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/ServiceUtilities.h> 78 #include <mediautils/Synchronization.h> 79 #include <mediautils/ThreadSnapshot.h> 80 81 #include <audio_utils/clock.h> 82 #include <audio_utils/FdToString.h> 83 #include <audio_utils/LinearMap.h> 84 #include <audio_utils/SimpleLog.h> 85 #include <audio_utils/TimestampVerifier.h> 86 87 #include "FastCapture.h" 88 #include "FastMixer.h" 89 #include <media/nbaio/NBAIO.h> 90 #include "AudioWatchdog.h" 91 #include "AudioStreamOut.h" 92 #include "SpdifStreamOut.h" 93 #include "AudioHwDevice.h" 94 #include "NBAIO_Tee.h" 95 #include "ThreadMetrics.h" 96 #include "TrackMetrics.h" 97 98 #include <android/os/IPowerManager.h> 99 100 #include <media/nblog/NBLog.h> 101 #include <private/media/AudioEffectShared.h> 102 #include <private/media/AudioTrackShared.h> 103 104 #include <vibrator/ExternalVibration.h> 105 #include <vibrator/ExternalVibrationUtils.h> 106 107 #include "android/media/BnAudioRecord.h" 108 #include "android/media/BnEffect.h" 109 110 namespace android { 111 112 class AudioMixer; 113 class AudioBuffer; 114 class AudioResampler; 115 class DeviceHalInterface; 116 class DevicesFactoryHalCallback; 117 class DevicesFactoryHalInterface; 118 class EffectsFactoryHalInterface; 119 class FastMixer; 120 class PassthruBufferProvider; 121 class RecordBufferConverter; 122 class ServerProxy; 123 124 // ---------------------------------------------------------------------------- 125 126 static const nsecs_t kDefaultStandbyTimeInNsecs = seconds(3); 127 128 #define INCLUDING_FROM_AUDIOFLINGER_H 129 130 using android::content::AttributionSourceState; 131 132 class AudioFlinger : public AudioFlingerServerAdapter::Delegate 133 { 134 public: 135 static void instantiate() ANDROID_API; 136 137 static AttributionSourceState checkAttributionSourcePackage( 138 const AttributionSourceState& attributionSource); 139 140 status_t dump(int fd, const Vector<String16>& args) override; 141 142 // IAudioFlinger interface, in binder opcode order 143 status_t createTrack(const media::CreateTrackRequest& input, 144 media::CreateTrackResponse& output) override; 145 146 status_t createRecord(const media::CreateRecordRequest& input, 147 media::CreateRecordResponse& output) override; 148 149 virtual uint32_t sampleRate(audio_io_handle_t ioHandle) const; 150 virtual audio_format_t format(audio_io_handle_t output) const; 151 virtual size_t frameCount(audio_io_handle_t ioHandle) const; 152 virtual size_t frameCountHAL(audio_io_handle_t ioHandle) const; 153 virtual uint32_t latency(audio_io_handle_t output) const; 154 155 virtual status_t setMasterVolume(float value); 156 virtual status_t setMasterMute(bool muted); 157 158 virtual float masterVolume() const; 159 virtual bool masterMute() const; 160 161 // Balance value must be within -1.f (left only) to 1.f (right only) inclusive. 162 status_t setMasterBalance(float balance) override; 163 status_t getMasterBalance(float *balance) const override; 164 165 virtual status_t setStreamVolume(audio_stream_type_t stream, float value, 166 audio_io_handle_t output); 167 virtual status_t setStreamMute(audio_stream_type_t stream, bool muted); 168 169 virtual float streamVolume(audio_stream_type_t stream, 170 audio_io_handle_t output) const; 171 virtual bool streamMute(audio_stream_type_t stream) const; 172 173 virtual status_t setMode(audio_mode_t mode); 174 175 virtual status_t setMicMute(bool state); 176 virtual bool getMicMute() const; 177 178 virtual void setRecordSilenced(audio_port_handle_t portId, bool silenced); 179 180 virtual status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs); 181 virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) const; 182 183 virtual void registerClient(const sp<media::IAudioFlingerClient>& client); 184 185 virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, 186 audio_channel_mask_t channelMask) const; 187 188 virtual status_t openOutput(const media::OpenOutputRequest& request, 189 media::OpenOutputResponse* response); 190 191 virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1, 192 audio_io_handle_t output2); 193 194 virtual status_t closeOutput(audio_io_handle_t output); 195 196 virtual status_t suspendOutput(audio_io_handle_t output); 197 198 virtual status_t restoreOutput(audio_io_handle_t output); 199 200 virtual status_t openInput(const media::OpenInputRequest& request, 201 media::OpenInputResponse* response); 202 203 virtual status_t closeInput(audio_io_handle_t input); 204 205 virtual status_t invalidateStream(audio_stream_type_t stream); 206 207 virtual status_t setVoiceVolume(float volume); 208 209 virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 210 audio_io_handle_t output) const; 211 212 virtual uint32_t getInputFramesLost(audio_io_handle_t ioHandle) const; 213 214 // This is the binder API. For the internal API see nextUniqueId(). 215 virtual audio_unique_id_t newAudioUniqueId(audio_unique_id_use_t use); 216 217 void acquireAudioSessionId(audio_session_t audioSession, pid_t pid, uid_t uid) override; 218 219 virtual void releaseAudioSessionId(audio_session_t audioSession, pid_t pid); 220 221 virtual status_t queryNumberEffects(uint32_t *numEffects) const; 222 223 virtual status_t queryEffect(uint32_t index, effect_descriptor_t *descriptor) const; 224 225 virtual status_t getEffectDescriptor(const effect_uuid_t *pUuid, 226 const effect_uuid_t *pTypeUuid, 227 uint32_t preferredTypeFlag, 228 effect_descriptor_t *descriptor) const; 229 230 virtual status_t createEffect(const media::CreateEffectRequest& request, 231 media::CreateEffectResponse* response); 232 233 virtual status_t moveEffects(audio_session_t sessionId, audio_io_handle_t srcOutput, 234 audio_io_handle_t dstOutput); 235 236 void setEffectSuspended(int effectId, 237 audio_session_t sessionId, 238 bool suspended) override; 239 240 virtual audio_module_handle_t loadHwModule(const char *name); 241 242 virtual uint32_t getPrimaryOutputSamplingRate(); 243 virtual size_t getPrimaryOutputFrameCount(); 244 245 virtual status_t setLowRamDevice(bool isLowRamDevice, int64_t totalMemory) override; 246 247 /* List available audio ports and their attributes */ 248 virtual status_t listAudioPorts(unsigned int *num_ports, 249 struct audio_port *ports); 250 251 /* Get attributes for a given audio port */ 252 virtual status_t getAudioPort(struct audio_port_v7 *port); 253 254 /* Create an audio patch between several source and sink ports */ 255 virtual status_t createAudioPatch(const struct audio_patch *patch, 256 audio_patch_handle_t *handle); 257 258 /* Release an audio patch */ 259 virtual status_t releaseAudioPatch(audio_patch_handle_t handle); 260 261 /* List existing audio patches */ 262 virtual status_t listAudioPatches(unsigned int *num_patches, 263 struct audio_patch *patches); 264 265 /* Set audio port configuration */ 266 virtual status_t setAudioPortConfig(const struct audio_port_config *config); 267 268 /* Get the HW synchronization source used for an audio session */ 269 virtual audio_hw_sync_t getAudioHwSyncForSession(audio_session_t sessionId); 270 271 /* Indicate JAVA services are ready (scheduling, power management ...) */ 272 virtual status_t systemReady(); audioPolicyReady()273 virtual status_t audioPolicyReady() { mAudioPolicyReady.store(true); return NO_ERROR; } isAudioPolicyReady()274 bool isAudioPolicyReady() const { return mAudioPolicyReady.load(); } 275 276 277 virtual status_t getMicrophones(std::vector<media::MicrophoneInfo> *microphones); 278 279 virtual status_t setAudioHalPids(const std::vector<pid_t>& pids); 280 281 virtual status_t setVibratorInfos(const std::vector<media::AudioVibratorInfo>& vibratorInfos); 282 283 virtual status_t updateSecondaryOutputs( 284 const TrackSecondaryOutputsMap& trackSecondaryOutputs); 285 286 virtual status_t getMmapPolicyInfos( 287 media::audio::common::AudioMMapPolicyType policyType, 288 std::vector<media::audio::common::AudioMMapPolicyInfo> *policyInfos); 289 290 virtual int32_t getAAudioMixerBurstCount(); 291 292 virtual int32_t getAAudioHardwareBurstMinUsec(); 293 294 virtual status_t setDeviceConnectedState(const struct audio_port_v7 *port, bool connected); 295 296 virtual status_t setRequestedLatencyMode( 297 audio_io_handle_t output, audio_latency_mode_t mode); 298 299 virtual status_t getSupportedLatencyModes(audio_io_handle_t output, 300 std::vector<audio_latency_mode_t>* modes); 301 302 status_t onTransactWrapper(TransactionCode code, const Parcel& data, uint32_t flags, 303 const std::function<status_t()>& delegate) override; 304 305 // end of IAudioFlinger interface 306 307 sp<NBLog::Writer> newWriter_l(size_t size, const char *name); 308 void unregisterWriter(const sp<NBLog::Writer>& writer); 309 sp<EffectsFactoryHalInterface> getEffectsFactory(); 310 311 status_t openMmapStream(MmapStreamInterface::stream_direction_t direction, 312 const audio_attributes_t *attr, 313 audio_config_base_t *config, 314 const AudioClient& client, 315 audio_port_handle_t *deviceId, 316 audio_session_t *sessionId, 317 const sp<MmapStreamCallback>& callback, 318 sp<MmapStreamInterface>& interface, 319 audio_port_handle_t *handle); 320 321 static int onExternalVibrationStart(const sp<os::ExternalVibration>& externalVibration); 322 static void onExternalVibrationStop(const sp<os::ExternalVibration>& externalVibration); 323 324 status_t addEffectToHal(audio_port_handle_t deviceId, 325 audio_module_handle_t hwModuleId, sp<EffectHalInterface> effect); 326 status_t removeEffectFromHal(audio_port_handle_t deviceId, 327 audio_module_handle_t hwModuleId, sp<EffectHalInterface> effect); 328 329 void updateDownStreamPatches_l(const struct audio_patch *patch, 330 const std::set<audio_io_handle_t> streams); 331 332 std::optional<media::AudioVibratorInfo> getDefaultVibratorInfo_l(); 333 334 private: 335 // FIXME The 400 is temporarily too high until a leak of writers in media.log is fixed. 336 static const size_t kLogMemorySize = 400 * 1024; 337 sp<MemoryDealer> mLogMemoryDealer; // == 0 when NBLog is disabled 338 // When a log writer is unregistered, it is done lazily so that media.log can continue to see it 339 // for as long as possible. The memory is only freed when it is needed for another log writer. 340 Vector< sp<NBLog::Writer> > mUnregisteredWriters; 341 Mutex mUnregisteredWritersLock; 342 343 public: 344 // Life cycle of gAudioFlinger and AudioFlinger: 345 // 346 // AudioFlinger is created once and survives until audioserver crashes 347 // irrespective of sp<> and wp<> as it is refcounted by ServiceManager and we 348 // don't issue a ServiceManager::tryUnregisterService(). 349 // 350 // gAudioFlinger is an atomic pointer set on AudioFlinger::onFirstRef(). 351 // After this is set, it is safe to obtain a wp<> or sp<> from it as the 352 // underlying object does not go away. 353 // 354 // Note: For most inner classes, it is acceptable to hold a reference to the outer 355 // AudioFlinger instance as creation requires AudioFlinger to exist in the first place. 356 // 357 // An atomic here ensures underlying writes have completed before setting 358 // the pointer. Access by memory_order_seq_cst. 359 // 360 361 static inline std::atomic<AudioFlinger *> gAudioFlinger = nullptr; 362 363 class SyncEvent; 364 365 typedef void (*sync_event_callback_t)(const wp<SyncEvent>& event) ; 366 367 class SyncEvent : public RefBase { 368 public: SyncEvent(AudioSystem::sync_event_t type,audio_session_t triggerSession,audio_session_t listenerSession,sync_event_callback_t callBack,wp<RefBase> cookie)369 SyncEvent(AudioSystem::sync_event_t type, 370 audio_session_t triggerSession, 371 audio_session_t listenerSession, 372 sync_event_callback_t callBack, 373 wp<RefBase> cookie) 374 : mType(type), mTriggerSession(triggerSession), mListenerSession(listenerSession), 375 mCallback(callBack), mCookie(cookie) 376 {} 377 ~SyncEvent()378 virtual ~SyncEvent() {} 379 trigger()380 void trigger() { 381 Mutex::Autolock _l(mLock); 382 if (mCallback) mCallback(wp<SyncEvent>(this)); 383 } isCancelled()384 bool isCancelled() const { Mutex::Autolock _l(mLock); return (mCallback == NULL); } cancel()385 void cancel() { Mutex::Autolock _l(mLock); mCallback = NULL; } type()386 AudioSystem::sync_event_t type() const { return mType; } triggerSession()387 audio_session_t triggerSession() const { return mTriggerSession; } listenerSession()388 audio_session_t listenerSession() const { return mListenerSession; } cookie()389 wp<RefBase> cookie() const { return mCookie; } 390 391 private: 392 const AudioSystem::sync_event_t mType; 393 const audio_session_t mTriggerSession; 394 const audio_session_t mListenerSession; 395 sync_event_callback_t mCallback; 396 const wp<RefBase> mCookie; 397 mutable Mutex mLock; 398 }; 399 400 sp<SyncEvent> createSyncEvent(AudioSystem::sync_event_t type, 401 audio_session_t triggerSession, 402 audio_session_t listenerSession, 403 sync_event_callback_t callBack, 404 const wp<RefBase>& cookie); 405 btNrecIsOff()406 bool btNrecIsOff() const { return mBtNrecIsOff.load(); } 407 408 409 private: 410 getMode()411 audio_mode_t getMode() const { return mMode; } 412 413 AudioFlinger() ANDROID_API; 414 virtual ~AudioFlinger(); 415 416 // call in any IAudioFlinger method that accesses mPrimaryHardwareDev initCheck()417 status_t initCheck() const { return mPrimaryHardwareDev == NULL ? 418 NO_INIT : NO_ERROR; } 419 420 // RefBase 421 virtual void onFirstRef(); 422 423 AudioHwDevice* findSuitableHwDev_l(audio_module_handle_t module, 424 audio_devices_t deviceType); 425 426 // Set kEnableExtendedChannels to true to enable greater than stereo output 427 // for the MixerThread and device sink. Number of channels allowed is 428 // FCC_2 <= channels <= AudioMixer::MAX_NUM_CHANNELS. 429 static const bool kEnableExtendedChannels = true; 430 431 // Returns true if channel mask is permitted for the PCM sink in the MixerThread isValidPcmSinkChannelMask(audio_channel_mask_t channelMask)432 static inline bool isValidPcmSinkChannelMask(audio_channel_mask_t channelMask) { 433 switch (audio_channel_mask_get_representation(channelMask)) { 434 case AUDIO_CHANNEL_REPRESENTATION_POSITION: { 435 // Haptic channel mask is only applicable for channel position mask. 436 const uint32_t channelCount = audio_channel_count_from_out_mask( 437 static_cast<audio_channel_mask_t>(channelMask & ~AUDIO_CHANNEL_HAPTIC_ALL)); 438 const uint32_t maxChannelCount = kEnableExtendedChannels 439 ? AudioMixer::MAX_NUM_CHANNELS : FCC_2; 440 if (channelCount < FCC_2 // mono is not supported at this time 441 || channelCount > maxChannelCount) { 442 return false; 443 } 444 // check that channelMask is the "canonical" one we expect for the channelCount. 445 return audio_channel_position_mask_is_out_canonical(channelMask); 446 } 447 case AUDIO_CHANNEL_REPRESENTATION_INDEX: 448 if (kEnableExtendedChannels) { 449 const uint32_t channelCount = audio_channel_count_from_out_mask(channelMask); 450 if (channelCount >= FCC_2 // mono is not supported at this time 451 && channelCount <= AudioMixer::MAX_NUM_CHANNELS) { 452 return true; 453 } 454 } 455 return false; 456 default: 457 return false; 458 } 459 } 460 461 // Set kEnableExtendedPrecision to true to use extended precision in MixerThread 462 static const bool kEnableExtendedPrecision = true; 463 464 // Returns true if format is permitted for the PCM sink in the MixerThread isValidPcmSinkFormat(audio_format_t format)465 static inline bool isValidPcmSinkFormat(audio_format_t format) { 466 switch (format) { 467 case AUDIO_FORMAT_PCM_16_BIT: 468 return true; 469 case AUDIO_FORMAT_PCM_FLOAT: 470 case AUDIO_FORMAT_PCM_24_BIT_PACKED: 471 case AUDIO_FORMAT_PCM_32_BIT: 472 case AUDIO_FORMAT_PCM_8_24_BIT: 473 return kEnableExtendedPrecision; 474 default: 475 return false; 476 } 477 } 478 479 // standby delay for MIXER and DUPLICATING playback threads is read from property 480 // ro.audio.flinger_standbytime_ms or defaults to kDefaultStandbyTimeInNsecs 481 static nsecs_t mStandbyTimeInNsecs; 482 483 // incremented by 2 when screen state changes, bit 0 == 1 means "off" 484 // AudioFlinger::setParameters() updates, other threads read w/o lock 485 static uint32_t mScreenState; 486 487 // Internal dump utilities. 488 static const int kDumpLockTimeoutNs = 1 * NANOS_PER_SECOND; 489 static bool dumpTryLock(Mutex& mutex); 490 void dumpPermissionDenial(int fd, const Vector<String16>& args); 491 void dumpClients(int fd, const Vector<String16>& args); 492 void dumpInternals(int fd, const Vector<String16>& args); 493 494 SimpleLog mThreadLog{16}; // 16 Thread history limit 495 496 class ThreadBase; 497 void dumpToThreadLog_l(const sp<ThreadBase> &thread); 498 499 // --- Client --- 500 class Client : public RefBase { 501 public: 502 Client(const sp<AudioFlinger>& audioFlinger, pid_t pid); 503 virtual ~Client(); 504 sp<MemoryDealer> heap() const; pid()505 pid_t pid() const { return mPid; } audioFlinger()506 sp<AudioFlinger> audioFlinger() const { return mAudioFlinger; } 507 508 private: 509 DISALLOW_COPY_AND_ASSIGN(Client); 510 511 const sp<AudioFlinger> mAudioFlinger; 512 sp<MemoryDealer> mMemoryDealer; 513 const pid_t mPid; 514 }; 515 516 // --- Notification Client --- 517 class NotificationClient : public IBinder::DeathRecipient { 518 public: 519 NotificationClient(const sp<AudioFlinger>& audioFlinger, 520 const sp<media::IAudioFlingerClient>& client, 521 pid_t pid, 522 uid_t uid); 523 virtual ~NotificationClient(); 524 audioFlingerClient()525 sp<media::IAudioFlingerClient> audioFlingerClient() const { return mAudioFlingerClient; } getPid()526 pid_t getPid() const { return mPid; } getUid()527 uid_t getUid() const { return mUid; } 528 529 // IBinder::DeathRecipient 530 virtual void binderDied(const wp<IBinder>& who); 531 532 private: 533 DISALLOW_COPY_AND_ASSIGN(NotificationClient); 534 535 const sp<AudioFlinger> mAudioFlinger; 536 const pid_t mPid; 537 const uid_t mUid; 538 const sp<media::IAudioFlingerClient> mAudioFlingerClient; 539 }; 540 541 // --- MediaLogNotifier --- 542 // Thread in charge of notifying MediaLogService to start merging. 543 // Receives requests from AudioFlinger's binder activity. It is used to reduce the amount of 544 // binder calls to MediaLogService in case of bursts of AudioFlinger binder calls. 545 class MediaLogNotifier : public Thread { 546 public: 547 MediaLogNotifier(); 548 549 // Requests a MediaLogService notification. It's ignored if there has recently been another 550 void requestMerge(); 551 private: 552 // Every iteration blocks waiting for a request, then interacts with MediaLogService to 553 // start merging. 554 // As every MediaLogService binder call is expensive, once it gets a request it ignores the 555 // following ones for a period of time. 556 virtual bool threadLoop() override; 557 558 bool mPendingRequests; 559 560 // Mutex and condition variable around mPendingRequests' value 561 Mutex mMutex; 562 Condition mCond; 563 564 // Duration of the sleep period after a processed request 565 static const int kPostTriggerSleepPeriod = 1000000; 566 }; 567 568 const sp<MediaLogNotifier> mMediaLogNotifier; 569 570 // This is a helper that is called during incoming binder calls. 571 // Requests media.log to start merging log buffers 572 void requestLogMerge(); 573 574 class TrackHandle; 575 class RecordHandle; 576 class RecordThread; 577 class PlaybackThread; 578 class MixerThread; 579 class DirectOutputThread; 580 class OffloadThread; 581 class DuplicatingThread; 582 class AsyncCallbackThread; 583 class Track; 584 class RecordTrack; 585 class EffectBase; 586 class EffectModule; 587 class EffectHandle; 588 class EffectChain; 589 class DeviceEffectProxy; 590 class DeviceEffectManager; 591 class PatchPanel; 592 class DeviceEffectManagerCallback; 593 594 struct AudioStreamIn; 595 struct TeePatch; 596 using TeePatches = std::vector<TeePatch>; 597 598 599 struct stream_type_t { stream_type_tstream_type_t600 stream_type_t() 601 : volume(1.0f), 602 mute(false) 603 { 604 } 605 float volume; 606 bool mute; 607 }; 608 609 // Abstraction for the Audio Source for the RecordThread (HAL or PassthruPatchRecord). 610 struct Source 611 { 612 virtual ~Source() = default; 613 // The following methods have the same signatures as in StreamHalInterface. 614 virtual status_t read(void *buffer, size_t bytes, size_t *read) = 0; 615 virtual status_t getCapturePosition(int64_t *frames, int64_t *time) = 0; 616 virtual status_t standby() = 0; 617 }; 618 619 // --- PlaybackThread --- 620 #ifdef FLOAT_EFFECT_CHAIN 621 #define EFFECT_BUFFER_FORMAT AUDIO_FORMAT_PCM_FLOAT 622 using effect_buffer_t = float; 623 #else 624 #define EFFECT_BUFFER_FORMAT AUDIO_FORMAT_PCM_16_BIT 625 using effect_buffer_t = int16_t; 626 #endif 627 628 #include "Threads.h" 629 630 #include "PatchPanel.h" 631 632 #include "Effects.h" 633 634 #include "DeviceEffectManager.h" 635 636 // Find io handle by session id. 637 // Preference is given to an io handle with a matching effect chain to session id. 638 // If none found, AUDIO_IO_HANDLE_NONE is returned. 639 template <typename T> findIoHandleBySessionId_l(audio_session_t sessionId,const T & threads)640 static audio_io_handle_t findIoHandleBySessionId_l( 641 audio_session_t sessionId, const T& threads) { 642 audio_io_handle_t io = AUDIO_IO_HANDLE_NONE; 643 644 for (size_t i = 0; i < threads.size(); i++) { 645 const uint32_t sessionType = threads.valueAt(i)->hasAudioSession(sessionId); 646 if (sessionType != 0) { 647 io = threads.keyAt(i); 648 if ((sessionType & AudioFlinger::ThreadBase::EFFECT_SESSION) != 0) { 649 break; // effect chain here. 650 } 651 } 652 } 653 return io; 654 } 655 656 // server side of the client's IAudioTrack 657 class TrackHandle : public android::media::BnAudioTrack { 658 public: 659 explicit TrackHandle(const sp<PlaybackThread::Track>& track); 660 virtual ~TrackHandle(); 661 662 binder::Status getCblk(std::optional<media::SharedFileRegion>* _aidl_return) override; 663 binder::Status start(int32_t* _aidl_return) override; 664 binder::Status stop() override; 665 binder::Status flush() override; 666 binder::Status pause() override; 667 binder::Status attachAuxEffect(int32_t effectId, int32_t* _aidl_return) override; 668 binder::Status setParameters(const std::string& keyValuePairs, 669 int32_t* _aidl_return) override; 670 binder::Status selectPresentation(int32_t presentationId, int32_t programId, 671 int32_t* _aidl_return) override; 672 binder::Status getTimestamp(media::AudioTimestampInternal* timestamp, 673 int32_t* _aidl_return) override; 674 binder::Status signal() override; 675 binder::Status applyVolumeShaper(const media::VolumeShaperConfiguration& configuration, 676 const media::VolumeShaperOperation& operation, 677 int32_t* _aidl_return) override; 678 binder::Status getVolumeShaperState( 679 int32_t id, 680 std::optional<media::VolumeShaperState>* _aidl_return) override; 681 binder::Status getDualMonoMode(media::AudioDualMonoMode* _aidl_return) override; 682 binder::Status setDualMonoMode(media::AudioDualMonoMode mode) override; 683 binder::Status getAudioDescriptionMixLevel(float* _aidl_return) override; 684 binder::Status setAudioDescriptionMixLevel(float leveldB) override; 685 binder::Status getPlaybackRateParameters( 686 media::AudioPlaybackRate* _aidl_return) override; 687 binder::Status setPlaybackRateParameters( 688 const media::AudioPlaybackRate& playbackRate) override; 689 690 private: 691 const sp<PlaybackThread::Track> mTrack; 692 }; 693 694 // server side of the client's IAudioRecord 695 class RecordHandle : public android::media::BnAudioRecord { 696 public: 697 explicit RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack); 698 virtual ~RecordHandle(); 699 virtual binder::Status start(int /*AudioSystem::sync_event_t*/ event, 700 int /*audio_session_t*/ triggerSession); 701 virtual binder::Status stop(); 702 virtual binder::Status getActiveMicrophones( 703 std::vector<media::MicrophoneInfoData>* activeMicrophones); 704 virtual binder::Status setPreferredMicrophoneDirection( 705 int /*audio_microphone_direction_t*/ direction); 706 virtual binder::Status setPreferredMicrophoneFieldDimension(float zoom); 707 virtual binder::Status shareAudioHistory(const std::string& sharedAudioPackageName, 708 int64_t sharedAudioStartMs); 709 710 private: 711 const sp<RecordThread::RecordTrack> mRecordTrack; 712 713 // for use from destructor 714 void stop_nonvirtual(); 715 }; 716 717 // Mmap stream control interface implementation. Each MmapThreadHandle controls one 718 // MmapPlaybackThread or MmapCaptureThread instance. 719 class MmapThreadHandle : public MmapStreamInterface { 720 public: 721 explicit MmapThreadHandle(const sp<MmapThread>& thread); 722 virtual ~MmapThreadHandle(); 723 724 // MmapStreamInterface virtuals 725 virtual status_t createMmapBuffer(int32_t minSizeFrames, 726 struct audio_mmap_buffer_info *info); 727 virtual status_t getMmapPosition(struct audio_mmap_position *position); 728 virtual status_t getExternalPosition(uint64_t *position, int64_t *timeNanos); 729 virtual status_t start(const AudioClient& client, 730 const audio_attributes_t *attr, 731 audio_port_handle_t *handle); 732 virtual status_t stop(audio_port_handle_t handle); 733 virtual status_t standby(); 734 735 private: 736 const sp<MmapThread> mThread; 737 }; 738 739 ThreadBase *checkThread_l(audio_io_handle_t ioHandle) const; 740 PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const; 741 MixerThread *checkMixerThread_l(audio_io_handle_t output) const; 742 RecordThread *checkRecordThread_l(audio_io_handle_t input) const; 743 MmapThread *checkMmapThread_l(audio_io_handle_t io) const; 744 VolumeInterface *getVolumeInterface_l(audio_io_handle_t output) const; 745 Vector <VolumeInterface *> getAllVolumeInterfaces_l() const; 746 747 sp<ThreadBase> openInput_l(audio_module_handle_t module, 748 audio_io_handle_t *input, 749 audio_config_t *config, 750 audio_devices_t device, 751 const char* address, 752 audio_source_t source, 753 audio_input_flags_t flags, 754 audio_devices_t outputDevice, 755 const String8& outputDeviceAddress); 756 sp<ThreadBase> openOutput_l(audio_module_handle_t module, 757 audio_io_handle_t *output, 758 audio_config_t *halConfig, 759 audio_config_base_t *mixerConfig, 760 audio_devices_t deviceType, 761 const String8& address, 762 audio_output_flags_t flags); 763 764 void closeOutputFinish(const sp<PlaybackThread>& thread); 765 void closeInputFinish(const sp<RecordThread>& thread); 766 767 // no range check, AudioFlinger::mLock held streamMute_l(audio_stream_type_t stream)768 bool streamMute_l(audio_stream_type_t stream) const 769 { return mStreamTypes[stream].mute; } 770 void ioConfigChanged(audio_io_config_event_t event, 771 const sp<AudioIoDescriptor>& ioDesc, 772 pid_t pid = 0); 773 void onSupportedLatencyModesChanged( 774 audio_io_handle_t output, const std::vector<audio_latency_mode_t>& modes); 775 776 // Allocate an audio_unique_id_t. 777 // Specific types are audio_io_handle_t, audio_session_t, effect ID (int), 778 // audio_module_handle_t, and audio_patch_handle_t. 779 // They all share the same ID space, but the namespaces are actually independent 780 // because there are separate KeyedVectors for each kind of ID. 781 // The return value is cast to the specific type depending on how the ID will be used. 782 // FIXME This API does not handle rollover to zero (for unsigned IDs), 783 // or from positive to negative (for signed IDs). 784 // Thus it may fail by returning an ID of the wrong sign, 785 // or by returning a non-unique ID. 786 // This is the internal API. For the binder API see newAudioUniqueId(). 787 audio_unique_id_t nextUniqueId(audio_unique_id_use_t use); 788 789 status_t moveEffectChain_l(audio_session_t sessionId, 790 PlaybackThread *srcThread, 791 PlaybackThread *dstThread); 792 793 status_t moveAuxEffectToIo(int EffectId, 794 const sp<PlaybackThread>& dstThread, 795 sp<PlaybackThread> *srcThread); 796 797 // return thread associated with primary hardware device, or NULL 798 PlaybackThread *primaryPlaybackThread_l() const; 799 DeviceTypeSet primaryOutputDevice_l() const; 800 801 // return the playback thread with smallest HAL buffer size, and prefer fast 802 PlaybackThread *fastPlaybackThread_l() const; 803 804 sp<ThreadBase> getEffectThread_l(audio_session_t sessionId, int effectId); 805 806 ThreadBase *hapticPlaybackThread_l() const; 807 808 void updateSecondaryOutputsForTrack_l( 809 PlaybackThread::Track* track, 810 PlaybackThread* thread, 811 const std::vector<audio_io_handle_t>& secondaryOutputs) const; 812 813 814 void removeClient_l(pid_t pid); 815 void removeNotificationClient(pid_t pid); 816 bool isNonOffloadableGlobalEffectEnabled_l(); 817 void onNonOffloadableGlobalEffectEnable(); 818 bool isSessionAcquired_l(audio_session_t audioSession); 819 820 // Store an effect chain to mOrphanEffectChains keyed vector. 821 // Called when a thread exits and effects are still attached to it. 822 // If effects are later created on the same session, they will reuse the same 823 // effect chain and same instances in the effect library. 824 // return ALREADY_EXISTS if a chain with the same session already exists in 825 // mOrphanEffectChains. Note that this should never happen as there is only one 826 // chain for a given session and it is attached to only one thread at a time. 827 status_t putOrphanEffectChain_l(const sp<EffectChain>& chain); 828 // Get an effect chain for the specified session in mOrphanEffectChains and remove 829 // it if found. Returns 0 if not found (this is the most common case). 830 sp<EffectChain> getOrphanEffectChain_l(audio_session_t session); 831 // Called when the last effect handle on an effect instance is removed. If this 832 // effect belongs to an effect chain in mOrphanEffectChains, the chain is updated 833 // and removed from mOrphanEffectChains if it does not contain any effect. 834 // Return true if the effect was found in mOrphanEffectChains, false otherwise. 835 bool updateOrphanEffectChains(const sp<EffectModule>& effect); 836 837 std::vector< sp<EffectModule> > purgeStaleEffects_l(); 838 839 void broadcastParametersToRecordThreads_l(const String8& keyValuePairs); 840 void updateOutDevicesForRecordThreads_l(const DeviceDescriptorBaseVector& devices); 841 void forwardParametersToDownstreamPatches_l( 842 audio_io_handle_t upStream, const String8& keyValuePairs, 843 std::function<bool(const sp<PlaybackThread>&)> useThread = nullptr); 844 845 // AudioStreamIn is immutable, so their fields are const. 846 // For emphasis, we could also make all pointers to them be "const *", 847 // but that would clutter the code unnecessarily. 848 849 struct AudioStreamIn : public Source { 850 AudioHwDevice* const audioHwDev; 851 sp<StreamInHalInterface> stream; 852 audio_input_flags_t flags; 853 hwDevAudioStreamIn854 sp<DeviceHalInterface> hwDev() const { return audioHwDev->hwDevice(); } 855 AudioStreamInAudioStreamIn856 AudioStreamIn(AudioHwDevice *dev, sp<StreamInHalInterface> in, audio_input_flags_t flags) : 857 audioHwDev(dev), stream(in), flags(flags) {} readAudioStreamIn858 status_t read(void *buffer, size_t bytes, size_t *read) override { 859 return stream->read(buffer, bytes, read); 860 } getCapturePositionAudioStreamIn861 status_t getCapturePosition(int64_t *frames, int64_t *time) override { 862 return stream->getCapturePosition(frames, time); 863 } standbyAudioStreamIn864 status_t standby() override { return stream->standby(); } 865 }; 866 867 struct TeePatch { 868 sp<RecordThread::PatchRecord> patchRecord; 869 sp<PlaybackThread::PatchTrack> patchTrack; 870 }; 871 872 // for mAudioSessionRefs only 873 struct AudioSessionRef { AudioSessionRefAudioSessionRef874 AudioSessionRef(audio_session_t sessionid, pid_t pid, uid_t uid) : 875 mSessionid(sessionid), mPid(pid), mUid(uid), mCnt(1) {} 876 const audio_session_t mSessionid; 877 const pid_t mPid; 878 const uid_t mUid; 879 int mCnt; 880 }; 881 882 mutable Mutex mLock; 883 // protects mClients and mNotificationClients. 884 // must be locked after mLock and ThreadBase::mLock if both must be locked 885 // avoids acquiring AudioFlinger::mLock from inside thread loop. 886 mutable Mutex mClientLock; 887 // protected by mClientLock 888 DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client() 889 890 mutable Mutex mHardwareLock; 891 // NOTE: If both mLock and mHardwareLock mutexes must be held, 892 // always take mLock before mHardwareLock 893 894 // guarded by mHardwareLock 895 AudioHwDevice* mPrimaryHardwareDev; 896 DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*> mAudioHwDevs; 897 898 // These two fields are immutable after onFirstRef(), so no lock needed to access 899 sp<DevicesFactoryHalInterface> mDevicesFactoryHal; 900 sp<DevicesFactoryHalCallback> mDevicesFactoryHalCallback; 901 902 // for dump, indicates which hardware operation is currently in progress (but not stream ops) 903 enum hardware_call_state { 904 AUDIO_HW_IDLE = 0, // no operation in progress 905 AUDIO_HW_INIT, // init_check 906 AUDIO_HW_OUTPUT_OPEN, // open_output_stream 907 AUDIO_HW_OUTPUT_CLOSE, // unused 908 AUDIO_HW_INPUT_OPEN, // unused 909 AUDIO_HW_INPUT_CLOSE, // unused 910 AUDIO_HW_STANDBY, // unused 911 AUDIO_HW_SET_MASTER_VOLUME, // set_master_volume 912 AUDIO_HW_GET_ROUTING, // unused 913 AUDIO_HW_SET_ROUTING, // unused 914 AUDIO_HW_GET_MODE, // unused 915 AUDIO_HW_SET_MODE, // set_mode 916 AUDIO_HW_GET_MIC_MUTE, // get_mic_mute 917 AUDIO_HW_SET_MIC_MUTE, // set_mic_mute 918 AUDIO_HW_SET_VOICE_VOLUME, // set_voice_volume 919 AUDIO_HW_SET_PARAMETER, // set_parameters 920 AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size 921 AUDIO_HW_GET_MASTER_VOLUME, // get_master_volume 922 AUDIO_HW_GET_PARAMETER, // get_parameters 923 AUDIO_HW_SET_MASTER_MUTE, // set_master_mute 924 AUDIO_HW_GET_MASTER_MUTE, // get_master_mute 925 AUDIO_HW_GET_MICROPHONES, // getMicrophones 926 AUDIO_HW_SET_CONNECTED_STATE, // setConnectedState 927 }; 928 929 mutable hardware_call_state mHardwareStatus; // for dump only 930 931 932 DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads; 933 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 934 935 // member variables below are protected by mLock 936 float mMasterVolume; 937 bool mMasterMute; 938 float mMasterBalance = 0.f; 939 // end of variables protected by mLock 940 941 DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads; 942 943 // protected by mClientLock 944 DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients; 945 946 // updated by atomic_fetch_add_explicit 947 volatile atomic_uint_fast32_t mNextUniqueIds[AUDIO_UNIQUE_ID_USE_MAX]; 948 949 audio_mode_t mMode; 950 std::atomic_bool mBtNrecIsOff; 951 952 // protected by mLock 953 Vector<AudioSessionRef*> mAudioSessionRefs; 954 955 float masterVolume_l() const; 956 float getMasterBalance_l() const; 957 bool masterMute_l() const; 958 audio_module_handle_t loadHwModule_l(const char *name); 959 960 Vector < sp<SyncEvent> > mPendingSyncEvents; // sync events awaiting for a session 961 // to be created 962 963 // Effect chains without a valid thread 964 DefaultKeyedVector< audio_session_t , sp<EffectChain> > mOrphanEffectChains; 965 966 // list of sessions for which a valid HW A/V sync ID was retrieved from the HAL 967 DefaultKeyedVector< audio_session_t , audio_hw_sync_t >mHwAvSyncIds; 968 969 // list of MMAP stream control threads. Those threads allow for wake lock, routing 970 // and volume control for activity on the associated MMAP stream at the HAL. 971 // Audio data transfer is directly handled by the client creating the MMAP stream 972 DefaultKeyedVector< audio_io_handle_t, sp<MmapThread> > mMmapThreads; 973 974 private: 975 sp<Client> registerPid(pid_t pid); // always returns non-0 976 977 // for use from destructor 978 status_t closeOutput_nonvirtual(audio_io_handle_t output); 979 void closeThreadInternal_l(const sp<PlaybackThread>& thread); 980 status_t closeInput_nonvirtual(audio_io_handle_t input); 981 void closeThreadInternal_l(const sp<RecordThread>& thread); 982 void setAudioHwSyncForSession_l(PlaybackThread *thread, audio_session_t sessionId); 983 984 status_t checkStreamType(audio_stream_type_t stream) const; 985 986 void filterReservedParameters(String8& keyValuePairs, uid_t callingUid); 987 void logFilteredParameters(size_t originalKVPSize, const String8& originalKVPs, 988 size_t rejectedKVPSize, const String8& rejectedKVPs, 989 uid_t callingUid); 990 991 public: 992 // These methods read variables atomically without mLock, 993 // though the variables are updated with mLock. isLowRamDevice()994 bool isLowRamDevice() const { return mIsLowRamDevice; } 995 size_t getClientSharedHeapSize() const; 996 997 private: 998 std::atomic<bool> mIsLowRamDevice; 999 bool mIsDeviceTypeKnown; 1000 int64_t mTotalMemory; 1001 std::atomic<size_t> mClientSharedHeapSize; 1002 static constexpr size_t kMinimumClientSharedHeapSizeBytes = 1024 * 1024; // 1MB 1003 1004 nsecs_t mGlobalEffectEnableTime; // when a global effect was last enabled 1005 1006 // protected by mLock 1007 PatchPanel mPatchPanel; 1008 sp<EffectsFactoryHalInterface> mEffectsFactoryHal; 1009 1010 DeviceEffectManager mDeviceEffectManager; 1011 1012 bool mSystemReady; 1013 std::atomic_bool mAudioPolicyReady{}; 1014 1015 mediautils::UidInfo mUidInfo; 1016 1017 SimpleLog mRejectedSetParameterLog; 1018 SimpleLog mAppSetParameterLog; 1019 SimpleLog mSystemSetParameterLog; 1020 1021 std::vector<media::AudioVibratorInfo> mAudioVibratorInfos; 1022 1023 static inline constexpr const char *mMetricsId = AMEDIAMETRICS_KEY_AUDIO_FLINGER; 1024 1025 // Keep in sync with java definition in media/java/android/media/AudioRecord.java 1026 static constexpr int32_t kMaxSharedAudioHistoryMs = 5000; 1027 1028 std::map<media::audio::common::AudioMMapPolicyType, 1029 std::vector<media::audio::common::AudioMMapPolicyInfo>> mPolicyInfos; 1030 int32_t mAAudioBurstsPerBuffer = 0; 1031 int32_t mAAudioHwBurstMinMicros = 0; 1032 }; 1033 1034 #undef INCLUDING_FROM_AUDIOFLINGER_H 1035 1036 std::string formatToString(audio_format_t format); 1037 std::string inputFlagsToString(audio_input_flags_t flags); 1038 std::string outputFlagsToString(audio_output_flags_t flags); 1039 std::string devicesToString(audio_devices_t devices); 1040 const char *sourceToString(audio_source_t source); 1041 1042 // ---------------------------------------------------------------------------- 1043 1044 } // namespace android 1045 1046 #endif // ANDROID_AUDIO_FLINGER_H 1047