AudioFlinger.h revision 886deb506ea2938cfec40fc0dd2bff072850386b
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 <deque> 23#include <map> 24#include <stdint.h> 25#include <sys/types.h> 26#include <limits.h> 27 28#include <cutils/compiler.h> 29#include <cutils/properties.h> 30 31#include <media/IAudioFlinger.h> 32#include <media/IAudioFlingerClient.h> 33#include <media/IAudioTrack.h> 34#include <media/IAudioRecord.h> 35#include <media/AudioSystem.h> 36#include <media/AudioTrack.h> 37#include <media/MmapStreamInterface.h> 38#include <media/MmapStreamCallback.h> 39 40#include <utils/Atomic.h> 41#include <utils/Errors.h> 42#include <utils/threads.h> 43#include <utils/SortedVector.h> 44#include <utils/TypeHelpers.h> 45#include <utils/Vector.h> 46 47#include <binder/BinderService.h> 48#include <binder/MemoryDealer.h> 49 50#include <system/audio.h> 51#include <system/audio_policy.h> 52 53#include <media/audiohal/EffectBufferHalInterface.h> 54#include <media/audiohal/StreamHalInterface.h> 55#include <media/AudioBufferProvider.h> 56#include <media/AudioMixer.h> 57#include <media/ExtendedAudioBufferProvider.h> 58#include <media/LinearMap.h> 59#include <media/VolumeShaper.h> 60 61#include "FastCapture.h" 62#include "FastMixer.h" 63#include <media/nbaio/NBAIO.h> 64#include "AudioWatchdog.h" 65#include "AudioStreamOut.h" 66#include "SpdifStreamOut.h" 67#include "AudioHwDevice.h" 68 69#include <powermanager/IPowerManager.h> 70 71#include <media/nbaio/NBLog.h> 72#include <private/media/AudioTrackShared.h> 73 74namespace android { 75 76struct audio_track_cblk_t; 77struct effect_param_cblk_t; 78class AudioMixer; 79class AudioBuffer; 80class AudioResampler; 81class DeviceHalInterface; 82class DevicesFactoryHalInterface; 83class EffectsFactoryHalInterface; 84class FastMixer; 85class PassthruBufferProvider; 86class RecordBufferConverter; 87class ServerProxy; 88 89// ---------------------------------------------------------------------------- 90 91static const nsecs_t kDefaultStandbyTimeInNsecs = seconds(3); 92 93 94// Max shared memory size for audio tracks and audio records per client process 95static const size_t kClientSharedHeapSizeBytes = 1024*1024; 96// Shared memory size multiplier for non low ram devices 97static const size_t kClientSharedHeapSizeMultiplier = 4; 98 99#define INCLUDING_FROM_AUDIOFLINGER_H 100 101class AudioFlinger : 102 public BinderService<AudioFlinger>, 103 public BnAudioFlinger 104{ 105 friend class BinderService<AudioFlinger>; // for AudioFlinger() 106 107public: 108 static const char* getServiceName() ANDROID_API { return "media.audio_flinger"; } 109 110 virtual status_t dump(int fd, const Vector<String16>& args); 111 112 // IAudioFlinger interface, in binder opcode order 113 virtual sp<IAudioTrack> createTrack( 114 audio_stream_type_t streamType, 115 uint32_t sampleRate, 116 audio_format_t format, 117 audio_channel_mask_t channelMask, 118 size_t *pFrameCount, 119 audio_output_flags_t *flags, 120 const sp<IMemory>& sharedBuffer, 121 audio_io_handle_t output, 122 pid_t pid, 123 pid_t tid, 124 audio_session_t *sessionId, 125 int clientUid, 126 status_t *status /*non-NULL*/, 127 audio_port_handle_t portId); 128 129 virtual sp<IAudioRecord> openRecord( 130 audio_io_handle_t input, 131 uint32_t sampleRate, 132 audio_format_t format, 133 audio_channel_mask_t channelMask, 134 const String16& opPackageName, 135 size_t *pFrameCount, 136 audio_input_flags_t *flags, 137 pid_t pid, 138 pid_t tid, 139 int clientUid, 140 audio_session_t *sessionId, 141 size_t *notificationFrames, 142 sp<IMemory>& cblk, 143 sp<IMemory>& buffers, 144 status_t *status /*non-NULL*/, 145 audio_port_handle_t portId); 146 147 virtual uint32_t sampleRate(audio_io_handle_t ioHandle) const; 148 virtual audio_format_t format(audio_io_handle_t output) const; 149 virtual size_t frameCount(audio_io_handle_t ioHandle) const; 150 virtual size_t frameCountHAL(audio_io_handle_t ioHandle) const; 151 virtual uint32_t latency(audio_io_handle_t output) const; 152 153 virtual status_t setMasterVolume(float value); 154 virtual status_t setMasterMute(bool muted); 155 156 virtual float masterVolume() const; 157 virtual bool masterMute() const; 158 159 virtual status_t setStreamVolume(audio_stream_type_t stream, float value, 160 audio_io_handle_t output); 161 virtual status_t setStreamMute(audio_stream_type_t stream, bool muted); 162 163 virtual float streamVolume(audio_stream_type_t stream, 164 audio_io_handle_t output) const; 165 virtual bool streamMute(audio_stream_type_t stream) const; 166 167 virtual status_t setMode(audio_mode_t mode); 168 169 virtual status_t setMicMute(bool state); 170 virtual bool getMicMute() const; 171 172 virtual status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs); 173 virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) const; 174 175 virtual void registerClient(const sp<IAudioFlingerClient>& client); 176 177 virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, 178 audio_channel_mask_t channelMask) const; 179 180 virtual status_t openOutput(audio_module_handle_t module, 181 audio_io_handle_t *output, 182 audio_config_t *config, 183 audio_devices_t *devices, 184 const String8& address, 185 uint32_t *latencyMs, 186 audio_output_flags_t flags); 187 188 virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1, 189 audio_io_handle_t output2); 190 191 virtual status_t closeOutput(audio_io_handle_t output); 192 193 virtual status_t suspendOutput(audio_io_handle_t output); 194 195 virtual status_t restoreOutput(audio_io_handle_t output); 196 197 virtual status_t openInput(audio_module_handle_t module, 198 audio_io_handle_t *input, 199 audio_config_t *config, 200 audio_devices_t *device, 201 const String8& address, 202 audio_source_t source, 203 audio_input_flags_t flags); 204 205 virtual status_t closeInput(audio_io_handle_t input); 206 207 virtual status_t invalidateStream(audio_stream_type_t stream); 208 209 virtual status_t setVoiceVolume(float volume); 210 211 virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 212 audio_io_handle_t output) const; 213 214 virtual uint32_t getInputFramesLost(audio_io_handle_t ioHandle) const; 215 216 // This is the binder API. For the internal API see nextUniqueId(). 217 virtual audio_unique_id_t newAudioUniqueId(audio_unique_id_use_t use); 218 219 virtual void acquireAudioSessionId(audio_session_t audioSession, pid_t pid); 220 221 virtual void releaseAudioSessionId(audio_session_t audioSession, pid_t pid); 222 223 virtual status_t queryNumberEffects(uint32_t *numEffects) const; 224 225 virtual status_t queryEffect(uint32_t index, effect_descriptor_t *descriptor) const; 226 227 virtual status_t getEffectDescriptor(const effect_uuid_t *pUuid, 228 effect_descriptor_t *descriptor) const; 229 230 virtual sp<IEffect> createEffect( 231 effect_descriptor_t *pDesc, 232 const sp<IEffectClient>& effectClient, 233 int32_t priority, 234 audio_io_handle_t io, 235 audio_session_t sessionId, 236 const String16& opPackageName, 237 pid_t pid, 238 status_t *status /*non-NULL*/, 239 int *id, 240 int *enabled); 241 242 virtual status_t moveEffects(audio_session_t sessionId, audio_io_handle_t srcOutput, 243 audio_io_handle_t dstOutput); 244 245 virtual audio_module_handle_t loadHwModule(const char *name); 246 247 virtual uint32_t getPrimaryOutputSamplingRate(); 248 virtual size_t getPrimaryOutputFrameCount(); 249 250 virtual status_t setLowRamDevice(bool isLowRamDevice); 251 252 /* List available audio ports and their attributes */ 253 virtual status_t listAudioPorts(unsigned int *num_ports, 254 struct audio_port *ports); 255 256 /* Get attributes for a given audio port */ 257 virtual status_t getAudioPort(struct audio_port *port); 258 259 /* Create an audio patch between several source and sink ports */ 260 virtual status_t createAudioPatch(const struct audio_patch *patch, 261 audio_patch_handle_t *handle); 262 263 /* Release an audio patch */ 264 virtual status_t releaseAudioPatch(audio_patch_handle_t handle); 265 266 /* List existing audio patches */ 267 virtual status_t listAudioPatches(unsigned int *num_patches, 268 struct audio_patch *patches); 269 270 /* Set audio port configuration */ 271 virtual status_t setAudioPortConfig(const struct audio_port_config *config); 272 273 /* Get the HW synchronization source used for an audio session */ 274 virtual audio_hw_sync_t getAudioHwSyncForSession(audio_session_t sessionId); 275 276 /* Indicate JAVA services are ready (scheduling, power management ...) */ 277 virtual status_t systemReady(); 278 279 virtual status_t onTransact( 280 uint32_t code, 281 const Parcel& data, 282 Parcel* reply, 283 uint32_t flags); 284 285 // end of IAudioFlinger interface 286 287 sp<NBLog::Writer> newWriter_l(size_t size, const char *name); 288 void unregisterWriter(const sp<NBLog::Writer>& writer); 289 sp<EffectsFactoryHalInterface> getEffectsFactory(); 290 291 status_t openMmapStream(MmapStreamInterface::stream_direction_t direction, 292 const audio_attributes_t *attr, 293 audio_config_base_t *config, 294 const MmapStreamInterface::Client& client, 295 audio_port_handle_t *deviceId, 296 const sp<MmapStreamCallback>& callback, 297 sp<MmapStreamInterface>& interface); 298private: 299 static const size_t kLogMemorySize = 40 * 1024; 300 sp<MemoryDealer> mLogMemoryDealer; // == 0 when NBLog is disabled 301 // When a log writer is unregistered, it is done lazily so that media.log can continue to see it 302 // for as long as possible. The memory is only freed when it is needed for another log writer. 303 Vector< sp<NBLog::Writer> > mUnregisteredWriters; 304 Mutex mUnregisteredWritersLock; 305 306public: 307 308 class SyncEvent; 309 310 typedef void (*sync_event_callback_t)(const wp<SyncEvent>& event) ; 311 312 class SyncEvent : public RefBase { 313 public: 314 SyncEvent(AudioSystem::sync_event_t type, 315 audio_session_t triggerSession, 316 audio_session_t listenerSession, 317 sync_event_callback_t callBack, 318 wp<RefBase> cookie) 319 : mType(type), mTriggerSession(triggerSession), mListenerSession(listenerSession), 320 mCallback(callBack), mCookie(cookie) 321 {} 322 323 virtual ~SyncEvent() {} 324 325 void trigger() { Mutex::Autolock _l(mLock); if (mCallback) mCallback(this); } 326 bool isCancelled() const { Mutex::Autolock _l(mLock); return (mCallback == NULL); } 327 void cancel() { Mutex::Autolock _l(mLock); mCallback = NULL; } 328 AudioSystem::sync_event_t type() const { return mType; } 329 audio_session_t triggerSession() const { return mTriggerSession; } 330 audio_session_t listenerSession() const { return mListenerSession; } 331 wp<RefBase> cookie() const { return mCookie; } 332 333 private: 334 const AudioSystem::sync_event_t mType; 335 const audio_session_t mTriggerSession; 336 const audio_session_t mListenerSession; 337 sync_event_callback_t mCallback; 338 const wp<RefBase> mCookie; 339 mutable Mutex mLock; 340 }; 341 342 sp<SyncEvent> createSyncEvent(AudioSystem::sync_event_t type, 343 audio_session_t triggerSession, 344 audio_session_t listenerSession, 345 sync_event_callback_t callBack, 346 const wp<RefBase>& cookie); 347 348private: 349 350 audio_mode_t getMode() const { return mMode; } 351 352 bool btNrecIsOff() const { return mBtNrecIsOff; } 353 354 AudioFlinger() ANDROID_API; 355 virtual ~AudioFlinger(); 356 357 // call in any IAudioFlinger method that accesses mPrimaryHardwareDev 358 status_t initCheck() const { return mPrimaryHardwareDev == NULL ? 359 NO_INIT : NO_ERROR; } 360 361 // RefBase 362 virtual void onFirstRef(); 363 364 AudioHwDevice* findSuitableHwDev_l(audio_module_handle_t module, 365 audio_devices_t devices); 366 void purgeStaleEffects_l(); 367 368 // Set kEnableExtendedChannels to true to enable greater than stereo output 369 // for the MixerThread and device sink. Number of channels allowed is 370 // FCC_2 <= channels <= AudioMixer::MAX_NUM_CHANNELS. 371 static const bool kEnableExtendedChannels = true; 372 373 // Returns true if channel mask is permitted for the PCM sink in the MixerThread 374 static inline bool isValidPcmSinkChannelMask(audio_channel_mask_t channelMask) { 375 switch (audio_channel_mask_get_representation(channelMask)) { 376 case AUDIO_CHANNEL_REPRESENTATION_POSITION: { 377 uint32_t channelCount = FCC_2; // stereo is default 378 if (kEnableExtendedChannels) { 379 channelCount = audio_channel_count_from_out_mask(channelMask); 380 if (channelCount < FCC_2 // mono is not supported at this time 381 || channelCount > AudioMixer::MAX_NUM_CHANNELS) { 382 return false; 383 } 384 } 385 // check that channelMask is the "canonical" one we expect for the channelCount. 386 return channelMask == audio_channel_out_mask_from_count(channelCount); 387 } 388 case AUDIO_CHANNEL_REPRESENTATION_INDEX: 389 if (kEnableExtendedChannels) { 390 const uint32_t channelCount = audio_channel_count_from_out_mask(channelMask); 391 if (channelCount >= FCC_2 // mono is not supported at this time 392 && channelCount <= AudioMixer::MAX_NUM_CHANNELS) { 393 return true; 394 } 395 } 396 return false; 397 default: 398 return false; 399 } 400 } 401 402 // Set kEnableExtendedPrecision to true to use extended precision in MixerThread 403 static const bool kEnableExtendedPrecision = true; 404 405 // Returns true if format is permitted for the PCM sink in the MixerThread 406 static inline bool isValidPcmSinkFormat(audio_format_t format) { 407 switch (format) { 408 case AUDIO_FORMAT_PCM_16_BIT: 409 return true; 410 case AUDIO_FORMAT_PCM_FLOAT: 411 case AUDIO_FORMAT_PCM_24_BIT_PACKED: 412 case AUDIO_FORMAT_PCM_32_BIT: 413 case AUDIO_FORMAT_PCM_8_24_BIT: 414 return kEnableExtendedPrecision; 415 default: 416 return false; 417 } 418 } 419 420 // standby delay for MIXER and DUPLICATING playback threads is read from property 421 // ro.audio.flinger_standbytime_ms or defaults to kDefaultStandbyTimeInNsecs 422 static nsecs_t mStandbyTimeInNsecs; 423 424 // incremented by 2 when screen state changes, bit 0 == 1 means "off" 425 // AudioFlinger::setParameters() updates, other threads read w/o lock 426 static uint32_t mScreenState; 427 428 // Internal dump utilities. 429 static const int kDumpLockRetries = 50; 430 static const int kDumpLockSleepUs = 20000; 431 static bool dumpTryLock(Mutex& mutex); 432 void dumpPermissionDenial(int fd, const Vector<String16>& args); 433 void dumpClients(int fd, const Vector<String16>& args); 434 void dumpInternals(int fd, const Vector<String16>& args); 435 436 // --- Client --- 437 class Client : public RefBase { 438 public: 439 Client(const sp<AudioFlinger>& audioFlinger, pid_t pid); 440 virtual ~Client(); 441 sp<MemoryDealer> heap() const; 442 pid_t pid() const { return mPid; } 443 sp<AudioFlinger> audioFlinger() const { return mAudioFlinger; } 444 445 private: 446 Client(const Client&); 447 Client& operator = (const Client&); 448 const sp<AudioFlinger> mAudioFlinger; 449 sp<MemoryDealer> mMemoryDealer; 450 const pid_t mPid; 451 }; 452 453 // --- Notification Client --- 454 class NotificationClient : public IBinder::DeathRecipient { 455 public: 456 NotificationClient(const sp<AudioFlinger>& audioFlinger, 457 const sp<IAudioFlingerClient>& client, 458 pid_t pid); 459 virtual ~NotificationClient(); 460 461 sp<IAudioFlingerClient> audioFlingerClient() const { return mAudioFlingerClient; } 462 463 // IBinder::DeathRecipient 464 virtual void binderDied(const wp<IBinder>& who); 465 466 private: 467 NotificationClient(const NotificationClient&); 468 NotificationClient& operator = (const NotificationClient&); 469 470 const sp<AudioFlinger> mAudioFlinger; 471 const pid_t mPid; 472 const sp<IAudioFlingerClient> mAudioFlingerClient; 473 }; 474 475 // --- MediaLogNotifier --- 476 // Thread in charge of notifying MediaLogService to start merging. 477 // Receives requests from AudioFlinger's binder activity. It is used to reduce the amount of 478 // binder calls to MediaLogService in case of bursts of AudioFlinger binder calls. 479 class MediaLogNotifier : public Thread { 480 public: 481 MediaLogNotifier(); 482 483 // Requests a MediaLogService notification. It's ignored if there has recently been another 484 void requestMerge(); 485 private: 486 // Every iteration blocks waiting for a request, then interacts with MediaLogService to 487 // start merging. 488 // As every MediaLogService binder call is expensive, once it gets a request it ignores the 489 // following ones for a period of time. 490 virtual bool threadLoop() override; 491 492 bool mPendingRequests; 493 494 // Mutex and condition variable around mPendingRequests' value 495 Mutex mMutex; 496 Condition mCond; 497 498 // Duration of the sleep period after a processed request 499 static const int kPostTriggerSleepPeriod = 1000000; 500 }; 501 502 const sp<MediaLogNotifier> mMediaLogNotifier; 503 504 // This is a helper that is called during incoming binder calls. 505 void requestLogMerge(); 506 507 class TrackHandle; 508 class RecordHandle; 509 class RecordThread; 510 class PlaybackThread; 511 class MixerThread; 512 class DirectOutputThread; 513 class OffloadThread; 514 class DuplicatingThread; 515 class AsyncCallbackThread; 516 class Track; 517 class RecordTrack; 518 class EffectModule; 519 class EffectHandle; 520 class EffectChain; 521 522 struct AudioStreamIn; 523 524 struct stream_type_t { 525 stream_type_t() 526 : volume(1.0f), 527 mute(false) 528 { 529 } 530 float volume; 531 bool mute; 532 }; 533 534 // --- PlaybackThread --- 535 536#include "Threads.h" 537 538#include "Effects.h" 539 540#include "PatchPanel.h" 541 542 // server side of the client's IAudioTrack 543 class TrackHandle : public android::BnAudioTrack { 544 public: 545 explicit TrackHandle(const sp<PlaybackThread::Track>& track); 546 virtual ~TrackHandle(); 547 virtual sp<IMemory> getCblk() const; 548 virtual status_t start(); 549 virtual void stop(); 550 virtual void flush(); 551 virtual void pause(); 552 virtual status_t attachAuxEffect(int effectId); 553 virtual status_t setParameters(const String8& keyValuePairs); 554 virtual VolumeShaper::Status applyVolumeShaper( 555 const sp<VolumeShaper::Configuration>& configuration, 556 const sp<VolumeShaper::Operation>& operation) override; 557 virtual sp<VolumeShaper::State> getVolumeShaperState(int id) override; 558 virtual status_t getTimestamp(AudioTimestamp& timestamp); 559 virtual void signal(); // signal playback thread for a change in control block 560 561 virtual status_t onTransact( 562 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 563 564 private: 565 const sp<PlaybackThread::Track> mTrack; 566 }; 567 568 // server side of the client's IAudioRecord 569 class RecordHandle : public android::BnAudioRecord { 570 public: 571 explicit RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack); 572 virtual ~RecordHandle(); 573 virtual status_t start(int /*AudioSystem::sync_event_t*/ event, 574 audio_session_t triggerSession); 575 virtual void stop(); 576 virtual status_t onTransact( 577 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 578 private: 579 const sp<RecordThread::RecordTrack> mRecordTrack; 580 581 // for use from destructor 582 void stop_nonvirtual(); 583 }; 584 585 // Mmap stream control interface implementation. Each MmapThreadHandle controls one 586 // MmapPlaybackThread or MmapCaptureThread instance. 587 class MmapThreadHandle : public MmapStreamInterface { 588 public: 589 explicit MmapThreadHandle(const sp<MmapThread>& thread); 590 virtual ~MmapThreadHandle(); 591 592 // MmapStreamInterface virtuals 593 virtual status_t createMmapBuffer(int32_t minSizeFrames, 594 struct audio_mmap_buffer_info *info); 595 virtual status_t getMmapPosition(struct audio_mmap_position *position); 596 virtual status_t start(const MmapStreamInterface::Client& client, 597 audio_port_handle_t *handle); 598 virtual status_t stop(audio_port_handle_t handle); 599 virtual status_t standby(); 600 601 private: 602 sp<MmapThread> mThread; 603 }; 604 605 ThreadBase *checkThread_l(audio_io_handle_t ioHandle) const; 606 PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const; 607 MixerThread *checkMixerThread_l(audio_io_handle_t output) const; 608 RecordThread *checkRecordThread_l(audio_io_handle_t input) const; 609 MmapThread *checkMmapThread_l(audio_io_handle_t io) const; 610 VolumeInterface *getVolumeInterface_l(audio_io_handle_t output) const; 611 Vector <VolumeInterface *> getAllVolumeInterfaces_l() const; 612 613 sp<ThreadBase> openInput_l(audio_module_handle_t module, 614 audio_io_handle_t *input, 615 audio_config_t *config, 616 audio_devices_t device, 617 const String8& address, 618 audio_source_t source, 619 audio_input_flags_t flags); 620 sp<ThreadBase> openOutput_l(audio_module_handle_t module, 621 audio_io_handle_t *output, 622 audio_config_t *config, 623 audio_devices_t devices, 624 const String8& address, 625 audio_output_flags_t flags); 626 627 void closeOutputFinish(const sp<PlaybackThread>& thread); 628 void closeInputFinish(const sp<RecordThread>& thread); 629 630 // no range check, AudioFlinger::mLock held 631 bool streamMute_l(audio_stream_type_t stream) const 632 { return mStreamTypes[stream].mute; } 633 // no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held 634 float streamVolume_l(audio_stream_type_t stream) const 635 { return mStreamTypes[stream].volume; } 636 void ioConfigChanged(audio_io_config_event event, 637 const sp<AudioIoDescriptor>& ioDesc, 638 pid_t pid = 0); 639 640 // Allocate an audio_unique_id_t. 641 // Specific types are audio_io_handle_t, audio_session_t, effect ID (int), 642 // audio_module_handle_t, and audio_patch_handle_t. 643 // They all share the same ID space, but the namespaces are actually independent 644 // because there are separate KeyedVectors for each kind of ID. 645 // The return value is cast to the specific type depending on how the ID will be used. 646 // FIXME This API does not handle rollover to zero (for unsigned IDs), 647 // or from positive to negative (for signed IDs). 648 // Thus it may fail by returning an ID of the wrong sign, 649 // or by returning a non-unique ID. 650 // This is the internal API. For the binder API see newAudioUniqueId(). 651 audio_unique_id_t nextUniqueId(audio_unique_id_use_t use); 652 653 status_t moveEffectChain_l(audio_session_t sessionId, 654 PlaybackThread *srcThread, 655 PlaybackThread *dstThread, 656 bool reRegister); 657 658 // return thread associated with primary hardware device, or NULL 659 PlaybackThread *primaryPlaybackThread_l() const; 660 audio_devices_t primaryOutputDevice_l() const; 661 662 // return the playback thread with smallest HAL buffer size, and prefer fast 663 PlaybackThread *fastPlaybackThread_l() const; 664 665 sp<PlaybackThread> getEffectThread_l(audio_session_t sessionId, int EffectId); 666 667 668 void removeClient_l(pid_t pid); 669 void removeNotificationClient(pid_t pid); 670 bool isNonOffloadableGlobalEffectEnabled_l(); 671 void onNonOffloadableGlobalEffectEnable(); 672 bool isSessionAcquired_l(audio_session_t audioSession); 673 674 // Store an effect chain to mOrphanEffectChains keyed vector. 675 // Called when a thread exits and effects are still attached to it. 676 // If effects are later created on the same session, they will reuse the same 677 // effect chain and same instances in the effect library. 678 // return ALREADY_EXISTS if a chain with the same session already exists in 679 // mOrphanEffectChains. Note that this should never happen as there is only one 680 // chain for a given session and it is attached to only one thread at a time. 681 status_t putOrphanEffectChain_l(const sp<EffectChain>& chain); 682 // Get an effect chain for the specified session in mOrphanEffectChains and remove 683 // it if found. Returns 0 if not found (this is the most common case). 684 sp<EffectChain> getOrphanEffectChain_l(audio_session_t session); 685 // Called when the last effect handle on an effect instance is removed. If this 686 // effect belongs to an effect chain in mOrphanEffectChains, the chain is updated 687 // and removed from mOrphanEffectChains if it does not contain any effect. 688 // Return true if the effect was found in mOrphanEffectChains, false otherwise. 689 bool updateOrphanEffectChains(const sp<EffectModule>& effect); 690 691 void broacastParametersToRecordThreads_l(const String8& keyValuePairs); 692 693 // AudioStreamIn is immutable, so their fields are const. 694 // For emphasis, we could also make all pointers to them be "const *", 695 // but that would clutter the code unnecessarily. 696 697 struct AudioStreamIn { 698 AudioHwDevice* const audioHwDev; 699 sp<StreamInHalInterface> stream; 700 audio_input_flags_t flags; 701 702 sp<DeviceHalInterface> hwDev() const { return audioHwDev->hwDevice(); } 703 704 AudioStreamIn(AudioHwDevice *dev, sp<StreamInHalInterface> in, audio_input_flags_t flags) : 705 audioHwDev(dev), stream(in), flags(flags) {} 706 }; 707 708 // for mAudioSessionRefs only 709 struct AudioSessionRef { 710 AudioSessionRef(audio_session_t sessionid, pid_t pid) : 711 mSessionid(sessionid), mPid(pid), mCnt(1) {} 712 const audio_session_t mSessionid; 713 const pid_t mPid; 714 int mCnt; 715 }; 716 717 mutable Mutex mLock; 718 // protects mClients and mNotificationClients. 719 // must be locked after mLock and ThreadBase::mLock if both must be locked 720 // avoids acquiring AudioFlinger::mLock from inside thread loop. 721 mutable Mutex mClientLock; 722 // protected by mClientLock 723 DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client() 724 725 mutable Mutex mHardwareLock; 726 // NOTE: If both mLock and mHardwareLock mutexes must be held, 727 // always take mLock before mHardwareLock 728 729 // These two fields are immutable after onFirstRef(), so no lock needed to access 730 AudioHwDevice* mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL 731 DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*> mAudioHwDevs; 732 733 sp<DevicesFactoryHalInterface> mDevicesFactoryHal; 734 735 // for dump, indicates which hardware operation is currently in progress (but not stream ops) 736 enum hardware_call_state { 737 AUDIO_HW_IDLE = 0, // no operation in progress 738 AUDIO_HW_INIT, // init_check 739 AUDIO_HW_OUTPUT_OPEN, // open_output_stream 740 AUDIO_HW_OUTPUT_CLOSE, // unused 741 AUDIO_HW_INPUT_OPEN, // unused 742 AUDIO_HW_INPUT_CLOSE, // unused 743 AUDIO_HW_STANDBY, // unused 744 AUDIO_HW_SET_MASTER_VOLUME, // set_master_volume 745 AUDIO_HW_GET_ROUTING, // unused 746 AUDIO_HW_SET_ROUTING, // unused 747 AUDIO_HW_GET_MODE, // unused 748 AUDIO_HW_SET_MODE, // set_mode 749 AUDIO_HW_GET_MIC_MUTE, // get_mic_mute 750 AUDIO_HW_SET_MIC_MUTE, // set_mic_mute 751 AUDIO_HW_SET_VOICE_VOLUME, // set_voice_volume 752 AUDIO_HW_SET_PARAMETER, // set_parameters 753 AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size 754 AUDIO_HW_GET_MASTER_VOLUME, // get_master_volume 755 AUDIO_HW_GET_PARAMETER, // get_parameters 756 AUDIO_HW_SET_MASTER_MUTE, // set_master_mute 757 AUDIO_HW_GET_MASTER_MUTE, // get_master_mute 758 }; 759 760 mutable hardware_call_state mHardwareStatus; // for dump only 761 762 763 DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads; 764 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 765 766 // member variables below are protected by mLock 767 float mMasterVolume; 768 bool mMasterMute; 769 // end of variables protected by mLock 770 771 DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads; 772 773 // protected by mClientLock 774 DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients; 775 776 // updated by atomic_fetch_add_explicit 777 volatile atomic_uint_fast32_t mNextUniqueIds[AUDIO_UNIQUE_ID_USE_MAX]; 778 779 audio_mode_t mMode; 780 bool mBtNrecIsOff; 781 782 // protected by mLock 783 Vector<AudioSessionRef*> mAudioSessionRefs; 784 785 float masterVolume_l() const; 786 bool masterMute_l() const; 787 audio_module_handle_t loadHwModule_l(const char *name); 788 789 Vector < sp<SyncEvent> > mPendingSyncEvents; // sync events awaiting for a session 790 // to be created 791 792 // Effect chains without a valid thread 793 DefaultKeyedVector< audio_session_t , sp<EffectChain> > mOrphanEffectChains; 794 795 // list of sessions for which a valid HW A/V sync ID was retrieved from the HAL 796 DefaultKeyedVector< audio_session_t , audio_hw_sync_t >mHwAvSyncIds; 797 798 // list of MMAP stream control threads. Those threads allow for wake lock, routing 799 // and volume control for activity on the associated MMAP stream at the HAL. 800 // Audio data transfer is directly handled by the client creating the MMAP stream 801 DefaultKeyedVector< audio_io_handle_t, sp<MmapThread> > mMmapThreads; 802 803private: 804 sp<Client> registerPid(pid_t pid); // always returns non-0 805 806 // for use from destructor 807 status_t closeOutput_nonvirtual(audio_io_handle_t output); 808 void closeOutputInternal_l(const sp<PlaybackThread>& thread); 809 status_t closeInput_nonvirtual(audio_io_handle_t input); 810 void closeInputInternal_l(const sp<RecordThread>& thread); 811 void setAudioHwSyncForSession_l(PlaybackThread *thread, audio_session_t sessionId); 812 813 status_t checkStreamType(audio_stream_type_t stream) const; 814 815#ifdef TEE_SINK 816 // all record threads serially share a common tee sink, which is re-created on format change 817 sp<NBAIO_Sink> mRecordTeeSink; 818 sp<NBAIO_Source> mRecordTeeSource; 819#endif 820 821public: 822 823#ifdef TEE_SINK 824 // tee sink, if enabled by property, allows dumpsys to write most recent audio to .wav file 825 static void dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id = 0); 826 827 // whether tee sink is enabled by property 828 static bool mTeeSinkInputEnabled; 829 static bool mTeeSinkOutputEnabled; 830 static bool mTeeSinkTrackEnabled; 831 832 // runtime configured size of each tee sink pipe, in frames 833 static size_t mTeeSinkInputFrames; 834 static size_t mTeeSinkOutputFrames; 835 static size_t mTeeSinkTrackFrames; 836 837 // compile-time default size of tee sink pipes, in frames 838 // 0x200000 stereo 16-bit PCM frames = 47.5 seconds at 44.1 kHz, 8 megabytes 839 static const size_t kTeeSinkInputFramesDefault = 0x200000; 840 static const size_t kTeeSinkOutputFramesDefault = 0x200000; 841 static const size_t kTeeSinkTrackFramesDefault = 0x200000; 842#endif 843 844 // This method reads from a variable without mLock, but the variable is updated under mLock. So 845 // we might read a stale value, or a value that's inconsistent with respect to other variables. 846 // In this case, it's safe because the return value isn't used for making an important decision. 847 // The reason we don't want to take mLock is because it could block the caller for a long time. 848 bool isLowRamDevice() const { return mIsLowRamDevice; } 849 850private: 851 bool mIsLowRamDevice; 852 bool mIsDeviceTypeKnown; 853 nsecs_t mGlobalEffectEnableTime; // when a global effect was last enabled 854 855 sp<PatchPanel> mPatchPanel; 856 sp<EffectsFactoryHalInterface> mEffectsFactoryHal; 857 858 bool mSystemReady; 859}; 860 861#undef INCLUDING_FROM_AUDIOFLINGER_H 862 863std::string formatToString(audio_format_t format); 864std::string inputFlagsToString(audio_input_flags_t flags); 865std::string outputFlagsToString(audio_output_flags_t flags); 866std::string devicesToString(audio_devices_t devices); 867const char *sourceToString(audio_source_t source); 868 869// ---------------------------------------------------------------------------- 870 871} // namespace android 872 873#endif // ANDROID_AUDIO_FLINGER_H 874