 744 // compile-time default size of tee sink pipes, in frames 745 // 0x200000 stereo 16-bit PCM frames = 47.5 seconds at 44.1 kHz, 8 megabytes 746 static const size_t kTeeSinkInputFramesDefault = 0x200000; 747 static const size_t kTeeSinkOutputFramesDefault = 0x200000; 748 static const size_t kTeeSinkTrackFramesDefault = 0x200000; 749#endif 750 751 // This method reads from a variable without mLock, but the variable is updated under mLock. So 752 // we might read a stale value, or a value that's inconsistent with respect to other variables. 753 // In this case, it's safe because the return value isn't used for making an important decision. 754 // The reason we don't want to take mLock is because it could block the caller for a long time. 755 bool isLowRamDevice() const { return mIsLowRamDevice; } 756 757private: 758 bool mIsLowRamDevice; 759 bool mIsDeviceTypeKnown; 760 nsecs_t mGlobalEffectEnableTime; // when a global effect was last enabled 761 762 sp<PatchPanel> mPatchPanel; 763 764 bool mSystemReady; 765}; 766 767#undef INCLUDING_FROM_AUDIOFLINGER_H 768 769const char *formatToString(audio_format_t format); 770String8 inputFlagsToString(audio_input_flags_t flags); 771String8 outputFlagsToString(audio_output_flags_t flags); 772String8 devicesToString(audio_devices_t devices); 773const char *sourceToString(audio_source_t source); 774 775// ---------------------------------------------------------------------------- 776 777} // namespace android 778 779#endif // ANDROID_AUDIO_FLINGER_H 780