AudioFlinger.h revision 82f5b811b2767289ebb8a9e6af1919c3b72b5121
1/* 2** 3** Copyright 2007, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18#ifndef ANDROID_AUDIO_FLINGER_H 19#define ANDROID_AUDIO_FLINGER_H 20 21#include "Configuration.h" 22#include <stdint.h> 23#include <sys/types.h> 24#include <limits.h> 25 26#include <cutils/compiler.h> 27 28#include <media/IAudioFlinger.h> 29#include <media/IAudioFlingerClient.h> 30#include <media/IAudioTrack.h> 31#include <media/IAudioRecord.h> 32#include <media/AudioSystem.h> 33#include <media/AudioTrack.h> 34 35#include <utils/Atomic.h> 36#include <utils/Errors.h> 37#include <utils/threads.h> 38#include <utils/SortedVector.h> 39#include <utils/TypeHelpers.h> 40#include <utils/Vector.h> 41 42#include <binder/BinderService.h> 43#include <binder/MemoryDealer.h> 44 45#include <system/audio.h> 46#include <hardware/audio.h> 47#include <hardware/audio_policy.h> 48 49#include <media/AudioBufferProvider.h> 50#include <media/ExtendedAudioBufferProvider.h> 51 52#include "FastCapture.h" 53#include "FastMixer.h" 54#include <media/nbaio/NBAIO.h> 55#include "AudioWatchdog.h" 56#include "AudioMixer.h" 57#include "AudioStreamOut.h" 58#include "SpdifStreamOut.h" 59#include "AudioHwDevice.h" 60#include "LinearMap.h" 61#include "LockWatch.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 audio_output_flags_t *flags, 109 const sp<IMemory>& sharedBuffer, 110 audio_io_handle_t output, 111 pid_t pid, 112 pid_t tid, 113 audio_session_t *sessionId, 114 int clientUid, 115 status_t *status /*non-NULL*/); 116 117 virtual sp<IAudioRecord> openRecord( 118 audio_io_handle_t input, 119 uint32_t sampleRate, 120 audio_format_t format, 121 audio_channel_mask_t channelMask, 122 const String16& opPackageName, 123 size_t *pFrameCount, 124 audio_input_flags_t *flags, 125 pid_t pid, 126 pid_t tid, 127 int clientUid, 128 audio_session_t *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 ioHandle) const; 135 virtual audio_format_t format(audio_io_handle_t output) const; 136 virtual size_t frameCount(audio_io_handle_t ioHandle) const; 137 virtual size_t frameCountHAL(audio_io_handle_t ioHandle) const; 138 virtual uint32_t latency(audio_io_handle_t output) const; 139 140 virtual status_t setMasterVolume(float value); 141 virtual status_t setMasterMute(bool muted); 142 143 virtual float masterVolume() const; 144 virtual bool masterMute() const; 145 146 virtual status_t setStreamVolume(audio_stream_type_t stream, float value, 147 audio_io_handle_t output); 148 virtual status_t setStreamMute(audio_stream_type_t stream, bool muted); 149 150 virtual float streamVolume(audio_stream_type_t stream, 151 audio_io_handle_t output) const; 152 virtual bool streamMute(audio_stream_type_t stream) const; 153 154 virtual status_t setMode(audio_mode_t mode); 155 156 virtual status_t setMicMute(bool state); 157 virtual bool getMicMute() const; 158 159 virtual status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs); 160 virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) const; 161 162 virtual void registerClient(const sp<IAudioFlingerClient>& client); 163 164 virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, 165 audio_channel_mask_t channelMask) const; 166 167 virtual status_t openOutput(audio_module_handle_t module, 168 audio_io_handle_t *output, 169 audio_config_t *config, 170 audio_devices_t *devices, 171 const String8& address, 172 uint32_t *latencyMs, 173 audio_output_flags_t flags); 174 175 virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1, 176 audio_io_handle_t output2); 177 178 virtual status_t closeOutput(audio_io_handle_t output); 179 180 virtual status_t suspendOutput(audio_io_handle_t output); 181 182 virtual status_t restoreOutput(audio_io_handle_t output); 183 184 virtual status_t openInput(audio_module_handle_t module, 185 audio_io_handle_t *input, 186 audio_config_t *config, 187 audio_devices_t *device, 188 const String8& address, 189 audio_source_t source, 190 audio_input_flags_t flags); 191 192 virtual status_t closeInput(audio_io_handle_t input); 193 194 virtual status_t invalidateStream(audio_stream_type_t stream); 195 196 virtual status_t setVoiceVolume(float volume); 197 198 virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 199 audio_io_handle_t output) const; 200 201 virtual uint32_t getInputFramesLost(audio_io_handle_t ioHandle) const; 202 203 // This is the binder API. For the internal API see nextUniqueId(). 204 virtual audio_unique_id_t newAudioUniqueId(audio_unique_id_use_t use); 205 206 virtual void acquireAudioSessionId(audio_session_t audioSession, pid_t pid); 207 208 virtual void releaseAudioSessionId(audio_session_t audioSession, pid_t pid); 209 210 virtual status_t queryNumberEffects(uint32_t *numEffects) const; 211 212 virtual status_t queryEffect(uint32_t index, effect_descriptor_t *descriptor) const; 213 214 virtual status_t getEffectDescriptor(const effect_uuid_t *pUuid, 215 effect_descriptor_t *descriptor) const; 216 217 virtual sp<IEffect> createEffect( 218 effect_descriptor_t *pDesc, 219 const sp<IEffectClient>& effectClient, 220 int32_t priority, 221 audio_io_handle_t io, 222 audio_session_t sessionId, 223 const String16& opPackageName, 224 status_t *status /*non-NULL*/, 225 int *id, 226 int *enabled); 227 228 virtual status_t moveEffects(audio_session_t sessionId, audio_io_handle_t srcOutput, 229 audio_io_handle_t dstOutput); 230 231 virtual audio_module_handle_t loadHwModule(const char *name); 232 233 virtual uint32_t getPrimaryOutputSamplingRate(); 234 virtual size_t getPrimaryOutputFrameCount(); 235 236 virtual status_t setLowRamDevice(bool isLowRamDevice); 237 238 /* List available audio ports and their attributes */ 239 virtual status_t listAudioPorts(unsigned int *num_ports, 240 struct audio_port *ports); 241 242 /* Get attributes for a given audio port */ 243 virtual status_t getAudioPort(struct audio_port *port); 244 245 /* Create an audio patch between several source and sink ports */ 246 virtual status_t createAudioPatch(const struct audio_patch *patch, 247 audio_patch_handle_t *handle); 248 249 /* Release an audio patch */ 250 virtual status_t releaseAudioPatch(audio_patch_handle_t handle); 251 252 /* List existing audio patches */ 253 virtual status_t listAudioPatches(unsigned int *num_patches, 254 struct audio_patch *patches); 255 256 /* Set audio port configuration */ 257 virtual status_t setAudioPortConfig(const struct audio_port_config *config); 258 259 /* Get the HW synchronization source used for an audio session */ 260 virtual audio_hw_sync_t getAudioHwSyncForSession(audio_session_t sessionId); 261 262 /* Indicate JAVA services are ready (scheduling, power management ...) */ 263 virtual status_t systemReady(); 264 265 virtual status_t onTransact( 266 uint32_t code, 267 const Parcel& data, 268 Parcel* reply, 269 uint32_t flags); 270 271 // end of IAudioFlinger interface 272 273 sp<NBLog::Writer> newWriter_l(size_t size, const char *name); 274 void unregisterWriter(const sp<NBLog::Writer>& writer); 275private: 276 static const size_t kLogMemorySize = 40 * 1024; 277 sp<MemoryDealer> mLogMemoryDealer; // == 0 when NBLog is disabled 278 // When a log writer is unregistered, it is done lazily so that media.log can continue to see it 279 // for as long as possible. The memory is only freed when it is needed for another log writer. 280 Vector< sp<NBLog::Writer> > mUnregisteredWriters; 281 Mutex mUnregisteredWritersLock; 282public: 283 284 class SyncEvent; 285 286 typedef void (*sync_event_callback_t)(const wp<SyncEvent>& event) ; 287 288 class SyncEvent : public RefBase { 289 public: 290 SyncEvent(AudioSystem::sync_event_t type, 291 audio_session_t triggerSession, 292 audio_session_t listenerSession, 293 sync_event_callback_t callBack, 294 wp<RefBase> cookie) 295 : mType(type), mTriggerSession(triggerSession), mListenerSession(listenerSession), 296 mCallback(callBack), mCookie(cookie) 297 {} 298 299 virtual ~SyncEvent() {} 300 301 void trigger() { Mutex::Autolock _l(mLock); if (mCallback) mCallback(this); } 302 bool isCancelled() const { Mutex::Autolock _l(mLock); return (mCallback == NULL); } 303 void cancel() { Mutex::Autolock _l(mLock); mCallback = NULL; } 304 AudioSystem::sync_event_t type() const { return mType; } 305 audio_session_t triggerSession() const { return mTriggerSession; } 306 audio_session_t listenerSession() const { return mListenerSession; } 307 wp<RefBase> cookie() const { return mCookie; } 308 309 private: 310 const AudioSystem::sync_event_t mType; 311 const audio_session_t mTriggerSession; 312 const audio_session_t mListenerSession; 313 sync_event_callback_t mCallback; 314 const wp<RefBase> mCookie; 315 mutable Mutex mLock; 316 }; 317 318 sp<SyncEvent> createSyncEvent(AudioSystem::sync_event_t type, 319 audio_session_t triggerSession, 320 audio_session_t listenerSession, 321 sync_event_callback_t callBack, 322 wp<RefBase> cookie); 323 324private: 325 326 audio_mode_t getMode() const { return mMode; } 327 328 bool btNrecIsOff() const { return mBtNrecIsOff; } 329 330 AudioFlinger() ANDROID_API; 331 virtual ~AudioFlinger(); 332 333 // call in any IAudioFlinger method that accesses mPrimaryHardwareDev 334 status_t initCheck() const { return mPrimaryHardwareDev == NULL ? 335 NO_INIT : NO_ERROR; } 336 337 // RefBase 338 virtual void onFirstRef(); 339 340 AudioHwDevice* findSuitableHwDev_l(audio_module_handle_t module, 341 audio_devices_t devices); 342 void purgeStaleEffects_l(); 343 344 // Set kEnableExtendedChannels to true to enable greater than stereo output 345 // for the MixerThread and device sink. Number of channels allowed is 346 // FCC_2 <= channels <= AudioMixer::MAX_NUM_CHANNELS. 347 static const bool kEnableExtendedChannels = true; 348 349 // Returns true if channel mask is permitted for the PCM sink in the MixerThread 350 static inline bool isValidPcmSinkChannelMask(audio_channel_mask_t channelMask) { 351 switch (audio_channel_mask_get_representation(channelMask)) { 352 case AUDIO_CHANNEL_REPRESENTATION_POSITION: { 353 uint32_t channelCount = FCC_2; // stereo is default 354 if (kEnableExtendedChannels) { 355 channelCount = audio_channel_count_from_out_mask(channelMask); 356 if (channelCount < FCC_2 // mono is not supported at this time 357 || channelCount > AudioMixer::MAX_NUM_CHANNELS) { 358 return false; 359 } 360 } 361 // check that channelMask is the "canonical" one we expect for the channelCount. 362 return channelMask == audio_channel_out_mask_from_count(channelCount); 363 } 364 case AUDIO_CHANNEL_REPRESENTATION_INDEX: 365 if (kEnableExtendedChannels) { 366 const uint32_t channelCount = audio_channel_count_from_out_mask(channelMask); 367 if (channelCount >= FCC_2 // mono is not supported at this time 368 && channelCount <= AudioMixer::MAX_NUM_CHANNELS) { 369 return true; 370 } 371 } 372 return false; 373 default: 374 return false; 375 } 376 } 377 378 // Set kEnableExtendedPrecision to true to use extended precision in MixerThread 379 static const bool kEnableExtendedPrecision = true; 380 381 // Returns true if format is permitted for the PCM sink in the MixerThread 382 static inline bool isValidPcmSinkFormat(audio_format_t format) { 383 switch (format) { 384 case AUDIO_FORMAT_PCM_16_BIT: 385 return true; 386 case AUDIO_FORMAT_PCM_FLOAT: 387 case AUDIO_FORMAT_PCM_24_BIT_PACKED: 388 case AUDIO_FORMAT_PCM_32_BIT: 389 case AUDIO_FORMAT_PCM_8_24_BIT: 390 return kEnableExtendedPrecision; 391 default: 392 return false; 393 } 394 } 395 396 // standby delay for MIXER and DUPLICATING playback threads is read from property 397 // ro.audio.flinger_standbytime_ms or defaults to kDefaultStandbyTimeInNsecs 398 static nsecs_t mStandbyTimeInNsecs; 399 400 // incremented by 2 when screen state changes, bit 0 == 1 means "off" 401 // AudioFlinger::setParameters() updates, other threads read w/o lock 402 static uint32_t mScreenState; 403 404 // Internal dump utilities. 405 static const int kDumpLockRetries = 50; 406 static const int kDumpLockSleepUs = 20000; 407 static bool dumpTryLock(Mutex& mutex); 408 void dumpPermissionDenial(int fd, const Vector<String16>& args); 409 void dumpClients(int fd, const Vector<String16>& args); 410 void dumpInternals(int fd, const Vector<String16>& args); 411 412 // --- Client --- 413 class Client : public RefBase { 414 public: 415 Client(const sp<AudioFlinger>& audioFlinger, pid_t pid); 416 virtual ~Client(); 417 sp<MemoryDealer> heap() const; 418 pid_t pid() const { return mPid; } 419 sp<AudioFlinger> audioFlinger() const { return mAudioFlinger; } 420 421 private: 422 Client(const Client&); 423 Client& operator = (const Client&); 424 const sp<AudioFlinger> mAudioFlinger; 425 sp<MemoryDealer> mMemoryDealer; 426 const pid_t mPid; 427 }; 428 429 // --- Notification Client --- 430 class NotificationClient : public IBinder::DeathRecipient { 431 public: 432 NotificationClient(const sp<AudioFlinger>& audioFlinger, 433 const sp<IAudioFlingerClient>& client, 434 pid_t pid); 435 virtual ~NotificationClient(); 436 437 sp<IAudioFlingerClient> audioFlingerClient() const { return mAudioFlingerClient; } 438 439 // IBinder::DeathRecipient 440 virtual void binderDied(const wp<IBinder>& who); 441 442 private: 443 NotificationClient(const NotificationClient&); 444 NotificationClient& operator = (const NotificationClient&); 445 446 const sp<AudioFlinger> mAudioFlinger; 447 const pid_t mPid; 448 const sp<IAudioFlingerClient> mAudioFlingerClient; 449 }; 450 451 class TrackHandle; 452 class RecordHandle; 453 class RecordThread; 454 class PlaybackThread; 455 class MixerThread; 456 class DirectOutputThread; 457 class OffloadThread; 458 class DuplicatingThread; 459 class AsyncCallbackThread; 460 class Track; 461 class RecordTrack; 462 class EffectModule; 463 class EffectHandle; 464 class EffectChain; 465 466 struct AudioStreamIn; 467 468 struct stream_type_t { 469 stream_type_t() 470 : volume(1.0f), 471 mute(false) 472 { 473 } 474 float volume; 475 bool mute; 476 }; 477 478 // --- PlaybackThread --- 479 480#include "Threads.h" 481 482#include "Effects.h" 483 484#include "PatchPanel.h" 485 486 // server side of the client's IAudioTrack 487 class TrackHandle : public android::BnAudioTrack { 488 public: 489 TrackHandle(const sp<PlaybackThread::Track>& track); 490 virtual ~TrackHandle(); 491 virtual sp<IMemory> getCblk() const; 492 virtual status_t start(); 493 virtual void stop(); 494 virtual void flush(); 495 virtual void pause(); 496 virtual status_t attachAuxEffect(int effectId); 497 virtual status_t setParameters(const String8& keyValuePairs); 498 virtual status_t getTimestamp(AudioTimestamp& timestamp); 499 virtual void signal(); // signal playback thread for a change in control block 500 501 virtual status_t onTransact( 502 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 503 504 private: 505 const sp<PlaybackThread::Track> mTrack; 506 }; 507 508 // server side of the client's IAudioRecord 509 class RecordHandle : public android::BnAudioRecord { 510 public: 511 RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack); 512 virtual ~RecordHandle(); 513 virtual status_t start(int /*AudioSystem::sync_event_t*/ event, 514 audio_session_t triggerSession); 515 virtual void stop(); 516 virtual status_t onTransact( 517 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 518 private: 519 const sp<RecordThread::RecordTrack> mRecordTrack; 520 521 // for use from destructor 522 void stop_nonvirtual(); 523 }; 524 525 526 ThreadBase *checkThread_l(audio_io_handle_t ioHandle) const; 527 PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const; 528 MixerThread *checkMixerThread_l(audio_io_handle_t output) const; 529 RecordThread *checkRecordThread_l(audio_io_handle_t input) const; 530 sp<RecordThread> openInput_l(audio_module_handle_t module, 531 audio_io_handle_t *input, 532 audio_config_t *config, 533 audio_devices_t device, 534 const String8& address, 535 audio_source_t source, 536 audio_input_flags_t flags); 537 sp<PlaybackThread> openOutput_l(audio_module_handle_t module, 538 audio_io_handle_t *output, 539 audio_config_t *config, 540 audio_devices_t devices, 541 const String8& address, 542 audio_output_flags_t flags); 543 544 void closeOutputFinish(sp<PlaybackThread> thread); 545 void closeInputFinish(sp<RecordThread> thread); 546 547 // no range check, AudioFlinger::mLock held 548 bool streamMute_l(audio_stream_type_t stream) const 549 { return mStreamTypes[stream].mute; } 550 // no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held 551 float streamVolume_l(audio_stream_type_t stream) const 552 { return mStreamTypes[stream].volume; } 553 void ioConfigChanged(audio_io_config_event event, 554 const sp<AudioIoDescriptor>& ioDesc, 555 pid_t pid = 0); 556 557 // Allocate an audio_unique_id_t. 558 // Specific types are audio_io_handle_t, audio_session_t, effect ID (int), 559 // audio_module_handle_t, and audio_patch_handle_t. 560 // They all share the same ID space, but the namespaces are actually independent 561 // because there are separate KeyedVectors for each kind of ID. 562 // The return value is cast to the specific type depending on how the ID will be used. 563 // FIXME This API does not handle rollover to zero (for unsigned IDs), 564 // or from positive to negative (for signed IDs). 565 // Thus it may fail by returning an ID of the wrong sign, 566 // or by returning a non-unique ID. 567 // This is the internal API. For the binder API see newAudioUniqueId(). 568 audio_unique_id_t nextUniqueId(audio_unique_id_use_t use); 569 570 status_t moveEffectChain_l(audio_session_t sessionId, 571 PlaybackThread *srcThread, 572 PlaybackThread *dstThread, 573 bool reRegister); 574 575 // return thread associated with primary hardware device, or NULL 576 PlaybackThread *primaryPlaybackThread_l() const; 577 audio_devices_t primaryOutputDevice_l() const; 578 579 // return the playback thread with smallest HAL buffer size, and prefer fast 580 PlaybackThread *fastPlaybackThread_l() const; 581 582 sp<PlaybackThread> getEffectThread_l(audio_session_t sessionId, int EffectId); 583 584 585 void removeClient_l(pid_t pid); 586 void removeNotificationClient(pid_t pid); 587 bool isNonOffloadableGlobalEffectEnabled_l(); 588 void onNonOffloadableGlobalEffectEnable(); 589 590 // Store an effect chain to mOrphanEffectChains keyed vector. 591 // Called when a thread exits and effects are still attached to it. 592 // If effects are later created on the same session, they will reuse the same 593 // effect chain and same instances in the effect library. 594 // return ALREADY_EXISTS if a chain with the same session already exists in 595 // mOrphanEffectChains. Note that this should never happen as there is only one 596 // chain for a given session and it is attached to only one thread at a time. 597 status_t putOrphanEffectChain_l(const sp<EffectChain>& chain); 598 // Get an effect chain for the specified session in mOrphanEffectChains and remove 599 // it if found. Returns 0 if not found (this is the most common case). 600 sp<EffectChain> getOrphanEffectChain_l(audio_session_t session); 601 // Called when the last effect handle on an effect instance is removed. If this 602 // effect belongs to an effect chain in mOrphanEffectChains, the chain is updated 603 // and removed from mOrphanEffectChains if it does not contain any effect. 604 // Return true if the effect was found in mOrphanEffectChains, false otherwise. 605 bool updateOrphanEffectChains(const sp<EffectModule>& effect); 606 607 void broacastParametersToRecordThreads_l(const String8& keyValuePairs); 608 609 // AudioStreamIn is immutable, so their fields are const. 610 // For emphasis, we could also make all pointers to them be "const *", 611 // but that would clutter the code unnecessarily. 612 613 struct AudioStreamIn { 614 AudioHwDevice* const audioHwDev; 615 audio_stream_in_t* const stream; 616 audio_input_flags_t flags; 617 618 audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); } 619 620 AudioStreamIn(AudioHwDevice *dev, audio_stream_in_t *in, audio_input_flags_t flags) : 621 audioHwDev(dev), stream(in), flags(flags) {} 622 }; 623 624 // for mAudioSessionRefs only 625 struct AudioSessionRef { 626 AudioSessionRef(audio_session_t sessionid, pid_t pid) : 627 mSessionid(sessionid), mPid(pid), mCnt(1) {} 628 const audio_session_t mSessionid; 629 const pid_t mPid; 630 int mCnt; 631 }; 632 633 mutable Mutex mLock; 634 sp<LockWatch> mLockWatch; 635 // protects mClients and mNotificationClients. 636 // must be locked after mLock and ThreadBase::mLock if both must be locked 637 // avoids acquiring AudioFlinger::mLock from inside thread loop. 638 mutable Mutex mClientLock; 639 // protected by mClientLock 640 DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client() 641 642 mutable Mutex mHardwareLock; 643 // NOTE: If both mLock and mHardwareLock mutexes must be held, 644 // always take mLock before mHardwareLock 645 646 // These two fields are immutable after onFirstRef(), so no lock needed to access 647 AudioHwDevice* mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL 648 DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*> mAudioHwDevs; 649 650 // for dump, indicates which hardware operation is currently in progress (but not stream ops) 651 enum hardware_call_state { 652 AUDIO_HW_IDLE = 0, // no operation in progress 653 AUDIO_HW_INIT, // init_check 654 AUDIO_HW_OUTPUT_OPEN, // open_output_stream 655 AUDIO_HW_OUTPUT_CLOSE, // unused 656 AUDIO_HW_INPUT_OPEN, // unused 657 AUDIO_HW_INPUT_CLOSE, // unused 658 AUDIO_HW_STANDBY, // unused 659 AUDIO_HW_SET_MASTER_VOLUME, // set_master_volume 660 AUDIO_HW_GET_ROUTING, // unused 661 AUDIO_HW_SET_ROUTING, // unused 662 AUDIO_HW_GET_MODE, // unused 663 AUDIO_HW_SET_MODE, // set_mode 664 AUDIO_HW_GET_MIC_MUTE, // get_mic_mute 665 AUDIO_HW_SET_MIC_MUTE, // set_mic_mute 666 AUDIO_HW_SET_VOICE_VOLUME, // set_voice_volume 667 AUDIO_HW_SET_PARAMETER, // set_parameters 668 AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size 669 AUDIO_HW_GET_MASTER_VOLUME, // get_master_volume 670 AUDIO_HW_GET_PARAMETER, // get_parameters 671 AUDIO_HW_SET_MASTER_MUTE, // set_master_mute 672 AUDIO_HW_GET_MASTER_MUTE, // get_master_mute 673 }; 674 675 mutable hardware_call_state mHardwareStatus; // for dump only 676 677 678 DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads; 679 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 680 681 // member variables below are protected by mLock 682 float mMasterVolume; 683 bool mMasterMute; 684 // end of variables protected by mLock 685 686 DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads; 687 688 // protected by mClientLock 689 DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients; 690 691 // updated by atomic_fetch_add_explicit 692 volatile atomic_uint_fast32_t mNextUniqueIds[AUDIO_UNIQUE_ID_USE_MAX]; 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