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 <atomic> 23 #include <mutex> 24 #include <chrono> 25 #include <deque> 26 #include <map> 27 #include <numeric> 28 #include <optional> 29 #include <set> 30 #include <string> 31 #include <vector> 32 #include <stdint.h> 33 #include <sys/types.h> 34 #include <limits.h> 35 36 #include <android/os/BnExternalVibrationController.h> 37 #include <android-base/macros.h> 38 39 #include <cutils/atomic.h> 40 #include <cutils/compiler.h> 41 #include <cutils/properties.h> 42 43 #include <media/IAudioFlinger.h> 44 #include <media/IAudioFlingerClient.h> 45 #include <media/IAudioTrack.h> 46 #include <media/AudioSystem.h> 47 #include <media/AudioTrack.h> 48 #include <media/MmapStreamInterface.h> 49 #include <media/MmapStreamCallback.h> 50 51 #include <utils/Errors.h> 52 #include <utils/threads.h> 53 #include <utils/SortedVector.h> 54 #include <utils/TypeHelpers.h> 55 #include <utils/Vector.h> 56 57 #include <binder/AppOpsManager.h> 58 #include <binder/BinderService.h> 59 #include <binder/IAppOpsCallback.h> 60 #include <binder/MemoryDealer.h> 61 62 #include <system/audio.h> 63 #include <system/audio_policy.h> 64 65 #include <media/audiohal/EffectBufferHalInterface.h> 66 #include <media/audiohal/StreamHalInterface.h> 67 #include <media/AudioBufferProvider.h> 68 #include <media/AudioMixer.h> 69 #include <media/ExtendedAudioBufferProvider.h> 70 #include <media/LinearMap.h> 71 #include <media/VolumeShaper.h> 72 73 #include <audio_utils/clock.h> 74 #include <audio_utils/FdToString.h> 75 #include <audio_utils/SimpleLog.h> 76 #include <audio_utils/TimestampVerifier.h> 77 78 #include "FastCapture.h" 79 #include "FastMixer.h" 80 #include <media/nbaio/NBAIO.h> 81 #include "AudioWatchdog.h" 82 #include "AudioStreamOut.h" 83 #include "SpdifStreamOut.h" 84 #include "AudioHwDevice.h" 85 #include "NBAIO_Tee.h" 86 87 #include <powermanager/IPowerManager.h> 88 89 #include <media/nblog/NBLog.h> 90 #include <private/media/AudioEffectShared.h> 91 #include <private/media/AudioTrackShared.h> 92 93 #include <vibrator/ExternalVibration.h> 94 95 #include "android/media/BnAudioRecord.h" 96 97 namespace android { 98 99 class AudioMixer; 100 class AudioBuffer; 101 class AudioResampler; 102 class DeviceHalInterface; 103 class DevicesFactoryHalInterface; 104 class EffectsFactoryHalInterface; 105 class FastMixer; 106 class PassthruBufferProvider; 107 class RecordBufferConverter; 108 class ServerProxy; 109 110 // ---------------------------------------------------------------------------- 111 112 static const nsecs_t kDefaultStandbyTimeInNsecs = seconds(3); 113 114 #define INCLUDING_FROM_AUDIOFLINGER_H 115 116 class AudioFlinger : 117 public BinderService<AudioFlinger>, 118 public BnAudioFlinger 119 { 120 friend class BinderService<AudioFlinger>; // for AudioFlinger() 121 122 public: 123 static const char* getServiceName() ANDROID_API { return "media.audio_flinger"; } 124 125 virtual status_t dump(int fd, const Vector<String16>& args); 126 127 // IAudioFlinger interface, in binder opcode order 128 virtual sp<IAudioTrack> createTrack(const CreateTrackInput& input, 129 CreateTrackOutput& output, 130 status_t *status); 131 132 virtual sp<media::IAudioRecord> createRecord(const CreateRecordInput& input, 133 CreateRecordOutput& output, 134 status_t *status); 135 136 virtual uint32_t sampleRate(audio_io_handle_t ioHandle) const; 137 virtual audio_format_t format(audio_io_handle_t output) const; 138 virtual size_t frameCount(audio_io_handle_t ioHandle) const; 139 virtual size_t frameCountHAL(audio_io_handle_t ioHandle) const; 140 virtual uint32_t latency(audio_io_handle_t output) const; 141 142 virtual status_t setMasterVolume(float value); 143 virtual status_t setMasterMute(bool muted); 144 145 virtual float masterVolume() const; 146 virtual bool masterMute() const; 147 148 // Balance value must be within -1.f (left only) to 1.f (right only) inclusive. 149 status_t setMasterBalance(float balance) override; 150 status_t getMasterBalance(float *balance) const override; 151 152 virtual status_t setStreamVolume(audio_stream_type_t stream, float value, 153 audio_io_handle_t output); 154 virtual status_t setStreamMute(audio_stream_type_t stream, bool muted); 155 156 virtual float streamVolume(audio_stream_type_t stream, 157 audio_io_handle_t output) const; 158 virtual bool streamMute(audio_stream_type_t stream) const; 159 160 virtual status_t setMode(audio_mode_t mode); 161 162 virtual status_t setMicMute(bool state); 163 virtual bool getMicMute() const; 164 165 virtual void setRecordSilenced(uid_t uid, bool silenced); 166 167 virtual status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs); 168 virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) const; 169 170 virtual void registerClient(const sp<IAudioFlingerClient>& client); 171 172 virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, 173 audio_channel_mask_t channelMask) const; 174 175 virtual status_t openOutput(audio_module_handle_t module, 176 audio_io_handle_t *output, 177 audio_config_t *config, 178 audio_devices_t *devices, 179 const String8& address, 180 uint32_t *latencyMs, 181 audio_output_flags_t flags); 182 183 virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1, 184 audio_io_handle_t output2); 185 186 virtual status_t closeOutput(audio_io_handle_t output); 187 188 virtual status_t suspendOutput(audio_io_handle_t output); 189 190 virtual status_t restoreOutput(audio_io_handle_t output); 191 192 virtual status_t openInput(audio_module_handle_t module, 193 audio_io_handle_t *input, 194 audio_config_t *config, 195 audio_devices_t *device, 196 const String8& address, 197 audio_source_t source, 198 audio_input_flags_t flags); 199 200 virtual status_t closeInput(audio_io_handle_t input); 201 202 virtual status_t invalidateStream(audio_stream_type_t stream); 203 204 virtual status_t setVoiceVolume(float volume); 205 206 virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 207 audio_io_handle_t output) const; 208 209 virtual uint32_t getInputFramesLost(audio_io_handle_t ioHandle) const; 210 211 // This is the binder API. For the internal API see nextUniqueId(). 212 virtual audio_unique_id_t newAudioUniqueId(audio_unique_id_use_t use); 213 214 virtual void acquireAudioSessionId(audio_session_t audioSession, pid_t pid); 215 216 virtual void releaseAudioSessionId(audio_session_t audioSession, pid_t pid); 217 218 virtual status_t queryNumberEffects(uint32_t *numEffects) const; 219 220 virtual status_t queryEffect(uint32_t index, effect_descriptor_t *descriptor) const; 221 222 virtual status_t getEffectDescriptor(const effect_uuid_t *pUuid, 223 const effect_uuid_t *pTypeUuid, 224 uint32_t preferredTypeFlag, 225 effect_descriptor_t *descriptor) const; 226 227 virtual sp<IEffect> createEffect( 228 effect_descriptor_t *pDesc, 229 const sp<IEffectClient>& effectClient, 230 int32_t priority, 231 audio_io_handle_t io, 232 audio_session_t sessionId, 233 const String16& opPackageName, 234 pid_t pid, 235 status_t *status /*non-NULL*/, 236 int *id, 237 int *enabled); 238 239 virtual status_t moveEffects(audio_session_t sessionId, audio_io_handle_t srcOutput, 240 audio_io_handle_t dstOutput); 241 242 void setEffectSuspended(int effectId, 243 audio_session_t sessionId, 244 bool suspended) override; 245 246 virtual audio_module_handle_t loadHwModule(const char *name); 247 248 virtual uint32_t getPrimaryOutputSamplingRate(); 249 virtual size_t getPrimaryOutputFrameCount(); 250 251 virtual status_t setLowRamDevice(bool isLowRamDevice, int64_t totalMemory) override; 252 253 /* List available audio ports and their attributes */ 254 virtual status_t listAudioPorts(unsigned int *num_ports, 255 struct audio_port *ports); 256 257 /* Get attributes for a given audio port */ 258 virtual status_t getAudioPort(struct audio_port *port); 259 260 /* Create an audio patch between several source and sink ports */ 261 virtual status_t createAudioPatch(const struct audio_patch *patch, 262 audio_patch_handle_t *handle); 263 264 /* Release an audio patch */ 265 virtual status_t releaseAudioPatch(audio_patch_handle_t handle); 266 267 /* List existing audio patches */ 268 virtual status_t listAudioPatches(unsigned int *num_patches, 269 struct audio_patch *patches); 270 271 /* Set audio port configuration */ 272 virtual status_t setAudioPortConfig(const struct audio_port_config *config); 273 274 /* Get the HW synchronization source used for an audio session */ 275 virtual audio_hw_sync_t getAudioHwSyncForSession(audio_session_t sessionId); 276 277 /* Indicate JAVA services are ready (scheduling, power management ...) */ 278 virtual status_t systemReady(); 279 280 virtual status_t getMicrophones(std::vector<media::MicrophoneInfo> *microphones); 281 282 virtual status_t onTransact( 283 uint32_t code, 284 const Parcel& data, 285 Parcel* reply, 286 uint32_t flags); 287 288 // end of IAudioFlinger interface 289 290 sp<NBLog::Writer> newWriter_l(size_t size, const char *name); 291 void unregisterWriter(const sp<NBLog::Writer>& writer); 292 sp<EffectsFactoryHalInterface> getEffectsFactory(); 293 294 status_t openMmapStream(MmapStreamInterface::stream_direction_t direction, 295 const audio_attributes_t *attr, 296 audio_config_base_t *config, 297 const AudioClient& client, 298 audio_port_handle_t *deviceId, 299 audio_session_t *sessionId, 300 const sp<MmapStreamCallback>& callback, 301 sp<MmapStreamInterface>& interface, 302 audio_port_handle_t *handle); 303 304 static int onExternalVibrationStart(const sp<os::ExternalVibration>& externalVibration); 305 static void onExternalVibrationStop(const sp<os::ExternalVibration>& externalVibration); 306 private: 307 // FIXME The 400 is temporarily too high until a leak of writers in media.log is fixed. 308 static const size_t kLogMemorySize = 400 * 1024; 309 sp<MemoryDealer> mLogMemoryDealer; // == 0 when NBLog is disabled 310 // When a log writer is unregistered, it is done lazily so that media.log can continue to see it 311 // for as long as possible. The memory is only freed when it is needed for another log writer. 312 Vector< sp<NBLog::Writer> > mUnregisteredWriters; 313 Mutex mUnregisteredWritersLock; 314 315 public: 316 317 class SyncEvent; 318 319 typedef void (*sync_event_callback_t)(const wp<SyncEvent>& event) ; 320 321 class SyncEvent : public RefBase { 322 public: 323 SyncEvent(AudioSystem::sync_event_t type, 324 audio_session_t triggerSession, 325 audio_session_t listenerSession, 326 sync_event_callback_t callBack, 327 wp<RefBase> cookie) 328 : mType(type), mTriggerSession(triggerSession), mListenerSession(listenerSession), 329 mCallback(callBack), mCookie(cookie) 330 {} 331 332 virtual ~SyncEvent() {} 333 334 void trigger() { Mutex::Autolock _l(mLock); if (mCallback) mCallback(this); } 335 bool isCancelled() const { Mutex::Autolock _l(mLock); return (mCallback == NULL); } 336 void cancel() { Mutex::Autolock _l(mLock); mCallback = NULL; } 337 AudioSystem::sync_event_t type() const { return mType; } 338 audio_session_t triggerSession() const { return mTriggerSession; } 339 audio_session_t listenerSession() const { return mListenerSession; } 340 wp<RefBase> cookie() const { return mCookie; } 341 342 private: 343 const AudioSystem::sync_event_t mType; 344 const audio_session_t mTriggerSession; 345 const audio_session_t mListenerSession; 346 sync_event_callback_t mCallback; 347 const wp<RefBase> mCookie; 348 mutable Mutex mLock; 349 }; 350 351 sp<SyncEvent> createSyncEvent(AudioSystem::sync_event_t type, 352 audio_session_t triggerSession, 353 audio_session_t listenerSession, 354 sync_event_callback_t callBack, 355 const wp<RefBase>& cookie); 356 357 bool btNrecIsOff() const { return mBtNrecIsOff.load(); } 358 359 360 private: 361 362 audio_mode_t getMode() const { return mMode; } 363 364 AudioFlinger() ANDROID_API; 365 virtual ~AudioFlinger(); 366 367 // call in any IAudioFlinger method that accesses mPrimaryHardwareDev 368 status_t initCheck() const { return mPrimaryHardwareDev == NULL ? 369 NO_INIT : NO_ERROR; } 370 371 // RefBase 372 virtual void onFirstRef(); 373 374 AudioHwDevice* findSuitableHwDev_l(audio_module_handle_t module, 375 audio_devices_t devices); 376 377 // Set kEnableExtendedChannels to true to enable greater than stereo output 378 // for the MixerThread and device sink. Number of channels allowed is 379 // FCC_2 <= channels <= AudioMixer::MAX_NUM_CHANNELS. 380 static const bool kEnableExtendedChannels = true; 381 382 // Returns true if channel mask is permitted for the PCM sink in the MixerThread 383 static inline bool isValidPcmSinkChannelMask(audio_channel_mask_t channelMask) { 384 switch (audio_channel_mask_get_representation(channelMask)) { 385 case AUDIO_CHANNEL_REPRESENTATION_POSITION: { 386 // Haptic channel mask is only applicable for channel position mask. 387 const uint32_t channelCount = audio_channel_count_from_out_mask( 388 channelMask & ~AUDIO_CHANNEL_HAPTIC_ALL); 389 const uint32_t maxChannelCount = kEnableExtendedChannels 390 ? AudioMixer::MAX_NUM_CHANNELS : FCC_2; 391 if (channelCount < FCC_2 // mono is not supported at this time 392 || channelCount > maxChannelCount) { 393 return false; 394 } 395 // check that channelMask is the "canonical" one we expect for the channelCount. 396 return audio_channel_position_mask_is_out_canonical(channelMask); 397 } 398 case AUDIO_CHANNEL_REPRESENTATION_INDEX: 399 if (kEnableExtendedChannels) { 400 const uint32_t channelCount = audio_channel_count_from_out_mask(channelMask); 401 if (channelCount >= FCC_2 // mono is not supported at this time 402 && channelCount <= AudioMixer::MAX_NUM_CHANNELS) { 403 return true; 404 } 405 } 406 return false; 407 default: 408 return false; 409 } 410 } 411 412 // Set kEnableExtendedPrecision to true to use extended precision in MixerThread 413 static const bool kEnableExtendedPrecision = true; 414 415 // Returns true if format is permitted for the PCM sink in the MixerThread 416 static inline bool isValidPcmSinkFormat(audio_format_t format) { 417 switch (format) { 418 case AUDIO_FORMAT_PCM_16_BIT: 419 return true; 420 case AUDIO_FORMAT_PCM_FLOAT: 421 case AUDIO_FORMAT_PCM_24_BIT_PACKED: 422 case AUDIO_FORMAT_PCM_32_BIT: 423 case AUDIO_FORMAT_PCM_8_24_BIT: 424 return kEnableExtendedPrecision; 425 default: 426 return false; 427 } 428 } 429 430 // standby delay for MIXER and DUPLICATING playback threads is read from property 431 // ro.audio.flinger_standbytime_ms or defaults to kDefaultStandbyTimeInNsecs 432 static nsecs_t mStandbyTimeInNsecs; 433 434 // incremented by 2 when screen state changes, bit 0 == 1 means "off" 435 // AudioFlinger::setParameters() updates, other threads read w/o lock 436 static uint32_t mScreenState; 437 438 // Internal dump utilities. 439 static const int kDumpLockTimeoutNs = 1 * NANOS_PER_SECOND; 440 static bool dumpTryLock(Mutex& mutex); 441 void dumpPermissionDenial(int fd, const Vector<String16>& args); 442 void dumpClients(int fd, const Vector<String16>& args); 443 void dumpInternals(int fd, const Vector<String16>& args); 444 445 SimpleLog mThreadLog{16}; // 16 Thread history limit 446 447 class ThreadBase; 448 void dumpToThreadLog_l(const sp<ThreadBase> &thread); 449 450 // --- Client --- 451 class Client : public RefBase { 452 public: 453 Client(const sp<AudioFlinger>& audioFlinger, pid_t pid); 454 virtual ~Client(); 455 sp<MemoryDealer> heap() const; 456 pid_t pid() const { return mPid; } 457 sp<AudioFlinger> audioFlinger() const { return mAudioFlinger; } 458 459 private: 460 DISALLOW_COPY_AND_ASSIGN(Client); 461 462 const sp<AudioFlinger> mAudioFlinger; 463 sp<MemoryDealer> mMemoryDealer; 464 const pid_t mPid; 465 }; 466 467 // --- Notification Client --- 468 class NotificationClient : public IBinder::DeathRecipient { 469 public: 470 NotificationClient(const sp<AudioFlinger>& audioFlinger, 471 const sp<IAudioFlingerClient>& client, 472 pid_t pid); 473 virtual ~NotificationClient(); 474 475 sp<IAudioFlingerClient> audioFlingerClient() const { return mAudioFlingerClient; } 476 477 // IBinder::DeathRecipient 478 virtual void binderDied(const wp<IBinder>& who); 479 480 private: 481 DISALLOW_COPY_AND_ASSIGN(NotificationClient); 482 483 const sp<AudioFlinger> mAudioFlinger; 484 const pid_t mPid; 485 const sp<IAudioFlingerClient> mAudioFlingerClient; 486 }; 487 488 // --- MediaLogNotifier --- 489 // Thread in charge of notifying MediaLogService to start merging. 490 // Receives requests from AudioFlinger's binder activity. It is used to reduce the amount of 491 // binder calls to MediaLogService in case of bursts of AudioFlinger binder calls. 492 class MediaLogNotifier : public Thread { 493 public: 494 MediaLogNotifier(); 495 496 // Requests a MediaLogService notification. It's ignored if there has recently been another 497 void requestMerge(); 498 private: 499 // Every iteration blocks waiting for a request, then interacts with MediaLogService to 500 // start merging. 501 // As every MediaLogService binder call is expensive, once it gets a request it ignores the 502 // following ones for a period of time. 503 virtual bool threadLoop() override; 504 505 bool mPendingRequests; 506 507 // Mutex and condition variable around mPendingRequests' value 508 Mutex mMutex; 509 Condition mCond; 510 511 // Duration of the sleep period after a processed request 512 static const int kPostTriggerSleepPeriod = 1000000; 513 }; 514 515 const sp<MediaLogNotifier> mMediaLogNotifier; 516 517 // This is a helper that is called during incoming binder calls. 518 void requestLogMerge(); 519 520 class TrackHandle; 521 class RecordHandle; 522 class RecordThread; 523 class PlaybackThread; 524 class MixerThread; 525 class DirectOutputThread; 526 class OffloadThread; 527 class DuplicatingThread; 528 class AsyncCallbackThread; 529 class Track; 530 class RecordTrack; 531 class EffectModule; 532 class EffectHandle; 533 class EffectChain; 534 535 struct AudioStreamIn; 536 struct TeePatch; 537 using TeePatches = std::vector<TeePatch>; 538 539 540 struct stream_type_t { 541 stream_type_t() 542 : volume(1.0f), 543 mute(false) 544 { 545 } 546 float volume; 547 bool mute; 548 }; 549 550 // --- PlaybackThread --- 551 #ifdef FLOAT_EFFECT_CHAIN 552 #define EFFECT_BUFFER_FORMAT AUDIO_FORMAT_PCM_FLOAT 553 using effect_buffer_t = float; 554 #else 555 #define EFFECT_BUFFER_FORMAT AUDIO_FORMAT_PCM_16_BIT 556 using effect_buffer_t = int16_t; 557 #endif 558 559 #include "Threads.h" 560 561 #include "Effects.h" 562 563 #include "PatchPanel.h" 564 565 // Find io handle by session id. 566 // Preference is given to an io handle with a matching effect chain to session id. 567 // If none found, AUDIO_IO_HANDLE_NONE is returned. 568 template <typename T> 569 static audio_io_handle_t findIoHandleBySessionId_l( 570 audio_session_t sessionId, const T& threads) { 571 audio_io_handle_t io = AUDIO_IO_HANDLE_NONE; 572 573 for (size_t i = 0; i < threads.size(); i++) { 574 const uint32_t sessionType = threads.valueAt(i)->hasAudioSession(sessionId); 575 if (sessionType != 0) { 576 io = threads.keyAt(i); 577 if ((sessionType & AudioFlinger::ThreadBase::EFFECT_SESSION) != 0) { 578 break; // effect chain here. 579 } 580 } 581 } 582 return io; 583 } 584 585 // server side of the client's IAudioTrack 586 class TrackHandle : public android::BnAudioTrack { 587 public: 588 explicit TrackHandle(const sp<PlaybackThread::Track>& track); 589 virtual ~TrackHandle(); 590 virtual sp<IMemory> getCblk() const; 591 virtual status_t start(); 592 virtual void stop(); 593 virtual void flush(); 594 virtual void pause(); 595 virtual status_t attachAuxEffect(int effectId); 596 virtual status_t setParameters(const String8& keyValuePairs); 597 virtual status_t selectPresentation(int presentationId, int programId); 598 virtual media::VolumeShaper::Status applyVolumeShaper( 599 const sp<media::VolumeShaper::Configuration>& configuration, 600 const sp<media::VolumeShaper::Operation>& operation) override; 601 virtual sp<media::VolumeShaper::State> getVolumeShaperState(int id) override; 602 virtual status_t getTimestamp(AudioTimestamp& timestamp); 603 virtual void signal(); // signal playback thread for a change in control block 604 605 virtual status_t onTransact( 606 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 607 608 private: 609 const sp<PlaybackThread::Track> mTrack; 610 }; 611 612 // server side of the client's IAudioRecord 613 class RecordHandle : public android::media::BnAudioRecord { 614 public: 615 explicit RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack); 616 virtual ~RecordHandle(); 617 virtual binder::Status start(int /*AudioSystem::sync_event_t*/ event, 618 int /*audio_session_t*/ triggerSession); 619 virtual binder::Status stop(); 620 virtual binder::Status getActiveMicrophones( 621 std::vector<media::MicrophoneInfo>* activeMicrophones); 622 virtual binder::Status setPreferredMicrophoneDirection( 623 int /*audio_microphone_direction_t*/ direction); 624 virtual binder::Status setPreferredMicrophoneFieldDimension(float zoom); 625 626 private: 627 const sp<RecordThread::RecordTrack> mRecordTrack; 628 629 // for use from destructor 630 void stop_nonvirtual(); 631 }; 632 633 // Mmap stream control interface implementation. Each MmapThreadHandle controls one 634 // MmapPlaybackThread or MmapCaptureThread instance. 635 class MmapThreadHandle : public MmapStreamInterface { 636 public: 637 explicit MmapThreadHandle(const sp<MmapThread>& thread); 638 virtual ~MmapThreadHandle(); 639 640 // MmapStreamInterface virtuals 641 virtual status_t createMmapBuffer(int32_t minSizeFrames, 642 struct audio_mmap_buffer_info *info); 643 virtual status_t getMmapPosition(struct audio_mmap_position *position); 644 virtual status_t start(const AudioClient& client, 645 audio_port_handle_t *handle); 646 virtual status_t stop(audio_port_handle_t handle); 647 virtual status_t standby(); 648 649 private: 650 const sp<MmapThread> mThread; 651 }; 652 653 ThreadBase *checkThread_l(audio_io_handle_t ioHandle) const; 654 PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const; 655 MixerThread *checkMixerThread_l(audio_io_handle_t output) const; 656 RecordThread *checkRecordThread_l(audio_io_handle_t input) const; 657 MmapThread *checkMmapThread_l(audio_io_handle_t io) const; 658 VolumeInterface *getVolumeInterface_l(audio_io_handle_t output) const; 659 Vector <VolumeInterface *> getAllVolumeInterfaces_l() const; 660 661 sp<ThreadBase> openInput_l(audio_module_handle_t module, 662 audio_io_handle_t *input, 663 audio_config_t *config, 664 audio_devices_t device, 665 const String8& address, 666 audio_source_t source, 667 audio_input_flags_t flags, 668 audio_devices_t outputDevice, 669 const String8& outputDeviceAddress); 670 sp<ThreadBase> openOutput_l(audio_module_handle_t module, 671 audio_io_handle_t *output, 672 audio_config_t *config, 673 audio_devices_t devices, 674 const String8& address, 675 audio_output_flags_t flags); 676 677 void closeOutputFinish(const sp<PlaybackThread>& thread); 678 void closeInputFinish(const sp<RecordThread>& thread); 679 680 // no range check, AudioFlinger::mLock held 681 bool streamMute_l(audio_stream_type_t stream) const 682 { return mStreamTypes[stream].mute; } 683 void ioConfigChanged(audio_io_config_event event, 684 const sp<AudioIoDescriptor>& ioDesc, 685 pid_t pid = 0); 686 687 // Allocate an audio_unique_id_t. 688 // Specific types are audio_io_handle_t, audio_session_t, effect ID (int), 689 // audio_module_handle_t, and audio_patch_handle_t. 690 // They all share the same ID space, but the namespaces are actually independent 691 // because there are separate KeyedVectors for each kind of ID. 692 // The return value is cast to the specific type depending on how the ID will be used. 693 // FIXME This API does not handle rollover to zero (for unsigned IDs), 694 // or from positive to negative (for signed IDs). 695 // Thus it may fail by returning an ID of the wrong sign, 696 // or by returning a non-unique ID. 697 // This is the internal API. For the binder API see newAudioUniqueId(). 698 audio_unique_id_t nextUniqueId(audio_unique_id_use_t use); 699 700 status_t moveEffectChain_l(audio_session_t sessionId, 701 PlaybackThread *srcThread, 702 PlaybackThread *dstThread); 703 704 status_t moveAuxEffectToIo(int EffectId, 705 const sp<PlaybackThread>& dstThread, 706 sp<PlaybackThread> *srcThread); 707 708 // return thread associated with primary hardware device, or NULL 709 PlaybackThread *primaryPlaybackThread_l() const; 710 audio_devices_t primaryOutputDevice_l() const; 711 712 // return the playback thread with smallest HAL buffer size, and prefer fast 713 PlaybackThread *fastPlaybackThread_l() const; 714 715 sp<ThreadBase> getEffectThread_l(audio_session_t sessionId, int effectId); 716 717 718 void removeClient_l(pid_t pid); 719 void removeNotificationClient(pid_t pid); 720 bool isNonOffloadableGlobalEffectEnabled_l(); 721 void onNonOffloadableGlobalEffectEnable(); 722 bool isSessionAcquired_l(audio_session_t audioSession); 723 724 // Store an effect chain to mOrphanEffectChains keyed vector. 725 // Called when a thread exits and effects are still attached to it. 726 // If effects are later created on the same session, they will reuse the same 727 // effect chain and same instances in the effect library. 728 // return ALREADY_EXISTS if a chain with the same session already exists in 729 // mOrphanEffectChains. Note that this should never happen as there is only one 730 // chain for a given session and it is attached to only one thread at a time. 731 status_t putOrphanEffectChain_l(const sp<EffectChain>& chain); 732 // Get an effect chain for the specified session in mOrphanEffectChains and remove 733 // it if found. Returns 0 if not found (this is the most common case). 734 sp<EffectChain> getOrphanEffectChain_l(audio_session_t session); 735 // Called when the last effect handle on an effect instance is removed. If this 736 // effect belongs to an effect chain in mOrphanEffectChains, the chain is updated 737 // and removed from mOrphanEffectChains if it does not contain any effect. 738 // Return true if the effect was found in mOrphanEffectChains, false otherwise. 739 bool updateOrphanEffectChains(const sp<EffectModule>& effect); 740 741 std::vector< sp<EffectModule> > purgeStaleEffects_l(); 742 743 void broacastParametersToRecordThreads_l(const String8& keyValuePairs); 744 void forwardParametersToDownstreamPatches_l( 745 audio_io_handle_t upStream, const String8& keyValuePairs, 746 std::function<bool(const sp<PlaybackThread>&)> useThread = nullptr); 747 748 // AudioStreamIn is immutable, so their fields are const. 749 // For emphasis, we could also make all pointers to them be "const *", 750 // but that would clutter the code unnecessarily. 751 752 struct AudioStreamIn { 753 AudioHwDevice* const audioHwDev; 754 sp<StreamInHalInterface> stream; 755 audio_input_flags_t flags; 756 757 sp<DeviceHalInterface> hwDev() const { return audioHwDev->hwDevice(); } 758 759 AudioStreamIn(AudioHwDevice *dev, sp<StreamInHalInterface> in, audio_input_flags_t flags) : 760 audioHwDev(dev), stream(in), flags(flags) {} 761 }; 762 763 struct TeePatch { 764 sp<RecordThread::PatchRecord> patchRecord; 765 sp<PlaybackThread::PatchTrack> patchTrack; 766 }; 767 768 // for mAudioSessionRefs only 769 struct AudioSessionRef { 770 AudioSessionRef(audio_session_t sessionid, pid_t pid) : 771 mSessionid(sessionid), mPid(pid), mCnt(1) {} 772 const audio_session_t mSessionid; 773 const pid_t mPid; 774 int mCnt; 775 }; 776 777 mutable Mutex mLock; 778 // protects mClients and mNotificationClients. 779 // must be locked after mLock and ThreadBase::mLock if both must be locked 780 // avoids acquiring AudioFlinger::mLock from inside thread loop. 781 mutable Mutex mClientLock; 782 // protected by mClientLock 783 DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client() 784 785 mutable Mutex mHardwareLock; 786 // NOTE: If both mLock and mHardwareLock mutexes must be held, 787 // always take mLock before mHardwareLock 788 789 // These two fields are immutable after onFirstRef(), so no lock needed to access 790 AudioHwDevice* mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL 791 DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*> mAudioHwDevs; 792 793 sp<DevicesFactoryHalInterface> mDevicesFactoryHal; 794 795 // for dump, indicates which hardware operation is currently in progress (but not stream ops) 796 enum hardware_call_state { 797 AUDIO_HW_IDLE = 0, // no operation in progress 798 AUDIO_HW_INIT, // init_check 799 AUDIO_HW_OUTPUT_OPEN, // open_output_stream 800 AUDIO_HW_OUTPUT_CLOSE, // unused 801 AUDIO_HW_INPUT_OPEN, // unused 802 AUDIO_HW_INPUT_CLOSE, // unused 803 AUDIO_HW_STANDBY, // unused 804 AUDIO_HW_SET_MASTER_VOLUME, // set_master_volume 805 AUDIO_HW_GET_ROUTING, // unused 806 AUDIO_HW_SET_ROUTING, // unused 807 AUDIO_HW_GET_MODE, // unused 808 AUDIO_HW_SET_MODE, // set_mode 809 AUDIO_HW_GET_MIC_MUTE, // get_mic_mute 810 AUDIO_HW_SET_MIC_MUTE, // set_mic_mute 811 AUDIO_HW_SET_VOICE_VOLUME, // set_voice_volume 812 AUDIO_HW_SET_PARAMETER, // set_parameters 813 AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size 814 AUDIO_HW_GET_MASTER_VOLUME, // get_master_volume 815 AUDIO_HW_GET_PARAMETER, // get_parameters 816 AUDIO_HW_SET_MASTER_MUTE, // set_master_mute 817 AUDIO_HW_GET_MASTER_MUTE, // get_master_mute 818 }; 819 820 mutable hardware_call_state mHardwareStatus; // for dump only 821 822 823 DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads; 824 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 825 826 // member variables below are protected by mLock 827 float mMasterVolume; 828 bool mMasterMute; 829 float mMasterBalance = 0.f; 830 // end of variables protected by mLock 831 832 DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads; 833 834 // protected by mClientLock 835 DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients; 836 837 // updated by atomic_fetch_add_explicit 838 volatile atomic_uint_fast32_t mNextUniqueIds[AUDIO_UNIQUE_ID_USE_MAX]; 839 840 audio_mode_t mMode; 841 std::atomic_bool mBtNrecIsOff; 842 843 // protected by mLock 844 Vector<AudioSessionRef*> mAudioSessionRefs; 845 846 float masterVolume_l() const; 847 float getMasterBalance_l() const; 848 bool masterMute_l() const; 849 audio_module_handle_t loadHwModule_l(const char *name); 850 851 Vector < sp<SyncEvent> > mPendingSyncEvents; // sync events awaiting for a session 852 // to be created 853 854 // Effect chains without a valid thread 855 DefaultKeyedVector< audio_session_t , sp<EffectChain> > mOrphanEffectChains; 856 857 // list of sessions for which a valid HW A/V sync ID was retrieved from the HAL 858 DefaultKeyedVector< audio_session_t , audio_hw_sync_t >mHwAvSyncIds; 859 860 // list of MMAP stream control threads. Those threads allow for wake lock, routing 861 // and volume control for activity on the associated MMAP stream at the HAL. 862 // Audio data transfer is directly handled by the client creating the MMAP stream 863 DefaultKeyedVector< audio_io_handle_t, sp<MmapThread> > mMmapThreads; 864 865 private: 866 sp<Client> registerPid(pid_t pid); // always returns non-0 867 868 // for use from destructor 869 status_t closeOutput_nonvirtual(audio_io_handle_t output); 870 void closeThreadInternal_l(const sp<PlaybackThread>& thread); 871 status_t closeInput_nonvirtual(audio_io_handle_t input); 872 void closeThreadInternal_l(const sp<RecordThread>& thread); 873 void setAudioHwSyncForSession_l(PlaybackThread *thread, audio_session_t sessionId); 874 875 status_t checkStreamType(audio_stream_type_t stream) const; 876 877 void filterReservedParameters(String8& keyValuePairs, uid_t callingUid); 878 void logFilteredParameters(size_t originalKVPSize, const String8& originalKVPs, 879 size_t rejectedKVPSize, const String8& rejectedKVPs, 880 uid_t callingUid); 881 882 public: 883 // These methods read variables atomically without mLock, 884 // though the variables are updated with mLock. 885 bool isLowRamDevice() const { return mIsLowRamDevice; } 886 size_t getClientSharedHeapSize() const; 887 888 private: 889 std::atomic<bool> mIsLowRamDevice; 890 bool mIsDeviceTypeKnown; 891 int64_t mTotalMemory; 892 std::atomic<size_t> mClientSharedHeapSize; 893 static constexpr size_t kMinimumClientSharedHeapSizeBytes = 1024 * 1024; // 1MB 894 895 nsecs_t mGlobalEffectEnableTime; // when a global effect was last enabled 896 897 // protected by mLock 898 PatchPanel mPatchPanel; 899 sp<EffectsFactoryHalInterface> mEffectsFactoryHal; 900 901 bool mSystemReady; 902 903 SimpleLog mRejectedSetParameterLog; 904 SimpleLog mAppSetParameterLog; 905 SimpleLog mSystemSetParameterLog; 906 }; 907 908 #undef INCLUDING_FROM_AUDIOFLINGER_H 909 910 std::string formatToString(audio_format_t format); 911 std::string inputFlagsToString(audio_input_flags_t flags); 912 std::string outputFlagsToString(audio_output_flags_t flags); 913 std::string devicesToString(audio_devices_t devices); 914 const char *sourceToString(audio_source_t source); 915 916 // ---------------------------------------------------------------------------- 917 918 } // namespace android 919 920 #endif // ANDROID_AUDIO_FLINGER_H 921
