@byteplus/rtc 4.54.2 → 4.55.0

package/index.d.ts

@@ -26,7 +26,8 @@ export interface IAudioMixingManager {
 	 * @notes + If you have already loaded the sound effect into memory through [preloadAudioMixing](#preloadaudiomixing), make sure that the ID here is the same as the ID set by [preloadAudioMixing](#preloadaudiomixing).
 	 *        + After you start playing the audio file and mixing, you can call the [stopAudioMixing](#stopaudiomixing) method to stop playing the audio file.
 	 *        + The mixing data of this method comes from an external file, while the mixing data of [enableAudioMixingBuffer](#enableaudiomixingbuffer) comes from an external cache and the audio format is PCM. These two mixing methods can coexist.
-	 * @region Advanced Feature
+	 * @group Audio Mixing
+	 * @order 5
 	 */
 	startAudioMixing: (id: number, filePath: string, options: MixOption) => Promise<void>;
 	
@@ -35,7 +36,8 @@ export interface IAudioMixingManager {
 	 * @param id ID of the mixing
 	 * @notes + After calling [startAudioMixing](#startaudiomixing) method to start playing audio files and mixes, you can call this method to stop playing audio files and mixes.
 	 *        + After calling this method to stop playing the audio file, the audio file will be automatically uninstalled.
-	 * @region Advanced Feature
+	 * @group Audio Mixing
+	 * @order 6
 	 */
 	stopAudioMixing: (id: number) => void;
 	
@@ -44,7 +46,8 @@ export interface IAudioMixingManager {
 	 * @param id  ID of the mixing
 	 * @notes + After calling [startAudioMixing](#startaudiomixing) method to start playing audio files and mixes, you can call this method to pause playing audio files and mixes.
 	 *        + After calling this method to pause playing audio files and mixing, you can call the [resumeAudioMixing](#resumeaudiomixing) method to resume playing and mixing.
-	 * @region Advanced Feature
+	 * @group Audio Mixing
+	 * @order 7
 	 */
 	pauseAudioMixing: (id: number) => void;
 	
@@ -52,7 +55,8 @@ export interface IAudioMixingManager {
 	 * @brief Resume playing audio files and mixing
 	 * @param id  Mixing ID
 	 * @notes After calling the [pauseAudioMixing](#pauseaudiomixing) method to pause playing audio files and mixing, you can resume playing and mixing by calling this method.
-	 * @region Advanced Feature
+	 * @group Audio Mixing
+	 * @order 8
 	 */
 	resumeAudioMixing: (id: number) => void;
 	
@@ -67,14 +71,16 @@ export interface IAudioMixingManager {
 	 * @notes + When you need to play an audio file frequently, call this method to preload the file. When playing, you can only load the file once to reduce CPU usage.
 	 *        + This method pre-loads the specified audio file, and only calling [startAudioMixing](#startaudiomixing) API can you start playing the specified audio file.
 	 *        + The specified audio file preloaded by calling this method can be unloaded through [unloadAudioMixing](#unloadaudiomixing).
-	 * @region Advanced Feature
+	 * @group Audio Mixing
+	 * @order 9
 	 */
 	preloadAudioMixing: (id: number, filePath: string) => Promise<void>;
 	
 	/** 
 	 * @brief Uninstall the specified audio file
 	 * @param id Mix ID
-	 * @region Advanced Feature
+	 * @group Audio Mixing
+	 * @order 10
 	 */
 	unloadAudioMixing: (id: number) => void;
 	
@@ -85,7 +91,8 @@ export interface IAudioMixingManager {
 	 *         + 0: Mute
 	 *         + 100: Original volume
 	 *         + 400: Maximum volume (with overflow protection)
-	 * @region Advanced Feature
+	 * @group Audio Mixing
+	 * @order 16
 	 */
 	setAudioMixingVolume: (id: number, volume: number) => void;
 	
@@ -94,7 +101,8 @@ export interface IAudioMixingManager {
 	 * @param id Audio mix id
 	 * @return + `0`: Failure
 	 *         + `>0`: Successful. Volume
-	 * @region Advanced Feature
+	 * @group Audio Mixing
+	 * @order 15
 	 */
 	getAudioMixingVolume: (id: number) => number;
 	
@@ -104,7 +112,8 @@ export interface IAudioMixingManager {
 	 * @return + `> 0`: Success, audio file duration, in milliseconds.
 	 *         + `< 0`: Failure
 	 * @notes  Before calling this method to obtain the length of the audio file, you need to call [preloadAudioMixing](#preloadaudiomixing) or [startAudioMixing](#startaudiomixing).
-	 * @region Advanced Feature
+	 * @group Audio Mixing
+	 * @order 20
 	 */
 	getAudioMixingDuration: (id: number) => number;
 	
@@ -114,7 +123,8 @@ export interface IAudioMixingManager {
 	 * @return + `> 0`: Success, audio file playback progress, in milliseconds.
 	 *         + `< 0`: Failed
 	 * @notes Before calling this method to get the audio file playback progress, you need to call [startAudioMixing](#startaudiomixing) to start playing the audio file.
-	 * @region Advanced Feature
+	 * @group Audio Mixing
+	 * @order 25
 	 */
 	getAudioMixingCurrentPosition: (id: number) => number;
 	
@@ -124,8 +134,9 @@ export interface IAudioMixingManager {
 	 * @param position The starting playback position of the audio file in milliseconds.
 	 *                 Call [getAudioMixingDuration](#getaudiomixingduration) to get the duration of the audio file. The value of  `position` must not be greater than the duration of the audio file.
 	 * @notes + This method allows you to play an audio file from a specified position as appropriate, without having to play an audio file in its entirety from the beginning to finish.
-	 *       + Before calling this method to set the playback position of the audio file, you need to call [startAudioMixing](#startaudiomixing) to start playing the audio file.
-	 * @region Advanced Feature
+	 *        + Before calling this method to set the playback position of the audio file, you need to call [startAudioMixing](#startaudiomixing) to start playing the audio file.
+	 * @group Audio Mixing
+	 * @order 26
 	 */
 	setAudioMixingPosition: (id: number, position: number) => void;
 	
@@ -136,13 +147,15 @@ export interface IAudioMixingManager {
 	 *        Whether the PCM data is mixed locally and sent to the remotes.
 	 * @notes + After this API, you must call [pushAudioMixingBuffer](#pushaudiomixingbuffer) to start mixing.
 	 *        + To end PCM mixing, use [disableAudioMixingBuffer](#disableaudiomixingbuffer).
-	 * @region Advanced Feature
+	 * @group Audio Mixing
+	 * @order 30
 	 */
 	enableAudioMixingBuffer: (type: AudioMixingType) => void;
 	
 	/** 
 	 * @brief Disable PCM mixing.
-	 * @region Advanced Feature
+	 * @group Audio Mixing
+	 * @order 31
 	 */
 	disableAudioMixingBuffer: () => void;
 	
@@ -150,7 +163,8 @@ export interface IAudioMixingManager {
 	 * @brief Push PCM audio frame data for mixing
 	 * @param buffer Refer to [Browser documentation](https://developer.mozilla.org/en-US/docs/Web/API/AudioBuffer)
 	 * @notes Before calling this method, you must start the external audio stream mixing through [enableAudioMixingBuffer](#enableaudiomixingbuffer).
-	 * @region Advanced Feature
+	 * @group Audio Mixing
+	 * @order 32
 	 */
 	pushAudioMixingBuffer: (buffer: AudioBuffer) => void;
 }
@@ -666,6 +680,7 @@ declare enum ErrorCode {
 	RECONNECT_FAILED = "RECONNECT_FAILED",
 	
 	/** 
+	 * @hidden
 	 * @brief Subtitles are already on. There is no need to call [startSubtitle](104478#startsubtitle) again.
 	 */
 	SUBTITLE_ALREADY_ON = "SUBTITLE_ALREADY_ON",
@@ -677,21 +692,25 @@ declare enum ErrorCode {
 	SUBTITLE_NOT_TURNED_ON = "SUBTITLE_NOT_TURNED_ON",
 	
 	/** 
+	 * @hidden
 	 * @brief Errors occurred concerning cloud media processing. Please contact the technical support.
 	 */
 	SUBTITLE_ERR_POSTPROCESS = "SUBTITLE_ERR_POSTPROCESS",
 	
 	/** 
+	 * @hidden
 	 * @brief Failed to connect to the thrid-party service. Please contact the technical support.
 	 */
 	SUBTITLE_ERR_CONNECTION_ERROR = "SUBTITLE_ERR_CONNECTION_ERROR",
 	
 	/** 
+	 * @hidden
 	 * @brief Cloud media processing is timeout. Please contact the technical support.
 	 */
 	SUBTITLE_ERR_PROCESS_ERROR = "SUBTITLE_ERR_PROCESS_ERROR",
 	
 	/** 
+	 * @hidden
 	 * @brief The client side failed to identity error codes sent by cloud media processing. Please contact the technical support.
 	 */
 	SUBTITLE_ERR_UNKNOWN = "SUBTITLE_ERR_UNKNOWN",
@@ -1931,28 +1950,32 @@ export interface IEngineEvents {
 	/** 
 	 * @brief Player event
 	 * @param event
-	 * @region Video Management Event
+	 * @group Video Management
+	 * @order 65
 	 */
 	[EngineEventsTypes.onPlayerEvent]: (event: PlayerEvent) => void;
 	
 	/** 
 	 * @brief Callback for a remote visible user joining the room, or an invisible user in the room switching to visible.
 	 * @param event
-	 * @region Room Management Event
+	 * @group Room Management
+	 * @order 30
 	 */
 	[EngineEventsTypes.onUserJoined]: (event: onUserJoinedEvent) => void;
 	
 	/** 
 	 * @brief Callback for a remote visible user leaving the room, or a visible user in the room switching to invisible.
 	 * @param event
-	 * @region Room Management Event
+	 * @group Room Management
+	 * @order 31
 	 */
 	[EngineEventsTypes.onUserLeave]: (event: onUserLeaveEvent) => void;
 	
 	/** 
 	 * @brief Connection state changed
 	 * @param event
-	 * @region Network Event
+	 * @group Room Management
+	 * @order 35
 	 */
 	[EngineEventsTypes.onConnectionStateChanged]: (event: ConnectionStateChangeEvent) => void;
 	
@@ -1960,13 +1983,14 @@ export interface IEngineEvents {
 	 * @hidden
 	 * @brief Vendor ICE connection state changed
 	 * @param event
-	 * @region Network Event
 	 */
 	[EngineEventsTypes.onVendorConnectionStateChanged]: (event: VendorConnectionStateChangeEvent) => void;
 	
 	/** 
 	 * @brief Callback on new media streams captured by camera/microphone in the room.
 	 * @notes You will receive this callback after a remote user successfully publishes media streams captured by camera/microphone in the room with [publishStream](104478#publishstream). Then you can choose whether to call [subscribeStream](104478#subscribestream) to subscribe to the streams or not.
+	 * @group Audio & Video Transport
+	 * @order 60
 	 */
 	[EngineEventsTypes.onUserPublishStream]: (event: {
 		
@@ -1984,6 +2008,8 @@ export interface IEngineEvents {
 	/** 
 	 * @brief Callback on removal of remote media stream captured by camera/microphone.
 	 * @notes You will receive this callback after a remote user successfully unpublishes media streams captured by camera/microphone in the room with [unpublishStream](104478#unpublishstream). Then you can choose whether to call [unsubscribeStream](104478#unsubscribestream) to unsubscribe from the streams or not.
+	 * @group Audio & Video Transport
+	 * @order 61
 	 */
 	[EngineEventsTypes.onUserUnpublishStream]: (event: {
 		
@@ -2006,6 +2032,8 @@ export interface IEngineEvents {
 	/** 
 	 * @brief Callback on new screen sharing media streams from remote users in the room.
 	 * @notes You will receive this callback after a remote user successfully published screen sharing streams in the room with [publishScreen](104478#publishscreen). Then you can choose whether to call [subscribeScreen](104478#subscribescreen) to subscribe to the streams or not.
+	 * @group Screen Sharing
+	 * @order 30
 	 */
 	[EngineEventsTypes.onUserPublishScreen]: (event: {
 		
@@ -2023,6 +2051,8 @@ export interface IEngineEvents {
 	/** 
 	 * @brief Callback on removal of screen sharing media streams from remote users in the room.
 	 * @notes You will receive this callback after a remote user successfully unpublishes media streams captured by camera/microphone in the room with [unpublishScreen](104478#unpublishscreen). Then you can choose whether to call [unsubscribeScreen](104478#unsubscribescreen) to unsubscribe from the streams or not.
+	 * @group Screen Sharing
+	 * @order 31
 	 */
 	[EngineEventsTypes.onUserUnpublishScreen]: (event: {
 		
@@ -2047,7 +2077,6 @@ export interface IEngineEvents {
 	 * @deprecated since 4.47, use on onRemoteVideoFirstFrame instead.
 	 * @brief Rendering of the first video frame
 	 * @param event
-	 * @region Video Management Event
 	 */
 	[EngineEventsTypes.onVideoFirstFrameRendered]: (event: onVideoFirstFrameRenderedEvent) => void;
 	
@@ -2056,14 +2085,14 @@ export interface IEngineEvents {
 	 * @deprecated since 4.47, use on onRemoteVideoFirstFrame instead.
 	 * @brief Rendering of the first video frame
 	 * @param event
-	 * @region Video Management Event
 	 */
 	[EngineEventsTypes.onVideoFirstFrameDecoded]: (event: onVideoFirstFrameDecodedEvent) => void;
 	
 	/** 
 	 * @brief Decoding of the first video frame
 	 * @param event
-	 * @region Video Management Event
+	 * @group Audio & Video Transport
+	 * @order 65
 	 */
 	[EngineEventsTypes.onRemoteVideoFirstFrame]: (event: onVideoFirstFrameRenderedEvent) => void;
 	
@@ -2072,40 +2101,38 @@ export interface IEngineEvents {
 	 * @deprecated since 4.47, use on onRemoteAudioFirstFrame instead.
 	 * @brief Decoding of the first audio frame
 	 * @param event
-	 * @region Audio Management Event
 	 */
 	[EngineEventsTypes.onAudioFirstFrameDecoded]: (event: onAudioFirstFrameDecodedEvent) => void;
 	
 	/** 
 	 * @brief first remote audio frame Received
 	 * @param event
-	 * @region Audio Management Event
+	 * @group Audio & Video Transport
+	 * @order 66
 	 */
 	[EngineEventsTypes.onRemoteAudioFirstFrame]: (event: onAudioFirstFrameDecodedEvent) => void;
 	
 	/** 
 	 * @brief Event of video device state changing.
 	 * @param event Information and state of a media device.
+	 * @group Video Management
+	 * @order 50
 	 */
 	[EngineEventsTypes.onVideoDeviceStateChanged]: (event: DeviceInfo) => void;
 	
 	/** 
 	 * @brief Event of audio device state changing.
 	 * @param event Information and state of a media device.
+	 * @group Audio Management
+	 * @order 50
 	 */
 	[EngineEventsTypes.onAudioDeviceStateChanged]: (event: DeviceInfo) => void;
 	
-	/** 
-	 * @brief Failure of automatically playing
-	 * @param event
-	 * @region Video Management Event
-	 */
-	[EngineEventsTypes.onAutoplayFailed]: (event: AutoPlayFailedEvent) => void;
-	
 	/** 
 	 * @brief Event of media track ending
 	 * @param event
-	 * @region Video Management Event
+	 * @group Video Management
+	 * @order 75
 	 */
 	[EngineEventsTypes.onTrackEnded]: (event: {
 		
@@ -2155,7 +2182,8 @@ export interface IEngineEvents {
 	/** 
 	 * @brief Event of receiving a broadcast text message in the room
 	 * @param event
-	 * @region Message Event
+	 * @group Message
+	 * @order 55
 	 */
 	[EngineEventsTypes.onRoomMessageReceived]: (event: {
 		
@@ -2173,7 +2201,8 @@ export interface IEngineEvents {
 	/** 
 	 * @brief Event of receiving a broadcast binary message in the room
 	 * @param event
-	 * @region Message Event
+	 * @group Message
+	 * @order 56
 	 */
 	[EngineEventsTypes.onRoomBinaryMessageReceived]: (event: {
 		
@@ -2191,28 +2220,32 @@ export interface IEngineEvents {
 	/** 
 	 * @brief Receiving a point-to-point text message from one of the users in the room by calling [sendUserMessage](104478#sendusermessage)
 	 * @param event
-	 * @region Message Event
+	 * @group Message
+	 * @order 50
 	 */
 	[EngineEventsTypes.onUserMessageReceived]: (event: UserMessageEvent) => void;
 	
 	/** 
 	 * @brief Receiving a point-to-point binary message from one of the users in the room
 	 * @param event
-	 * @region Message Event
+	 * @group Message
+	 * @order 51
 	 */
 	[EngineEventsTypes.onUserBinaryMessageReceived]: (event: UserBinaryMessageEvent) => void;
 	
 	/** 
 	 * @brief Statistics of subscribed media streams and network states in a cycle of 2s
 	 * @param stats
-	 * @region Room Event
+	 * @group Audio & Video Transport
+	 * @order 70
 	 */
 	[EngineEventsTypes.onRemoteStreamStats]: (stats: RemoteStreamStats) => void;
 	
 	/** 
 	 * @brief Statistics of local media streams and network states in a cycle of 2s
 	 * @param stats
-	 * @region Room Event
+	 * @group Audio & Video Transport
+	 * @order 71
 	 */
 	[EngineEventsTypes.onLocalStreamStats]: (stats: LocalStreamStats) => void;
 	
@@ -2220,6 +2253,8 @@ export interface IEngineEvents {
 	 * @hidden
 	 * @brief This 2 s periodical callback notifies information of the public stream after calling [startPlayPublicStream](104478#startplaypublicstream).
 	 * @param stats
+	 * @group Public Stream
+	 * @order 20
 	 */
 	[EngineEventsTypes.onPublicStreamStats]: (stats: PublicStreamStats) => void;
 	
@@ -2229,7 +2264,6 @@ export interface IEngineEvents {
 	 * @brief Volume of each of the users in the room
 	 * If the user has not joined a room, the list will only include volume of `local_user`.
 	 * @param event
-	 * @region Room Event
 	 */
 	[EngineEventsTypes.onAudioVolumeIndication]: (event: {
 		
@@ -2243,7 +2277,8 @@ export interface IEngineEvents {
 	 * @brief After calling [enableAudioPropertiesReport](104478#enableaudiopropertiesreport), you will periodically receive this callback for the information about local audio.
 	 *        Local audio includes the microphone audio and the screen audio captured using RTC SDK internal mechanisms.
 	 * @param event Information about local audio.
-	 * @region Audio Managment
+	 * @group Audio Management
+	 * @order 55
 	 */
 	[EngineEventsTypes.onLocalAudioPropertiesReport]: (event: LocalAudioPropertiesInfo[]) => void;
 	
@@ -2251,27 +2286,32 @@ export interface IEngineEvents {
 	 * @brief After calling [enableAudioPropertiesReport](104478#enableaudiopropertiesreport), you will periodically receive this callback for the information about the subscribed remote audio streams.
 	 *        The remote audio streams includes the microphone audio and screen audio collected using the RTC SDK internal mechanism/custom mechanism.
 	 * @param event Information about remote audio.
-	 * @region Audio Management
+	 * @group Audio Management
+	 * @order 56
 	 */
 	[EngineEventsTypes.onRemoteAudioPropertiesReport]: (event: RemoteAudioPropertiesInfo[]) => void;
 	
 	/** 
 	 * @brief After calling [enableAudioPropertiesReport](104478#enableaudiopropertiesreport), you will periodically receive this callback for the information of the user with the highest voice volume.
 	 * @param event The information of the user with the highest voice volume.
-	 * @region Audio Management
+	 * @group Audio Management
+	 * @order 60
 	 */
 	[EngineEventsTypes.onActiveSpeaker]: (userInfo: UserInfo) => void;
 	
 	/** 
-	 * @brief The current playback device has been changed successfully by calling [setAudioPlaybackDevice](104478#setAudioPlaybackDevice).
+	 * @brief The current playback device has been changed successfully by calling [setAudioPlaybackDevice](104478#setaudioplaybackdevice).
 	 * @param event Information of the media devices. See [MediaDeviceInfo](https://developer.mozilla.org/en-US/docs/Web/API/MediaDeviceInfo).
+	 * @group Audio Management
+	 * @order 65
 	 */
 	[EngineEventsTypes.onAudioPlaybackDeviceChanged]: (event: MediaDeviceInfo) => void;
 	
 	/** 
 	 * @brief The remote clients in the room will be informed of the state change via this event after the publisher starts audio capture by calling [startAudioCapture](104478#startaudiocapture).
 	 * @param event
-	 * @region Audio Management Event
+	 * @group Audio Management
+	 * @order 70
 	 */
 	[EngineEventsTypes.onUserStartAudioCapture]: (event: {
 		
@@ -2284,7 +2324,8 @@ export interface IEngineEvents {
 	/** 
 	 * @brief The remote clients in the room will be informed of the state change via this event after the publisher stops audio capture by calling [stopAudioCapture](104478#stopaudiocapture).
 	 * @param event
-	 * @region Room Management Event
+	 * @group Audio Management
+	 * @order 71
 	 */
 	[EngineEventsTypes.onUserStopAudioCapture]: (event: {
 		
@@ -2297,7 +2338,8 @@ export interface IEngineEvents {
 	/** 
 	 * @brief The remote clients in the room will be informed of the state change via this event after the publisher starts video capture by calling [startVideoCapture](104478#startvideocapture).
 	 * @param event
-	 * @region Room Management Event
+	 * @group Video Management
+	 * @order 55
 	 */
 	[EngineEventsTypes.onUserStartVideoCapture]: (event: {
 		
@@ -2310,7 +2352,8 @@ export interface IEngineEvents {
 	/** 
 	 The remote clients in the room will be informed of the state change via  this event after the publisher stops video capture by calling [stopVideoCapture](104478#stopvideocapture).
 	 * @param event
-	 * @region Room Management Event
+	 * @group Video Management
+	 * @order 56
 	 */
 	[EngineEventsTypes.onUserStopVideoCapture]: (event: {
 		
@@ -2323,14 +2366,16 @@ export interface IEngineEvents {
 	/** 
 	 * @brief Event of receiving SEI data from video frames
 	 * @param event Event of receiving SEI data
-	 * @region Message Event
+	 * @group Message
+	 * @order 60
 	 */
 	[EngineEventsTypes.onSEIMessageReceived]: (event: onSEIMessageEvent) => void;
 	
 	/** 
 	 * @brief Result of the Auto-Subscription
 	 * @param event
-	 * @region Room Management Event
+	 * @group Audio & Video Transport
+	 * @order 75
 	 */
 	[EngineEventsTypes.onAutoSubscribeResult]: (event: {
 		
@@ -2353,7 +2398,8 @@ export interface IEngineEvents {
 	/** 
 	 * @brief Result of Auto-publishing. You will receive this event if you enable Auto-publish when calling `joinRoom`
 	 * @param event
-	 * @region Room Management Event
+	 * @group Audio & Video Transport
+	 * @order 76
 	 */
 	[EngineEventsTypes.onAutoPublishResult]: (event: {
 		
@@ -2372,6 +2418,8 @@ export interface IEngineEvents {
 	 * @brief Auto-play fails.
 	 * @param event Event related to the player
 	 * @region Room Management Event
+	 * @group Video Management
+	 * @order 70
 	 */
 	[EngineEventsTypes.onAutoplayFailed]: (event: AutoPlayFailedEvent) => void;
 	
@@ -2380,14 +2428,14 @@ export interface IEngineEvents {
 	 * @deprecated since 4.44, use onStreamMixingEvent instead.
 	 * @brief Result of the Pushing stream to CDN
 	 * @param event
-	 * @region Advanced Feature Event
 	 */
 	[EngineEventsTypes.onLiveTranscodingResult]: (event: LiveTranscodingResultEvent) => void;
 	
 	/** 
 	 * @brief event of error
 	 * @param event
-	 * @region Room Management Event
+	 * @group Room Management
+	 * @order 40
 	 */
 	[EngineEventsTypes.onError]: (event: {
 		
@@ -2400,27 +2448,32 @@ export interface IEngineEvents {
 	/** 
 	 * @brief Event of the state change of audio mixing
 	 * @param event
-	 * @region Advanced Feature
+	 * @group Audio Mixing
+	 * @order 40
 	 */
 	[EngineEventsTypes.onAudioMixingStateChanged]: (event: AudioMixingStateChangedEvent) => void;
 	
 	/** 
 	 * @brief Event of receiving a text message
 	 * @param event
-	 * @region Message Event
+	 * @group Message
+	 * @order 65
 	 */
 	[EngineEventsTypes.onUserMessageReceivedOutsideRoom]: (event: UserMessageEvent) => void;
 	
 	/** 
 	 * @brief Event of receiving a binary message
 	 * @param event
-	 * @region Message Event
+	 * @group Message
+	 * @order 66
 	 */
 	[EngineEventsTypes.onUserBinaryMessageReceivedOutsideRoom]: (event: UserBinaryMessageEvent) => void;
 	
 	/** 
 	 * @brief You will receive this callback 30 s before the Token expires.
 	 * Call [updateToken](104478#updatetoken) to submit a new Token。The user will be removed from the room when the Token expires.
+	 * @group Room Management
+	 * @order 45
 	 */
 	[EngineEventsTypes.onTokenWillExpire]: () => void;
 	
@@ -2435,6 +2488,8 @@ export interface IEngineEvents {
 	 * @brief Callback of the result of publishing the public stream.
 	 *        After calling [startPushPublicStream](104478#startpushpublicstream), you will be informed of the result through this callback.
 	 * @param event
+	 * @group Public Stream
+	 * @order 25
 	 */
 	[EngineEventsTypes.onPushPublicStreamResult]: (event: PublicStreamResultEvent) => void;
 	
@@ -2446,6 +2501,8 @@ export interface IEngineEvents {
 	 * @notes When multiple streams in the public stream contain SEI:
 	 *        + You will receive SEI from all the video frames if the SEI do not have conflicts.
 	 *        + However, if SEI from video frames have conflicts, you will receive only one of them.
+	 * @group Public Stream
+	 * @order 30
 	 */
 	[EngineEventsTypes.onPublicStreamSEIMessageReceived]: (event: PublicStreamSEIMessageEvent) => void;
 	
@@ -2454,6 +2511,8 @@ export interface IEngineEvents {
 	 * @brief Callback of the successful decoding of the first video frame of the public stream.
 	 *        Refer to [startPlayPublicStream](104478#startplaypublicstream) for details about subscribing to a public stream.
 	 * @param event
+	 * @group Public Stream
+	 * @order 36
 	 */
 	[EngineEventsTypes.onFirstPublicStreamVideoFrameDecoded]: (event: PublicStreamVideoFirstFrameDecodedEvent) => void;
 	
@@ -2462,6 +2521,8 @@ export interface IEngineEvents {
 	 * @brief Callback of the successful decoding of the first audio frame of the public stream.
 	 *        Refer to [startPlayPublicStream](104478#startplaypublicstream) for details about subscribing to a public stream.
 	 * @param event
+	 * @group Public Stream
+	 * @order 35
 	 */
 	[EngineEventsTypes.onFirstPublicStreamAudioFrameDecoded]: (event: PublicStreamAudioFirstFrameDecodedEvent) => void;
 	
@@ -2470,6 +2531,8 @@ export interface IEngineEvents {
 	 * @brief Callback of the successful rendering of the first video frame of the public stream.
 	 *         Refer to [startPlayPublicStream](104478#startplaypublicstream) for details about subscribing to a public stream.
 	 * @param event
+	 * @group Public Stream
+	 * @order 37
 	 */
 	[EngineEventsTypes.onFirstPublicStreamVideoFrameRendered]: (event: PublicStreamAudioFirstFrameDecodedEvent) => void;
 	
@@ -2477,6 +2540,8 @@ export interface IEngineEvents {
 	 * @brief Notification on the playing volume during the test for the local audio devices
 	 * @param volume playing volume during the test for the local audio devices in milliseconds. We recommend to set it to 200 ms.The range is [0,255].
 	 * @notes Start an audio-device test by calling [startAudioPlaybackDeviceTest](104478#startaudioplaybackdevicetest) or [startAudioDeviceRecordTest](104478#startaudiodevicerecordtest) will register this callback for regular notification on playing volume. You can set the time interval between each callback by passing a proper value when calling the API above.
+	 * @group Device Management
+	 * @order 40
 	 */
 	[EngineEventsTypes.onAudioPlaybackDeviceTestVolume]: (volume: number) => void;
 	
@@ -2484,17 +2549,23 @@ export interface IEngineEvents {
 	 * @brief The callback reports the upstream and downstream network quality of the local user every 2s after the user enters the room.
 	 * @param uplinkNetworkQuality Upstream network quality.
 	 * @param downlinkNetworkQuality Downstream network quality.
+	 * @group Audio & Video Transport
+	 * @order 80
 	 */
 	[EngineEventsTypes.onNetworkQuality]: (uplinkNetworkQuality: NetworkQuality, downlinkNetworkQuality: NetworkQuality) => void;
 	
 	/** 
 	 * @brief Receives this callback when fallback happens or the stream recovers from fallback.
 	 * @param event Information about stream switching
+	 * @group Audio & Video Transport
+	 * @order 85
 	 */
 	[EngineEventsTypes.onSimulcastSubscribeFallback]: (event: RemoteStreamSwitch) => void;
 	
 	/** 
 	 * @brief When the resolution of remote video changes, users who subscribed to the stream will receive this callback.
+	 * @group Audio & Video Transport
+	 * @order 90
 	 */
 	[EngineEventsTypes.onRemoteVideoSizeChanged]: (
 	
@@ -2522,7 +2593,8 @@ export interface IEngineEvents {
 	/** 
 	 * @brief Pushing a mixed stream to CDN events
 	 * @param event
-	 * @region Push to CDN
+	 * @group Push to CDN
+	 * @order 15
 	 */
 	[EngineEventsTypes.onStreamMixingEvent]: (event: StreamMixingEvent) => void;
 	
@@ -2534,6 +2606,8 @@ export interface IEngineEvents {
 	 *        + If a user is banned, other users receive this callback again when they leave and re-enter the room.
 	 *        + Only banned/unbanned user will receive this callback if the conference mode is enabled in the console.
 	 *        + If the room is dissolved and created again, the status in the room is cleared.
+	 * @group Audio & Video Transport
+	 * @order 95
 	 */
 	[EngineEventsTypes.onVideoStreamBanned]: (event: VideoStreamBannedEvent) => void;
 	
@@ -2545,11 +2619,15 @@ export interface IEngineEvents {
 	 *        + If a user is banned, other users receive this callback again when they leave and re-enter the room.
 	 *        + Only banned/unbanned user will receive this callback if the conference mode is enabled in the console.
 	 *        + If the room is dissolved and created again, the status in the room is cleared.
+	 * @group Audio & Video Transport
+	 * @order 96
 	 */
 	[EngineEventsTypes.onAudioStreamBanned]: (event: AudioStreamBannedEvent) => void;
 	
 	/** 
 	 * @brief The callback of the actual video capturing configuration.
+	 * @group Video Management
+	 * @order 60
 	 */
 	[EngineEventsTypes.onLocalVideoSizeChanged]: (e: {
 		
@@ -2577,17 +2655,21 @@ export interface IEngineEvents {
 	
 	/** 
 	 * @hidden currently not available
-	 * @brief  Callback on subtitle states.
-	 *         After you call [startSubtitle](104478#startsubtitle) and [stopSubtitle](104478#stopsubtitle), you will receive this callback which informs you of the states and error codes of the subtitling task, as well as detailed information on the third party services' errors.
-	 * @param event
+	 * @brief Callback on subtitle states.
+	 *        After you call [startSubtitle](104478#startsubtitle) and [stopSubtitle](104478#stopsubtitle), you will receive this callback which informs you of the states and error codes of the subtitling task, as well as detailed information on the third party services' errors.
+	 * @param e Subtitle status.
+	 * @group Live Caption and Translation
+	 * @order 10
 	 */
 	[EngineEventsTypes.onSubtitleStateChanged]: (e: SubtitleEvent) => void;
 	
 	/** 
 	 * @hidden currently not available
-	 * @brief  Callback on subtitle messages.
-	 *         After calling [startSubtitle](104478#startsubtitle) successfully, you will receive this callback which informs you of the related information on subtitles.
-	 * @param event
+	 * @brief Callback on subtitle messages.
+	 *        After calling [startSubtitle](104478#startsubtitle) successfully, you will receive this callback which informs you of the related information on subtitles.
+	 * @param e Subtitle message.
+	 * @group Live Caption and Translation
+	 * @order 11
 	 */
 	[EngineEventsTypes.onSubtitleMessageReceived]: (e: SubtitleMessage[]) => void;
 	
@@ -2595,7 +2677,9 @@ export interface IEngineEvents {
 	 * @brief  Event of whether the setting for the message server successes
 	 * @param error Error code
 	 * @notes + This event notifies you the result after the call of [setServerParams](104478#setserverparams).
-	 * + If the client losts connection after the call of [setServerParams](104478#setserverparams) this event notifies once the client reconnects.
+	 *        + If you lose internet connection after calling [setServerParams](104478#setserverparams), you will receive this event once the internet reconnects.
+	 * @group Message
+	 * @order 70
 	 */
 	[EngineEventsTypes.onServerParamsSetResult]: (error?: ErrorCode) => void;
 	
@@ -2610,6 +2694,8 @@ export interface IEngineEvents {
 	/** 
 	 * @brief Event of failing to forward stream across rooms.
 	 * @param e Status information for forwarding media stream across rooms.
+	 * @group Audio & Video Transport
+	 * @order 100
 	 */
 	[EngineEventsTypes.onForwardStreamError]: (e: ForwardStreamStateInfo) => void;
 	
@@ -2645,7 +2731,8 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @notes + In the same room with the same App ID, the user ID of each user must be unique. The user will be kicked out of the room when a user with the same userID joins the room because of the error of `DUPLICATE_LOGIN`.
 	 *        + Remote users will receive [onUserJoined](104479#onuserjoined) callback when a visible user joins the room. Call [setUserVisibility](#setuservisibility) to switch the visibility of the user.
 	 *        + RTC SDK will try reconnect after losing connection to the server caused by a poor network connection. At this point, [onConnectionStateChanged](104479#onconnectionstatechanged) callback will be triggered and the SDK will automatically retry until it successfully reconnects to the server. The remote user will receive [onUserJoined](104479#onuserjoined) if a visible user reconnected successfully.
-	 * @region Room Management
+	 * @group Room Management
+	 * @order 0
 	 */
 	joinRoom: (token: string | null, roomId: string, userInfo: UserInfo, roomConfig?: RoomConfig) => Promise<void>;
 	
@@ -2654,7 +2741,8 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * The user calls this API to leave the room, end the call process, and release all call-related resources.
 	 * @notes + After visible users leave the room, other users in the room will receive [onUserLeave](104479#onuserleave) callback notification.
 	 *     + You must call this API to end the call, after calling the [joinRoom](#joinroom) method to join the room. Otherwise you cannot start the next call. Repeatedly call this method have no negative impact.
-	 * @region Room Management
+	 * @group Room Management
+	 * @order 1
 	 */
 	leaveRoom: (waitAck?: boolean) => Promise<void>;
 	
@@ -2665,7 +2753,6 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @notes + Make sure that the business scenarios related to the `RTCEngine` instances that need to be destroyed are all finished before calling this API.
 	 *       + Calling this API will destroy all memory related to this `RTCEngine` instance, and stop any interaction with the media server.
 	 *       + This API is a blocking call which will block the current thread until the SDK completes the exit logic. Therefore, do NOT to call this API directly in the callback thread. Besides do NOT wait for the execution of the main thread including this API in any callback, thus causing a deadlock.
-	 * @region Engine Management
 	 */
 	destroy: () => void;
 	
@@ -2676,7 +2763,8 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 *        + Call [publishScreen](#publishscreen) to start screen sharing.
 	 *        + After you call this API, the other users in the room will receive [onUserPublishStream](104479#onuserpublishstream).
 	 *        + Call [unpublishStream](#unpublishstream) to stop publishing streams.
-	 * @region Room Management
+	 * @group Audio & Video Transport
+	 * @order 0
 	 */
 	publishStream: (mediaType: MediaType) => Promise<void>;
 	
@@ -2685,7 +2773,8 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @param mediaType Media stream type, used for specifying whether to stop publishing audio stream or video stream.
 	 * @notes + After calling [publishStream](#publishstream), call this API to stop publishing streams.
 	 *        + After calling this API, the other users in the room will receive [onUserUnpublishStream](104479#onuserunpublishstream).
-	 * @region Room Management
+	 * @group Audio & Video Transport
+	 * @order 1
 	 */
 	unpublishStream: (mediaType: MediaType) => Promise<void>;
 	
@@ -2695,7 +2784,8 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @notes + An invisible user cannot publish media streams. Call [setUserVisibility](#setuservisibility) to change your visibility in the room.
 	 *        + After you called this API, the other users in the room will receive [onUserPublishScreen](104479#onuserpublishscreen).
 	 *        + Call [unpublishScreen](#unpublishscreen) to stop publishing screen sharing streams.
-	 * @region Screen Sharing
+	 * @group Screen Sharing
+	 * @order 10
 	 */
 	publishScreen: (mediaType: MediaType) => Promise<void>;
 	
@@ -2704,7 +2794,8 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @param mediaType Media stream type, used for specifying whether to stop publishing audio stream or video stream.
 	 * @notes  + After calling [publishScreen](#publishscreen), call this API to stop publishing streams.
 	 *         + After calling this API, the other users in the room will receive [onUserUnpublishScreen](104479#onuserunpublishscreen).
-	 * @region Screen Sharing
+	 * @group Screen Sharing
+	 * @order 11
 	 */
 	unpublishScreen: (mediaType: MediaType) => Promise<void>;
 	
@@ -2715,7 +2806,8 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @param mediaType Media stream type, used for specifying whether to subscribe to the audio stream or the video stream.
 	 * @notes You must first get the remote stream
 	 * information through [onUserPublishStream](104479#onuserpublishstream) before calling this API to subscribe to streams accordingly.
-	 * @region Room Management
+	 * @group Audio & Video Transport
+	 * @order 5
 	 */
 	subscribeStream: (userId: string, mediaType: MediaType) => Promise<void>;
 	
@@ -2724,7 +2816,8 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 *        You can call this API in both automatic subscription mode and manual subscription mode.
 	 * @param userId The ID of the remote user who published the target audio/video stream.
 	 * @param mediaType Media stream type, used for specifying whether to unsubscribe from the audio stream or the video stream.
-	 * @region Room Management
+	 * @group Audio & Video Transport
+	 * @order 6
 	 */
 	unsubscribeStream: (userId: string, mediaType: MediaType) => Promise<void>;
 	
@@ -2734,7 +2827,8 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @param userId The ID of the remote user who published the target screen audio/video stream.
 	 * @param mediaType Media stream type, used for specifying whether to subscribe to the audio stream or the video stream.
 	 * @notes You must first get the remote stream information through [onUserPublishScreen](104479#onuserpublishscreen) before calling this API to subscribe to streams accordingly.
-	 * @region Room Management
+	 * @group Screen Sharing
+	 * @order 15
 	 */
 	subscribeScreen: (userId: string, mediaType: MediaType) => Promise<void>;
 	
@@ -2743,7 +2837,8 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 *        You can call this API in both automatic subscription mode and manual subscription mode.
 	 * @param uid The ID of the remote user who published the target screen audio/video stream.
 	 * @param type Media stream type, used for specifying whether to unsubscribe from the audio stream or the video stream.
-	 * @region Room Management
+	 * @group Screen Sharing
+	 * @order 16
 	 */
 	unsubscribeScreen: (userId: string, mediaType: MediaType) => Promise<void>;
 	
@@ -2754,7 +2849,8 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @notes + If you call [joinRoom](#joinroom) with an expired token, the joining will fail with an error code `INVALID_TOKEN`. And you need to obtain a valid token and call this API to update the token.
 	 *      + If joining a room failed because the token expired, call this API with an updated token. And then RTC SDK will automatically rejoin the room without calling [joinRoom](#joinroom) again.
 	 *      + The user will be removed from the room when the Token expires.
-	 * @region Room Management
+	 * @group Room Management
+	 * @order 5
 	 */
 	updateToken: (token: string) => void;
 	
@@ -2763,6 +2859,8 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @param deviceId Device ID. Call [enumerateDevices](#enumeratedevices) to get the device list.
 	 * @return
 	 * @notes  If the internal audio capturing is on, the switch is effective once you call this API. If the internal audio capturing is off, the setting will be effective when capture starts.
+	 * @group Audio Management
+	 * @order 0
 	 */
 	setAudioCaptureDevice: (deviceId: string) => Promise<void>;
 	
@@ -2773,6 +2871,8 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @notes + If the internal video capturing is on, the switch is effective once you call this API. If the internal video capturing is off, the setting will be effective when capture starts.
 	 *        + For mobile devices, set deviceId as `"user"` to use the front camera. Set deviceId as `"environment"` to use the rear camera.
 	 *        + The default camera is set by the browser.
+	 * @group Video Management
+	 * @order 0
 	 */
 	setVideoCaptureDevice: (deviceId: string) => Promise<void>;
 	
@@ -2781,35 +2881,39 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 *        Internal video capture refers to: capturing video using the built-in module.
 	 *        The remote clients in the room will be informed of the state change via [onUserStartVideoCapture](104479#onuserstartvideocapture) after the publisher starts video capture by calling this API.
 	 * @param deviceId ID of the capture device which you need to pass in to avoid exceptions such as no screen or a screen that is not the desired one. You can call [enumerateDevices](#enumeratedevices) to obtain the list of devices.
-	 * @return The actual effective video capture parameters. See [MediaTrackSettings](https://developer.mozilla.org/en-US/docs/Web/API/MediaTrackConstraints). Pay attention to the properties: `frameRate`, `height`, and `width`.
+	 * @return The actual effective video capture parameters. See [MediaTrackSettings](https://developer.mozilla.org/en-US/docs/Web/API/MediaTrackSettings). Pay attention to the properties: `frameRate`, `height`, and `width`.
 	 * @notes + Call [stopVideoCapture](#stopvideocapture) to stop the internal video capture. Otherwise, the internal video capture will sustain until you destroy the engine instance.
 	 *        + Once you create the engine instance, you can start internal video capture regardless of the video publishing state. The video stream will start publishing only after the video capture starts.
 	 *        + Call [setVideoCaptureDevice](#setvideocapturedevice) to switch the camera used by the internal video capture module.
+	 * @group Video Management
+	 * @order 5
 	 */
 	startVideoCapture: (deviceId?: string) => Promise<MediaTrackSettings>;
 	
 	/** 
 	 * @brief Enable internal video capture immediately. The default setting is off.
-	 * Internal video capture refers to: capturing video using the built-in module.
-	 * The remote clients in the room will be informed of the state change via [onUserStopVideoCapture](104479#onuserstopvideocapture) after the publisher stops video capture by calling this API.
+	 *        Internal video capture refers to: capturing video using the built-in module.
+	 *        The remote clients in the room will be informed of the state change via [onUserStopVideoCapture](104479#onuserstopvideocapture) after the publisher stops video capture by calling this API.
 	 * @return
 	 * @notes + Call [startVideoCapture](#startvideocapture) to enable the internal video capture.
-	 *     + Without calling this API the internal video capture will sustain until you destroy the engine instance.
-	 * @region Video Management
+	 *        + Without calling this API the internal video capture will sustain until you destroy the engine instance.
+	 * @group Video Management
+	 * @order 6
 	 */
 	stopVideoCapture: () => Promise<void>;
 	
 	/** 
 	 * @brief Start internal audio capture. The default is off.
-	 * Internal audio capture refers to: capturing audio using the built-in module.
-	 * The remote clients in the room will be informed of the state change via [onUserStartAudioCapture](104479#onuserstartaudiocapture) after the publisher starts audio capture by calling this API.
+	 *        Internal audio capture refers to: capturing audio using the built-in module.
+	 *        The remote clients in the room will be informed of the state change via [onUserStartAudioCapture](104479#onuserstartaudiocapture) after the publisher starts audio capture by calling this API.
 	 * @param deviceId ID of the capture device which you need to pass in to avoid exceptions such as no sound. You can call [enumerateDevices](#enumeratedevices) to obtain the list of devices.
-	 * @return
+	 * @return The actual effective audio capture parameters. See [MediaTrackSettings](https://developer.mozilla.org/en-US/docs/Web/API/MediaTrackSettings).
 	 * @notes + Call [stopAudioCapture](#stopaudiocapture) to stop the internal audio capture. Otherwise, the internal audio capture will sustain until you destroy the engine instance.
-	 *     + Once you create the engine instance, you can start internal audio capture regardless of the audio publishing state. The audio stream will start publishing only after the audio capture starts.
-	 * @region Audio Management
+	 *        + Once you create the engine instance, you can start internal audio capture regardless of the audio publishing state. The audio stream will start publishing only after the audio capture starts.
+	 * @group Audio Management
+	 * @order 5
 	 */
-	startAudioCapture: (deviceId?: string) => Promise<void>;
+	startAudioCapture: (deviceId?: string) => Promise<MediaTrackSettings>;
 	
 	/** 
 	 * @brief Stop internal audio capture. The default is off.
@@ -2818,7 +2922,8 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @return
 	 * @notes + Call [startAudioCapture](#startaudiocapture) to enable the internal audio capture.
 	 *        + Without calling this API the internal audio capture will sustain until you destroy the engine instance.
-	 * @region Audio Management
+	 * @group Audio Management
+	 * @order 6
 	 */
 	stopAudioCapture: () => Promise<void>;
 	
@@ -2828,15 +2933,17 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @return
 	 * @notes + Call this API will start screen stream video capture but will not publish the captured video. Publishing a screen-sharing stream by calling [publishScreen](#publishscreen).
 	 *     + To turn off screen video source capture, call [stopScreenCapture](#stopscreencapture).
-	 *     + Before calling this API, you can call [setVideoEncoderConfig](#setvideoencoderconfig) to set the capture frame rate and encoding resolution of the screen video stream.
-	 * @region Advanced Feature
+	 *     + Before calling this API, you can call [setScreenEncoderConfig](#setscreenencoderconfig) to set the capture frame rate and encoding resolution of the screen video stream.
+	 * @group Screen Sharing
+	 * @order 0
 	 */
 	startScreenCapture: (screenConfig?: ScreenConfig) => Promise<void>;
 	
 	/** 
 	 * @brief Stop screen sharing.
 	 * @return
-	 * @region Advanced Feature
+	 * @group Screen Sharing
+	 * @order 1
 	 */
 	stopScreenCapture: () => Promise<void>;
 	
@@ -2847,7 +2954,8 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @param streamIndex Video stream properties
 	 * @param videoPlayerOption View information and rendering mode
 	 * @notes You should bind the view after the user joins a room. This setting will sustain when the user leaves the room.
-	 * @region Video Management
+	 * @group Video Management
+	 * @order 20
 	 */
 	setLocalVideoPlayer: (streamIndex: StreamIndex, videoPlayerOption?: Partial<VideoPlayerOption>) => HTMLVideoElement | undefined;
 	
@@ -2858,14 +2966,16 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @param streamIndex Video stream properties
 	 * @param videoPlayer View information and rendering mode
 	 * @notes You should bind the view after the user joins a room. This setting will sustain when the user leaves the room.
-	 * @region Video Management
+	 * @group Video Management
+	 * @order 21
 	 */
 	setRemoteVideoPlayer: (streamIndex: StreamIndex, videoPlayerOption: VideoPlayerOption) => HTMLVideoElement | undefined;
 	
 	/** 
 	 * @brief Enable or disable the mirror mode for the local video stream.
 	 * @param mirrorType Whether to turn on mirror mode.
-	 * @region Media Processing
+	 * @group Video Management
+	 * @order 30
 	 */
 	setLocalVideoMirrorType: (mirrorType: MirrorType) => void;
 	
@@ -2874,7 +2984,8 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @param deviceId ID of the audio playback device. Call [enumerateDevices](#enumeratedevices) to get device ID.
 	 * @return
 	 * @notes  You can call this API before or after the user joins a room.
-	 * @region Audio Management
+	 * @group Audio Management
+	 * @order 1
 	 */
 	setAudioPlaybackDevice: (deviceId: string) => Promise<void>;
 	
@@ -2886,9 +2997,10 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @param mediaType Media stream type, used for specifying whether to publish audio stream or video stream.
 	 * @param streamIndex Video stream properties
 	 * @return Calling this API when the audio and video are playing will also return a Promise.
-	 * @region Room Management
 	 * @note + Call this API to start playback if the audio and/or video is not playing.
 	 * + Call this API once to start both audio and video playbacks when they do not start.
+	 * @group Video Management
+	 * @order 25
 	 */
 	play: (userId?: string, mediaType?: MediaType, streamIndex?: StreamIndex) => Promise<void>;
 	
@@ -2899,7 +3011,8 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @notes + Calling this API does not change the capture state and the transmission state of the remote clients.
 	 *         + Calling this API does not cancel the subscription or change any subscription configuration.
 	 *         + To resume, call [resumeAllSubscribedStream](#resumeallsubscribedstream).
-	 * @region Media Stream Transmission
+	 * @group Audio & Video Transport
+	 * @order 15
 	 */
 	pauseAllSubscribedStream: (mediaType: MediaType) => Promise<void>;
 	
@@ -2909,18 +3022,20 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @return
 	 *  @notes + Calling this API does not change the capture state and the transmission state of the remote clients.
 	 *         + Calling this API does not change any subscription configuration.
-	 * @region Media Transmission
+	 * @group Audio & Video Transport
+	 * @order 16
 	 */
 	resumeAllSubscribedStream: (mediaType: MediaType) => Promise<void>;
 	
 	/** 
 	 * @brief Send point-to-point text messages to a specified user in the room.
-	 * The recipient receives the message via [onUserMessageReceived](104479#onusermessagereceived)  once the message is sent successfully.
+	 *        The recipient receives the message via [onUserMessageReceived](104479#onusermessagereceived)  once the message is sent successfully.
 	 * @param userId User ID of the recipient
 	 * @param message Text to be sent within 64 KB
 	 * @return
 	 * @notes Call [joinRoom](#joinroom) to join the room before calling this API.
-	 * @region Message
+	 * @group Message
+	 * @order 0
 	 */
 	sendUserMessage: (userId: string, message: string) => Promise<unknown> | undefined;
 	
@@ -2931,25 +3046,28 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @param message Binary to be sent within 64 KB
 	 * @return
 	 * @notes Call [joinRoom](#joinroom) to join the room before calling this API.
-	 * @region Message
+	 * @group Message
+	 * @order 1
 	 */
 	sendUserBinaryMessage: (userId: string, message: ArrayBuffer) => Promise<unknown>;
 	
 	/** 
 	 * @brief Send text messaging to all other users in the room.
-	 * Users in the room will receive message via [onRoomMessageReceived](104479#onroommessagereceived) once the message is sent successfully.
+	 *        Users in the room will receive message via [onRoomMessageReceived](104479#onroommessagereceived) once the message is sent successfully.
 	 * @param message Text to be sent within 64 KB
 	 * @return
-	 * @region Message
+	 * @group Message
+	 * @order 5
 	 */
 	sendRoomMessage: (message: string) => Promise<any> | undefined;
 	
 	/** 
 	 * @brief Send binary messaging to all other users in the room.
-	 * Users in the room will receive binary message via [onRoomBinaryMessageReceived](104479#onroombinarymessagereceived) once the message is sent successfully.
+	 *        Users in the room will receive binary message via [onRoomBinaryMessageReceived](104479#onroombinarymessagereceived) once the message is sent successfully.
 	 * @param message Binary message to be sent within 64 KB
 	 * @return
-	 * @region Message
+	 * @group Message
+	 * @order 6
 	 */
 	sendRoomBinaryMessage: (message: ArrayBuffer) => Promise<any> | undefined;
 	
@@ -2962,24 +3080,26 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @notes + Call this API to add SEI message to the video frame captured by the internal video capture module.
 	 *        + SDK will abandon the SEI message if no video frame in 2s after you call this API.
 	 *        + Users receiving the stream will receive the SEI message via [onSEIMessageReceived](104479#onseimessagereceived) once the SEI message is sent successfully.
-	 * @region Message
+	 * @group Message
+	 * @order 10
 	 */
 	sendSEIMessage: (streamIndex: StreamIndex, message: Uint8Array | string, repeatCount: number) => void;
 	
 	/** 
-	 * @brief Sets the video encoder configuration of the publishing streams.
-	 * @param descriptions List of configurations for multiple streams. You should note that the resolution you set is the maximum resolution of each stream.
-	 *        You can set parameters for up to 3 streams, SDK will always take the value of the first 3 streams when parameters for more streams are set.
-	 *        The resolutions and the frame rates must be in descending order.
-	 *        When the resolution of the max stream is in [540P, 1080P], you can simulcast 3 streams; when in [270P, 450P], you can simulcast 2 streams; when in [0, 180P], you can not enable simulcast.
-	 * @notes + The default parameters are 640 × 480 px, 15 FPS, with 600 Kbps max bitrate.
-	 *        + You can call this API before and after publishing video streams. After the video stream is published, you cannot switch from single stream to multi-stream, or modify the number of published multi-stream layers.
-	 *        + When using mobile devices to publish video streams in the vertical holding state, the set resolution width and height should be the opposite of the actual sending parameters. For example, when setting the send resolution to 640 × 480 px, the actual send resolution is 480 × 640 px.
-	 *        + When simulcasting, it is recommended to cancel automatic publishing and publish audio/video streams manually by calling [publishStream](#publishstream), otherwise simulcast will not take effect.
-	 *        + If the set parameters are not supported by the browser, RTC SDK will capture and encode according to the range of parameters supported by the browser, and report the actual parameters of the capture by [onLocalVideoSizeChanged](104479#onlocalvideosizechanged).
-	 *        + You need to call [enableSimulcastMode](#enablesimulcastmode) to start simulcasting. If you set the parameters for multiple streams without calling [enableSimulcastMode](#enablesimulcastmode), only the stream with the largest resolution will be published, and the setting will take effect after simulcasting is enabled. If simulcasting is enabled, but only one stream's parameters are set, RTC SDK will automatically fill 1~2 groups of smaller stream parameters.
-	 *        + This API is applicable to the video stream captured by the camera, see [setScreenEncoderConfig](#setscreenencoderconfig) for setting parameters for screen sharing video stream.
-	 *        + If simulcasting is started, for better compatibility, it is recommended to set the ratio of the maximum level resolution to other level resolutions as an integer multiple.
+	 * @brief Set the video stream configurations from the video publishing end.
+	 * @param descriptions The configurations of multiple video streams to be published. The resolution set is the maximum resolution for each stream.
+	 *                     Up to 3 sets of configurations can be set. When there are more than 3 sets, the values of the first 3 sets will be used.
+	 *                     When multiple configurations are set, the resolution and frame rate must be arranged in descending order. For better compatibility, it is recommended that the ratio between the maximum level resolution and other level resolutions be an integer multiple.
+	 *                     When the resolution of the first stream is greater than or equal to 540P, it supports publishing 3 streams; when it is between 270P (inclusive) and 450P (inclusive), it supports publishing 2 streams; when it is 180P or below, you cannot enable simulcasting.
+	 * @notes + Before calling this method, the SDK defaults to publishing a stream with a resolution of 640 × 480 pixels, a frame rate of 15 FPS, and a maximum encoding bitrate of 600 Kbps.
+	 *        + When publishing a single stream, you can call this method before or after publishing the video stream. After the video stream is published, dynamic switching between single stream and multiple streams is not supported, and modifying the parameters of the published multiple streams is not supported.
+	 *        + When using a mobile device to publish a video stream in portrait mode, the set resolution width and height are reversed in the actual streaming parameters. For example, when setting the sending resolution to 640 × 480 pixels, the actual streaming parameters are 480 × 640 pixels.
+	 *        + If the set parameters are not supported by the browser, the RTC SDK will capture and encode based on the parameter range supported by the browser, and notify the actual parameters through the [onLocalVideoSizeChanged](104479#onlocalvideosizechanged) callback.
+	 *        + If you want to simulcast, you need to disable automatic publishing, call this method and [enableSimulcastMode](#enablesimulcastmode) to enable multiple stream mode, and then call [publishStream](#publishstream) to manually publish the audio and video streams.
+	 *        + If this method is called to set multiple resolutions but multiple stream mode is not enabled, the SDK will default to publishing the stream with the highest resolution. If multiple stream publishing is enabled but only one set of stream parameters is set, the SDK will automatically fill in 1-2 sets of small stream parameters.
+	 *        + This method is applicable to video streams captured by the camera. For setting the screen sharing stream parameters, refer to [setScreenEncoderConfig](#setscreenencoderconfig).
+	 * @group Video Management
+	 * @order 35
 	 */
 	setVideoEncoderConfig: (descriptions: VideoEncoderConfig | VideoEncoderConfig[]) => Promise<void>;
 	
@@ -2989,6 +3109,8 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @param description The encoding configuration of the screen-sharing stream.
 	 * @notes + The default parameters are 1920 × 1080 px, 15 FPS, with 3000 Kbps max bitrate.
 	 *        + We recommend that you set the encoding configuration before video capture. Otherwise, the video will be captured based on the default configuration.
+	 * @group Screen Sharing
+	 * @order 5
 	 */
 	setScreenEncoderConfig: (description: ScreenEncoderConfig) => Promise<void>;
 	
@@ -2998,17 +3120,19 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @param remoteVideoConfig The parameters you expect to configure for the remote video stream.
 	 * @notes + This API only works after the publisher calls [enableSimulcastMode](#enablesimulcastmode) to enable publishing multiple video streams, in which case the subscriber will receive a stream that is closest to the set configuration.
 	 *        + If you don't call this API after the publisher enables the function of publishing multiple streams, you will receive by default the video stream with the largest resolution set by the publisher.
-	 *        + You should call this API in the room. If you want to call it before entering the room, you should set the `remoteVideoConfig` in the roomConfig` when calling [joinRoom](#joinroom).
-	 * @region Video Management
+	 *        + You should call this API in the room. If you want to call it before entering the room, you should set the `remoteVideoConfig` in the `roomConfig` when calling [joinRoom](#joinroom).
+	 * @group Audio & Video Transport
+	 * @order 20
 	 */
 	setRemoteVideoConfig: (userId: string, remoteVideoConfig: RemoteVideoConfig) => Promise<void>;
 	
 	/** 
-	 * @region Video Management
 	 * @brief Enables the simulcast mode of the video stream captured by the camera at the publishing end.imulcasting, including resolution, frame rate, bitrate, scale mode, fallback strategy under poor network conditions.
 	 * @param enabled
 	 * @param + You must call this API before publishing the video streams.
-	 *    + After enabling the simulcast mode, SDK automatically sets the encoding parameters for the video streams. For manual setting, call [setVideoEncoderConfig](#setvideoencoderconfig).
+	 *        + After enabling the simulcast mode, SDK automatically sets the encoding parameters for the video streams. For manual setting, call [setVideoEncoderConfig](#setvideoencoderconfig).
+	 * @group Audio & Video Transport
+	 * @order 25
 	 */
 	enableSimulcastMode: (enabled: boolean) => boolean;
 	
@@ -3022,7 +3146,6 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 *        It is recommended to set to greater than or equal to 200 milliseconds; when less than 10 milliseconds, the behavior is undefined.
 	 *         + `≤ 0`: Disable the volume prompt function.
 	 *         + `> 0`: Enable the volume prompt function and set the time interval for receiving the volume prompt callback. In milliseconds.
-	 * @region Room Management
 	 */
 	setAudioVolumeIndicationInterval: (interval?: number) => void;
 	
@@ -3032,7 +3155,8 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @notes With the prompt on, you can:
 	 *       + Get the information of the audio stream captured by the local microphone and screen audio stream through [onLocalAudioPropertiesReport](104479#onlocalaudiopropertiesreport).
 	 *       + Get the information of the subscribed remote audio streams through [onRemoteAudioPropertiesReport](104479#onremoteaudiopropertiesreport).
-	 * @region Audio Management
+	 * @group Audio Management
+	 * @order 35
 	 */
 	enableAudioPropertiesReport: (config?: AudioPropertiesConfig) => void;
 	
@@ -3040,9 +3164,12 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @brief Starts pushing media streams to CDN and sets the relevant configurations.
 	 * @param transcode Configurations to be set when pushing media streams to CDN.
 	 * @return
-	 * @notes + After calling this API, you will receive the pushing a mixed stream to CDN events via [onStreamMixingEvent](104479#onstreammixingevent).
+	 * @notes + Before calling this API，you need to enable Push to CDN on the [console](https://console.byteplus.com/rtc/workplaceRTC).
+	 *        + After calling this API, you will receive the pushing a mixed stream to CDN events via [onStreamMixingEvent](104479#onstreammixingevent).
+	 *        + If you have subscribed to the push-to-CDN server callback in the [console](https://console.byteplus.com/rtc/cloudRTC?tab=callback), calling this API will receive a [TranscodeStarted](https://docs.byteplus.com/en/byteplus-rtc/docs/75125#transcodestarted) server callback notification. When calling this API repeatedly, subsequent calls to this API will trigger both [TranscodeStarted](https://docs.byteplus.com/en/byteplus-rtc/docs/75125#transcodestarted) and [TranscodeUpdated](https://docs.byteplus.com/en/byteplus-rtc/docs/75125#transcodeupdated) callbacks.
 	 *        + Call [stopLiveTranscoding](#stoplivetranscoding) to stop pushing mixed streams to CDN.
-	 * @region Advanced Feature
+	 * @group Push to CDN
+	 * @order 5
 	 */
 	startLiveTranscoding: (transcode: LiveTranscodeConfig) => Promise<void>;
 	
@@ -3051,6 +3178,8 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 *        After calling [startLiveTranscoding](104478#startlivetranscoding) to enable the function of pushing streams to CDN, you can call this API to update the relevant configurations.
 	 * @param transcode Updated configurations. Some of the configurations cannot be updated. See [LiveTranscodeConfig](104481#livetranscodeconfig) for specific indications.
 	 * @return
+	 * @group Push to CDN
+	 * @order 6
 	 */
 	updateLiveTranscoding: (transcode: Partial<LiveTranscodeConfig>) => Promise<void>;
 	
@@ -3058,7 +3187,8 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @brief Stop pushing media streams to CDN
 	 * For starting pushing media streams to CDN. Refer to [startLiveTranscoding](104478#startlivetranscoding).
 	 * @return
-	 * @region Advanced Feature
+	 * @group Push to CDN
+	 * @order 7
 	 */
 	stopLiveTranscoding: () => Promise<void>;
 	
@@ -3068,7 +3198,8 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @param businessId Your customized businessId
 	 *                   BusinessId is a tag, and you can customize its granularity.
 	 * @notes  You must call this API before the [joinRoom](#joinroom) API, otherwise it will be invalid.
-	 * @region Engine Management
+	 * @group Engine Management
+	 * @order 5
 	 */
 	setBusinessId: (businessId: string) => boolean;
 	
@@ -3082,7 +3213,8 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 *        + When you call this API, the other users in the room will be informed via the related callback:
 	 *            - Switch from `false` to `true`: [onUserJoined](104479#onuserjoined)
 	 *            - Switch from `true` to `false`: [onUserLeave](104479#onuserleave)
-	 * @region Room Management
+	 * @group Room Management
+	 * @order 10
 	 */
 	setUserVisibility: (enable: boolean) => Promise<void>;
 	
@@ -3092,7 +3224,8 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @param index specify a stream index.
 	 * @param track specify a video track. See [MediaStreamTrack](https://developer.mozilla.org/en-US/docs/Web/API/MediaStreamTrack).
 	 * @return
-	 * @region Video Management
+	 * @group Video Management
+	 * @order 15
 	 */
 	setExternalVideoTrack: (index: StreamIndex, track: MediaStreamTrack) => Promise<void>;
 	
@@ -3102,7 +3235,8 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @param index specify a stream index.
 	 * @param track specify an audio track. See [MediaStreamTrack](https://developer.mozilla.org/en-US/docs/Web/API/MediaStreamTrack).
 	 * @return
-	 * @region Audio Management
+	 * @group Audio Management
+	 * @order 15
 	 */
 	setExternalAudioTrack: (index: StreamIndex, track: MediaStreamTrack) => Promise<void>;
 	
@@ -3112,7 +3246,8 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @param index audio stream index
 	 * @param videoSourceType Type of the module for video capture
 	 * @return
-	 * @region Video Management
+	 * @group Video Management
+	 * @order 10
 	 */
 	setVideoSourceType: (index: StreamIndex, videoSourceType: VideoSourceType) => Promise<void>;
 	
@@ -3122,13 +3257,16 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @param index audio stream index
 	 * @param audioSourceType Type of the module for video capture
 	 * @return
+	 * @group Audio Management
+	 * @order 10
 	 */
 	setAudioSourceType: (index: StreamIndex, audioSourceType: AudioSourceType) => Promise<void>;
 	
 	/** 
 	 * @brief Create an audio mixing manager object
 	 * @return audio mixing manager object
-	 * @region Advanced Feature
+	 * @group Audio Mixing
+	 * @order 0
 	 */
 	getAudioMixingManager: () => IAudioMixingManager;
 	
@@ -3137,6 +3275,8 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @param config Configuration of the internal audio capture such as, disabling audio 3A processing. Refer to [MediaTrackConstraints](https://developer.mozilla.org/zh-CN/docs/Web/API/MediaTrackConstraints) for more details.
 	 * @return
 	 * @notes You should call this API before starting audio capture and publishing streams.
+	 * @group Audio Management
+	 * @order 30
 	 */
 	setAudioCaptureConfig: (config: TrackCaptureConfig) => Promise<void>;
 	
@@ -3154,7 +3294,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	
 	/** 
 	 * @brief Log in to the RTM server.
-	 * Call this method before users within the same `appID` can communicate with each other via messages.
+	 *        Call this method before users within the same `appID` can communicate with each other via messages.
 	 * @param token Token for authentication.
 	 *         During testing, you can use generate a temporary Token on [Console](https://console.byteplus.com/auth/login), and fill in any value with `roomId`.
 	 *         To officially go live, you need to use the implement token generation at your server. The `roomId` field should be empty. See [Authentication with Token](70121).
@@ -3162,12 +3302,16 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @return
 	 * @notes + Call [logout](#logout) to log out the IM server.
 	 *        + In the same App ID, the user ID of each user must be unique. The user will be logged out when a user with the same userID joins the room because of the error of `RTM_DUPLICATE_LOGIN`.
+	 * @group Message
+	 * @order 15
 	 */
 	login: (token: string | null, userId: string) => Promise<void>;
 	
 	/** 
-	 * After calling this API to logout you are not able to call APIs to send messages and server messages.
+	 * @brief After calling this API to logout you are not able to call APIs to send messages and server messages.
 	 * @return
+	 * @group Message
+	 * @order 16
 	 */
 	logout: () => Promise<void>;
 	
@@ -3176,7 +3320,9 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @param token updated token
 	 * @return
 	 * @notes + Call this API to refresh the token after the token has expired with no need for calling [login](#login).
-	 *  + You will be informed of the expiration of the token if you tried to login with an expired token or during a reconnection. If the token expired after the user has logged in, no actions need to be taken.
+	 *        + You will be informed of the expiration of the token if you tried to login with an expired token or during a reconnection. If the token expired after the user has logged in, no actions need to be taken.
+	 * @group Message
+	 * @order 20
 	 */
 	updateLoginToken: (token: string | null) => Promise<void>;
 	
@@ -3185,6 +3331,8 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @param userId ID of the user whose status intended to know
 	 * @return
 	 * @notes Call [login](#login) before calling this API.
+	 * @group Message
+	 * @order 25
 	 */
 	getPeerOnlineStatus: (userId: string) => Promise<USER_ONLINE_STATUS>;
 	
@@ -3194,7 +3342,9 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @param message Text message within 64 KB.
 	 * @return
 	 * @notes + Call [login](#login) before calling this API.
-	 * + The recipient must be logged into the server and will receive the text message via [onUserMessageReceivedOutsideRoom](104479#onusermessagereceivedoutsideroom) if the message has been sent successfully.
+	 *        + The recipient must be logged into the server and will receive the text message via [onUserMessageReceivedOutsideRoom](104479#onusermessagereceivedoutsideroom) if the message has been sent successfully.
+	 * @group Message
+	 * @order 30
 	 */
 	sendUserMessageOutsideRoom: (userId: string, message: string) => Promise<any>;
 	
@@ -3202,9 +3352,11 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @brief Send point-to-point (P2P) binary message to a user.
 	 * @param userId ID of the recipient
 	 * @param message Binary message within 64 KB
-	 * @return + Call [login](#login) before calling this API.
-	 *  + The recipient must be logged into the server and will receive the text message via [EngineEventsTypes.onUserBinaryMessageReceivedOutsideRoom](104479#onuserbinarymessagereceivedoutsideroom) if the message has been sent successfully.
-	 * @notes
+	 * @return
+	 * @notes + Call [login](#login) before calling this API.
+	 *        + The recipient must be logged into the server and will receive the text message via [EngineEventsTypes.onUserBinaryMessageReceivedOutsideRoom](104479#onuserbinarymessagereceivedoutsideroom) if the message has been sent successfully.
+	 * @group Message
+	 * @order 31
 	 */
 	sendUserBinaryMessageOutsideRoom: (userId: string, message: ArrayBuffer) => Promise<any>;
 	
@@ -3214,7 +3366,9 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @param url Address of the IM server
 	 * @return
 	 * @notes + Call [login](#login) before calling this API.
-	 * + [onServerParamsSetResult](104479#onserverparamssetresult) notifies you whether the setting successes.
+	 *        + [onServerParamsSetResult](104479#onserverparamssetresult) notifies you whether the setting successes.
+	 * @group Message
+	 * @order 35
 	 */
 	setServerParams: (signature: string, url: string) => Promise<any>;
 	
@@ -3222,7 +3376,9 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @brief Send point-to-point (P2P) text message to the server.
 	 * @param message Text message within 64 KB.
 	 * @return
-	 * @notes
+	 * @notes Before sending a message to the server, you must first call [login](#login) to complete the login process.
+	 * @group Message
+	 * @order 36
 	 */
 	sendServerMessage: (message: string) => Promise<any>;
 	
@@ -3230,7 +3386,9 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @brief Send point-to-point (P2P) binary message to the server.
 	 * @param message Binary message within 64 KB.
 	 * @return
-	 * @notes
+	 * @notes Before sending a message to the server, you must first call [login](#login) to complete the login process.
+	 * @group Message
+	 * @order 37
 	 */
 	sendServerBinaryMessage: (message: ArrayBuffer) => Promise<any>;
 	
@@ -3253,7 +3411,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	/** 
 	 * @hidden
 	 * @brief Starts publishing a public media stream.
-	 * @param publicStreamId The public stream's ID.
+	 * @param publicStreamId The public stream's ID. The string must conform to the regular expression: `[a-zA-Z0-9_@\-\.]{1,128}`.
 	 * @param publicStreamParam The public stream's parameters.
 	 *                 A public stream can contain several media streams and will appears as the designated layout.
 	 *                 If the designated media stream is not published yet, the public stream will automatically update once the media stream is published.
@@ -3263,6 +3421,8 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 *        + Users with different `userID` can call this API with the same stream ID to update the parameters.
 	 *        + Call this API with different stream ID to publish multiple public streams.
 	 *        + Call [stopPushPublicStream](#stoppushpublicstream) to stop publishing the public stream.
+	 * @group Public Stream
+	 * @order 0
 	 */
 	startPushPublicStream: (publicStreamId: string, publicStreamParam: PublicStreamConfig) => Promise<void>;
 	
@@ -3274,6 +3434,8 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 *                       The public stream must be published by the current user.
 	 * @param publicStreamParam The public stream's parameters.
 	 * @return
+	 * @group Public Stream
+	 * @order 1
 	 */
 	updatePublicStreamParam: (publicStreamId: string, publicStreamParam: Partial<PublicStreamConfig>) => Promise<void>;
 	
@@ -3285,6 +3447,8 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @param publicStreamId The public stream's ID.
 	 *                       The public stream must be published by the current user.
 	 * @return
+	 * @group Public Stream
+	 * @order 2
 	 */
 	stopPushPublicStream: (publicStreamId: string) => Promise<void>;
 	
@@ -3299,6 +3463,8 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 *        + After calling this API, you will receive [onFirstPublicStreamVideoFrameDecoded](104479#onfirstpublicstreamvideoframedecoded) and [onFirstPublicStreamAudioFrameDecoded](104479#onfirstpublicstreamaudioframedecoded) once the first frame has been decoded.
 	 *        + If the public stream contains SEI message, you will receive the message by be informed through [onPublicStreamSEIMessageReceived](104479#onpublicstreamseimessagereceived).
 	 *        + Call [stopPlayPublicStream](#stopplaypublicstream) to unsubscribing from the public stream.
+	 * @group Public Stream
+	 * @order 5
 	 */
 	startPlayPublicStream: (publicStreamId: string) => Promise<void>;
 	
@@ -3308,6 +3474,8 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 *        Call [startPlayPublicStream](104478#startplaypublicstream) to subscribe to the public stream.
 	 * @param publicStreamId The public stream's ID.
 	 * @return
+	 * @group Public Stream
+	 * @order 6
 	 */
 	stopPlayPublicStream: (publicStreamId: string) => Promise<void>;
 	
@@ -3316,11 +3484,12 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @brief Sets the player information and rendering mode for the public stream.
 	 * @param publicStreamId The public stream's ID.
 	 * @param videoPlayerOption Player information and rendering mode.
+	 * @group Public Stream
+	 * @order 10
 	 */
 	setPublicStreamVideoPlayer: (publicStreamId: string, videoPlayerOption: Omit<VideoPlayerOption, "userId">) => void;
 	
 	/** 
-	 * @region Video Management
 	 * @brief Set the placeholder image during local video capture pause
 	 * @param index
 	 * @param filePath Image path in Http protocal; or Blob object.
@@ -3329,12 +3498,16 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @notes + Only available to Chrome browser on Mac
 	 * + It's recommended to call this API before stopping video capture to avoid blank frame in video recording.
 	 * + You cannot see the image during local preview.
+	 * @group Video Management
+	 * @order 40
 	 */
 	setDummyCaptureImagePath: (index: StreamIndex, filePath: string) => Promise<void>;
 	
 	/** 
 	 * @brief Get the default parameters of `LiveTranscodeConfig`.
-	 * @return The default parameters of `LiveTranscodeConfig`
+	 * @return The default parameters of `LiveTranscodeConfig`.
+	 * @group Push to CDN
+	 * @order 0
 	 */
 	defaultTranscoding: () => Required<LiveTranscodeConfig>;
 	
@@ -3347,6 +3520,8 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @notes + You must call this API before joining the room.
 	 *        + After you enable the fallback, you will receive [onSimulcastSubscribeFallback](104479#onsimulcastsubscribefallback) and [onRemoteVideoSizeChanged](104479#onremotevideosizechanged) when the resolution of your subscribed stream is lowered or restored.
 	 *        + You can set media stream fallback using API or cloud configuration. The latter takes a higher priority.
+	 * @group Audio & Video Transport
+	 * @order 30
 	 */
 	setSubscribeFallbackOption: (option: SubscribeFallbackOption) => boolean;
 	
@@ -3358,6 +3533,8 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @param priority The priority of the remote user's media stream
 	 * @return + `true` Success
 	 *         + `false` Failure
+	 * @group Audio & Video Transport
+	 * @order 35
 	 */
 	setRemoteUserPriority: (userId: string, priority: RemoteUserPriority) => boolean;
 	
@@ -3368,39 +3545,50 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @return Resolve when the test ends.
 	 * @notes + You can call this API whether the user is in the room.
 	 * + Call [stopAudioPlaybackDeviceTest](#stopaudioplaybackdevicetest) to stop the playback test before moving on to the other device tests.
+	 * @group Device Management
+	 * @order 20
 	 */
 	startAudioPlaybackDeviceTest: (filePath: string, indicationInterval: number) => Promise<void>;
 	
 	/** 
 	 * @brief Stop the playback test for the local audio device.
 	 * @notes Call this API to stop the playback test started by calling [startAudioPlaybackDeviceTest](#startaudioplaybackdevicetest) before moving on to the other device tests.
+	 * @group Device Management
+	 * @order 21
 	 */
 	stopAudioPlaybackDeviceTest: () => void;
 	
 	/** 
 	 * @brief Start the capture and playback test for local audio devices.
-	 * We recommend to register [onLocalAudioPropertiesReport](104479#onlocalaudiopropertiesreport) callback via [enableAudioPropertiesReport](104478#enableaudiopropertiesreport). Once the test starts, you can get the volume of the capturing audio by this callback.
+	 *        We recommend to register [onLocalAudioPropertiesReport](104479#onlocalaudiopropertiesreport) callback via [enableAudioPropertiesReport](104478#enableaudiopropertiesreport). Once the test starts, you can get the volume of the capturing audio by this callback.
 	 * @param indicationInterval The time interval between each callback of [onAudioPlaybackDeviceTestVolume](104479#onaudioplaybackdevicetestvolume) via which you get informed of the playback volume during the test.  We recommend setting it to 200 ms. The minimal value is 100 ms.
 	 * @param onAutoplayFailed Autoplay fails due to the limitation of the browser. We recommend to execute `resume` as soon as the user clicks on the UI element.
 	 * @return Resolve when the test ends.
 	 * @notes + You can call this API whether the user is in a room.
 	 *        + You can call [setCaptureVolume]({{api}}#RTCEngine.setCaptureVolume) to adjust the capture volume during audio route testing. You cannot call [setPlaybackVolume]({{api}}#RTCEngine.setPlaybackVolume) during audio route testing.
-	 *        + The audio capturing will stop and begins to play the recording in 30 s after calling this API. Before that, call [stopAudioDeviceRecordAndPlayTest](#stopaudiodevicerecordandplaytest) to stop capturing and start playing the recording.
+	 *        + The audio capturing will stop and begins to play the recording in 30 seconds after calling this API. Before that, call [stopAudioDeviceRecordAndPlayTest](#stopaudiodevicerecordandplaytest) to stop capturing and start playing the recording.
 	 *        + You must stop the test before starting another test for audio devices.
 	 *        + Call [stopAudioDevicePlayTest](#stopaudiodeviceplaytest) to stop the test, including capturing and playing the recording.
 	 *        + This test performs locally and does not involve network connection testing.
+	 * @group Device Management
+	 * @order 25
 	 */
 	startAudioDeviceRecordTest: (indicationInterval: number, onAutoplayFailed?: (resume: () => Promise<any>) => any) => Promise<void>;
 	
 	/** 
 	 * @brief Call this API to stop recording in the test and start to play the recording.
+	 *        After calling [startAudioDeviceRecordTest](104478#startaudiodevicerecordtest), you need to call this API within 30 seconds. Otherwise, the recording will automatically stop and begin playing back the recorded audio.
 	 * @notes After calling this API, the recording starts playing during which you can call [stopAudioDevicePlayTest](104478#stopaudiodeviceplaytest) to stop playing.
+	 * @group Device Management
+	 * @order 26
 	 */
 	stopAudioDeviceRecordAndPlayTest: () => void;
 	
 	/** 
 	 * @brief Stop the capture and playback test for local audio devices which is started by calling [startAudioDeviceRecordTest](104478#startaudiodevicerecordtest).
-	 * Before the test ends by itself, you can call this API to stop the recording or playing.
+	 *        Before the test ends by itself, you can call this API to stop the recording or playing.
+	 * @group Device Management
+	 * @order 27
 	 */
 	stopAudioDevicePlayTest: () => void;
 	
@@ -3409,23 +3597,32 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @param profile  A sound quality option consists of a set of audio profiles and applying each brings a different listening experience. Choose the appropriate audio quality to meet the requirement of the scenario.
 	 * @notes You should call this API before starting audio capture and publishing streams.
 	 * @return
+	 * @group Audio Management
+	 * @order 20
 	 */
 	setAudioProfile: (profile: AudioProfileType) => Promise<void>;
 	
+	/** 
+	 * @hidden
+	 */
+	setAudioEncodeMaxBitrate: (maxBitrate: number) => Promise<void>;
+	
 	/** 
 	 * @brief Get the [MediaStreamTrack](https://developer.mozilla.org/en-US/docs/Web/API/MediaStreamTrack) of the local audios or videos.
 	 * @param streamIndex Mainstream or screen-sharing stream.
 	 * @param type Video or audio
-	 * Audio: Track of the sounds collected from the microphone, customized audio track, audio files, and the sound card.
-	 * Video: Track of the videos collected from the camera, customized video track and screen-sharings.
+	 *             Audio: Track of the sounds collected from the microphone, customized audio track, audio files, and the sound card.
+	 *             Video: Track of the videos collected from the camera, customized video track and screen-sharings.
 	 * @return `undefined` for the MediaStreamTrack does not exist. For example, the local media capturing has not started.
 	 * @notes + Call this API after you start the local media capture or designate a customized MediaStreamTrack.
-	 * + Callback of the MediaStreamTrack will inform you by the event of `ended` that the local media capture stops, probably caused either by an API call or the removal of devices.
-	 * + You can get the following info from the MediaStreamTrack: video resolution, device ID, and volume. You can manage the track actions by the method of `stop` or `mute`. You can also change the properties of the stream, such as audio volume.
-	 * + We recommend to call RTC APIs to enable your app with the following capabilities instead of using mediaStreamTrack.<br/>
-	 *   &ensp; 1. Get the video resolution or audio volume of the local streams: [onLocalStreamStats](104479#onlocalstreamstats)
-	 *   &ensp; 2. Get device IDs: [enumerateDevices](#enumeratedevices)
-	 *   &ensp; 3. stop: [stopAudioCapture](#stopaudiocapture), [stopVideoCapture](#stopvideocapture)
+	 *        + Callback of the MediaStreamTrack will inform you by the event of `ended` that the local media capture stops, probably caused either by an API call or the removal of devices.
+	 *        + You can get the following info from the MediaStreamTrack: video resolution, device ID, and volume. You can manage the track actions by the method of `stop` or `mute`. You can also change the properties of the stream, such as audio volume.
+	 *        + We recommend to call RTC APIs to enable your app with the following capabilities instead of using mediaStreamTrack.<br/>
+	 *          &ensp; - Get the video resolution or audio volume of the local streams: [onLocalStreamStats](104479#onlocalstreamstats)
+	 *          &ensp; - Get device IDs: [enumerateDevices](#enumeratedevices)
+	 *          &ensp; - Stop capturing: [stopAudioCapture](#stopaudiocapture), [stopVideoCapture](#stopvideocapture)
+	 * @group Custom Stream Processing
+	 * @order 0
 	 */
 	getLocalStreamTrack: (streamIndex: StreamIndex, type: "video" | "audio") => MediaStreamTrack | undefined;
 	
@@ -3436,11 +3633,13 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @param type Video or audio
 	 * @return `undefined` for the MediaStreamTrack does not exist. For example, the local user has not subscribed to the remote stream.
 	 * @notes + Call this API after the local user subscribe to the remote media stream.
-	 * + Callback of the MediaStreamTrack will inform you by the event of `ended` that the remote media stream stops, probably caused by the API call of [unsubscribeStream](#unsubscribestream) or [unsubscribeScreen](#unsubscribescreen).
-	 * + You can get video resolution and audio volume from the MediaStreamTrack. You can manage the track actions by the method of `stop` or `mute`. You can also change the properties of the stream, such as audio volume.
-	 * + We recommend to call RTC APIs to enable your app with the following capabilities instead of using mediaStreamTrack.<br/>
-	 *   &ensp; 1. Get the video resolution or audio volume of the remote streams: [onRemoteStreamStats]({{callback}}#onremotestreamstats)
-	 *   &ensp; 2. Unsubscribe: [unsubscribeStream](#unsubscribestream), [unsubscribeScreen](#unsubscribescreen)
+	 *        + Callback of the MediaStreamTrack will inform you by the event of `ended` that the remote media stream stops, probably caused by the API call of [unsubscribeStream](#unsubscribestream) or [unsubscribeScreen](#unsubscribescreen).
+	 *        + You can get video resolution and audio volume from the MediaStreamTrack. You can manage the track actions by the method of `stop` or `mute`. You can also change the properties of the stream, such as audio volume.
+	 *        + We recommend to call RTC APIs to enable your app with the following capabilities instead of using mediaStreamTrack.<br/>
+	 *          &ensp; - Get the video resolution or audio volume of the remote streams: [onRemoteStreamStats]({{callback}}#onremotestreamstats)
+	 *          &ensp; - Unsubscribe: [unsubscribeStream](#unsubscribestream), [unsubscribeScreen](#unsubscribescreen)
+	 * @group Custom Stream Processing
+	 * @order 1
 	 */
 	getRemoteStreamTrack: (userId: string, streamIndex: StreamIndex, type: "video" | "audio") => MediaStreamTrack | undefined;
 	
@@ -3450,7 +3649,9 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @param type Video or audio
 	 * @return `undefined` for the MediaStreamTrack does not exist. For example, the public stream does not exist.
 	 * @notes + Call this API after you start playing the public stream by calling [`stopPlayPublicStream`](#stopplaypublicstream).
-	 * + Callback of the MediaStreamTrack will inform you the event of `ended` that the public stream stops, probably caused by the API call of [`stopPlayPublicStream`](#stopplaypublicstream).
+	 *        + Callback of the MediaStreamTrack will inform you the event of `ended` that the public stream stops, probably caused by the API call of [`stopPlayPublicStream`](#stopplaypublicstream).
+	 * @group Custom Stream Processing
+	 * @order 5
 	 */
 	getPublicStreamTrack: (id: string, type: "video" | "audio") => MediaStreamTrack | undefined;
 	
@@ -3459,6 +3660,8 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @param streamIndex The specified video stream.
 	 * @notes + The snapshot is taken with all video effects on, like rotation, and mirroring.
 	 *        + You can take the snapshot either using SDK internal video capture or customized capture.
+	 * @group Video Snapshot
+	 * @order 0
 	 */
 	takeLocalSnapshot: (streamIndex: StreamIndex) => Promise<ImageData>;
 	
@@ -3466,17 +3669,18 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @brief Take a snapshot of the remote video.
 	 * @param id The index of the user's stream or the index of the public stream.
 	 * @param streamIndex The specified video stream.
+	 * @group Video Snapshot
+	 * @order 1
 	 */
 	takeRemoteSnapshot: (id: string, streamIndex: StreamIndex) => Promise<ImageData>;
 	
 	/** 
 	 * @brief Registers an extension.
+	 *        You can use the extension to implement features such as [Beauty Effects](https://docs.byteplus.com/byteplus-rtc/docs/148646), [Virtual Background](https://docs.byteplus.com/byteplus-rtc/docs/697417), and [AI Noise-Cancelling](https://docs.byteplus.com/byteplus-rtc/docs/148647).
 	 * @param extension The extension instance.
 	 * @return
-	 * @notes You can use the extension to implement the following features. </br>
-	 *     &ensp; + [Beauty Effects](https://docs.byteplus.com/byteplus-rtc/docs/148646)
-	 *     &ensp; + [Virtual Background](https://docs.byteplus.com/byteplus-rtc/docs/697417)
-	 *     &ensp; + [AI Noise-Cancelling](https://docs.byteplus.com/byteplus-rtc/docs/148647)
+	 * @group Extensions
+	 * @order 0
 	 */
 	registerExtension: (extension: RTCExtension) => Promise<void>;
 	
@@ -3487,10 +3691,11 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 *        After calling this method, you will receive the [onSubtitleStateChanged](104479#onsubtitlestatechanged) to inform you of whether subtitles are on.
 	 * @param config Subtitle configrations.
 	 * @return
-	 * @notes
-	 *        You must call this method after joining the room.
-	 *        You can set your source language by calling `joinRoom`  and importing a json formatted string `"source_language": "zh"` through the paramater of extraInfo. Only chinese, English and Japanese are supported.
-	 *        The language of your browser will be automatically set as the source language if you don't set it.
+	 * @notes + You must call this method after joining the room.
+	 *        + You can set your source language to Chinese by calling `joinRoom`  and importing a json formatted string `"source_language": "zh"` through the parameter of extraInfo, to English by importing `"source_language": "en"` , and to Japanese by importing `"source_language": "ja"`.
+	 *        + The language of your browser will be automatically set as the source language if you don't set it. If the language of your browser is not Chinese, English or Japanese, SDK will set Chinese as the source language.
+	 * @group Live Caption and Translation
+	 * @order 0
 	 */
 	startSubtitle: (config: RTCSubtitleConfig) => Promise<void>;
 	
@@ -3498,6 +3703,8 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @hidden currently not available
 	 * @brief Turns off subtitles.
 	 *        After calling this method, you will receive the [onSubtitleStateChanged](104479#onsubtitlestatechanged) to inform you of whether subtitles are off.
+	 * @group Live Caption and Translation
+	 * @order 1
 	 */
 	stopSubtitle: () => void;
 	
@@ -3510,6 +3717,8 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @return + `true`: Success.
 	 *         + `false`: Failure.
 	 * @notes You must call this API before joining the room. Calling it afterward is invalid.
+	 * @group Custom Stream Processing
+	 * @order 10
 	 */
 	setRemoteStreamRenderSync: (sync: boolean) => boolean;
 	
@@ -3521,6 +3730,8 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @notes + It is recommended to set the capture volume after the capture has started.
 	 *        + If capture has not yet started when you call this API, the setting will take effect after it starts.
 	 *        + The capture volume adjustment controls the audio stream before audio mixing.
+	 * @group Audio Management
+	 * @order 25
 	 */
 	setCaptureVolume: (streamIndex: StreamIndex, volume: number) => void;
 	
@@ -3533,16 +3744,21 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @notes + It is recommended to set the playback volume after the playing has started.
 	 *        + If the playing has not yet started when you call this API, the setting will take effect after it starts.
 	 *        + The playback volume cannot be adjusted in [Conference Mode](113547).
+	 * @group Audio Management
+	 * @order 26
 	 */
 	setPlaybackVolume: (userId: string, streamIndex: StreamIndex, volume: number) => void;
 	
 	/** 
+	 * @hidden
 	 * @brief Adjust audio playback volume of a public stream.
 	 * @param publicStreamId The Id of a specified public stream.
 	 * @param volume The percentage of playback volume to the original volume in range of [0, 400]. The default value is 100%.
 	 *               For better call quality, it is recommended to set the value in range of [0, 100].
 	 * @notes + It is recommended to set the playback volume after the playing has started.
 	 *        + If the playing has not yet started when you call this API, the setting will take effect after it starts.
+	 * @group Public Stream
+	 * @order 15
 	 */
 	setPublicStreamVolume: (publicStreamId: string, volume: number) => void;
 	
@@ -3555,6 +3771,8 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 *        + Call [updateForwardStreamToRooms]({{api}}#RTCEngine.updateForwardStreamToRooms) to add or remove the target room(s).
 	 *        + Call [stopForwardStreamToRooms]({{api}}#RTCEngine.stopForwardStreamToRooms) to stop forwarding media stream to all rooms.
 	 *        + Call [pauseForwardStreamToAllRooms]({{api}}#RTCEngine.pauseForwardStreamToAllRooms) to pause forwarding media stream to all rooms.
+	 * @group Audio & Video Transport
+	 * @order 40
 	 */
 	startForwardStreamToRooms: (forwardStreamInfos: ForwardStreamInfo[]) => Promise<ForwardStreamStateInfo[]>;
 	
@@ -3564,6 +3782,8 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @return Status information for forwarding media stream across rooms.
 	 * @notes + Users in the added room will receive [onUserJoined]({{callback}}#EngineEvents.onUserJoined) and [onUserPublishStream]({{callback}}#EngineEvents.onUserPublishStream)/[onUserPublishScreen]({{callback}}#EngineEvents.onUserPublishScreen).
 	 *        + Users in the removed room will receive [onUserUnpublishStream]({{callback}}#EngineEvents.onUserUnpublishStream)/[onUserUnpublishScreen]({{callback}}#EngineEvents.onUserUnpublishScreen) and [onUserLeave]({{callback}}#EngineEvents.onUserLeave).
+	 * @group Audio & Video Transport
+	 * @order 41
 	 */
 	updateForwardStreamToRooms: (forwardStreamInfos: ForwardStreamInfo[]) => Promise<ForwardStreamStateInfo[]>;
 	
@@ -3573,6 +3793,8 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @notes + Users in the former target rooms will receive [onUserUnpublishStream]({{callback}}#EngineEvents.onUserUnpublishStream)/[onUserUnpublishScreen]({{callback}}#EngineEvents.onUserUnpublishScreen) and [onUserLeave]({{callback}}#EngineEvents.onUserLeave).
 	 *        + Call [updateForwardStreamToRooms]({{api}}#RTCEngine.updateForwardStreamToRooms) to stop forwarding media stream to specific rooms.
 	 *        + Call [pauseForwardStreamToAllRooms]({{api}}#RTCEngine.pauseForwardStreamToAllRooms) and [resumeForwardStreamToAllRooms]({{api}}#RTCEngine.resumeForwardStreamToAllRooms) to pause and resume fowarding media stream to all rooms.
+	 * @group Audio & Video Transport
+	 * @order 42
 	 */
 	stopForwardStreamToRooms: () => Promise<ForwardStreamStateInfo[]>;
 	
@@ -3581,6 +3803,8 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 *        After calling this API, call [resumeForwardStreamToAllRooms]({{api}}#RTCEngine.resumeForwardStreamToAllRooms) to resume.
 	 * @return Status information for forwarding media stream across rooms.
 	 * @notes Users in the target rooms will receive [onUserUnpublishStream]({{callback}}#EngineEvents.onUserUnpublishStream)/[onUserUnpublishScreen]({{callback}}#EngineEvents.onUserUnpublishScreen) and [onUserLeave]({{callback}}#EngineEvents.onUserLeave).
+	 * @group Audio & Video Transport
+	 * @order 43
 	 */
 	pauseForwardStreamToAllRooms: () => Promise<ForwardStreamStateInfo[]>;
 	
@@ -3588,6 +3812,8 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @brief Resume forwarding media stream to all rooms.
 	 * @return Status information for forwarding media stream across rooms.
 	 * @notes Users in the target room will receive [onUserJoined]({{callback}}#EngineEvents.onUserJoined) and [onUserPublishStream]({{callback}}#EngineEvents.onUserPublishStream)/[onUserPublishScreen]({{callback}}#EngineEvents.onUserPublishScreen).
+	 * @group Audio & Video Transport
+	 * @order 44
 	 */
 	resumeForwardStreamToAllRooms: () => Promise<ForwardStreamStateInfo[]>;
 }
@@ -4149,7 +4375,7 @@ export interface ScreenEncoderConfig {
 export declare type TrackCaptureConfig = Omit<MediaTrackConstraints, "deviceId">;
 
 /** 
- * @brief The mirror type
+ * @brief The local mirror type. You cannot use mirror effect during transmission and encoding due to the restrictions of web explorer.
  */
 export declare enum MirrorType {
 	
@@ -4665,7 +4891,12 @@ export declare enum NetworkQuality {
 	/** 
 	 * @brief Network quality is very bad, users can barely communicate. It is recommended to notify users of poor network connection through UI.
 	 */
-	VBAD = 5
+	VBAD = 5,
+	
+	/** 
+	 * @brief The call was terminated due to a network disconnection.
+	 */
+	DOWN = 6
 }
 
 /** 
@@ -4897,6 +5128,7 @@ export declare enum TRANSCODING_VIDEO_CODEC {
 }
 
 /** 
+ * @hidden
  * @brief Subtitle configrations.
  */
 export interface RTCSubtitleConfig {
@@ -4913,6 +5145,7 @@ export interface RTCSubtitleConfig {
 }
 
 /** 
+ * @hidden
  * @brief Subtitle mode.
  */
 export declare enum SUBTITLE_MODE {
@@ -4929,6 +5162,7 @@ export declare enum SUBTITLE_MODE {
 }
 
 /** 
+ * @hidden
  * @brief The states of the subtitling task.
  */
 export interface SubtitleEvent {
@@ -4950,6 +5184,7 @@ export interface SubtitleEvent {
 }
 
 /** 
+ * @hidden
  * @brief Types of the states of the subtitling task.
  */
 export declare enum SubtitleEventType {
@@ -4971,6 +5206,7 @@ export declare enum SubtitleEventType {
 }
 
 /** 
+ * @hidden
  * @brief Related information about subtitles.
  */
 export interface SubtitleMessage {
@@ -4994,11 +5230,20 @@ export interface SubtitleMessage {
 	 * @brief Incremental sequence numbers assigned to transcribed or translated texts of the speaker's speech. Complete and incomplete sentences are numbered individually.
 	 */
 	sequence: number;
+	
+	/** 
+	 * @brief Subtitle language code.
+	 */
 	language: string;
+	
+	/** 
+	 * @brief Subtitle mode.
+	 */
 	mode: SUBTITLE_MODE;
 }
+
 /** 
- * @brief Cloud proxy informatio
+ * @brief Cloud proxy information.
  */
 export interface CloudProxyInfo {
 	
@@ -5210,7 +5455,8 @@ export interface IVERTC {
 	 * @return Return an `IRTCEngine` instance if you call this API successfully.
 	 * @notes + Call [destroyEngine](#destroyengine) respectively to destroy each IRTCEngine instance.
 	 *         + As each IRTCEngine instance is resource consuming, we recommend creating no more than 2 engine instances on each device at the same time.
-	 * @region Engine Management
+	 * @group Engine Management
+	 * @order 0
 	 */
 	createEngine: (appId: string, config?: EngineConfig) => IRTCEngine;
 	
@@ -5219,14 +5465,16 @@ export interface IVERTC {
 	 * @param engine By calling this API, this `IRTCEngine` instance and resource related will be wiped from memory. And any interaction with the media server will end.
 	 * @notes + Before calling this API， ensure all business scenarios related to this `IRTCEngine` instance have ended.
 	 *       + Calling this API will block the main thread until it completes the exiting process. Therefore, do not include this API in a callback. Nor wait in a callback for the execution of this API in the main thread, thus causing a deadlock.
-	 * @region Engine Management
+	 * @group Engine Management
+	 * @order 1
 	 */
 	destroyEngine: (engine: IRTCEngine) => void;
 	
 	/** 
 	 * @brief Get the current version of the SDK.
 	 * @return The current SDK version.
-	 * @region Engine Management
+	 * @group Engine Management
+	 * @order 10
 	 */
 	getSdkVersion: () => string;
 	
@@ -5235,6 +5483,8 @@ export interface IVERTC {
 	 *        Call this API to get deviceId used to specify the device when calling [setAudioPlaybackDevice](104478#setaudioplaybackdevice), [startVideoCapture](104478#startvideocapture), and [startAudioCapture](104478#startaudiocapture).
 	 * @return A list of all media devices.
 	 * @notes Only when the browser has permission to the media devices can it enumerate the full device list. Make sure the media device access is turned on before calling this API.
+	 * @group Device Management
+	 * @order 0
 	 */
 	enumerateDevices: () => Promise<MediaDeviceInfo[]>;
 	
@@ -5243,7 +5493,8 @@ export interface IVERTC {
 	 * @param options The device type you need
 	 * @return + `video: boolean` Video devices
 	 *         + `audio:boolean` Audio devices
-	 * @region Device Management
+	 * @group Device Management
+	 * @order 5
 	 */
 	enableDevices: (options?: {
 		
@@ -5266,6 +5517,8 @@ export interface IVERTC {
 	 *        Call this API to get Device IDs used to specify the device when calling [startAudioCapture](104478#startaudiocapture) and [setAudioCaptureDevice](104478#setaudiocapturedevice).
 	 * @return A list including all microphone devices.
 	 * @notes Only when the browser has permission to the media devices can it enumerate the full device list. Make sure the media device access is turned on before calling this API.
+	 * @group Device Management
+	 * @order 10
 	 */
 	enumerateAudioCaptureDevices: () => Promise<MediaDeviceInfo[]>;
 	
@@ -5274,6 +5527,8 @@ export interface IVERTC {
 	 *        Call this API to get Device IDs used to specify the device when calling [startVideoCapture](104478#startvideocapture) and [setVideoCaptureDevice](104478#setvideocapturedevice).
 	 * @return A list including all camera devices.
 	 * @notes Only when the browser has permission to the media devices can it enumerate the full device list. Make sure the media device access is turned on before calling this API.
+	 * @group Device Management
+	 * @order 11
 	 */
 	enumerateVideoCaptureDevices: () => Promise<MediaDeviceInfo[]>;
 	
@@ -5282,6 +5537,8 @@ export interface IVERTC {
 	 *        Call this API to get Device IDs used to specify the device when calling [setAudioPlaybackDevice](104478#setaudioplaybackdevice).
 	 * @return A list including all audio playback devices.
 	 * @notes Only when the browser has permission to the media devices can it enumerate the full device list. Make sure the media device access is turned on before calling this API.
+	 * @group Device Management
+	 * @order 12
 	 */
 	enumerateAudioPlaybackDevices: () => Promise<MediaDeviceInfo[]>;
 	
@@ -5289,7 +5546,8 @@ export interface IVERTC {
 	 * @brief Call this API to check whether the browser is supported.
 	 * @return + `True`: Supported
 	 *         + `False`: Not Supported
-	 * @region Engine Management
+	 * @group Engine Management
+	 * @order 15
 	 */
 	isSupported: () => Promise<boolean>;
 	
@@ -5297,15 +5555,18 @@ export interface IVERTC {
 	 * @brief Gets the codec type supported by current browser.
 	 * If you have enabled vp8 codec, you can call the API to check whether the current browser supports vp8 encoding type before entering the room.
 	 * @return Codec types supported by the current browser
-	 * @region Engine Management
+	 * @group Engine Management
+	 * @order 20
 	 */
 	getSupportedCodecs: () => Promise<string[]>;
 	
 	/** 
 	 * @brief Configures the logging level of local cache, and the maximum cache space that can be used.
 	 * @notes This API must be called before [createEngine](#createengine). If called after [createEngine](#createengine), it will take effect after the next [createEngine](#createengine) after [destroyEngine](#destroyengine).
+	 * @group Engine Management
+	 * @order 25
 	 */
-	setLogConfig: ({ logLevel, LogfileSize }: {
+	setLogConfig: ({ logLevel, LogfileSize, }: {
 		
 		/** 
 		 * @brief The logging level of local cache.
@@ -5326,6 +5587,8 @@ export interface IVERTC {
 	/** 
 	 * @brief Download local cache generated file.
 	 * @param key The key of the local cache. If empty, download the logs from the current cache.
+	 * @group Engine Management
+	 * @order 26
 	 */
 	downloadLog: (key?: string) => void;
 }
@@ -5353,7 +5616,7 @@ declare class VERTC implements IVERTC {
 	platform: string;
 	commitInfo: string;
 	downloadLog: (key?: string | undefined) => void;
-	setLogConfig: ({ logLevel, LogfileSize }: {
+	setLogConfig: ({ logLevel, LogfileSize, }: {
 		logLevel?: "none" | "error" | "info" | "warning" | "trace" | undefined;
 		LogfileSize?: number | undefined;
 	}) => void;