AudioFlinger.h revision 64b6cb2cd5646a5ffcdb2cccee09a170ef2882d5
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 virtual status_t onTransact( 261 uint32_t code, 262 const Parcel& data, 263 Parcel* reply, 264 uint32_t flags); 265 266 // end of IAudioFlinger interface 267 268 sp<NBLog::Writer> newWriter_l(size_t size, const char *name); 269 void unregisterWriter(const sp<NBLog::Writer>& writer); 270private: 271 static const size_t kLogMemorySize = 40 * 1024; 272 sp<MemoryDealer> mLogMemoryDealer; // == 0 when NBLog is disabled 273 // When a log writer is unregistered, it is done lazily so that media.log can continue to see it 274 // for as long as possible. The memory is only freed when it is needed for another log writer. 275 Vector< sp<NBLog::Writer> > mUnregisteredWriters; 276 Mutex mUnregisteredWritersLock; 277public: 278 279 class SyncEvent; 280 281 typedef void (*sync_event_callback_t)(const wp<SyncEvent>& event) ; 282 283 class SyncEvent : public RefBase { 284 public: 285 SyncEvent(AudioSystem::sync_event_t type, 286 int triggerSession, 287 int listenerSession, 288 sync_event_callback_t callBack, 289 wp<RefBase> cookie) 290 : mType(type), mTriggerSession(triggerSession), mListenerSession(listenerSession), 291 mCallback(callBack), mCookie(cookie) 292 {} 293 294 virtual ~SyncEvent() {} 295 296 void trigger() { Mutex::Autolock _l(mLock); if (mCallback) mCallback(this); } 297 bool isCancelled() const { Mutex::Autolock _l(mLock); return (mCallback == NULL); } 298 void cancel() { Mutex::Autolock _l(mLock); mCallback = NULL; } 299 AudioSystem::sync_event_t type() const { return mType; } 300 int triggerSession() const { return mTriggerSession; } 301 int listenerSession() const { return mListenerSession; } 302 wp<RefBase> cookie() const { return mCookie; } 303 304 private: 305 const AudioSystem::sync_event_t mType; 306 const int mTriggerSession; 307 const int mListenerSession; 308 sync_event_callback_t mCallback; 309 const wp<RefBase> mCookie; 310 mutable Mutex mLock; 311 }; 312 313 sp<SyncEvent> createSyncEvent(AudioSystem::sync_event_t type, 314 int triggerSession, 315 int listenerSession, 316 sync_event_callback_t callBack, 317 wp<RefBase> cookie); 318 319private: 320 321 audio_mode_t getMode() const { return mMode; } 322 323 bool btNrecIsOff() const { return mBtNrecIsOff; } 324 325 AudioFlinger() ANDROID_API; 326 virtual ~AudioFlinger(); 327 328 // call in any IAudioFlinger method that accesses mPrimaryHardwareDev 329 status_t initCheck() const { return mPrimaryHardwareDev == NULL ? 330 NO_INIT : NO_ERROR; } 331 332 // RefBase 333 virtual void onFirstRef(); 334 335 AudioHwDevice* findSuitableHwDev_l(audio_module_handle_t module, 336 audio_devices_t devices); 337 void purgeStaleEffects_l(); 338 339 // Set kEnableExtendedChannels to true to enable greater than stereo output 340 // for the MixerThread and device sink. Number of channels allowed is 341 // FCC_2 <= channels <= AudioMixer::MAX_NUM_CHANNELS. 342 static const bool kEnableExtendedChannels = true; 343 344 // Returns true if channel mask is permitted for the PCM sink in the MixerThread 345 static inline bool isValidPcmSinkChannelMask(audio_channel_mask_t channelMask) { 346 switch (audio_channel_mask_get_representation(channelMask)) { 347 case AUDIO_CHANNEL_REPRESENTATION_POSITION: { 348 uint32_t channelCount = FCC_2; // stereo is default 349 if (kEnableExtendedChannels) { 350 channelCount = audio_channel_count_from_out_mask(channelMask); 351 if (channelCount < FCC_2 // mono is not supported at this time 352 || channelCount > AudioMixer::MAX_NUM_CHANNELS) { 353 return false; 354 } 355 } 356 // check that channelMask is the "canonical" one we expect for the channelCount. 357 return channelMask == audio_channel_out_mask_from_count(channelCount); 358 } 359 case AUDIO_CHANNEL_REPRESENTATION_INDEX: 360 if (kEnableExtendedChannels) { 361 const uint32_t channelCount = audio_channel_count_from_out_mask(channelMask); 362 if (channelCount >= FCC_2 // mono is not supported at this time 363 && channelCount <= AudioMixer::MAX_NUM_CHANNELS) { 364 return true; 365 } 366 } 367 return false; 368 default: 369 return false; 370 } 371 } 372 373 // Set kEnableExtendedPrecision to true to use extended precision in MixerThread 374 static const bool kEnableExtendedPrecision = true; 375 376 // Returns true if format is permitted for the PCM sink in the MixerThread 377 static inline bool isValidPcmSinkFormat(audio_format_t format) { 378 switch (format) { 379 case AUDIO_FORMAT_PCM_16_BIT: 380 return true; 381 case AUDIO_FORMAT_PCM_FLOAT: 382 case AUDIO_FORMAT_PCM_24_BIT_PACKED: 383 case AUDIO_FORMAT_PCM_32_BIT: 384 case AUDIO_FORMAT_PCM_8_24_BIT: 385 return kEnableExtendedPrecision; 386 default: 387 return false; 388 } 389 } 390 391 // standby delay for MIXER and DUPLICATING playback threads is read from property 392 // ro.audio.flinger_standbytime_ms or defaults to kDefaultStandbyTimeInNsecs 393 static nsecs_t mStandbyTimeInNsecs; 394 395 // incremented by 2 when screen state changes, bit 0 == 1 means "off" 396 // AudioFlinger::setParameters() updates, other threads read w/o lock 397 static uint32_t mScreenState; 398 399 // Internal dump utilities. 400 static const int kDumpLockRetries = 50; 401 static const int kDumpLockSleepUs = 20000; 402 static bool dumpTryLock(Mutex& mutex); 403 void dumpPermissionDenial(int fd, const Vector<String16>& args); 404 void dumpClients(int fd, const Vector<String16>& args); 405 void dumpInternals(int fd, const Vector<String16>& args); 406 407 // --- Client --- 408 class Client : public RefBase { 409 public: 410 Client(const sp<AudioFlinger>& audioFlinger, pid_t pid); 411 virtual ~Client(); 412 sp<MemoryDealer> heap() const; 413 pid_t pid() const { return mPid; } 414 sp<AudioFlinger> audioFlinger() const { return mAudioFlinger; } 415 416 bool reserveTimedTrack(); 417 void releaseTimedTrack(); 418 419 private: 420 Client(const Client&); 421 Client& operator = (const Client&); 422 const sp<AudioFlinger> mAudioFlinger; 423 const sp<MemoryDealer> mMemoryDealer; 424 const pid_t mPid; 425 426 Mutex mTimedTrackLock; 427 int mTimedTrackCount; 428 }; 429 430 // --- Notification Client --- 431 class NotificationClient : public IBinder::DeathRecipient { 432 public: 433 NotificationClient(const sp<AudioFlinger>& audioFlinger, 434 const sp<IAudioFlingerClient>& client, 435 pid_t pid); 436 virtual ~NotificationClient(); 437 438 sp<IAudioFlingerClient> audioFlingerClient() const { return mAudioFlingerClient; } 439 440 // IBinder::DeathRecipient 441 virtual void binderDied(const wp<IBinder>& who); 442 443 private: 444 NotificationClient(const NotificationClient&); 445 NotificationClient& operator = (const NotificationClient&); 446 447 const sp<AudioFlinger> mAudioFlinger; 448 const pid_t mPid; 449 const sp<IAudioFlingerClient> mAudioFlingerClient; 450 }; 451 452 class TrackHandle; 453 class RecordHandle; 454 class RecordThread; 455 class PlaybackThread; 456 class MixerThread; 457 class DirectOutputThread; 458 class OffloadThread; 459 class DuplicatingThread; 460 class AsyncCallbackThread; 461 class Track; 462 class RecordTrack; 463 class EffectModule; 464 class EffectHandle; 465 class EffectChain; 466 467 struct AudioStreamIn; 468 469 struct stream_type_t { 470 stream_type_t() 471 : volume(1.0f), 472 mute(false) 473 { 474 } 475 float volume; 476 bool mute; 477 }; 478 479 // --- PlaybackThread --- 480 481#include "Threads.h" 482 483#include "Effects.h" 484 485#include "PatchPanel.h" 486 487 // server side of the client's IAudioTrack 488 class TrackHandle : public android::BnAudioTrack { 489 public: 490 TrackHandle(const sp<PlaybackThread::Track>& track); 491 virtual ~TrackHandle(); 492 virtual sp<IMemory> getCblk() const; 493 virtual status_t start(); 494 virtual void stop(); 495 virtual void flush(); 496 virtual void pause(); 497 virtual status_t attachAuxEffect(int effectId); 498 virtual status_t allocateTimedBuffer(size_t size, 499 sp<IMemory>* buffer); 500 virtual status_t queueTimedBuffer(const sp<IMemory>& buffer, 501 int64_t pts); 502 virtual status_t setMediaTimeTransform(const LinearTransform& xform, 503 int target); 504 virtual status_t setParameters(const String8& keyValuePairs); 505 virtual status_t getTimestamp(AudioTimestamp& timestamp); 506 virtual void signal(); // signal playback thread for a change in control block 507 508 virtual status_t onTransact( 509 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 510 511 private: 512 const sp<PlaybackThread::Track> mTrack; 513 }; 514 515 // server side of the client's IAudioRecord 516 class RecordHandle : public android::BnAudioRecord { 517 public: 518 RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack); 519 virtual ~RecordHandle(); 520 virtual status_t start(int /*AudioSystem::sync_event_t*/ event, int triggerSession); 521 virtual void stop(); 522 virtual status_t onTransact( 523 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 524 private: 525 const sp<RecordThread::RecordTrack> mRecordTrack; 526 527 // for use from destructor 528 void stop_nonvirtual(); 529 }; 530 531 532 PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const; 533 MixerThread *checkMixerThread_l(audio_io_handle_t output) const; 534 RecordThread *checkRecordThread_l(audio_io_handle_t input) const; 535 sp<RecordThread> openInput_l(audio_module_handle_t module, 536 audio_io_handle_t *input, 537 audio_config_t *config, 538 audio_devices_t device, 539 const String8& address, 540 audio_source_t source, 541 audio_input_flags_t flags); 542 sp<PlaybackThread> openOutput_l(audio_module_handle_t module, 543 audio_io_handle_t *output, 544 audio_config_t *config, 545 audio_devices_t devices, 546 const String8& address, 547 audio_output_flags_t flags); 548 549 void closeOutputFinish(sp<PlaybackThread> thread); 550 void closeInputFinish(sp<RecordThread> thread); 551 552 // no range check, AudioFlinger::mLock held 553 bool streamMute_l(audio_stream_type_t stream) const 554 { return mStreamTypes[stream].mute; } 555 // no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held 556 float streamVolume_l(audio_stream_type_t stream) const 557 { return mStreamTypes[stream].volume; } 558 void ioConfigChanged(audio_io_config_event event, 559 const sp<AudioIoDescriptor>& ioDesc); 560 561 // Allocate an audio_io_handle_t, session ID, effect ID, or audio_module_handle_t. 562 // They all share the same ID space, but the namespaces are actually independent 563 // because there are separate KeyedVectors for each kind of ID. 564 // The return value is uint32_t, but is cast to signed for some IDs. 565 // FIXME This API does not handle rollover to zero (for unsigned IDs), 566 // or from positive to negative (for signed IDs). 567 // Thus it may fail by returning an ID of the wrong sign, 568 // or by returning a non-unique ID. 569 uint32_t nextUniqueId(); 570 571 status_t moveEffectChain_l(int sessionId, 572 PlaybackThread *srcThread, 573 PlaybackThread *dstThread, 574 bool reRegister); 575 // return thread associated with primary hardware device, or NULL 576 PlaybackThread *primaryPlaybackThread_l() const; 577 audio_devices_t primaryOutputDevice_l() const; 578 579 sp<PlaybackThread> getEffectThread_l(int sessionId, int EffectId); 580 581 582 void removeClient_l(pid_t pid); 583 void removeNotificationClient(pid_t pid); 584 bool isNonOffloadableGlobalEffectEnabled_l(); 585 void onNonOffloadableGlobalEffectEnable(); 586 587 // Store an effect chain to mOrphanEffectChains keyed vector. 588 // Called when a thread exits and effects are still attached to it. 589 // If effects are later created on the same session, they will reuse the same 590 // effect chain and same instances in the effect library. 591 // return ALREADY_EXISTS if a chain with the same session already exists in 592 // mOrphanEffectChains. Note that this should never happen as there is only one 593 // chain for a given session and it is attached to only one thread at a time. 594 status_t putOrphanEffectChain_l(const sp<EffectChain>& chain); 595 // Get an effect chain for the specified session in mOrphanEffectChains and remove 596 // it if found. Returns 0 if not found (this is the most common case). 597 sp<EffectChain> getOrphanEffectChain_l(audio_session_t session); 598 // Called when the last effect handle on an effect instance is removed. If this 599 // effect belongs to an effect chain in mOrphanEffectChains, the chain is updated 600 // and removed from mOrphanEffectChains if it does not contain any effect. 601 // Return true if the effect was found in mOrphanEffectChains, false otherwise. 602 bool updateOrphanEffectChains(const sp<EffectModule>& effect); 603 604 void broacastParametersToRecordThreads_l(const String8& keyValuePairs); 605 606 // AudioStreamIn is immutable, so their fields are const. 607 // For emphasis, we could also make all pointers to them be "const *", 608 // but that would clutter the code unnecessarily. 609 610 struct AudioStreamIn { 611 AudioHwDevice* const audioHwDev; 612 audio_stream_in_t* const stream; 613 614 audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); } 615 616 AudioStreamIn(AudioHwDevice *dev, audio_stream_in_t *in) : 617 audioHwDev(dev), stream(in) {} 618 }; 619 620 // for mAudioSessionRefs only 621 struct AudioSessionRef { 622 AudioSessionRef(int sessionid, pid_t pid) : 623 mSessionid(sessionid), mPid(pid), mCnt(1) {} 624 const int mSessionid; 625 const pid_t mPid; 626 int mCnt; 627 }; 628 629 mutable Mutex mLock; 630 // protects mClients and mNotificationClients. 631 // must be locked after mLock and ThreadBase::mLock if both must be locked 632 // avoids acquiring AudioFlinger::mLock from inside thread loop. 633 mutable Mutex mClientLock; 634 // protected by mClientLock 635 DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client() 636 637 mutable Mutex mHardwareLock; 638 // NOTE: If both mLock and mHardwareLock mutexes must be held, 639 // always take mLock before mHardwareLock 640 641 // These two fields are immutable after onFirstRef(), so no lock needed to access 642 AudioHwDevice* mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL 643 DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*> mAudioHwDevs; 644 645 // for dump, indicates which hardware operation is currently in progress (but not stream ops) 646 enum hardware_call_state { 647 AUDIO_HW_IDLE = 0, // no operation in progress 648 AUDIO_HW_INIT, // init_check 649 AUDIO_HW_OUTPUT_OPEN, // open_output_stream 650 AUDIO_HW_OUTPUT_CLOSE, // unused 651 AUDIO_HW_INPUT_OPEN, // unused 652 AUDIO_HW_INPUT_CLOSE, // unused 653 AUDIO_HW_STANDBY, // unused 654 AUDIO_HW_SET_MASTER_VOLUME, // set_master_volume 655 AUDIO_HW_GET_ROUTING, // unused 656 AUDIO_HW_SET_ROUTING, // unused 657 AUDIO_HW_GET_MODE, // unused 658 AUDIO_HW_SET_MODE, // set_mode 659 AUDIO_HW_GET_MIC_MUTE, // get_mic_mute 660 AUDIO_HW_SET_MIC_MUTE, // set_mic_mute 661 AUDIO_HW_SET_VOICE_VOLUME, // set_voice_volume 662 AUDIO_HW_SET_PARAMETER, // set_parameters 663 AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size 664 AUDIO_HW_GET_MASTER_VOLUME, // get_master_volume 665 AUDIO_HW_GET_PARAMETER, // get_parameters 666 AUDIO_HW_SET_MASTER_MUTE, // set_master_mute 667 AUDIO_HW_GET_MASTER_MUTE, // get_master_mute 668 }; 669 670 mutable hardware_call_state mHardwareStatus; // for dump only 671 672 673 DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads; 674 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 675 676 // member variables below are protected by mLock 677 float mMasterVolume; 678 bool mMasterMute; 679 // end of variables protected by mLock 680 681 DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads; 682 683 // protected by mClientLock 684 DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients; 685 686 volatile int32_t mNextUniqueId; // updated by android_atomic_inc 687 // nextUniqueId() returns uint32_t, but this is declared int32_t 688 // because the atomic operations require an int32_t 689 690 audio_mode_t mMode; 691 bool mBtNrecIsOff; 692 693 // protected by mLock 694 Vector<AudioSessionRef*> mAudioSessionRefs; 695 696 float masterVolume_l() const; 697 bool masterMute_l() const; 698 audio_module_handle_t loadHwModule_l(const char *name); 699 700 Vector < sp<SyncEvent> > mPendingSyncEvents; // sync events awaiting for a session 701 // to be created 702 703 // Effect chains without a valid thread 704 DefaultKeyedVector< audio_session_t , sp<EffectChain> > mOrphanEffectChains; 705 706 // list of sessions for which a valid HW A/V sync ID was retrieved from the HAL 707 DefaultKeyedVector< audio_session_t , audio_hw_sync_t >mHwAvSyncIds; 708private: 709 sp<Client> registerPid(pid_t pid); // always returns non-0 710 711 // for use from destructor 712 status_t closeOutput_nonvirtual(audio_io_handle_t output); 713 void closeOutputInternal_l(sp<PlaybackThread> thread); 714 status_t closeInput_nonvirtual(audio_io_handle_t input); 715 void closeInputInternal_l(sp<RecordThread> thread); 716 void setAudioHwSyncForSession_l(PlaybackThread *thread, audio_session_t sessionId); 717 718 status_t checkStreamType(audio_stream_type_t stream) const; 719 720#ifdef TEE_SINK 721 // all record threads serially share a common tee sink, which is re-created on format change 722 sp<NBAIO_Sink> mRecordTeeSink; 723 sp<NBAIO_Source> mRecordTeeSource; 724#endif 725 726public: 727 728#ifdef TEE_SINK 729 // tee sink, if enabled by property, allows dumpsys to write most recent audio to .wav file 730 static void dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id = 0); 731 732 // whether tee sink is enabled by property 733 static bool mTeeSinkInputEnabled; 734 static bool mTeeSinkOutputEnabled; 735 static bool mTeeSinkTrackEnabled; 736 737 // runtime configured size of each tee sink pipe, in frames 738 static size_t mTeeSinkInputFrames; 739 static size_t mTeeSinkOutputFrames; 740 static size_t mTeeSinkTrackFrames; 741 742 // compile-time default size of tee sink pipes, in frames 743 // 0x200000 stereo 16-bit PCM frames = 47.5 seconds at 44.1 kHz, 8 megabytes 744 static const size_t kTeeSinkInputFramesDefault = 0x200000; 745 static const size_t kTeeSinkOutputFramesDefault = 0x200000; 746 static const size_t kTeeSinkTrackFramesDefault = 0x200000; 747#endif 748 749 // This method reads from a variable without mLock, but the variable is updated under mLock. So 750 // we might read a stale value, or a value that's inconsistent with respect to other variables. 751 // In this case, it's safe because the return value isn't used for making an important decision. 752 // The reason we don't want to take mLock is because it could block the caller for a long time. 753 bool isLowRamDevice() const { return mIsLowRamDevice; } 754 755private: 756 bool mIsLowRamDevice; 757 bool mIsDeviceTypeKnown; 758 nsecs_t mGlobalEffectEnableTime; // when a global effect was last enabled 759 760 sp<PatchPanel> mPatchPanel; 761 762 uint32_t mPrimaryOutputSampleRate; // sample rate of the primary output, or zero if none 763 // protected by mHardwareLock 764}; 765 766#undef INCLUDING_FROM_AUDIOFLINGER_H 767 768const char *formatToString(audio_format_t format); 769String8 inputFlagsToString(audio_input_flags_t flags); 770String8 outputFlagsToString(audio_output_flags_t flags); 771String8 devicesToString(audio_devices_t devices); 772const char *sourceToString(audio_source_t source); 773 774// ---------------------------------------------------------------------------- 775 776} // namespace android 777 778#endif // ANDROID_AUDIO_FLINGER_H 779