AudioFlinger.h revision eb9487e10294a4e73977f460f30eeaff503acd21
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 81// The macro FCC_2 highlights some (but not all) places where there are are 2-channel assumptions. 82// This is typically due to legacy implementation of stereo input or output. 83// Search also for "2", "left", "right", "[0]", "[1]", ">> 16", "<< 16", etc. 84#define FCC_2 2 // FCC_2 = Fixed Channel Count 2 85// The macro FCC_8 highlights places where there are 8-channel assumptions. 86// This is typically due to audio mixer and resampler limitations. 87#define FCC_8 8 // FCC_8 = Fixed Channel Count 8 88 89static const nsecs_t kDefaultStandbyTimeInNsecs = seconds(3); 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 IAudioFlinger::track_flags_t *flags, 111 const sp<IMemory>& sharedBuffer, 112 audio_io_handle_t output, 113 pid_t tid, 114 int *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 IAudioFlinger::track_flags_t *flags, 126 pid_t tid, 127 int clientUid, 128 int *sessionId, 129 size_t *notificationFrames, 130 sp<IMemory>& cblk, 131 sp<IMemory>& buffers, 132 status_t *status /*non-NULL*/); 133 134 virtual uint32_t sampleRate(audio_io_handle_t output) const; 135 virtual audio_format_t format(audio_io_handle_t output) const; 136 virtual size_t frameCount(audio_io_handle_t output) const; 137 virtual uint32_t latency(audio_io_handle_t output) const; 138 139 virtual status_t setMasterVolume(float value); 140 virtual status_t setMasterMute(bool muted); 141 142 virtual float masterVolume() const; 143 virtual bool masterMute() const; 144 145 virtual status_t setStreamVolume(audio_stream_type_t stream, float value, 146 audio_io_handle_t output); 147 virtual status_t setStreamMute(audio_stream_type_t stream, bool muted); 148 149 virtual float streamVolume(audio_stream_type_t stream, 150 audio_io_handle_t output) const; 151 virtual bool streamMute(audio_stream_type_t stream) const; 152 153 virtual status_t setMode(audio_mode_t mode); 154 155 virtual status_t setMicMute(bool state); 156 virtual bool getMicMute() const; 157 158 virtual status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs); 159 virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) const; 160 161 virtual void registerClient(const sp<IAudioFlingerClient>& client); 162 163 virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, 164 audio_channel_mask_t channelMask) const; 165 166 virtual status_t openOutput(audio_module_handle_t module, 167 audio_io_handle_t *output, 168 audio_config_t *config, 169 audio_devices_t *devices, 170 const String8& address, 171 uint32_t *latencyMs, 172 audio_output_flags_t flags); 173 174 virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1, 175 audio_io_handle_t output2); 176 177 virtual status_t closeOutput(audio_io_handle_t output); 178 179 virtual status_t suspendOutput(audio_io_handle_t output); 180 181 virtual status_t restoreOutput(audio_io_handle_t output); 182 183 virtual status_t openInput(audio_module_handle_t module, 184 audio_io_handle_t *input, 185 audio_config_t *config, 186 audio_devices_t *device, 187 const String8& address, 188 audio_source_t source, 189 audio_input_flags_t flags); 190 191 virtual status_t closeInput(audio_io_handle_t input); 192 193 virtual status_t invalidateStream(audio_stream_type_t stream); 194 195 virtual status_t setVoiceVolume(float volume); 196 197 virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 198 audio_io_handle_t output) const; 199 200 virtual uint32_t getInputFramesLost(audio_io_handle_t ioHandle) const; 201 202 virtual audio_unique_id_t newAudioUniqueId(); 203 204 virtual void acquireAudioSessionId(int audioSession, pid_t pid); 205 206 virtual void releaseAudioSessionId(int audioSession, pid_t pid); 207 208 virtual status_t queryNumberEffects(uint32_t *numEffects) const; 209 210 virtual status_t queryEffect(uint32_t index, effect_descriptor_t *descriptor) const; 211 212 virtual status_t getEffectDescriptor(const effect_uuid_t *pUuid, 213 effect_descriptor_t *descriptor) const; 214 215 virtual sp<IEffect> createEffect( 216 effect_descriptor_t *pDesc, 217 const sp<IEffectClient>& effectClient, 218 int32_t priority, 219 audio_io_handle_t io, 220 int sessionId, 221 const String16& opPackageName, 222 status_t *status /*non-NULL*/, 223 int *id, 224 int *enabled); 225 226 virtual status_t moveEffects(int sessionId, audio_io_handle_t srcOutput, 227 audio_io_handle_t dstOutput); 228 229 virtual audio_module_handle_t loadHwModule(const char *name); 230 231 virtual uint32_t getPrimaryOutputSamplingRate(); 232 virtual size_t getPrimaryOutputFrameCount(); 233 234 virtual status_t setLowRamDevice(bool isLowRamDevice); 235 236 /* List available audio ports and their attributes */ 237 virtual status_t listAudioPorts(unsigned int *num_ports, 238 struct audio_port *ports); 239 240 /* Get attributes for a given audio port */ 241 virtual status_t getAudioPort(struct audio_port *port); 242 243 /* Create an audio patch between several source and sink ports */ 244 virtual status_t createAudioPatch(const struct audio_patch *patch, 245 audio_patch_handle_t *handle); 246 247 /* Release an audio patch */ 248 virtual status_t releaseAudioPatch(audio_patch_handle_t handle); 249 250 /* List existing audio patches */ 251 virtual status_t listAudioPatches(unsigned int *num_patches, 252 struct audio_patch *patches); 253 254 /* Set audio port configuration */ 255 virtual status_t setAudioPortConfig(const struct audio_port_config *config); 256 257 /* Get the HW synchronization source used for an audio session */ 258 virtual audio_hw_sync_t getAudioHwSyncForSession(audio_session_t sessionId); 259 260 /* Indicate JAVA services are ready (scheduling, power management ...) */ 261 virtual status_t systemReady(); 262 263 virtual status_t onTransact( 264 uint32_t code, 265 const Parcel& data, 266 Parcel* reply, 267 uint32_t flags); 268 269 // end of IAudioFlinger interface 270 271 sp<NBLog::Writer> newWriter_l(size_t size, const char *name); 272 void unregisterWriter(const sp<NBLog::Writer>& writer); 273private: 274 static const size_t kLogMemorySize = 40 * 1024; 275 sp<MemoryDealer> mLogMemoryDealer; // == 0 when NBLog is disabled 276 // When a log writer is unregistered, it is done lazily so that media.log can continue to see it 277 // for as long as possible. The memory is only freed when it is needed for another log writer. 278 Vector< sp<NBLog::Writer> > mUnregisteredWriters; 279 Mutex mUnregisteredWritersLock; 280public: 281 282 class SyncEvent; 283 284 typedef void (*sync_event_callback_t)(const wp<SyncEvent>& event) ; 285 286 class SyncEvent : public RefBase { 287 public: 288 SyncEvent(AudioSystem::sync_event_t type, 289 int triggerSession, 290 int listenerSession, 291 sync_event_callback_t callBack, 292 wp<RefBase> cookie) 293 : mType(type), mTriggerSession(triggerSession), mListenerSession(listenerSession), 294 mCallback(callBack), mCookie(cookie) 295 {} 296 297 virtual ~SyncEvent() {} 298 299 void trigger() { Mutex::Autolock _l(mLock); if (mCallback) mCallback(this); } 300 bool isCancelled() const { Mutex::Autolock _l(mLock); return (mCallback == NULL); } 301 void cancel() { Mutex::Autolock _l(mLock); mCallback = NULL; } 302 AudioSystem::sync_event_t type() const { return mType; } 303 int triggerSession() const { return mTriggerSession; } 304 int listenerSession() const { return mListenerSession; } 305 wp<RefBase> cookie() const { return mCookie; } 306 307 private: 308 const AudioSystem::sync_event_t mType; 309 const int mTriggerSession; 310 const int mListenerSession; 311 sync_event_callback_t mCallback; 312 const wp<RefBase> mCookie; 313 mutable Mutex mLock; 314 }; 315 316 sp<SyncEvent> createSyncEvent(AudioSystem::sync_event_t type, 317 int triggerSession, 318 int listenerSession, 319 sync_event_callback_t callBack, 320 wp<RefBase> cookie); 321 322private: 323 324 audio_mode_t getMode() const { return mMode; } 325 326 bool btNrecIsOff() const { return mBtNrecIsOff; } 327 328 AudioFlinger() ANDROID_API; 329 virtual ~AudioFlinger(); 330 331 // call in any IAudioFlinger method that accesses mPrimaryHardwareDev 332 status_t initCheck() const { return mPrimaryHardwareDev == NULL ? 333 NO_INIT : NO_ERROR; } 334 335 // RefBase 336 virtual void onFirstRef(); 337 338 AudioHwDevice* findSuitableHwDev_l(audio_module_handle_t module, 339 audio_devices_t devices); 340 void purgeStaleEffects_l(); 341 342 // Set kEnableExtendedChannels to true to enable greater than stereo output 343 // for the MixerThread and device sink. Number of channels allowed is 344 // FCC_2 <= channels <= AudioMixer::MAX_NUM_CHANNELS. 345 static const bool kEnableExtendedChannels = true; 346 347 // Returns true if channel mask is permitted for the PCM sink in the MixerThread 348 static inline bool isValidPcmSinkChannelMask(audio_channel_mask_t channelMask) { 349 switch (audio_channel_mask_get_representation(channelMask)) { 350 case AUDIO_CHANNEL_REPRESENTATION_POSITION: { 351 uint32_t channelCount = FCC_2; // stereo is default 352 if (kEnableExtendedChannels) { 353 channelCount = audio_channel_count_from_out_mask(channelMask); 354 if (channelCount < FCC_2 // mono is not supported at this time 355 || channelCount > AudioMixer::MAX_NUM_CHANNELS) { 356 return false; 357 } 358 } 359 // check that channelMask is the "canonical" one we expect for the channelCount. 360 return channelMask == audio_channel_out_mask_from_count(channelCount); 361 } 362 case AUDIO_CHANNEL_REPRESENTATION_INDEX: 363 if (kEnableExtendedChannels) { 364 const uint32_t channelCount = audio_channel_count_from_out_mask(channelMask); 365 if (channelCount >= FCC_2 // mono is not supported at this time 366 && channelCount <= AudioMixer::MAX_NUM_CHANNELS) { 367 return true; 368 } 369 } 370 return false; 371 default: 372 return false; 373 } 374 } 375 376 // Set kEnableExtendedPrecision to true to use extended precision in MixerThread 377 static const bool kEnableExtendedPrecision = true; 378 379 // Returns true if format is permitted for the PCM sink in the MixerThread 380 static inline bool isValidPcmSinkFormat(audio_format_t format) { 381 switch (format) { 382 case AUDIO_FORMAT_PCM_16_BIT: 383 return true; 384 case AUDIO_FORMAT_PCM_FLOAT: 385 case AUDIO_FORMAT_PCM_24_BIT_PACKED: 386 case AUDIO_FORMAT_PCM_32_BIT: 387 case AUDIO_FORMAT_PCM_8_24_BIT: 388 return kEnableExtendedPrecision; 389 default: 390 return false; 391 } 392 } 393 394 // standby delay for MIXER and DUPLICATING playback threads is read from property 395 // ro.audio.flinger_standbytime_ms or defaults to kDefaultStandbyTimeInNsecs 396 static nsecs_t mStandbyTimeInNsecs; 397 398 // incremented by 2 when screen state changes, bit 0 == 1 means "off" 399 // AudioFlinger::setParameters() updates, other threads read w/o lock 400 static uint32_t mScreenState; 401 402 // Internal dump utilities. 403 static const int kDumpLockRetries = 50; 404 static const int kDumpLockSleepUs = 20000; 405 static bool dumpTryLock(Mutex& mutex); 406 void dumpPermissionDenial(int fd, const Vector<String16>& args); 407 void dumpClients(int fd, const Vector<String16>& args); 408 void dumpInternals(int fd, const Vector<String16>& args); 409 410 // --- Client --- 411 class Client : public RefBase { 412 public: 413 Client(const sp<AudioFlinger>& audioFlinger, pid_t pid); 414 virtual ~Client(); 415 sp<MemoryDealer> heap() const; 416 pid_t pid() const { return mPid; } 417 sp<AudioFlinger> audioFlinger() const { return mAudioFlinger; } 418 419 bool reserveTimedTrack(); 420 void releaseTimedTrack(); 421 422 private: 423 Client(const Client&); 424 Client& operator = (const Client&); 425 const sp<AudioFlinger> mAudioFlinger; 426 const sp<MemoryDealer> mMemoryDealer; 427 const pid_t mPid; 428 429 Mutex mTimedTrackLock; 430 int mTimedTrackCount; 431 }; 432 433 // --- Notification Client --- 434 class NotificationClient : public IBinder::DeathRecipient { 435 public: 436 NotificationClient(const sp<AudioFlinger>& audioFlinger, 437 const sp<IAudioFlingerClient>& client, 438 pid_t pid); 439 virtual ~NotificationClient(); 440 441 sp<IAudioFlingerClient> audioFlingerClient() const { return mAudioFlingerClient; } 442 443 // IBinder::DeathRecipient 444 virtual void binderDied(const wp<IBinder>& who); 445 446 private: 447 NotificationClient(const NotificationClient&); 448 NotificationClient& operator = (const NotificationClient&); 449 450 const sp<AudioFlinger> mAudioFlinger; 451 const pid_t mPid; 452 const sp<IAudioFlingerClient> mAudioFlingerClient; 453 }; 454 455 class TrackHandle; 456 class RecordHandle; 457 class RecordThread; 458 class PlaybackThread; 459 class MixerThread; 460 class DirectOutputThread; 461 class OffloadThread; 462 class DuplicatingThread; 463 class AsyncCallbackThread; 464 class Track; 465 class RecordTrack; 466 class EffectModule; 467 class EffectHandle; 468 class EffectChain; 469 470 struct AudioStreamIn; 471 472 struct stream_type_t { 473 stream_type_t() 474 : volume(1.0f), 475 mute(false) 476 { 477 } 478 float volume; 479 bool mute; 480 }; 481 482 // --- PlaybackThread --- 483 484#include "Threads.h" 485 486#include "Effects.h" 487 488#include "PatchPanel.h" 489 490 // server side of the client's IAudioTrack 491 class TrackHandle : public android::BnAudioTrack { 492 public: 493 TrackHandle(const sp<PlaybackThread::Track>& track); 494 virtual ~TrackHandle(); 495 virtual sp<IMemory> getCblk() const; 496 virtual status_t start(); 497 virtual void stop(); 498 virtual void flush(); 499 virtual void pause(); 500 virtual status_t attachAuxEffect(int effectId); 501 virtual status_t allocateTimedBuffer(size_t size, 502 sp<IMemory>* buffer); 503 virtual status_t queueTimedBuffer(const sp<IMemory>& buffer, 504 int64_t pts); 505 virtual status_t setMediaTimeTransform(const LinearTransform& xform, 506 int target); 507 virtual status_t setParameters(const String8& keyValuePairs); 508 virtual status_t getTimestamp(AudioTimestamp& timestamp); 509 virtual void signal(); // signal playback thread for a change in control block 510 511 virtual status_t onTransact( 512 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 513 514 private: 515 const sp<PlaybackThread::Track> mTrack; 516 }; 517 518 // server side of the client's IAudioRecord 519 class RecordHandle : public android::BnAudioRecord { 520 public: 521 RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack); 522 virtual ~RecordHandle(); 523 virtual status_t start(int /*AudioSystem::sync_event_t*/ event, int triggerSession); 524 virtual void stop(); 525 virtual status_t onTransact( 526 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 527 private: 528 const sp<RecordThread::RecordTrack> mRecordTrack; 529 530 // for use from destructor 531 void stop_nonvirtual(); 532 }; 533 534 535 PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const; 536 MixerThread *checkMixerThread_l(audio_io_handle_t output) const; 537 RecordThread *checkRecordThread_l(audio_io_handle_t input) const; 538 sp<RecordThread> openInput_l(audio_module_handle_t module, 539 audio_io_handle_t *input, 540 audio_config_t *config, 541 audio_devices_t device, 542 const String8& address, 543 audio_source_t source, 544 audio_input_flags_t flags); 545 sp<PlaybackThread> openOutput_l(audio_module_handle_t module, 546 audio_io_handle_t *output, 547 audio_config_t *config, 548 audio_devices_t devices, 549 const String8& address, 550 audio_output_flags_t flags); 551 552 void closeOutputFinish(sp<PlaybackThread> thread); 553 void closeInputFinish(sp<RecordThread> thread); 554 555 // no range check, AudioFlinger::mLock held 556 bool streamMute_l(audio_stream_type_t stream) const 557 { return mStreamTypes[stream].mute; } 558 // no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held 559 float streamVolume_l(audio_stream_type_t stream) const 560 { return mStreamTypes[stream].volume; } 561 void ioConfigChanged(audio_io_config_event event, 562 const sp<AudioIoDescriptor>& ioDesc, 563 pid_t pid = 0); 564 565 // Allocate an audio_io_handle_t, session ID, effect ID, or audio_module_handle_t. 566 // They all share the same ID space, but the namespaces are actually independent 567 // because there are separate KeyedVectors for each kind of ID. 568 // The return value is uint32_t, but is cast to signed for some IDs. 569 // FIXME This API does not handle rollover to zero (for unsigned IDs), 570 // or from positive to negative (for signed IDs). 571 // Thus it may fail by returning an ID of the wrong sign, 572 // or by returning a non-unique ID. 573 uint32_t nextUniqueId(); 574 575 status_t moveEffectChain_l(int sessionId, 576 PlaybackThread *srcThread, 577 PlaybackThread *dstThread, 578 bool reRegister); 579 // return thread associated with primary hardware device, or NULL 580 PlaybackThread *primaryPlaybackThread_l() const; 581 audio_devices_t primaryOutputDevice_l() const; 582 583 sp<PlaybackThread> getEffectThread_l(int sessionId, int EffectId); 584 585 586 void removeClient_l(pid_t pid); 587 void removeNotificationClient(pid_t pid); 588 bool isNonOffloadableGlobalEffectEnabled_l(); 589 void onNonOffloadableGlobalEffectEnable(); 590 591 // Store an effect chain to mOrphanEffectChains keyed vector. 592 // Called when a thread exits and effects are still attached to it. 593 // If effects are later created on the same session, they will reuse the same 594 // effect chain and same instances in the effect library. 595 // return ALREADY_EXISTS if a chain with the same session already exists in 596 // mOrphanEffectChains. Note that this should never happen as there is only one 597 // chain for a given session and it is attached to only one thread at a time. 598 status_t putOrphanEffectChain_l(const sp<EffectChain>& chain); 599 // Get an effect chain for the specified session in mOrphanEffectChains and remove 600 // it if found. Returns 0 if not found (this is the most common case). 601 sp<EffectChain> getOrphanEffectChain_l(audio_session_t session); 602 // Called when the last effect handle on an effect instance is removed. If this 603 // effect belongs to an effect chain in mOrphanEffectChains, the chain is updated 604 // and removed from mOrphanEffectChains if it does not contain any effect. 605 // Return true if the effect was found in mOrphanEffectChains, false otherwise. 606 bool updateOrphanEffectChains(const sp<EffectModule>& effect); 607 608 void broacastParametersToRecordThreads_l(const String8& keyValuePairs); 609 610 // AudioStreamIn is immutable, so their fields are const. 611 // For emphasis, we could also make all pointers to them be "const *", 612 // but that would clutter the code unnecessarily. 613 614 struct AudioStreamIn { 615 AudioHwDevice* const audioHwDev; 616 audio_stream_in_t* const stream; 617 618 audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); } 619 620 AudioStreamIn(AudioHwDevice *dev, audio_stream_in_t *in) : 621 audioHwDev(dev), stream(in) {} 622 }; 623 624 // for mAudioSessionRefs only 625 struct AudioSessionRef { 626 AudioSessionRef(int sessionid, pid_t pid) : 627 mSessionid(sessionid), mPid(pid), mCnt(1) {} 628 const int mSessionid; 629 const pid_t mPid; 630 int mCnt; 631 }; 632 633 mutable Mutex mLock; 634 // protects mClients and mNotificationClients. 635 // must be locked after mLock and ThreadBase::mLock if both must be locked 636 // avoids acquiring AudioFlinger::mLock from inside thread loop. 637 mutable Mutex mClientLock; 638 // protected by mClientLock 639 DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client() 640 641 mutable Mutex mHardwareLock; 642 // NOTE: If both mLock and mHardwareLock mutexes must be held, 643 // always take mLock before mHardwareLock 644 645 // These two fields are immutable after onFirstRef(), so no lock needed to access 646 AudioHwDevice* mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL 647 DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*> mAudioHwDevs; 648 649 // for dump, indicates which hardware operation is currently in progress (but not stream ops) 650 enum hardware_call_state { 651 AUDIO_HW_IDLE = 0, // no operation in progress 652 AUDIO_HW_INIT, // init_check 653 AUDIO_HW_OUTPUT_OPEN, // open_output_stream 654 AUDIO_HW_OUTPUT_CLOSE, // unused 655 AUDIO_HW_INPUT_OPEN, // unused 656 AUDIO_HW_INPUT_CLOSE, // unused 657 AUDIO_HW_STANDBY, // unused 658 AUDIO_HW_SET_MASTER_VOLUME, // set_master_volume 659 AUDIO_HW_GET_ROUTING, // unused 660 AUDIO_HW_SET_ROUTING, // unused 661 AUDIO_HW_GET_MODE, // unused 662 AUDIO_HW_SET_MODE, // set_mode 663 AUDIO_HW_GET_MIC_MUTE, // get_mic_mute 664 AUDIO_HW_SET_MIC_MUTE, // set_mic_mute 665 AUDIO_HW_SET_VOICE_VOLUME, // set_voice_volume 666 AUDIO_HW_SET_PARAMETER, // set_parameters 667 AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size 668 AUDIO_HW_GET_MASTER_VOLUME, // get_master_volume 669 AUDIO_HW_GET_PARAMETER, // get_parameters 670 AUDIO_HW_SET_MASTER_MUTE, // set_master_mute 671 AUDIO_HW_GET_MASTER_MUTE, // get_master_mute 672 }; 673 674 mutable hardware_call_state mHardwareStatus; // for dump only 675 676 677 DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads; 678 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 679 680 // member variables below are protected by mLock 681 float mMasterVolume; 682 bool mMasterMute; 683 // end of variables protected by mLock 684 685 DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads; 686 687 // protected by mClientLock 688 DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients; 689 690 volatile int32_t mNextUniqueId; // updated by android_atomic_inc 691 // nextUniqueId() returns uint32_t, but this is declared int32_t 692 // because the atomic operations require an int32_t 693 694 audio_mode_t mMode; 695 bool mBtNrecIsOff; 696 697 // protected by mLock 698 Vector<AudioSessionRef*> mAudioSessionRefs; 699 700 float masterVolume_l() const; 701 bool masterMute_l() const; 702 audio_module_handle_t loadHwModule_l(const char *name); 703 704 Vector < sp<SyncEvent> > mPendingSyncEvents; // sync events awaiting for a session 705 // to be created 706 707 // Effect chains without a valid thread 708 DefaultKeyedVector< audio_session_t , sp<EffectChain> > mOrphanEffectChains; 709 710 // list of sessions for which a valid HW A/V sync ID was retrieved from the HAL 711 DefaultKeyedVector< audio_session_t , audio_hw_sync_t >mHwAvSyncIds; 712private: 713 sp<Client> registerPid(pid_t pid); // always returns non-0 714 715 // for use from destructor 716 status_t closeOutput_nonvirtual(audio_io_handle_t output); 717 void closeOutputInternal_l(sp<PlaybackThread> thread); 718 status_t closeInput_nonvirtual(audio_io_handle_t input); 719 void closeInputInternal_l(sp<RecordThread> thread); 720 void setAudioHwSyncForSession_l(PlaybackThread *thread, audio_session_t sessionId); 721 722 status_t checkStreamType(audio_stream_type_t stream) const; 723 724#ifdef TEE_SINK 725 // all record threads serially share a common tee sink, which is re-created on format change 726 sp<NBAIO_Sink> mRecordTeeSink; 727 sp<NBAIO_Source> mRecordTeeSource; 728#endif 729 730public: 731 732#ifdef TEE_SINK 733 // tee sink, if enabled by property, allows dumpsys to write most recent audio to .wav file 734 static void dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id = 0); 735 736 // whether tee sink is enabled by property 737 static bool mTeeSinkInputEnabled; 738 static bool mTeeSinkOutputEnabled; 739 static bool mTeeSinkTrackEnabled; 740 741 // runtime configured size of each tee sink pipe, in frames 742 static size_t mTeeSinkInputFrames; 743 static size_t mTeeSinkOutputFrames; 744 static size_t mTeeSinkTrackFrames; 745 746 // compile-time default size of tee sink pipes, in frames 747 // 0x200000 stereo 16-bit PCM frames = 47.5 seconds at 44.1 kHz, 8 megabytes 748 static const size_t kTeeSinkInputFramesDefault = 0x200000; 749 static const size_t kTeeSinkOutputFramesDefault = 0x200000; 750 static const size_t kTeeSinkTrackFramesDefault = 0x200000; 751#endif 752 753 // This method reads from a variable without mLock, but the variable is updated under mLock. So 754 // we might read a stale value, or a value that's inconsistent with respect to other variables. 755 // In this case, it's safe because the return value isn't used for making an important decision. 756 // The reason we don't want to take mLock is because it could block the caller for a long time. 757 bool isLowRamDevice() const { return mIsLowRamDevice; } 758 759private: 760 bool mIsLowRamDevice; 761 bool mIsDeviceTypeKnown; 762 nsecs_t mGlobalEffectEnableTime; // when a global effect was last enabled 763 764 sp<PatchPanel> mPatchPanel; 765 766 bool mSystemReady; 767}; 768 769#undef INCLUDING_FROM_AUDIOFLINGER_H 770 771const char *formatToString(audio_format_t format); 772String8 inputFlagsToString(audio_input_flags_t flags); 773String8 outputFlagsToString(audio_output_flags_t flags); 774String8 devicesToString(audio_devices_t devices); 775const char *sourceToString(audio_source_t source); 776 777// ---------------------------------------------------------------------------- 778 779} // namespace android 780 781#endif // ANDROID_AUDIO_FLINGER_H 782