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/media/BnAudioTrack.h> 37 #include <android/media/IAudioFlingerClient.h> 38 #include <android/media/IAudioTrackCallback.h> 39 #include <android/os/BnExternalVibrationController.h> 40 #include <android/content/AttributionSourceState.h> 41 42 43 #include <android-base/macros.h> 44 #include <cutils/atomic.h> 45 #include <cutils/compiler.h> 46 47 #include <cutils/properties.h> 48 #include <media/IAudioFlinger.h> 49 #include <media/AudioSystem.h> 50 #include <media/AudioTrack.h> 51 #include <media/MmapStreamInterface.h> 52 #include <media/MmapStreamCallback.h> 53 54 #include <utils/Errors.h> 55 #include <utils/threads.h> 56 #include <utils/SortedVector.h> 57 #include <utils/TypeHelpers.h> 58 #include <utils/Vector.h> 59 60 #include <binder/AppOpsManager.h> 61 #include <binder/BinderService.h> 62 #include <binder/IAppOpsCallback.h> 63 #include <binder/MemoryDealer.h> 64 65 #include <system/audio.h> 66 #include <system/audio_policy.h> 67 68 #include <media/audiohal/EffectBufferHalInterface.h> 69 #include <media/audiohal/StreamHalInterface.h> 70 #include <media/AudioBufferProvider.h> 71 #include <media/AudioContainers.h> 72 #include <media/AudioDeviceTypeAddr.h> 73 #include <media/AudioMixer.h> 74 #include <media/DeviceDescriptorBase.h> 75 #include <media/ExtendedAudioBufferProvider.h> 76 #include <media/VolumeShaper.h> 77 #include <mediautils/BatteryNotifier.h> 78 #include <mediautils/ServiceUtilities.h> 79 #include <mediautils/SharedMemoryAllocator.h> 80 #include <mediautils/Synchronization.h> 81 #include <mediautils/ThreadSnapshot.h> 82 83 #include <audio_utils/clock.h> 84 #include <audio_utils/FdToString.h> 85 #include <audio_utils/LinearMap.h> 86 #include <audio_utils/MelAggregator.h> 87 #include <audio_utils/MelProcessor.h> 88 #include <audio_utils/SimpleLog.h> 89 #include <audio_utils/TimestampVerifier.h> 90 91 #include <sounddose/SoundDoseManager.h> 92 #include <timing/MonotonicFrameCounter.h> 93 94 #include "FastCapture.h" 95 #include "FastMixer.h" 96 #include <media/nbaio/NBAIO.h> 97 #include "AudioWatchdog.h" 98 #include "AudioStreamOut.h" 99 #include "SpdifStreamOut.h" 100 #include "AudioHwDevice.h" 101 #include "NBAIO_Tee.h" 102 #include "ThreadMetrics.h" 103 #include "TrackMetrics.h" 104 #include "AllocatorFactory.h" 105 #include <android/os/IPowerManager.h> 106 107 #include <media/nblog/NBLog.h> 108 #include <private/media/AudioEffectShared.h> 109 #include <private/media/AudioTrackShared.h> 110 111 #include <vibrator/ExternalVibration.h> 112 #include <vibrator/ExternalVibrationUtils.h> 113 114 #include "android/media/BnAudioRecord.h" 115 #include "android/media/BnEffect.h" 116 117 namespace android { 118 119 class AudioMixer; 120 class AudioBuffer; 121 class AudioResampler; 122 class DeviceHalInterface; 123 class DevicesFactoryHalCallback; 124 class DevicesFactoryHalInterface; 125 class EffectsFactoryHalInterface; 126 class FastMixer; 127 class IAudioManager; 128 class PassthruBufferProvider; 129 class RecordBufferConverter; 130 class ServerProxy; 131 132 // ---------------------------------------------------------------------------- 133 134 static const nsecs_t kDefaultStandbyTimeInNsecs = seconds(3); 135 136 #define INCLUDING_FROM_AUDIOFLINGER_H 137 138 using android::content::AttributionSourceState; 139 140 class AudioFlinger : public AudioFlingerServerAdapter::Delegate 141 { 142 friend class sp<AudioFlinger>; 143 public: 144 static void instantiate() ANDROID_API; 145 146 static AttributionSourceState checkAttributionSourcePackage( 147 const AttributionSourceState& attributionSource); 148 149 status_t dump(int fd, const Vector<String16>& args) override; 150 151 // IAudioFlinger interface, in binder opcode order 152 status_t createTrack(const media::CreateTrackRequest& input, 153 media::CreateTrackResponse& output) override; 154 155 status_t createRecord(const media::CreateRecordRequest& input, 156 media::CreateRecordResponse& output) override; 157 158 virtual uint32_t sampleRate(audio_io_handle_t ioHandle) const; 159 virtual audio_format_t format(audio_io_handle_t output) const; 160 virtual size_t frameCount(audio_io_handle_t ioHandle) const; 161 virtual size_t frameCountHAL(audio_io_handle_t ioHandle) const; 162 virtual uint32_t latency(audio_io_handle_t output) const; 163 164 virtual status_t setMasterVolume(float value); 165 virtual status_t setMasterMute(bool muted); 166 167 virtual float masterVolume() const; 168 virtual bool masterMute() const; 169 170 // Balance value must be within -1.f (left only) to 1.f (right only) inclusive. 171 status_t setMasterBalance(float balance) override; 172 status_t getMasterBalance(float *balance) const override; 173 174 virtual status_t setStreamVolume(audio_stream_type_t stream, float value, 175 audio_io_handle_t output); 176 virtual status_t setStreamMute(audio_stream_type_t stream, bool muted); 177 178 virtual float streamVolume(audio_stream_type_t stream, 179 audio_io_handle_t output) const; 180 virtual bool streamMute(audio_stream_type_t stream) const; 181 182 virtual status_t setMode(audio_mode_t mode); 183 184 virtual status_t setMicMute(bool state); 185 virtual bool getMicMute() const; 186 187 virtual void setRecordSilenced(audio_port_handle_t portId, bool silenced); 188 189 virtual status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs); 190 virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) const; 191 192 virtual void registerClient(const sp<media::IAudioFlingerClient>& client); 193 194 virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, 195 audio_channel_mask_t channelMask) const; 196 197 virtual status_t openOutput(const media::OpenOutputRequest& request, 198 media::OpenOutputResponse* response); 199 200 virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1, 201 audio_io_handle_t output2); 202 203 virtual status_t closeOutput(audio_io_handle_t output); 204 205 virtual status_t suspendOutput(audio_io_handle_t output); 206 207 virtual status_t restoreOutput(audio_io_handle_t output); 208 209 virtual status_t openInput(const media::OpenInputRequest& request, 210 media::OpenInputResponse* response); 211 212 virtual status_t closeInput(audio_io_handle_t input); 213 214 virtual status_t setVoiceVolume(float volume); 215 216 virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 217 audio_io_handle_t output) const; 218 219 virtual uint32_t getInputFramesLost(audio_io_handle_t ioHandle) const; 220 221 // This is the binder API. For the internal API see nextUniqueId(). 222 virtual audio_unique_id_t newAudioUniqueId(audio_unique_id_use_t use); 223 224 void acquireAudioSessionId(audio_session_t audioSession, pid_t pid, uid_t uid) override; 225 226 virtual void releaseAudioSessionId(audio_session_t audioSession, pid_t pid); 227 228 virtual status_t queryNumberEffects(uint32_t *numEffects) const; 229 230 virtual status_t queryEffect(uint32_t index, effect_descriptor_t *descriptor) const; 231 232 virtual status_t getEffectDescriptor(const effect_uuid_t *pUuid, 233 const effect_uuid_t *pTypeUuid, 234 uint32_t preferredTypeFlag, 235 effect_descriptor_t *descriptor) const; 236 237 virtual status_t createEffect(const media::CreateEffectRequest& request, 238 media::CreateEffectResponse* response); 239 240 virtual status_t moveEffects(audio_session_t sessionId, audio_io_handle_t srcOutput, 241 audio_io_handle_t dstOutput); 242 243 void setEffectSuspended(int effectId, 244 audio_session_t sessionId, 245 bool suspended) override; 246 247 virtual audio_module_handle_t loadHwModule(const char *name); 248 249 virtual uint32_t getPrimaryOutputSamplingRate(); 250 virtual size_t getPrimaryOutputFrameCount(); 251 252 virtual status_t setLowRamDevice(bool isLowRamDevice, int64_t totalMemory) override; 253 254 /* List available audio ports and their attributes */ 255 virtual status_t listAudioPorts(unsigned int *num_ports, 256 struct audio_port *ports); 257 258 /* Get attributes for a given audio port */ 259 virtual status_t getAudioPort(struct audio_port_v7 *port); 260 261 /* Create an audio patch between several source and sink ports */ 262 virtual status_t createAudioPatch(const struct audio_patch *patch, 263 audio_patch_handle_t *handle); 264 265 /* Release an audio patch */ 266 virtual status_t releaseAudioPatch(audio_patch_handle_t handle); 267 268 /* List existing audio patches */ 269 virtual status_t listAudioPatches(unsigned int *num_patches, 270 struct audio_patch *patches); 271 272 /* Set audio port configuration */ 273 virtual status_t setAudioPortConfig(const struct audio_port_config *config); 274 275 /* Get the HW synchronization source used for an audio session */ 276 virtual audio_hw_sync_t getAudioHwSyncForSession(audio_session_t sessionId); 277 278 /* Indicate JAVA services are ready (scheduling, power management ...) */ 279 virtual status_t systemReady(); audioPolicyReady()280 virtual status_t audioPolicyReady() { mAudioPolicyReady.store(true); return NO_ERROR; } isAudioPolicyReady()281 bool isAudioPolicyReady() const { return mAudioPolicyReady.load(); } 282 283 284 virtual status_t getMicrophones(std::vector<media::MicrophoneInfoFw> *microphones); 285 286 virtual status_t setAudioHalPids(const std::vector<pid_t>& pids); 287 288 virtual status_t setVibratorInfos(const std::vector<media::AudioVibratorInfo>& vibratorInfos); 289 290 virtual status_t updateSecondaryOutputs( 291 const TrackSecondaryOutputsMap& trackSecondaryOutputs); 292 293 virtual status_t getMmapPolicyInfos( 294 media::audio::common::AudioMMapPolicyType policyType, 295 std::vector<media::audio::common::AudioMMapPolicyInfo> *policyInfos); 296 297 virtual int32_t getAAudioMixerBurstCount(); 298 299 virtual int32_t getAAudioHardwareBurstMinUsec(); 300 301 virtual status_t setDeviceConnectedState(const struct audio_port_v7 *port, 302 media::DeviceConnectedState state); 303 304 virtual status_t setSimulateDeviceConnections(bool enabled); 305 306 virtual status_t setRequestedLatencyMode( 307 audio_io_handle_t output, audio_latency_mode_t mode); 308 309 virtual status_t getSupportedLatencyModes(audio_io_handle_t output, 310 std::vector<audio_latency_mode_t>* modes); 311 312 virtual status_t setBluetoothVariableLatencyEnabled(bool enabled); 313 314 virtual status_t isBluetoothVariableLatencyEnabled(bool* enabled); 315 316 virtual status_t supportsBluetoothVariableLatency(bool* support); 317 318 virtual status_t getSoundDoseInterface(const sp<media::ISoundDoseCallback>& callback, 319 sp<media::ISoundDose>* soundDose); 320 321 status_t invalidateTracks(const std::vector<audio_port_handle_t>& portIds) override; 322 323 virtual status_t getAudioPolicyConfig(media::AudioPolicyConfig* config); 324 325 status_t onTransactWrapper(TransactionCode code, const Parcel& data, uint32_t flags, 326 const std::function<status_t()>& delegate) override; 327 328 // end of IAudioFlinger interface 329 330 sp<NBLog::Writer> newWriter_l(size_t size, const char *name); 331 void unregisterWriter(const sp<NBLog::Writer>& writer); 332 sp<EffectsFactoryHalInterface> getEffectsFactory(); 333 334 status_t openMmapStream(MmapStreamInterface::stream_direction_t direction, 335 const audio_attributes_t *attr, 336 audio_config_base_t *config, 337 const AudioClient& client, 338 audio_port_handle_t *deviceId, 339 audio_session_t *sessionId, 340 const sp<MmapStreamCallback>& callback, 341 sp<MmapStreamInterface>& interface, 342 audio_port_handle_t *handle); 343 344 static os::HapticScale onExternalVibrationStart( 345 const sp<os::ExternalVibration>& externalVibration); 346 static void onExternalVibrationStop(const sp<os::ExternalVibration>& externalVibration); 347 348 status_t addEffectToHal( 349 const struct audio_port_config *device, const sp<EffectHalInterface>& effect); 350 status_t removeEffectFromHal( 351 const struct audio_port_config *device, const sp<EffectHalInterface>& effect); 352 353 void updateDownStreamPatches_l(const struct audio_patch *patch, 354 const std::set<audio_io_handle_t>& streams); 355 356 std::optional<media::AudioVibratorInfo> getDefaultVibratorInfo_l(); 357 358 private: 359 // FIXME The 400 is temporarily too high until a leak of writers in media.log is fixed. 360 static const size_t kLogMemorySize = 400 * 1024; 361 sp<MemoryDealer> mLogMemoryDealer; // == 0 when NBLog is disabled 362 // When a log writer is unregistered, it is done lazily so that media.log can continue to see it 363 // for as long as possible. The memory is only freed when it is needed for another log writer. 364 Vector< sp<NBLog::Writer> > mUnregisteredWriters; 365 Mutex mUnregisteredWritersLock; 366 367 public: 368 // Life cycle of gAudioFlinger and AudioFlinger: 369 // 370 // AudioFlinger is created once and survives until audioserver crashes 371 // irrespective of sp<> and wp<> as it is refcounted by ServiceManager and we 372 // don't issue a ServiceManager::tryUnregisterService(). 373 // 374 // gAudioFlinger is an atomic pointer set on AudioFlinger::onFirstRef(). 375 // After this is set, it is safe to obtain a wp<> or sp<> from it as the 376 // underlying object does not go away. 377 // 378 // Note: For most inner classes, it is acceptable to hold a reference to the outer 379 // AudioFlinger instance as creation requires AudioFlinger to exist in the first place. 380 // 381 // An atomic here ensures underlying writes have completed before setting 382 // the pointer. Access by memory_order_seq_cst. 383 // 384 385 static inline std::atomic<AudioFlinger *> gAudioFlinger = nullptr; 386 387 class SyncEvent; 388 389 typedef void (*sync_event_callback_t)(const wp<SyncEvent>& event) ; 390 391 class SyncEvent : public RefBase { 392 public: SyncEvent(AudioSystem::sync_event_t type,audio_session_t triggerSession,audio_session_t listenerSession,sync_event_callback_t callBack,const wp<RefBase> & cookie)393 SyncEvent(AudioSystem::sync_event_t type, 394 audio_session_t triggerSession, 395 audio_session_t listenerSession, 396 sync_event_callback_t callBack, 397 const wp<RefBase>& cookie) 398 : mType(type), mTriggerSession(triggerSession), mListenerSession(listenerSession), 399 mCallback(callBack), mCookie(cookie) 400 {} 401 ~SyncEvent()402 virtual ~SyncEvent() {} 403 trigger()404 void trigger() { 405 Mutex::Autolock _l(mLock); 406 if (mCallback) mCallback(wp<SyncEvent>(this)); 407 } isCancelled()408 bool isCancelled() const { Mutex::Autolock _l(mLock); return (mCallback == NULL); } cancel()409 void cancel() { Mutex::Autolock _l(mLock); mCallback = NULL; } type()410 AudioSystem::sync_event_t type() const { return mType; } triggerSession()411 audio_session_t triggerSession() const { return mTriggerSession; } listenerSession()412 audio_session_t listenerSession() const { return mListenerSession; } cookie()413 wp<RefBase> cookie() const { return mCookie; } 414 415 private: 416 const AudioSystem::sync_event_t mType; 417 const audio_session_t mTriggerSession; 418 const audio_session_t mListenerSession; 419 sync_event_callback_t mCallback; 420 const wp<RefBase> mCookie; 421 mutable Mutex mLock; 422 }; 423 424 sp<SyncEvent> createSyncEvent(AudioSystem::sync_event_t type, 425 audio_session_t triggerSession, 426 audio_session_t listenerSession, 427 sync_event_callback_t callBack, 428 const wp<RefBase>& cookie); 429 btNrecIsOff()430 bool btNrecIsOff() const { return mBtNrecIsOff.load(); } 431 lock()432 void lock() ACQUIRE(mLock) { mLock.lock(); } unlock()433 void unlock() RELEASE(mLock) { mLock.unlock(); } 434 435 private: 436 getMode()437 audio_mode_t getMode() const { return mMode; } 438 439 AudioFlinger() ANDROID_API; 440 virtual ~AudioFlinger(); 441 442 // call in any IAudioFlinger method that accesses mPrimaryHardwareDev initCheck()443 status_t initCheck() const { return mPrimaryHardwareDev == NULL ? 444 NO_INIT : NO_ERROR; } 445 446 // RefBase 447 virtual void onFirstRef(); 448 449 AudioHwDevice* findSuitableHwDev_l(audio_module_handle_t module, 450 audio_devices_t deviceType); 451 452 // Set kEnableExtendedChannels to true to enable greater than stereo output 453 // for the MixerThread and device sink. Number of channels allowed is 454 // FCC_2 <= channels <= AudioMixer::MAX_NUM_CHANNELS. 455 static const bool kEnableExtendedChannels = true; 456 457 // Returns true if channel mask is permitted for the PCM sink in the MixerThread isValidPcmSinkChannelMask(audio_channel_mask_t channelMask)458 static inline bool isValidPcmSinkChannelMask(audio_channel_mask_t channelMask) { 459 switch (audio_channel_mask_get_representation(channelMask)) { 460 case AUDIO_CHANNEL_REPRESENTATION_POSITION: { 461 // Haptic channel mask is only applicable for channel position mask. 462 const uint32_t channelCount = audio_channel_count_from_out_mask( 463 static_cast<audio_channel_mask_t>(channelMask & ~AUDIO_CHANNEL_HAPTIC_ALL)); 464 const uint32_t maxChannelCount = kEnableExtendedChannels 465 ? AudioMixer::MAX_NUM_CHANNELS : FCC_2; 466 if (channelCount < FCC_2 // mono is not supported at this time 467 || channelCount > maxChannelCount) { 468 return false; 469 } 470 // check that channelMask is the "canonical" one we expect for the channelCount. 471 return audio_channel_position_mask_is_out_canonical(channelMask); 472 } 473 case AUDIO_CHANNEL_REPRESENTATION_INDEX: 474 if (kEnableExtendedChannels) { 475 const uint32_t channelCount = audio_channel_count_from_out_mask(channelMask); 476 if (channelCount >= FCC_2 // mono is not supported at this time 477 && channelCount <= AudioMixer::MAX_NUM_CHANNELS) { 478 return true; 479 } 480 } 481 return false; 482 default: 483 return false; 484 } 485 } 486 487 // Set kEnableExtendedPrecision to true to use extended precision in MixerThread 488 static const bool kEnableExtendedPrecision = true; 489 490 // Returns true if format is permitted for the PCM sink in the MixerThread isValidPcmSinkFormat(audio_format_t format)491 static inline bool isValidPcmSinkFormat(audio_format_t format) { 492 switch (format) { 493 case AUDIO_FORMAT_PCM_16_BIT: 494 return true; 495 case AUDIO_FORMAT_PCM_FLOAT: 496 case AUDIO_FORMAT_PCM_24_BIT_PACKED: 497 case AUDIO_FORMAT_PCM_32_BIT: 498 case AUDIO_FORMAT_PCM_8_24_BIT: 499 return kEnableExtendedPrecision; 500 default: 501 return false; 502 } 503 } 504 505 // standby delay for MIXER and DUPLICATING playback threads is read from property 506 // ro.audio.flinger_standbytime_ms or defaults to kDefaultStandbyTimeInNsecs 507 static nsecs_t mStandbyTimeInNsecs; 508 509 // incremented by 2 when screen state changes, bit 0 == 1 means "off" 510 // AudioFlinger::setParameters() updates, other threads read w/o lock 511 static uint32_t mScreenState; 512 513 // Internal dump utilities. 514 static const int kDumpLockTimeoutNs = 1 * NANOS_PER_SECOND; 515 static bool dumpTryLock(Mutex& mutex); 516 void dumpPermissionDenial(int fd, const Vector<String16>& args); 517 void dumpClients(int fd, const Vector<String16>& args); 518 void dumpInternals(int fd, const Vector<String16>& args); 519 520 SimpleLog mThreadLog{16}; // 16 Thread history limit 521 522 class ThreadBase; 523 void dumpToThreadLog_l(const sp<ThreadBase> &thread); 524 525 // --- Client --- 526 class Client : public RefBase { 527 public: 528 Client(const sp<AudioFlinger>& audioFlinger, pid_t pid); 529 virtual ~Client(); 530 AllocatorFactory::ClientAllocator& allocator(); pid()531 pid_t pid() const { return mPid; } audioFlinger()532 sp<AudioFlinger> audioFlinger() const { return mAudioFlinger; } 533 534 private: 535 DISALLOW_COPY_AND_ASSIGN(Client); 536 537 const sp<AudioFlinger> mAudioFlinger; 538 const pid_t mPid; 539 AllocatorFactory::ClientAllocator mClientAllocator; 540 }; 541 542 // --- Notification Client --- 543 class NotificationClient : public IBinder::DeathRecipient { 544 public: 545 NotificationClient(const sp<AudioFlinger>& audioFlinger, 546 const sp<media::IAudioFlingerClient>& client, 547 pid_t pid, 548 uid_t uid); 549 virtual ~NotificationClient(); 550 audioFlingerClient()551 sp<media::IAudioFlingerClient> audioFlingerClient() const { return mAudioFlingerClient; } getPid()552 pid_t getPid() const { return mPid; } getUid()553 uid_t getUid() const { return mUid; } 554 555 // IBinder::DeathRecipient 556 virtual void binderDied(const wp<IBinder>& who); 557 558 private: 559 DISALLOW_COPY_AND_ASSIGN(NotificationClient); 560 561 const sp<AudioFlinger> mAudioFlinger; 562 const pid_t mPid; 563 const uid_t mUid; 564 const sp<media::IAudioFlingerClient> mAudioFlingerClient; 565 }; 566 567 // --- MediaLogNotifier --- 568 // Thread in charge of notifying MediaLogService to start merging. 569 // Receives requests from AudioFlinger's binder activity. It is used to reduce the amount of 570 // binder calls to MediaLogService in case of bursts of AudioFlinger binder calls. 571 class MediaLogNotifier : public Thread { 572 public: 573 MediaLogNotifier(); 574 575 // Requests a MediaLogService notification. It's ignored if there has recently been another 576 void requestMerge(); 577 private: 578 // Every iteration blocks waiting for a request, then interacts with MediaLogService to 579 // start merging. 580 // As every MediaLogService binder call is expensive, once it gets a request it ignores the 581 // following ones for a period of time. 582 virtual bool threadLoop() override; 583 584 bool mPendingRequests; 585 586 // Mutex and condition variable around mPendingRequests' value 587 Mutex mMutex; 588 Condition mCond; 589 590 // Duration of the sleep period after a processed request 591 static const int kPostTriggerSleepPeriod = 1000000; 592 }; 593 594 const sp<MediaLogNotifier> mMediaLogNotifier; 595 596 // This is a helper that is called during incoming binder calls. 597 // Requests media.log to start merging log buffers 598 void requestLogMerge(); 599 600 class TrackHandle; 601 class RecordHandle; 602 class RecordThread; 603 class PlaybackThread; 604 class MixerThread; 605 class DirectOutputThread; 606 class OffloadThread; 607 class DuplicatingThread; 608 class AsyncCallbackThread; 609 class BitPerfectThread; 610 class Track; 611 class RecordTrack; 612 class EffectBase; 613 class EffectModule; 614 class EffectHandle; 615 class EffectChain; 616 class DeviceEffectProxy; 617 class DeviceEffectManager; 618 class PatchPanel; 619 class DeviceEffectManagerCallback; 620 621 struct AudioStreamIn; 622 struct TeePatch; 623 using TeePatches = std::vector<TeePatch>; 624 625 626 struct stream_type_t { stream_type_tstream_type_t627 stream_type_t() 628 : volume(1.0f), 629 mute(false) 630 { 631 } 632 float volume; 633 bool mute; 634 }; 635 636 // Abstraction for the Audio Source for the RecordThread (HAL or PassthruPatchRecord). 637 struct Source 638 { 639 virtual ~Source() = default; 640 // The following methods have the same signatures as in StreamHalInterface. 641 virtual status_t read(void *buffer, size_t bytes, size_t *read) = 0; 642 virtual status_t getCapturePosition(int64_t *frames, int64_t *time) = 0; 643 virtual status_t standby() = 0; 644 }; 645 646 // --- PlaybackThread --- 647 #ifdef FLOAT_EFFECT_CHAIN 648 #define EFFECT_BUFFER_FORMAT AUDIO_FORMAT_PCM_FLOAT 649 using effect_buffer_t = float; 650 #else 651 #define EFFECT_BUFFER_FORMAT AUDIO_FORMAT_PCM_16_BIT 652 using effect_buffer_t = int16_t; 653 #endif 654 655 #include "Threads.h" 656 657 #include "PatchPanel.h" 658 659 #include "PatchCommandThread.h" 660 661 #include "Effects.h" 662 663 #include "DeviceEffectManager.h" 664 665 #include "MelReporter.h" 666 667 // Find io handle by session id. 668 // Preference is given to an io handle with a matching effect chain to session id. 669 // If none found, AUDIO_IO_HANDLE_NONE is returned. 670 template <typename T> findIoHandleBySessionId_l(audio_session_t sessionId,const T & threads)671 static audio_io_handle_t findIoHandleBySessionId_l( 672 audio_session_t sessionId, const T& threads) { 673 audio_io_handle_t io = AUDIO_IO_HANDLE_NONE; 674 675 for (size_t i = 0; i < threads.size(); i++) { 676 const uint32_t sessionType = threads.valueAt(i)->hasAudioSession(sessionId); 677 if (sessionType != 0) { 678 io = threads.keyAt(i); 679 if ((sessionType & AudioFlinger::ThreadBase::EFFECT_SESSION) != 0) { 680 break; // effect chain here. 681 } 682 } 683 } 684 return io; 685 } 686 687 // server side of the client's IAudioTrack 688 class TrackHandle : public android::media::BnAudioTrack { 689 public: 690 explicit TrackHandle(const sp<PlaybackThread::Track>& track); 691 virtual ~TrackHandle(); 692 693 binder::Status getCblk(std::optional<media::SharedFileRegion>* _aidl_return) override; 694 binder::Status start(int32_t* _aidl_return) override; 695 binder::Status stop() override; 696 binder::Status flush() override; 697 binder::Status pause() override; 698 binder::Status attachAuxEffect(int32_t effectId, int32_t* _aidl_return) override; 699 binder::Status setParameters(const std::string& keyValuePairs, 700 int32_t* _aidl_return) override; 701 binder::Status selectPresentation(int32_t presentationId, int32_t programId, 702 int32_t* _aidl_return) override; 703 binder::Status getTimestamp(media::AudioTimestampInternal* timestamp, 704 int32_t* _aidl_return) override; 705 binder::Status signal() override; 706 binder::Status applyVolumeShaper(const media::VolumeShaperConfiguration& configuration, 707 const media::VolumeShaperOperation& operation, 708 int32_t* _aidl_return) override; 709 binder::Status getVolumeShaperState( 710 int32_t id, 711 std::optional<media::VolumeShaperState>* _aidl_return) override; 712 binder::Status getDualMonoMode( 713 media::audio::common::AudioDualMonoMode* _aidl_return) override; 714 binder::Status setDualMonoMode( 715 media::audio::common::AudioDualMonoMode mode) override; 716 binder::Status getAudioDescriptionMixLevel(float* _aidl_return) override; 717 binder::Status setAudioDescriptionMixLevel(float leveldB) override; 718 binder::Status getPlaybackRateParameters( 719 media::audio::common::AudioPlaybackRate* _aidl_return) override; 720 binder::Status setPlaybackRateParameters( 721 const media::audio::common::AudioPlaybackRate& playbackRate) override; 722 723 private: 724 const sp<PlaybackThread::Track> mTrack; 725 }; 726 727 // server side of the client's IAudioRecord 728 class RecordHandle : public android::media::BnAudioRecord { 729 public: 730 explicit RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack); 731 virtual ~RecordHandle(); 732 virtual binder::Status start(int /*AudioSystem::sync_event_t*/ event, 733 int /*audio_session_t*/ triggerSession); 734 virtual binder::Status stop(); 735 virtual binder::Status getActiveMicrophones( 736 std::vector<media::MicrophoneInfoFw>* activeMicrophones); 737 virtual binder::Status setPreferredMicrophoneDirection( 738 int /*audio_microphone_direction_t*/ direction); 739 virtual binder::Status setPreferredMicrophoneFieldDimension(float zoom); 740 virtual binder::Status shareAudioHistory(const std::string& sharedAudioPackageName, 741 int64_t sharedAudioStartMs); 742 743 private: 744 const sp<RecordThread::RecordTrack> mRecordTrack; 745 746 // for use from destructor 747 void stop_nonvirtual(); 748 }; 749 750 // Mmap stream control interface implementation. Each MmapThreadHandle controls one 751 // MmapPlaybackThread or MmapCaptureThread instance. 752 class MmapThreadHandle : public MmapStreamInterface { 753 public: 754 explicit MmapThreadHandle(const sp<MmapThread>& thread); 755 virtual ~MmapThreadHandle(); 756 757 // MmapStreamInterface virtuals 758 virtual status_t createMmapBuffer(int32_t minSizeFrames, 759 struct audio_mmap_buffer_info *info); 760 virtual status_t getMmapPosition(struct audio_mmap_position *position); 761 virtual status_t getExternalPosition(uint64_t *position, int64_t *timeNanos); 762 virtual status_t start(const AudioClient& client, 763 const audio_attributes_t *attr, 764 audio_port_handle_t *handle); 765 virtual status_t stop(audio_port_handle_t handle); 766 virtual status_t standby(); 767 status_t reportData(const void* buffer, size_t frameCount) override; 768 769 private: 770 const sp<MmapThread> mThread; 771 }; 772 773 ThreadBase *checkThread_l(audio_io_handle_t ioHandle) const; 774 sp<AudioFlinger::ThreadBase> checkOutputThread_l(audio_io_handle_t ioHandle) const 775 REQUIRES(mLock); 776 PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const; 777 MixerThread *checkMixerThread_l(audio_io_handle_t output) const; 778 RecordThread *checkRecordThread_l(audio_io_handle_t input) const; 779 MmapThread *checkMmapThread_l(audio_io_handle_t io) const; 780 VolumeInterface *getVolumeInterface_l(audio_io_handle_t output) const; 781 Vector <VolumeInterface *> getAllVolumeInterfaces_l() const; 782 783 sp<ThreadBase> openInput_l(audio_module_handle_t module, 784 audio_io_handle_t *input, 785 audio_config_t *config, 786 audio_devices_t device, 787 const char* address, 788 audio_source_t source, 789 audio_input_flags_t flags, 790 audio_devices_t outputDevice, 791 const String8& outputDeviceAddress); 792 sp<ThreadBase> openOutput_l(audio_module_handle_t module, 793 audio_io_handle_t *output, 794 audio_config_t *halConfig, 795 audio_config_base_t *mixerConfig, 796 audio_devices_t deviceType, 797 const String8& address, 798 audio_output_flags_t flags); 799 800 void closeOutputFinish(const sp<PlaybackThread>& thread); 801 void closeInputFinish(const sp<RecordThread>& thread); 802 803 // no range check, AudioFlinger::mLock held streamMute_l(audio_stream_type_t stream)804 bool streamMute_l(audio_stream_type_t stream) const 805 { return mStreamTypes[stream].mute; } 806 void ioConfigChanged(audio_io_config_event_t event, 807 const sp<AudioIoDescriptor>& ioDesc, 808 pid_t pid = 0); 809 void onSupportedLatencyModesChanged( 810 audio_io_handle_t output, const std::vector<audio_latency_mode_t>& modes); 811 812 // Allocate an audio_unique_id_t. 813 // Specific types are audio_io_handle_t, audio_session_t, effect ID (int), 814 // audio_module_handle_t, and audio_patch_handle_t. 815 // They all share the same ID space, but the namespaces are actually independent 816 // because there are separate KeyedVectors for each kind of ID. 817 // The return value is cast to the specific type depending on how the ID will be used. 818 // FIXME This API does not handle rollover to zero (for unsigned IDs), 819 // or from positive to negative (for signed IDs). 820 // Thus it may fail by returning an ID of the wrong sign, 821 // or by returning a non-unique ID. 822 // This is the internal API. For the binder API see newAudioUniqueId(). 823 audio_unique_id_t nextUniqueId(audio_unique_id_use_t use); 824 825 status_t moveEffectChain_l(audio_session_t sessionId, 826 PlaybackThread *srcThread, 827 PlaybackThread *dstThread); 828 829 status_t moveAuxEffectToIo(int EffectId, 830 const sp<PlaybackThread>& dstThread, 831 sp<PlaybackThread> *srcThread); 832 833 // return thread associated with primary hardware device, or NULL 834 PlaybackThread *primaryPlaybackThread_l() const; 835 DeviceTypeSet primaryOutputDevice_l() const; 836 837 // return the playback thread with smallest HAL buffer size, and prefer fast 838 PlaybackThread *fastPlaybackThread_l() const; 839 840 sp<ThreadBase> getEffectThread_l(audio_session_t sessionId, int effectId); 841 842 ThreadBase *hapticPlaybackThread_l() const; 843 844 void updateSecondaryOutputsForTrack_l( 845 PlaybackThread::Track* track, 846 PlaybackThread* thread, 847 const std::vector<audio_io_handle_t>& secondaryOutputs) const; 848 849 850 void removeClient_l(pid_t pid); 851 void removeNotificationClient(pid_t pid); 852 bool isNonOffloadableGlobalEffectEnabled_l(); 853 void onNonOffloadableGlobalEffectEnable(); 854 bool isSessionAcquired_l(audio_session_t audioSession); 855 856 // Store an effect chain to mOrphanEffectChains keyed vector. 857 // Called when a thread exits and effects are still attached to it. 858 // If effects are later created on the same session, they will reuse the same 859 // effect chain and same instances in the effect library. 860 // return ALREADY_EXISTS if a chain with the same session already exists in 861 // mOrphanEffectChains. Note that this should never happen as there is only one 862 // chain for a given session and it is attached to only one thread at a time. 863 status_t putOrphanEffectChain_l(const sp<EffectChain>& chain); 864 // Get an effect chain for the specified session in mOrphanEffectChains and remove 865 // it if found. Returns 0 if not found (this is the most common case). 866 sp<EffectChain> getOrphanEffectChain_l(audio_session_t session); 867 // Called when the last effect handle on an effect instance is removed. If this 868 // effect belongs to an effect chain in mOrphanEffectChains, the chain is updated 869 // and removed from mOrphanEffectChains if it does not contain any effect. 870 // Return true if the effect was found in mOrphanEffectChains, false otherwise. 871 bool updateOrphanEffectChains(const sp<EffectModule>& effect); 872 873 std::vector< sp<EffectModule> > purgeStaleEffects_l(); 874 875 void broadcastParametersToRecordThreads_l(const String8& keyValuePairs); 876 void updateOutDevicesForRecordThreads_l(const DeviceDescriptorBaseVector& devices); 877 void forwardParametersToDownstreamPatches_l( 878 audio_io_handle_t upStream, const String8& keyValuePairs, 879 const std::function<bool(const sp<PlaybackThread>&)>& useThread = nullptr); 880 881 // AudioStreamIn is immutable, so their fields are const. 882 // For emphasis, we could also make all pointers to them be "const *", 883 // but that would clutter the code unnecessarily. 884 885 struct AudioStreamIn : public Source { 886 AudioHwDevice* const audioHwDev; 887 sp<StreamInHalInterface> stream; 888 audio_input_flags_t flags; 889 hwDevAudioStreamIn890 sp<DeviceHalInterface> hwDev() const { return audioHwDev->hwDevice(); } 891 AudioStreamInAudioStreamIn892 AudioStreamIn(AudioHwDevice *dev, const sp<StreamInHalInterface>& in, 893 audio_input_flags_t flags) : 894 audioHwDev(dev), stream(in), flags(flags) {} readAudioStreamIn895 status_t read(void *buffer, size_t bytes, size_t *read) override { 896 return stream->read(buffer, bytes, read); 897 } getCapturePositionAudioStreamIn898 status_t getCapturePosition(int64_t *frames, int64_t *time) override { 899 return stream->getCapturePosition(frames, time); 900 } standbyAudioStreamIn901 status_t standby() override { return stream->standby(); } 902 }; 903 904 struct TeePatch { 905 sp<RecordThread::PatchRecord> patchRecord; 906 sp<PlaybackThread::PatchTrack> patchTrack; 907 }; 908 909 // for mAudioSessionRefs only 910 struct AudioSessionRef { AudioSessionRefAudioSessionRef911 AudioSessionRef(audio_session_t sessionid, pid_t pid, uid_t uid) : 912 mSessionid(sessionid), mPid(pid), mUid(uid), mCnt(1) {} 913 const audio_session_t mSessionid; 914 const pid_t mPid; 915 const uid_t mUid; 916 int mCnt; 917 }; 918 919 mutable Mutex mLock; 920 // protects mClients and mNotificationClients. 921 // must be locked after mLock and ThreadBase::mLock if both must be locked 922 // avoids acquiring AudioFlinger::mLock from inside thread loop. 923 mutable Mutex mClientLock; 924 // protected by mClientLock 925 DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client() 926 927 mutable Mutex mHardwareLock; 928 // NOTE: If both mLock and mHardwareLock mutexes must be held, 929 // always take mLock before mHardwareLock 930 931 // guarded by mHardwareLock 932 AudioHwDevice* mPrimaryHardwareDev; 933 DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*> mAudioHwDevs; 934 935 // These two fields are immutable after onFirstRef(), so no lock needed to access 936 sp<DevicesFactoryHalInterface> mDevicesFactoryHal; 937 sp<DevicesFactoryHalCallback> mDevicesFactoryHalCallback; 938 939 // for dump, indicates which hardware operation is currently in progress (but not stream ops) 940 enum hardware_call_state { 941 AUDIO_HW_IDLE = 0, // no operation in progress 942 AUDIO_HW_INIT, // init_check 943 AUDIO_HW_OUTPUT_OPEN, // open_output_stream 944 AUDIO_HW_OUTPUT_CLOSE, // unused 945 AUDIO_HW_INPUT_OPEN, // unused 946 AUDIO_HW_INPUT_CLOSE, // unused 947 AUDIO_HW_STANDBY, // unused 948 AUDIO_HW_SET_MASTER_VOLUME, // set_master_volume 949 AUDIO_HW_GET_ROUTING, // unused 950 AUDIO_HW_SET_ROUTING, // unused 951 AUDIO_HW_GET_MODE, // unused 952 AUDIO_HW_SET_MODE, // set_mode 953 AUDIO_HW_GET_MIC_MUTE, // get_mic_mute 954 AUDIO_HW_SET_MIC_MUTE, // set_mic_mute 955 AUDIO_HW_SET_VOICE_VOLUME, // set_voice_volume 956 AUDIO_HW_SET_PARAMETER, // set_parameters 957 AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size 958 AUDIO_HW_GET_MASTER_VOLUME, // get_master_volume 959 AUDIO_HW_GET_PARAMETER, // get_parameters 960 AUDIO_HW_SET_MASTER_MUTE, // set_master_mute 961 AUDIO_HW_GET_MASTER_MUTE, // get_master_mute 962 AUDIO_HW_GET_MICROPHONES, // getMicrophones 963 AUDIO_HW_SET_CONNECTED_STATE, // setConnectedState 964 AUDIO_HW_SET_SIMULATE_CONNECTIONS, // setSimulateDeviceConnections 965 }; 966 967 mutable hardware_call_state mHardwareStatus; // for dump only 968 969 970 DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads; 971 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 972 973 // member variables below are protected by mLock 974 float mMasterVolume; 975 bool mMasterMute; 976 float mMasterBalance = 0.f; 977 // end of variables protected by mLock 978 979 DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads; 980 981 // protected by mClientLock 982 DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients; 983 984 // updated by atomic_fetch_add_explicit 985 volatile atomic_uint_fast32_t mNextUniqueIds[AUDIO_UNIQUE_ID_USE_MAX]; 986 987 audio_mode_t mMode; 988 std::atomic_bool mBtNrecIsOff; 989 990 // protected by mLock 991 Vector<AudioSessionRef*> mAudioSessionRefs; 992 993 float masterVolume_l() const; 994 float getMasterBalance_l() const; 995 bool masterMute_l() const; 996 AudioHwDevice* loadHwModule_l(const char *name); 997 998 Vector < sp<SyncEvent> > mPendingSyncEvents; // sync events awaiting for a session 999 // to be created 1000 1001 // Effect chains without a valid thread 1002 DefaultKeyedVector< audio_session_t , sp<EffectChain> > mOrphanEffectChains; 1003 1004 // list of sessions for which a valid HW A/V sync ID was retrieved from the HAL 1005 DefaultKeyedVector< audio_session_t , audio_hw_sync_t >mHwAvSyncIds; 1006 1007 // list of MMAP stream control threads. Those threads allow for wake lock, routing 1008 // and volume control for activity on the associated MMAP stream at the HAL. 1009 // Audio data transfer is directly handled by the client creating the MMAP stream 1010 DefaultKeyedVector< audio_io_handle_t, sp<MmapThread> > mMmapThreads; 1011 1012 private: 1013 sp<Client> registerPid(pid_t pid); // always returns non-0 1014 1015 // for use from destructor 1016 status_t closeOutput_nonvirtual(audio_io_handle_t output); 1017 void closeThreadInternal_l(const sp<PlaybackThread>& thread); 1018 status_t closeInput_nonvirtual(audio_io_handle_t input); 1019 void closeThreadInternal_l(const sp<RecordThread>& thread); 1020 void setAudioHwSyncForSession_l(PlaybackThread *thread, audio_session_t sessionId); 1021 1022 status_t checkStreamType(audio_stream_type_t stream) const; 1023 1024 void filterReservedParameters(String8& keyValuePairs, uid_t callingUid); 1025 void logFilteredParameters(size_t originalKVPSize, const String8& originalKVPs, 1026 size_t rejectedKVPSize, const String8& rejectedKVPs, 1027 uid_t callingUid); 1028 1029 sp<IAudioManager> getOrCreateAudioManager(); 1030 1031 public: 1032 // These methods read variables atomically without mLock, 1033 // though the variables are updated with mLock. isLowRamDevice()1034 bool isLowRamDevice() const { return mIsLowRamDevice; } 1035 size_t getClientSharedHeapSize() const; 1036 1037 private: 1038 std::atomic<bool> mIsLowRamDevice; 1039 bool mIsDeviceTypeKnown; 1040 int64_t mTotalMemory; 1041 std::atomic<size_t> mClientSharedHeapSize; 1042 static constexpr size_t kMinimumClientSharedHeapSizeBytes = 1024 * 1024; // 1MB 1043 1044 nsecs_t mGlobalEffectEnableTime; // when a global effect was last enabled 1045 1046 // protected by mLock 1047 PatchPanel mPatchPanel; 1048 sp<EffectsFactoryHalInterface> mEffectsFactoryHal; 1049 1050 const sp<PatchCommandThread> mPatchCommandThread; 1051 sp<DeviceEffectManager> mDeviceEffectManager; 1052 sp<MelReporter> mMelReporter; 1053 1054 bool mSystemReady; 1055 std::atomic_bool mAudioPolicyReady{}; 1056 1057 mediautils::UidInfo mUidInfo; 1058 1059 SimpleLog mRejectedSetParameterLog; 1060 SimpleLog mAppSetParameterLog; 1061 SimpleLog mSystemSetParameterLog; 1062 1063 std::vector<media::AudioVibratorInfo> mAudioVibratorInfos; 1064 1065 static inline constexpr const char *mMetricsId = AMEDIAMETRICS_KEY_AUDIO_FLINGER; 1066 1067 // Keep in sync with java definition in media/java/android/media/AudioRecord.java 1068 static constexpr int32_t kMaxSharedAudioHistoryMs = 5000; 1069 1070 std::map<media::audio::common::AudioMMapPolicyType, 1071 std::vector<media::audio::common::AudioMMapPolicyInfo>> mPolicyInfos; 1072 int32_t mAAudioBurstsPerBuffer = 0; 1073 int32_t mAAudioHwBurstMinMicros = 0; 1074 1075 /** Interface for interacting with the AudioService. */ 1076 mediautils::atomic_sp<IAudioManager> mAudioManager; 1077 1078 // Bluetooth Variable latency control logic is enabled or disabled 1079 std::atomic_bool mBluetoothLatencyModesEnabled; 1080 }; 1081 1082 #undef INCLUDING_FROM_AUDIOFLINGER_H 1083 1084 std::string formatToString(audio_format_t format); 1085 std::string inputFlagsToString(audio_input_flags_t flags); 1086 std::string outputFlagsToString(audio_output_flags_t flags); 1087 std::string devicesToString(audio_devices_t devices); 1088 const char *sourceToString(audio_source_t source); 1089 1090 // ---------------------------------------------------------------------------- 1091 1092 } // namespace android 1093 1094 #endif // ANDROID_AUDIO_FLINGER_H 1095