@byteplus/rtc 4.60.11 → 4.61.1

package/index.d.ts

@@ -4,6 +4,7 @@ import { EventArgs, EventEmitter, EventNames, ValidEventTypes } from 'eventemitt
 
 declare class EnhancedEventEmitter<EventTypes extends ValidEventTypes = string | symbol> extends EventEmitter<EventTypes> {
 	safeEmit<T extends EventNames<EventTypes>>(event: T, ...args: EventArgs<EventTypes, T>): boolean;
+	asyncEmit<T extends EventNames<EventTypes>>(event: T, ...args: EventArgs<EventTypes, T>): Promise<void>;
 }
 declare class VERTC implements IVERTC {
 	getSdkVersion: () => string;
@@ -64,6 +65,7 @@ declare const Config: {
 	FALLBACK_ENCODE_CODEC: string;
 	SEI_TIME_OUT: number;
 	SEI_COUNT_FPS: number;
+	PRE_ICE: boolean;
 };
 declare const _default: VERTC;
 declare enum AudioMixingError {
@@ -775,7 +777,7 @@ declare enum ErrorCode {
 	
 	/** 
 	 * @hidden
-	 * @brief Subtitles are already on. There is no need to call [startSubtitle](104478#startsubtitle) again.
+	 * @brief Subtitles are already on. There is no need to call [startSubtitle](docs-104478#startsubtitle) again.
 	 */
 	SUBTITLE_ALREADY_ON = "SUBTITLE_ALREADY_ON",
 	
@@ -828,13 +830,13 @@ declare enum SEIStreamEventType {
 	
 	/** 
 	 * @brief A black frame video stream is published from the remote user.
-	 *        In the voice call scenario, when a remote user sends SEI data by calling [sendSEIMessage](104478#sendseimessage), the SDK will publish a black frame video stream and trigger this callback.
+	 *        In the voice call scenario, when a remote user sends SEI data by calling [sendSEIMessage](docs-104478#sendseimessage), the SDK will publish a black frame video stream and trigger this callback.
 	 */
 	BLACK = 0,
 	
 	/** 
 	 * @brief The black frame video stream is removed. The triggers for this callback include:
-	 *        <ul><li> When the remote user starts camera capture, switching from voice call to video call, the black frame video stream stops publishing. </li><li> When the remote user calls [setVideoSourceType](104478#setvideosourcetype) to switch to custom video capture, black frame video stream stops publishing. </li><li> If no SEI data is sent within 1 minute after the remote user calls [sendSEIMessage](104478#sendseimessage), the black frame video stream stops publishing. </li></ul>
+	 *        <ul><li> When the remote user starts camera capture, switching from voice call to video call, the black frame video stream stops publishing. </li><li> When the remote user calls [setVideoSourceType](docs-104478#setvideosourcetype) to switch to custom video capture, black frame video stream stops publishing. </li><li> If no SEI data is sent within 1 minute after the remote user calls [sendSEIMessage](docs-104478#sendseimessage), the black frame video stream stops publishing. </li></ul>
 	 */
 	NORMAL = 1
 }
@@ -990,7 +992,7 @@ export declare enum AudioMixingType {
 
 /** 
  * @brief  Audio quality
- *         Call [setAudioProfile](104478#setaudioprofile) to choose an audio quality option.
+ *         Call [setAudioProfile](docs-104478#setaudioprofile) to choose an audio quality option.
  */
 export declare enum AudioProfileType {
 	
@@ -1043,7 +1045,7 @@ export declare enum AudioProfileType {
 
 /** 
  * @brief The data range of the local main stream audio callback. It controls whether the callback includes mixed audio.
- *        This feature affects the callback [onActiveSpeaker](104479#onactivespeaker). For example, if [AudioReportMode](#audioreportmode) is set to `AUDIOMIXING`, the callback may still include the local user's main stream audio even with zero microphone volume.
+ *        This feature affects the callback [onActiveSpeaker](docs-104479#onactivespeaker). For example, if [AudioReportMode](#audioreportmode) is set to `AUDIOMIXING`, the callback may still include the local user's main stream audio even with zero microphone volume.
  */
 export declare enum AudioReportMode {
 	
@@ -1209,7 +1211,7 @@ export declare enum ForwardStreamState {
 
 /** 
  * @brief Audio callback mode of local main stream.
- *        This feature affects the callback [onActiveSpeaker](104479#onactivespeaker). For example, if [LocalMainReportMode](#localmainreportmode) is set to `RESET` when the local main stream is unpublished, the callback will never include the local user's main stream audio.
+ *        This feature affects the callback [onActiveSpeaker](docs-104479#onactivespeaker). For example, if [LocalMainReportMode](#localmainreportmode) is set to `RESET` when the local main stream is unpublished, the callback will never include the local user's main stream audio.
  */
 export declare enum LocalMainReportMode {
 	
@@ -1408,6 +1410,16 @@ export declare enum RTCAutoPlayPolicy {
 	 */
 	PLAY_MANUALLY = 2
 }
+/**
+ * @hidden
+ * 发生重连的原因
+ */
+export declare enum ReconnectReason {
+	ICE_FAILED = "iceFailed",
+	NODE_CHANGE = "nodeChange",
+	JOIN_TIMEOUT = "joinTimeout",
+	NOTIFY_RECONNECT = "notifyReconnect"
+}
 
 /** 
  * @brief priority of the remote user
@@ -1432,8 +1444,8 @@ export declare enum RemoteUserPriority {
 
 /** 
  * @brief  Room profile
- *        While calling [joinRoom](104478#joinroom) choose an appropriate room-profile option to meet the requirement of the scenario for media encoding algorithm, video profiles, and network configurations.
- *        Call [setAudioProfile](104478#setaudioprofile) to choose an audio quality option if it is necessary.
+ *        While calling [joinRoom](docs-104478#joinroom) choose an appropriate room-profile option to meet the requirement of the scenario for media encoding algorithm, video profiles, and network configurations.
+ *        Call [setAudioProfile](docs-104478#setaudioprofile) to choose an audio quality option if it is necessary.
  */
 export declare enum RoomProfileType {
 	
@@ -1553,8 +1565,8 @@ export declare enum SubscribeFallbackOption {
 	DISABLE = 0,
 	
 	/** 
-	 * @brief When the downstream network quality is bad, the video stream fallbacks to stream of lower resolution and then to voice only. See [Media Stream Fallback](https://docs.byteplus.com/en/byteplus-rtc/docs/70137) for fallback rules.
-	 *        This setting takes effect only when the publisher enables simulcast mode through [EnableSimulcastMode](104478#enablesimulcastmode).
+	 * @brief When the downstream network quality is bad, the video stream fallbacks to stream of lower resolution and then to voice only. See [Media Stream Fallback](https://docs.byteplus.com/en/docs/byteplus-rtc/docs-70137) for fallback rules.
+	 *        This setting takes effect only when the publisher enables simulcast mode through [EnableSimulcastMode](docs-104478#enablesimulcastmode).
 	 */
 	VIDEO_STREAM_LOW = 1,
 	
@@ -1576,7 +1588,7 @@ export declare enum SubscribeMode {
 	AUTO_SUBSCRIBE_MODE = 0,
 	
 	/** 
-	 * @brief Manual. You can subscribe the stream in the room manually by calling [subscribeUserStream](104478#subscribeuserstream).
+	 * @brief Manual. You can subscribe the stream in the room manually by calling [subscribeUserStream](docs-104478#subscribeuserstream).
 	 */
 	MANUAL_SUBSCRIBE_MODE = 1
 }
@@ -1880,6 +1892,11 @@ export interface AutoPlayFailedEvent {
 	 * @brief Remote stream type
 	 */
 	mediaType: MediaType.AUDIO | MediaType.VIDEO;
+	
+	/** 
+	 * @brief Player ID. An empty value indicates the default player.
+	 */
+	playerId?: string;
 }
 
 /** 
@@ -1923,6 +1940,12 @@ export interface ConnectionStateChangeEvent {
 	 * @brief The internal connection state
 	 */
 	state: ConnectionState;
+	
+	/** 
+	 * @hidden
+	 * @brief The reason to the reconnection
+	 */
+	reason?: ReconnectReason;
 }
 export interface CoreConfigType {
 	UPLOAD_CONSOLE_ON: boolean;
@@ -2024,7 +2047,7 @@ export interface IAudioMixingManager {
 	 * @param id Mix ID. For identifying the mix, please ensure that the mix ID is unique.
 	 *         If this method is repeatedly called with the same ID, the previous mix stops and the next mix starts.
 	 * @param filePath  Specify the file path either the absolute path of a local file or the url of online audio.  Files with the extension of mp3 and aac are supported.
-	 * @param config  Mix Sound configuration.
+	 * @param options  Mix Sound configuration.
 	 *         You can set the number of times the mix is played, whether the mix is played locally, and whether the mix is sent to the remote end. For details.
 	 * @return `LOAD_RESOURCES_FAILED`: Loading resource failed. Please pass in the correct resource path.
 	 * @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).
@@ -2089,7 +2112,7 @@ export interface IAudioMixingManager {
 	unloadAudioMixing: (id: number) => void;
 	
 	/** 
-	 * @brief Adjust the volume of audio mixing, including audio files and PCM data.
+	 * @brief Adjust the volume of the audio-file mixing.
 	 * @param id Mixing ID
 	 * @param volume The ratio of the mixing volume to the original volume. The range is `[0, 400]`. The recommended range is `[0, 100]`.
 	 *         + 0: Mute
@@ -2228,7 +2251,7 @@ export interface IEngineEvents {
 	
 	/** 
 	 * @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.
+	 * @notes You will receive this callback after a remote user successfully publishes media streams captured by camera/microphone in the room with [publishStream](docs-104478#publishstream). Then you can choose whether to call [subscribeStream](docs-104478#subscribestream) to subscribe to the streams or not.
 	 * @group Audio & Video Transport
 	 * @order 60
 	 */
@@ -2247,7 +2270,7 @@ 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.
+	 * @notes You will receive this callback after a remote user successfully unpublishes media streams captured by camera/microphone in the room with [unpublishStream](docs-104478#unpublishstream). Then you can choose whether to call [unsubscribeStream](docs-104478#unsubscribestream) to unsubscribe from the streams or not.
 	 * @group Audio & Video Transport
 	 * @order 61
 	 */
@@ -2271,7 +2294,7 @@ 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.
+	 * @notes You will receive this callback after a remote user successfully published screen sharing streams in the room with [publishScreen](docs-104478#publishscreen). Then you can choose whether to call [subscribeScreen](docs-104478#subscribescreen) to subscribe to the streams or not.
 	 * @group Screen Sharing
 	 * @order 30
 	 */
@@ -2290,7 +2313,7 @@ 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.
+	 * @notes You will receive this callback after a remote user successfully unpublishes media streams captured by camera/microphone in the room with [unpublishScreen](docs-104478#unpublishscreen). Then you can choose whether to call [unsubscribeScreen](docs-104478#unsubscribescreen) to unsubscribe from the streams or not.
 	 * @group Screen Sharing
 	 * @order 31
 	 */
@@ -2377,7 +2400,7 @@ export interface IEngineEvents {
 	[EngineEventsTypes.onTrackEnded]: (event: {
 		
 		/** 
-		 * @brief Type of the trac
+		 * @brief Media stream kind. Refer to [kind](https://developer.mozilla.org/en-US/docs/Web/API/MediaStreamTrack/kind).
 		 */
 		kind: string;
 		
@@ -2458,7 +2481,7 @@ export interface IEngineEvents {
 	}) => void;
 	
 	/** 
-	 * @brief Receiving a point-to-point text message from one of the users in the room by calling [sendUserMessage](104478#sendusermessage)
+	 * @brief Receiving a point-to-point text message from one of the users in the room by calling [sendUserMessage](docs-104478#sendusermessage)
 	 * @param event
 	 * @group Message
 	 * @order 50
@@ -2490,7 +2513,7 @@ export interface IEngineEvents {
 	[EngineEventsTypes.onLocalStreamStats]: (stats: LocalStreamStats) => void;
 	
 	/** 
-	 * @brief This 2 s periodical callback notifies information of the public stream after calling [startPlayPublicStream](104478#startplaypublicstream).
+	 * @brief This 2 s periodical callback notifies information of the public stream after calling [startPlayPublicStream](docs-104478#startplaypublicstream).
 	 * @param stats
 	 * @group Public Stream
 	 * @order 20
@@ -2513,7 +2536,7 @@ export interface IEngineEvents {
 	}) => void;
 	
 	/** 
-	 * @brief After calling [enableAudioPropertiesReport](104478#enableaudiopropertiesreport), you will periodically receive this callback for the information about local audio.
+	 * @brief After calling [enableAudioPropertiesReport](docs-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.
 	 * @group Audio Management
@@ -2522,7 +2545,7 @@ export interface IEngineEvents {
 	[EngineEventsTypes.onLocalAudioPropertiesReport]: (event: LocalAudioPropertiesInfo[]) => void;
 	
 	/** 
-	 * @brief After calling [enableAudioPropertiesReport](104478#enableaudiopropertiesreport), you will periodically receive this callback for the information about the subscribed remote audio streams.
+	 * @brief After calling [enableAudioPropertiesReport](docs-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.
 	 * @group Audio Management
@@ -2531,7 +2554,7 @@ export interface IEngineEvents {
 	[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.
+	 * @brief After calling [enableAudioPropertiesReport](docs-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.
 	 * @group Audio Management
 	 * @order 60
@@ -2539,7 +2562,7 @@ export interface IEngineEvents {
 	[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](docs-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
@@ -2547,7 +2570,7 @@ export interface IEngineEvents {
 	[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).
+	 * @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](docs-104478#startaudiocapture).
 	 * @param event
 	 * @group Audio Management
 	 * @order 70
@@ -2561,7 +2584,7 @@ export interface IEngineEvents {
 	}) => void;
 	
 	/** 
-	 * @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).
+	 * @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](docs-104478#stopaudiocapture).
 	 * @param event
 	 * @group Audio Management
 	 * @order 71
@@ -2575,7 +2598,7 @@ export interface IEngineEvents {
 	}) => void;
 	
 	/** 
-	 * @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).
+	 * @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](docs-104478#startvideocapture).
 	 * @param event
 	 * @group Video Management
 	 * @order 55
@@ -2589,7 +2612,7 @@ export interface IEngineEvents {
 	}) => void;
 	
 	/** 
-	 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).
+	 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](docs-104478#stopvideocapture).
 	 * @param event
 	 * @group Video Management
 	 * @order 56
@@ -2612,7 +2635,7 @@ export interface IEngineEvents {
 	
 	/** 
 	 * @brief Stream update event of stream that contains SEI information.
-	 *        In a voice call scenario, when a remote user calls [sendSEIMessage](104478#sendseimessage) to send SEI data via the black frame video stream, the stream's sending status will be notified to the local user through this event callback.
+	 *        In a voice call scenario, when a remote user calls [sendSEIMessage](docs-104478#sendseimessage) to send SEI data via the black frame video stream, the stream's sending status will be notified to the local user through this event callback.
 	 *        If the video frame carrying SEI data is a black frame, you do not need to render the video frame.
 	 * @param event Stream containing SEI information update event.
 	 * @group Message
@@ -2689,7 +2712,7 @@ export interface IEngineEvents {
 	[EngineEventsTypes.onError]: (event: {
 		
 		/** 
-		 * @brief See [ErrorCode](104480#errorcode).
+		 * @brief See [ErrorCode](docs-104480#errorcode).
 		 */
 		errorCode: ErrorCode.DUPLICATE_LOGIN | ErrorCode.RTM_DUPLICATE_LOGIN | ErrorCode.RTM_TOKEN_ERROR | ErrorCode.TOKEN_EXPIRED | ErrorCode.RECONNECT_FAILED | ErrorCode.KICKED_OUT | ErrorCode.ROOM_DISMISS;
 		
@@ -2726,7 +2749,7 @@ export interface IEngineEvents {
 	
 	/** 
 	 * @brief You will receive this callback 30 seconds 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.
+	 *        Call [updateToken](docs-104478#updatetoken) to submit a new Token. The user will be removed from the room when the Token expires.
 	 * @group Room Management
 	 * @order 45
 	 */
@@ -2734,15 +2757,15 @@ export interface IEngineEvents {
 	
 	/** 
 	 * @brief This callback will be triggered 30 seconds before the token publishing permission expires.
-	 *        Upon receiving this callback, call [updateToken](104478#updatetoken) to update the token.
-	 * @notes If the Token is not updated in time, the publishing permission will expire and the stream that is being published will stop. The publisher will receive the [onTokenPublishPrivilegeDidExpired](104479#ontokenpublishprivilegedidexpired) callback, and the remote user will receive the [onUserUnpublishStream](104479#onuserunpublishstream)/[onUserUnpublishScreen](104479#onuserunpublishscreen) callback, with the reason being `STREAM_REMOVE_REASON_TOKEN_PRIVILEGE_EXPIRED`.
+	 *        Upon receiving this callback, call [updateToken](docs-104478#updatetoken) to update the token.
+	 * @notes If the Token is not updated in time, the publishing permission will expire and the stream that is being published will stop. The publisher will receive the [onTokenPublishPrivilegeDidExpired](docs-104479#ontokenpublishprivilegedidexpired) callback, and the remote user will receive the [onUserUnpublishStream](docs-104479#onuserunpublishstream)/[onUserUnpublishScreen](docs-104479#onuserunpublishscreen) callback, with the reason being `STREAM_REMOVE_REASON_TOKEN_PRIVILEGE_EXPIRED`.
 	 * @group Room Management
 	 * @order 50
 	 */
 	[EngineEventsTypes.onTokenPublishPrivilegeWillExpire]: () => void;
 	
 	/** 
-	 * @brief This callback is triggered when the token publishing permission expires. Call [updateToken](104478#updatetoken) to update the Token.
+	 * @brief This callback is triggered when the token publishing permission expires. Call [updateToken](docs-104478#updatetoken) to update the Token.
 	 * @group Room Management
 	 * @order 51
 	 */
@@ -2761,8 +2784,8 @@ export interface IEngineEvents {
 	
 	/** 
 	 * @brief This callback will be triggered 30 seconds before the token subscription permission expires.
-	 *        Upon receiving this callback, call [updateToken](104478#updatetoken) to update the token.
-	 * @notes If the Token is not updated in time after receiving this callback, the streams that have been subscribed to will be unsubscribed once the token subscription permission expires, and the subscriber will receive the [onTokenSubscribePrivilegeDidExpired](104479#ontokensubscribeprivilegedidexpired) callback.
+	 *        Upon receiving this callback, call [updateToken](docs-104478#updatetoken) to update the token.
+	 * @notes If the Token is not updated in time after receiving this callback, the streams that have been subscribed to will be unsubscribed once the token subscription permission expires, and the subscriber will receive the [onTokenSubscribePrivilegeDidExpired](docs-104479#ontokensubscribeprivilegedidexpired) callback.
 	 *        Also, attempts to subscribe to new streams will fail, returning an error code `TOKEN_NO_SUBSCRIBE_PERMISSION`, indicating that the token subscription permission has expired.
 	 * @group Room Management
 	 * @order 55
@@ -2770,7 +2793,7 @@ export interface IEngineEvents {
 	[EngineEventsTypes.onTokenSubscribePrivilegeWillExpire]: () => void;
 	
 	/** 
-	 * @brief This callback is triggered when the token subscription permission expires. Call [updateToken](104478#updatetoken) to update the Token.
+	 * @brief This callback is triggered when the token subscription permission expires. Call [updateToken](docs-104478#updatetoken) to update the Token.
 	 * @group Room Management
 	 * @order 56
 	 */
@@ -2788,14 +2811,14 @@ export interface IEngineEvents {
 	}) => void;
 	
 	/** 
-	 * @brief Receives the callback when you call [startCloudProxy](104478#startcloudproxy) to start cloud proxy, and the SDK connects the proxy server successfully.
+	 * @brief Receives the callback when you call [startCloudProxy](docs-104478#startcloudproxy) to start cloud proxy, and the SDK connects the proxy server successfully.
 	 * @param event
 	 */
 	[EngineEventsTypes.onCloudProxyConnected]: (event: CloudProxyConnectedEvent) => void;
 	
 	/** 
 	 * @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.
+	 *        After calling [startPushPublicStream](docs-104478#startpushpublicstream), you will be informed of the result through this callback.
 	 * @param event
 	 * @group Public Stream
 	 * @order 25
@@ -2804,7 +2827,7 @@ export interface IEngineEvents {
 	
 	/** 
 	 * @brief Callback of the SEI carried by the public video stream.
-	 *        You will receive SEI by this callback if the public stream you subscribed to by calling [startPlayPublicStream](104478#startplaypublicstream) is carrying SEI.
+	 *        You will receive SEI by this callback if the public stream you subscribed to by calling [startPlayPublicStream](docs-104478#startplaypublicstream) is carrying SEI.
 	 * @param event
 	 * @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.
@@ -2816,7 +2839,7 @@ 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.
+	 *        Refer to [startPlayPublicStream](docs-104478#startplaypublicstream) for details about subscribing to a public stream.
 	 * @param event
 	 * @group Public Stream
 	 * @order 36
@@ -2825,7 +2848,7 @@ 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.
+	 *        Refer to [startPlayPublicStream](docs-104478#startplaypublicstream) for details about subscribing to a public stream.
 	 * @param event
 	 * @group Public Stream
 	 * @order 35
@@ -2834,7 +2857,7 @@ 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.
+	 *         Refer to [startPlayPublicStream](docs-104478#startplaypublicstream) for details about subscribing to a public stream.
 	 * @param event
 	 * @group Public Stream
 	 * @order 37
@@ -2844,7 +2867,7 @@ 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.
+	 * @notes Start an audio-device test by calling [startAudioPlaybackDeviceTest](docs-104478#startaudioplaybackdevicetest) or [startAudioDeviceRecordTest](docs-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
 	 */
@@ -2961,7 +2984,7 @@ 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.
+	 *        After you call [startSubtitle](docs-104478#startsubtitle) and [stopSubtitle](docs-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
@@ -2971,7 +2994,7 @@ export interface IEngineEvents {
 	/** 
 	 * @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.
+	 *        After calling [startSubtitle](docs-104478#startsubtitle) successfully, you will receive this callback which informs you of the related information on subtitles.
 	 * @param e Subtitle message. The subtitle messages will include multiple languages as the task specified. You can collect the `text` by `userId` and `language`.
 	 * @group Live Caption and Translation
 	 * @order 11
@@ -2981,8 +3004,8 @@ 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 you lose internet connection after calling [setServerParams](104478#setserverparams), you will receive this event once the internet reconnects.
+	 * @notes + This event notifies you the result after the call of [setServerParams](docs-104478#setserverparams).
+	 *        + If you lose internet connection after calling [setServerParams](docs-104478#setserverparams), you will receive this event once the internet reconnects.
 	 * @group Message
 	 * @order 70
 	 */
@@ -3065,21 +3088,21 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @brief Join the room.
 	 * Call this method to join the room and make audio & video calls with other users in the room.
 	 * @param token Dynamic key. It is used for authentication and verification of users entering the room.
-	 *         You need to bring Token to enter the room. When testing, you can use the console to generate temporary tokens. The official launch requires the use of the key SDK to generate and issue tokens at your server level. See [Use Token to complete authentication](70121) for token validity and generation method.
+	 *         You need to bring Token to enter the room. When testing, you can use the console to generate temporary tokens. The official launch requires the use of the key SDK to generate and issue tokens at your server level. See [Use Token to complete authentication](docs-70121) for token validity and generation method.
 	 *         Apps with different AppIDs are not interoperable.
 	 *         Make sure that the AppID used to generate the Token is the same as the AppID used to create the engine, otherwise it will cause the join room to fail.
 	 * @param roomId  The Room ID. The string must conform to the regular expression: `[a-zA-Z0-9_@\-\.]{1,128}`.
 	 * @param userInfo  User information.
 	 * @param roomConfig  Room parameter configuration, set the room mode and whether to automatically publish or subscribe to the flow.
 	 * @return + `INVALID_TOKEN`: Failed to enter the room due to using an expired or invalid token.
-	 *         + `JOIN_ROOM_FAILED`: Failed to enter the room. Please refer to the corresponding message for specific reasons.
+	 *         + `JOIN_ROOM_FAILED`: Failed to enter the room.
 	 *         + `REPEAT_JOIN`: Join the same room repeatedly.
 	 *         + `ROOM_FORBIDDEN`: Failed to enter the room because the room has been banned.
 	 *         + `USER_FORBIDDEN`: Failed to enter the room because the user has been banned.
 	 * @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.
-	 *        + By default, the user is visible in an RTC room. Joining fails when the number of users in an RTC room reaches the upper limit. To avoid this, call [setUserVisibility](#setuservisibility) to change the visibility of the audience users to false by considering the capacity for the invisible users is much larger than that for visible users. An RTC room can accommodate a maximum of 50 visible users, and 30 media streams can be published simultaneously. For more information, see [Room Capability](257549).
-	 *        + 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.
+	 *        + Remote users will receive [onUserJoined](docs-104479#onuserjoined) callback when a visible user joins the room. Call [setUserVisibility](#setuservisibility) to switch the visibility of the user.
+	 *        + By default, the user is visible in an RTC room. Joining fails when the number of users in an RTC room reaches the upper limit. To avoid this, call [setUserVisibility](#setuservisibility) to change the visibility of the audience users to false by considering the capacity for the invisible users is much larger than that for visible users. An RTC room can accommodate a maximum of 50 visible users, and 30 media streams can be published simultaneously. For more information, see [Room Capability](docs-257549).
+	 *        + RTC SDK will try reconnect after losing connection to the server caused by a poor network connection. At this point, [onConnectionStateChanged](docs-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](docs-104479#onuserjoined) if a visible user reconnected successfully.
 	 * @group Room Management
 	 * @order 0
 	 */
@@ -3088,7 +3111,8 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	/** 
 	 * @brief Leave the room.
 	 * 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.
+	 * @param waitAck Reserved parameter, no actual significance.
+	 * @notes + After visible users leave the room, other users in the room will receive [onUserLeave](docs-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.
 	 * @group Room Management
 	 * @order 1
@@ -3098,7 +3122,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	/** 
 	 * @hidden
 	 * @brief Destroy engine
-	 * Destroy the engine instances created by calling [createEngine](104478#createengine) and release all related resources.
+	 * Destroy the engine instances created by calling [createEngine](docs-104478#createengine) and release all related resources.
 	 * @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.
@@ -3110,7 +3134,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @param mediaType Media stream type, used for specifying whether to publish audio stream or video stream.
 	 * @notes + An invisible user cannot publish media streams. Call [setUserVisibility](#setuservisibility) to change your visibility in the room.
 	 *        + Call [publishScreen](#publishscreen) to start screen sharing.
-	 *        + After you call this API, the other users in the room will receive [onUserPublishStream](104479#onuserpublishstream).
+	 *        + After you call this API, the other users in the room will receive [onUserPublishStream](docs-104479#onuserpublishstream).
 	 *        + Call [unpublishStream](#unpublishstream) to stop publishing streams.
 	 * @return + `NO_PUBLISH_PERMISSION`: No permission to publish streams. Please call [setUserVisibility](#setuservisibility) to switch the user to be visible first.
 	 *         + `TOKEN_NO_PUBLISH_PERMISSION`: The Token has no publishing permission or the publishing permission has expired, call [updateToke](#updatetoken) to update the Token's publishing permission.
@@ -3124,7 +3148,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @brief Stops publishing media streams captured by camera/microphone in the current room.
 	 * @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).
+	 *        + After calling this API, the other users in the room will receive [onUserUnpublishStream](docs-104479#onuserunpublishstream).
 	 * @group Audio & Video Transport
 	 * @order 1
 	 */
@@ -3134,7 +3158,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @brief Publishes local screen sharing streams in the current room.
 	 * @param mediaType Media stream type, used for specifying whether to publish audio stream or video stream.
 	 * @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).
+	 *        + After you called this API, the other users in the room will receive [onUserPublishScreen](docs-104479#onuserpublishscreen).
 	 *        + Call [unpublishScreen](#unpublishscreen) to stop publishing screen sharing streams.
 	 * @return + `NO_PUBLISH_PERMISSION`: No permission to publish streams. Please call [setUserVisibility](#setuservisibility) to switch the user to be visible first.
 	 *         + `TOKEN_NO_PUBLISH_PERMISSION`: The Token has no publishing permission or the publishing permission has expired, call [updateToke](#updatetoken) to update the Token's publishing permission.
@@ -3148,7 +3172,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @brief Stops publishing local screen sharing streams in the current room.
 	 * @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).
+	 *         + After calling this API, the other users in the room will receive [onUserUnpublishScreen](docs-104479#onuserunpublishscreen).
 	 * @group Screen Sharing
 	 * @order 11
 	 */
@@ -3159,7 +3183,7 @@ 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 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.
+	 * @notes You must first get the remote stream information through [onUserPublishStream](docs-104479#onuserpublishstream) before calling this API to subscribe to streams accordingly.
 	 * @return + `STREAM_NOT_EXIST`: The target stream does not exist. Please enter the correct ID or make sure that the target stream is in the room.
 	 *         + `TOKEN_NO_SUBSCRIBE_PERMISSION`: The Token has no subscription permission or the subscription permission has expired, call [updateToke](#updatetoken) to update the Token's subscription permission.
 	 * @group Audio & Video Transport
@@ -3183,7 +3207,7 @@ 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 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.
+	 * @notes You must first get the remote stream information through [onUserPublishScreen](docs-104479#onuserpublishscreen) before calling this API to subscribe to streams accordingly.
 	 * @return + `STREAM_NOT_EXIST`: The target stream does not exist. Please enter the correct ID or make sure that the target stream is in the room.
 	 *         + `TOKEN_NO_SUBSCRIBE_PERMISSION`: The Token has no subscription permission or the subscription permission has expired, call [updateToke](#updatetoken) to update the Token's subscription permission.
 	 * @group Screen Sharing
@@ -3194,8 +3218,8 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	/** 
 	 * @brief Unsubscribes from specific screen sharing media stream.
 	 *        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.
+	 * @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 unsubscribe from the audio stream or the video stream.
 	 * @group Screen Sharing
 	 * @order 16
 	 */
@@ -3203,7 +3227,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	
 	/** 
 	 * @brief Update Token.
-	 *        You will receive [onTokenWillExpire](104479#ontokenwillexpire) 30 s before the Token expires. Call this API to submit a new Token.
+	 *        You will receive [onTokenWillExpire](docs-104479#ontokenwillexpire) 30 s before the Token expires. Call this API to submit a new Token.
 	 * @param token A valid token.
 	 * @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.
@@ -3240,7 +3264,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	/** 
 	 * @brief Enable internal video capture. The default state 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 [onUserStartVideoCapture](104479#onuserstartvideocapture) after the publisher starts video capture by calling this API.
+	 *        The remote clients in the room will be informed of the state change via [onUserStartVideoCapture](docs-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/MediaTrackSettings). Pay attention to the properties: `frameRate`, `height`, and `width`.
 	 *         The following error codes may be returned.
@@ -3258,7 +3282,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	/** 
 	 * @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.
+	 *        The remote clients in the room will be informed of the state change via [onUserStopVideoCapture](docs-104479#onuserstopvideocapture) after the publisher stops video capture by calling this API.
 	 * @return `STREAM_TYPE_NOT_MATCH`: Wrong stream type. Triggered when you call internal capture APIs after switch to the custom stream source with setVideoSourceType.
 	 * @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.
@@ -3270,7 +3294,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	/** 
 	 * @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.
+	 *        The remote clients in the room will be informed of the state change via [onUserStartAudioCapture](docs-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 The actual effective audio capture parameters. See [MediaTrackSettings](https://developer.mozilla.org/en-US/docs/Web/API/MediaTrackSettings).
 	 *         The following error codes may be returned.
@@ -3287,7 +3311,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	/** 
 	 * @brief Stop 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 [onUserStopAudioCapture](104479#onuserstopaudiocapture) after stopping audio capture by calling this API.
+	 * The remote clients in the room will be informed of the state change via [onUserStopAudioCapture](docs-104479#onuserstopaudiocapture) after stopping audio capture by calling this API.
 	 * @return `STREAM_TYPE_NOT_MATCH`: Wrong stream type. Triggered when you call internal capture APIs after switch to the custom stream source with setAudioSourceType.
 	 * @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.
@@ -3298,7 +3322,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	
 	/** 
 	 * @brief Enable audio and video capture using the internal capture module. When calling this method, the browser will request microphone and camera permissions simultaneously, and the user only needs to grant permission once.
-	 *        When calling this method, other users in the room will receive the callback of [onUserStartVideoCapture](104479#onuserstartvideocapture) and [onUserStartAudioCapture](104479#onuserstartaudiocapture).
+	 *        When calling this method, other users in the room will receive the callback of [onUserStartVideoCapture](docs-104479#onuserstartvideocapture) and [onUserStartAudioCapture](docs-104479#onuserstartaudiocapture).
 	 * @param optionsOrAudioDeviceId When using a single parameter, both audio and video device IDs can be set simultaneously. When using two parameters, the audio device's ID can be set.
 	 * @param videoDeviceId Video device's ID.
 	 * @return The actual audio and video capture parameters that take effect. Refer to [MediaTrackSettings](https://developer.mozilla.org/en-US/docs/Web/API/MediaTrackSettings). Pay attention to the properties: `deviceId`, `channelCount`, `sampleRate`, `frameRate`, `height` and `width`.
@@ -3333,7 +3357,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	/** 
 	 * @hidden
 	 * @brief Enable audio and video capture using the internal capture module. When calling this method, the browser will request microphone and camera permissions simultaneously, and the user only needs to grant permission once.
-	 * When calling this method, other users in the room will receive the callback of [onUserStartVideoCapture](104479#onuserstartvideocapture) and [onUserStartAudioCapture](104479#onuserstartaudiocapture).
+	 * When calling this method, other users in the room will receive the callback of [onUserStartVideoCapture](docs-104479#onuserstartvideocapture) and [onUserStartAudioCapture](docs-104479#onuserstartaudiocapture).
 	 * @param videoDeviceId Video device's ID.
 	 * @param audioDeviceId Audio device's ID.
 	 * @return The actual audio and video capture parameters that take effect. Refer to [MediaTrackSettings](https://developer.mozilla.org/en-US/docs/Web/API/MediaTrackSettings). Pay attention to the properties: `deviceId`, `channelCount`, `sampleRate`, `frameRate`, `height` and `width`.
@@ -3368,23 +3392,25 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	stopScreenCapture: () => Promise<void>;
 	
 	/** 
-	 * @brief Bind a view to the local media stream and set the rendering mode for the rendering of the local stream
+	 * @brief Add the local media stream to a player and set the rendering mode for the rendering of the local stream.
 	 *        Listen to `onPlayerEvent` for the state after binding a view to the local video stream.
-	 *        Call this API again with a blank view to unbind the view from the local video stream.
+	 *        Call this API again with a blank view to remove the stream from the player. Doing this without a playerId while there are multiple players including a default player will remove the stream from the default player.
 	 * @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.
+	 * @param videoPlayerOption View information and rendering mode. Call this API repeatedly with different playerId to add the stream to different players.
+	 * @notes + You should bind the view after the user joins a room. This setting will sustain when the user leaves the room.
+	 * + If the stream has already been added to the specific player, repeated calls will not create duplicate players.
 	 * @group Video Management
 	 * @order 20
 	 */
 	setLocalVideoPlayer: (streamIndex: StreamIndex, videoPlayerOption?: Partial<Omit<VideoPlayerOption, "rotation">>) => HTMLVideoElement | undefined;
 	
 	/** 
-	 * @brief Bind a view to the remote media stream and set the rendering mode for the rendering of the remote stream
-	 *        It is recommended to call this API and bind a view once the event of [onUserPublishStream](104479#onuserpublishstream)/[onUserPublishScreen](104479#onuserpublishscreen) is triggered.
-	 *        Call this API again with a blank view to unbind the view from the remote video stream.
-	 * @param streamIndex Video stream properties
-	 * @param videoPlayerOption View information and rendering mode. Starting from version 4.56, you can set the rotation angle of the remote video rendering using `rotation`.
+	 * @brief Add the remote media stream to a player and set the rendering mode for the rendering of the remote stream.
+	 *        It is recommended to call this API and bind a view once the event of [onUserPublishStream](docs-104479#onuserpublishstream)/[onUserPublishScreen](docs-104479#onuserpublishscreen) is triggered.
+	 *        Call this API again with a blank view to remove the stream from the player. Doing this without a playerId while there are multiple players including a default player will remove the stream from the default player.
+	 * @param streamIndex Video stream properties.
+	 * @param videoPlayerOption View information and rendering mode. Since V4.61, call this API repeatedly with different playerIds to add the stream to different players.
+	 * Since V4.56, you can set the rotation of the stream via the `rotation` property.
 	 * @notes You should bind the view after the user joins a room. This setting will sustain when the user leaves the room.
 	 * @group Video Management
 	 * @order 21
@@ -3404,7 +3430,8 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @param deviceId ID of the audio playback device. Call [enumerateDevices](#enumeratedevices) to get device ID.
 	 * @return + `INVALID_DEVICE_ID`: Invalid device ID.
 	 *         + `NOT_SUPPORTED`: The browser does not support audio device test or audio capture.
-	 * @notes  You can call this API before or after the user joins a room.
+	 * @notes  + This method is available on Chrome.
+	 * + You can call this API before or after the user joins a room.
 	 * @group Audio Management
 	 * @order 1
 	 */
@@ -3412,18 +3439,59 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	
 	/** 
 	 * @brief Play audio and video stream manually.
-	 * Call this API when user need to take part in the audio and video reaction after calling [createEngine](104478#createengine) and set to autoPlay to False.
-	 * @param userId ID of the user whose stream is intended to play
-	 *               Call this API without a parameter to start playing the local medias.
-	 * @param mediaType Media stream type, used for specifying whether to publish audio stream or video stream.
-	 * @param streamIndex Video stream properties
+	 * Call this API when user need to take part in the audio and video reaction after calling [createEngine](docs-104478#createengine) and set to autoPlay to False.
+	 * @param userId The user ID or the public stream ID. The empty value indicates the local stream. When you specify a public stream ID, the `mediaType` or `streamIndex` will not take effect.
+	 * @param mediaType For audio or video. It defaults to `AUDIO_AND_VIDEO`. The local audio is not included. While the Audio selection is enabled, it can only be set to `VIDEO`.
+	 * @param streamIndex Main stream or screen-sharing stream. By default, the action will apply to both the main stream and the screen-sharing stream.
+	 * @param playerId Player ID. An empty value indicates the default player.
 	 * @return Calling this API when the audio and video are playing will also return a Promise.
-	 * @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.
+	 * @note + Call this methods to play the stream in the player where the stream has been added into via [`setLocalVideoPlayer`](docs-104478#rtcengine-setlocalvideoplayer) or [`setRemoteVideoPlayer`](docs-104478#rtcengine-setremotevideoplayer).
+	 * + Call this API to start playback when the audio and/or video is not playing.
+	 * + Refer to [How to Deal with Auto-play Restrictions of Browsers](https://docs.byteplus.com/en/docs/byteplus-rtc/docs-130302) for more information.
 	 * @group Video Management
 	 * @order 25
 	 */
-	play: (userId?: string, mediaType?: MediaType, streamIndex?: StreamIndex) => Promise<void>;
+	play: (userId?: string, mediaType?: MediaType, streamIndex?: StreamIndex, playerId?: string) => Promise<void>;
+	/** 
+	 * @brief Pause the specified media stream.
+	 * @param userId The user ID or the public stream ID. The empty value indicates the local stream. When you specify a public stream ID, the `mediaType` or `streamIndex` will not take effect.
+	 * @param mediaType For audio or video. It defaults to `AUDIO_AND_VIDEO`. The local audio is not included. While the Audio selection is enabled, it can only be set to `VIDEO`.
+	 * @param streamIndex Main stream or screen-sharing stream. By default, the action will apply to both the main stream and the screen-sharing stream.
+	 * @param playerId Player ID. An empty value indicates the default player.
+	 * @notes After using this method to pause the media stream,
+	 * + The billing duration will not stop counting since the subscription doesn't cancel.
+	 * + The stream will not remove from the player. You can call [play](#play) to resume the stream.
+	 * + The volume of the stream will not be `0`, which you can obtain via [getAudioLevel](getAudioLevel) or the callback of [onLocalAudioPropertiesReport](docs-104479#onlocalaudiopropertiesreport) or [onRemoteAudioPropertiesReport](docs-104479#onremoteaudiopropertiesreport).
+	 * @group Video Management
+	 * @order 26
+	 */
+	
+	stop: (userId?: string, mediaType?: MediaType, streamIndex?: StreamIndex, playerId?: string) => void;
+	
+	/** 
+	 * @brief Gets the volume of the remote audio stream or the local audio stream captured by the local microphone and screen audio stream.
+	 * @param streamIndex Main stream or screen-sharing stream
+	 * @param userId The user ID. The empty value indicates the local stream.
+	 * @return If the specified stream does not exist, the volume values are `0`.
+	 * + linear volume. The value is in linear relation to the original volume. The higher the value, the higher the volume. The range is [0,255]. <br>
+	 * &ensp; - [0, 25]: Silence
+	 * &ensp; - [26, 75]: Low volume
+	 * &ensp; - [76, 204]: Medium volume
+	 * &ensp; - [205, 255]: High volume
+	 * + non-linear volume in dB. The value is in proportion to the log value of the original volume. You can use the value to recognize the Active Speaker in the room. The range is [-127, 0].
+	 * &ensp; - [-127, -60]: Silence
+	 * &ensp; - [-59, -40]: Low volume
+	 * &ensp; - [-39, -20]: Medium volume
+	 * &ensp; - [-19, 0]: High volume
+	 * @notes + The interval for repeatedly calling this method should be at least 100 milliseconds.
+	 * + You can also get volume infor via [onLocalAudioPropertiesReport](docs-104479#onlocalaudiopropertiesreport) and [onRemoteAudioPropertiesReport](docs-104479#onremoteaudiopropertiesreport).
+	 * @group Audio Management
+	 * @order 27
+	 */
+	getAudioVolume: (streamIndex: StreamIndex, userId?: string) => {
+		linearVolume: number;
+		nonlinearVolume: number;
+	};
 	
 	/** 
 	 * @brief Pause receiving remote media streams.
@@ -3439,7 +3507,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	
 	/** 
 	 * @brief Resume receiving remote media streams
-	 * @param mediaType Media stream type subscribed to
+	 * @param mediaType Media stream type
 	 * @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.
@@ -3450,7 +3518,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	
 	/** 
 	 * @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](docs-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 + `USER_MESSAGE_TIMEOUT`: P2P message failed to be sent due to a timeout.
@@ -3466,7 +3534,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	
 	/** 
 	 * @brief Send point-to-point binary messages to a specified user in the room.
-	 * The recipient receives the message via [onUserBinaryMessageReceived](104479#onuserbinarymessagereceived) once the message is sent successfully.
+	 * The recipient receives the message via [onUserBinaryMessageReceived](docs-104479#onuserbinarymessagereceived) once the message is sent successfully.
 	 * @param userId User ID of the recipient
 	 * @param message Binary to be sent within 64 KB
 	 * @return + `USER_MESSAGE_TIMEOUT`: P2P message failed to be sent due to a timeout.
@@ -3482,7 +3550,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	
 	/** 
 	 * @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](docs-104479#onroommessagereceived) once the message is sent successfully.
 	 * @param message Text to be sent within 64 KB
 	 * @return
 	 * @group Message
@@ -3492,7 +3560,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	
 	/** 
 	 * @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](docs-104479#onroombinarymessagereceived) once the message is sent successfully.
 	 * @param message Binary message to be sent within 64 KB
 	 * @return
 	 * @group Message
@@ -3510,12 +3578,10 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 *         &ensp; - SEI messages only support H.264 encoding, and use on browsers with Chromium 86+.
 	 *         &ensp; - Android devices do not support sending SEI data in voice call scenarios.
 	 *         &ensp; - SEI messages cannot be used on iOS devices.
-	 *        + In voice call scenarios, you still need to publish and subscribe to video streams, instead of audio streams.<br/>
-	 *         &ensp; - The publisher can enable automatic publishing when entering the room, or call [publishStream](#publishstream) with the media stream type `AUDIO_AND_VIDEO`.
-	 *         &ensp; - The subsciber can enable automatic subscribing to video streams when entering the room, or call [subscribeStream](#subscribestream) with the media stream type `AUDIO_AND_VIDEO`.
+	 *        + In voice call scenarios, you still need to publish and subscribe to video streams, instead of audio streams. The subsciber can enable automatic subscribing to video streams when entering the room, or call [subscribeStream](#subscribestream) with the media stream type `AUDIO_AND_VIDEO`.
 	 *        + In voice call scenarios, if no SEI data is sent within 1 minute after calling this interface, the SDK will automatically cancel the publication of the black frame video stream.
-	 *        + After the message is sent successfully, the remote end will receive [onSEIMessageReceived](104479#onseimessagereceived).
-	 *        + In voice call scenarios, the sending status of the black frame video stream will be notified to remote users through the [onSEIStreamUpdate](104479#onseistreamupdate) event.
+	 *        + After the message is sent successfully, the remote end will receive [onSEIMessageReceived](docs-104479#onseimessagereceived).
+	 *        + In voice call scenarios, the sending status of the black frame video stream will be notified to remote users through the [onSEIStreamUpdate](docs-104479#onseistreamupdate) event.
 	 * @group Message
 	 * @order 10
 	 */
@@ -3530,7 +3596,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @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 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](docs-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).
@@ -3541,7 +3607,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	
 	/** 
 	 * @brief Set the encoding configuration of the screen-sharing stream，which is also applied in video capture.
-	 *        If the configuration is not supported by your browser, the video will be captured based on the supported configuration. You will also receive the callback [onLocalVideoSizeChanged](104479#onlocalvideosizechanged).
+	 *        If the configuration is not supported by your browser, the video will be captured based on the supported configuration. You will also receive the callback [onLocalVideoSizeChanged](docs-104479#onlocalvideosizechanged).
 	 * @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.
@@ -3569,7 +3635,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 *         + `false`: Failure.
 	 * @notes + 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).
-	 *        + Due to browser restrictions, the remote end can only subscribe to the small stream in Firefox version 80.
+	 *        + Simulcast is not available on Firefox.
 	 * @group Audio & Video Transport
 	 * @order 25
 	 */
@@ -3580,7 +3646,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @deprecated since 4.42, use enableAudioPropertiesReport instead
 	 * @brief Turn on/off volume prompts. Off by default.
 	 * Call this API to set the report period of the volume information of local user and each subscribed user.
-	 * Local client will get informed of the the volume of local user and each subscribed users via [onAudioVolumeIndication](104479#onaudiovolumeindication).
+	 * Local client will get informed of the the volume of local user and each subscribed users via [onAudioVolumeIndication](docs-104479#onaudiovolumeindication).
 	 * @param interval The time interval for receiving a volume prompt callback.
 	 *        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.
@@ -3592,8 +3658,8 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @brief Enable audio information prompts.
 	 * @param config Audio information prompts-related configurations.
 	 * @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).
+	 *       + Get the information of the audio stream captured by the local microphone and screen audio stream through [onLocalAudioPropertiesReport](docs-104479#onlocalaudiopropertiesreport).
+	 *       + Get the information of the subscribed remote audio streams through [onRemoteAudioPropertiesReport](docs-104479#onremoteaudiopropertiesreport).
 	 * @group Audio Management
 	 * @order 35
 	 */
@@ -3604,8 +3670,8 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @param transcode Configurations to be set when pushing media streams to CDN.
 	 * @return
 	 * @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.
+	 *        + After calling this API, you will receive the pushing a mixed stream to CDN events via [onStreamMixingEvent](docs-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/docs/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/docs/byteplus-rtc/docs-75125#transcodestarted) and [TranscodeUpdated](https://docs.byteplus.com/en/docs/byteplus-rtc/docs-75125#transcodeupdated) callbacks.
 	 *        + Call [stopLiveTranscoding](#stoplivetranscoding) to stop pushing mixed streams to CDN.
 	 * @group Push to CDN
 	 * @order 5
@@ -3614,8 +3680,8 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	
 	/** 
 	 * @brief Update configuration needed when pushing media streams to CDN.
-	 *        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.
+	 *        After calling [startLiveTranscoding](docs-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](docs-104481#livetranscodeconfig) for specific indications.
 	 * @return
 	 * @group Push to CDN
 	 * @order 6
@@ -3624,7 +3690,7 @@ 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).
+	 * For starting pushing media streams to CDN. Refer to [startLiveTranscoding](docs-104478#startlivetranscoding).
 	 * @return
 	 * @group Push to CDN
 	 * @order 7
@@ -3646,15 +3712,15 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	
 	/** 
 	 * @brief Set user visibility. The local user is visible to others by default before calling this API.
-	 *        An RTC room can accommodate a maximum of 50 visible users, and 30 media streams can be published simultaneously. For more information, see [Room Capability](257549).
+	 *        An RTC room can accommodate a maximum of 50 visible users, and 30 media streams can be published simultaneously. For more information, see [Room Capability](docs-257549).
 	 * @param enable  Visibility of the user in the room.
 	 *    &ensp;     + `true`: The user can publish media streams. And the other users in the room get informed of the behaviors of the user, such as joining room, starting video capture, and leaving room.
 	 *    &ensp;     + `false`: The user cannot publish media streams. And the other users in the room do not get informed of the behaviors of the user, such as joining, starting video capture, or leaving.
 	 * @return
 	 * @notes + You can call this API whether the user is in a room or not.
 	 *        + 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)
+	 *            - Switch from `false` to `true`: [onUserJoined](docs-104479#onuserjoined)
+	 *            - Switch from `true` to `false`: [onUserLeave](docs-104479#onuserleave)
 	 * @group Room Management
 	 * @order 10
 	 */
@@ -3662,22 +3728,22 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	
 	/** 
 	 * @brief Set a custom videoTrack as the video source.
-	 *        Before calling this API, you must call [setVideoSourceType](104478#setvideosourcetype) to enable custom video capture.
+	 *        Before calling this API, you must call [setVideoSourceType](docs-104478#setvideosourcetype) to enable custom video capture.
 	 * @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
-	 * @group Video Management
-	 * @order 15
+	 * @group Custom Stream Processing
+	 * @order 5
 	 */
 	setExternalVideoTrack: (index: StreamIndex, track: MediaStreamTrack) => Promise<void>;
 	
 	/** 
 	 * @brief Set a custom audioTrack as the audio source.
-	 *        Before calling this API, you must call [setAudioSourceType](104478#setaudiosourcetype) to enable custom video capture.
+	 *        Before calling this API, you must call [setAudioSourceType](docs-104478#setaudiosourcetype) to enable custom video capture.
 	 * @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
-	 * @group Audio Management
+	 * @group Custom Stream Processing
 	 * @order 15
 	 */
 	setExternalAudioTrack: (index: StreamIndex, track: MediaStreamTrack) => Promise<void>;
@@ -3688,8 +3754,8 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @param index audio stream index
 	 * @param videoSourceType Type of the module for video capture
 	 * @return
-	 * @group Video Management
-	 * @order 10
+	 * @group Custom Stream Processing
+	 * @order 0
 	 */
 	setVideoSourceType: (index: StreamIndex, videoSourceType: VideoSourceType) => Promise<void>;
 	
@@ -3699,7 +3765,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @param index audio stream index
 	 * @param audioSourceType Type of the module for video capture
 	 * @return
-	 * @group Audio Management
+	 * @group Custom Stream Processing
 	 * @order 10
 	 */
 	setAudioSourceType: (index: StreamIndex, audioSourceType: AudioSourceType) => Promise<void>;
@@ -3727,7 +3793,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @deprecated since v4.51, use setVideoEncoderConfig instead.
 	 * @brief Configure the internal video capture. Different browser has its own default settings.
 	 * @param config Configuration for internal video capture. See [MediaTrackConstraints](https://developer.mozilla.org/en-US/docs/Web/API/MediaTrackConstraints).
-	 *               The input `frameRate`, `height`, `width` are only desired values. The true resolution and framerate of the internal capture are the return values of [startVideoCapture](104478#startvideocapture).
+	 *               The input `frameRate`, `height`, `width` are only desired values. The true resolution and framerate of the internal capture are the return values of [startVideoCapture](docs-104478#startvideocapture).
 	 * @return
 	 * @note + You can call this API after creating the engine. We recommend to call this API before [startVideoCapture](#startvideocapture).
 	 *       + Since v4.51, the scaling ratio of capturing resolution to encoding resolution is fixed to 1. When the capturing resolution is modified, the actual sending resolution will be changed at the same time.
@@ -3739,7 +3805,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 *        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).
+	 *         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](docs-70121).
 	 * @param userId  User ID.
 	 * @return + `ALREADY_LOGIN`: The user has already logged in.
 	 *         + `LOGIN_FAILED`: Login failed. Please check the returned message for the specific reason.
@@ -3751,7 +3817,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	login: (token: string | null, userId: string) => Promise<void>;
 	
 	/** 
-	 * @brief After calling this API to logout you are not able to call APIs to send messages and server messages.
+	 * @brief Log out of the RTS service. After calling this API to logout you are not able to call APIs to send messages and server messages.
 	 * @return
 	 * @group Message
 	 * @order 16
@@ -3789,7 +3855,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 *         + `NOT_LOGIN`: The sender has not logged in yet.
 	 *         + `USER_MESSAGE_UNKNOWN`: Unknown error.
 	 * @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](docs-104479#onusermessagereceivedoutsideroom) if the message has been sent successfully.
 	 * @group Message
 	 * @order 30
 	 */
@@ -3805,21 +3871,21 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 *         + `NOT_LOGIN`: The sender has not logged in yet.
 	 *         + `USER_MESSAGE_UNKNOWN`: Unknown error.
 	 * @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.
+	 *        + The recipient must be logged into the server and will receive the text message via [EngineEventsTypes.onUserBinaryMessageReceivedOutsideRoom](docs-104479#onuserbinarymessagereceivedoutsideroom) if the message has been sent successfully.
 	 * @group Message
 	 * @order 31
 	 */
 	sendUserBinaryMessageOutsideRoom: (userId: string, message: ArrayBuffer) => Promise<any>;
 	
 	/** 
-	 * @brief Set the params of the server before calling [sendServerMessage](104478#sendservermessage) or [sendServerBinaryMessage](104478#sendserverbinarymessage) to send messages to the server.
+	 * @brief Set the params of the server before calling [sendServerMessage](docs-104478#sendservermessage) or [sendServerBinaryMessage](docs-104478#sendserverbinarymessage) to send messages to the server.
 	 * @param signature Dynamic signature. The App server may use the signature to verify the source of messages.
 	 *                  You need to define the signature yourself. It can be any non-empty string. It is recommended to encode information such as UID into the signature.
 	 *                  The signature will be sent to the address set through the "url" parameter in the form of a POST request.
 	 * @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](docs-104479#onserverparamssetresult) notifies you whether the setting successes.
 	 * @group Message
 	 * @order 35
 	 */
@@ -3864,7 +3930,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @param  cloudProxyInfo cloud proxy informarion list.
 	 * @notes  + Call this API before joining the room.
 	 *         + Start pre-call network detection after starting cloud proxy.
-	 *         + After starting cloud proxy and connects the cloud proxy server successfully, receives [onCloudProxyConnected](104479#oncloudproxyconnected).
+	 *         + After starting cloud proxy and connects the cloud proxy server successfully, receives [onCloudProxyConnected](docs-104479#oncloudproxyconnected).
 	 *         + To stop cloud proxy, call [stopCloudProxy](#stopcloudproxy).
 	 */
 	startCloudProxy: (cloudProxyInfo: CloudProxyInfo) => void;
@@ -3894,7 +3960,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	
 	/** 
 	 * @brief Update the properties of the public stream published by the current user.
-	 *        Call [startPushPublicStream](104478#startpushpublicstream) to start publishing a public stream.
+	 *        Call [startPushPublicStream](docs-104478#startpushpublicstream) to start publishing a public stream.
 	 * @param publicStreamId The public stream's ID.
 	 *                       The public stream must be published by the current user.
 	 * @param publicStreamParam The public stream's parameters.
@@ -3907,7 +3973,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	/** 
 	 * @type api
 	 * @brief Stops publishing the public stream.
-	 *        Call [startPushPublicStream](104478#startpushpublicstream) to start publishing a public stream.
+	 *        Call [startPushPublicStream](docs-104478#startpushpublicstream) to start publishing a public stream.
 	 * @param publicStreamId The public stream's ID.
 	 *                       The public stream must be published by the current user.
 	 * @return
@@ -3923,8 +3989,8 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 *                       If the designated stream is not published when you call this API, you will receive the public stream once it is published.
 	 * @return
 	 * @notes + Before calling this API, we recommend that you call [setPublicStreamVideoPlayer](#setpublicstreamvideoplayer) first to set the player information and rendering mode for the public stream.
-	 *        + 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).
+	 *        + After calling this API, you will receive [onFirstPublicStreamVideoFrameDecoded](docs-104479#onfirstpublicstreamvideoframedecoded) and [onFirstPublicStreamAudioFrameDecoded](docs-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](docs-104479#onpublicstreamseimessagereceived).
 	 *        + Call [stopPlayPublicStream](#stopplaypublicstream) to unsubscribing from the public stream.
 	 * @group Public Stream
 	 * @order 5
@@ -3933,7 +3999,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	
 	/** 
 	 * @brief Unsubscribes from a designated public stream.
-	 *        Call [startPlayPublicStream](104478#startplaypublicstream) to subscribe to the public stream.
+	 *        Call [startPlayPublicStream](docs-104478#startplaypublicstream) to subscribe to the public stream.
 	 * @param publicStreamId The public stream's ID.
 	 * @return
 	 * @group Public Stream
@@ -3948,7 +4014,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @group Public Stream
 	 * @order 10
 	 */
-	setPublicStreamVideoPlayer: (publicStreamId: string, videoPlayerOption: Omit<VideoPlayerOption, "userId">) => void;
+	setPublicStreamVideoPlayer: (publicStreamId: string, videoPlayerOption: Omit<VideoPlayerOption, "userId" | "playerId">) => void;
 	
 	/** 
 	 * @brief Set the placeholder image during local video capture pause
@@ -3979,7 +4045,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @return + `true` Success
 	 *         + `false` Failure
 	 * @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.
+	 *        + After you enable the fallback, you will receive [onSimulcastSubscribeFallback](docs-104479#onsimulcastsubscribefallback) and [onRemoteVideoSizeChanged](docs-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
@@ -3988,7 +4054,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	
 	/** 
 	 * @brief Set the priority of the remote user's media stream.
-	 *        Use this API with [setSubscribeFallbackOption](104478#setsubscribefallbackoption).
+	 *        Use this API with [setSubscribeFallbackOption](docs-104478#setsubscribefallbackoption).
 	 *        With video stream fallback enabled and user priority specified, the stream quality of higher priority user is the last to be compromised.
 	 * @param userId The ID of the remote user
 	 * @param priority The priority of the remote user's media stream
@@ -4000,7 +4066,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	setRemoteUserPriority: (userId: string, priority: RemoteUserPriority) => boolean;
 	
 	/** 
-	 * @brief Start the playback test for the local audio device. RTC will start playing the audio file specified. And RTC will notify the audio volume via the [onAudioPlaybackDeviceTestVolume](104479#onaudioplaybackdevicetestvolume) periodically.
+	 * @brief Start the playback test for the local audio device. RTC will start playing the audio file specified. And RTC will notify the audio volume via the [onAudioPlaybackDeviceTestVolume](docs-104479#onaudioplaybackdevicetestvolume) periodically.
 	 * @param filePath Specify the path of the audio file for the playback test, including *.wav and *.mp3.
 	 * @param indicationInterval The time interval between each callback in milliseconds. We recommend setting it to 200 ms. The minimal value is 100 ms.
 	 * @return Resolve when the test ends.
@@ -4022,8 +4088,8 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	
 	/** 
 	 * @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.
-	 * @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.
+	 *        We recommend to register [onLocalAudioPropertiesReport](docs-104479#onlocalaudiopropertiesreport) callback via [enableAudioPropertiesReport](docs-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](docs-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.
 	 *         The following error codes may be returned.
@@ -4043,15 +4109,15 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	
 	/** 
 	 * @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.
+	 *        After calling [startAudioDeviceRecordTest](docs-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](docs-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).
+	 * @brief Stop the capture and playback test for local audio devices which is started by calling [startAudioDeviceRecordTest](docs-104478#startaudiodevicerecordtest).
 	 *        Before the test ends by itself, you can call this API to stop the recording or playing.
 	 * @group Device Management
 	 * @order 27
@@ -4084,11 +4150,11 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 *        + 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 the video resolution or audio volume of the local streams: [onLocalStreamStats](docs-104479#onlocalstreamstats)
 	 *          &ensp; - Get device IDs: [enumerateDevices](#enumeratedevices)
 	 *          &ensp; - Stop capturing: [stopAudioCapture](#stopaudiocapture), [stopVideoCapture](#stopvideocapture)
 	 * @group Custom Stream Processing
-	 * @order 0
+	 * @order 20
 	 */
 	getLocalStreamTrack: (streamIndex: StreamIndex, type: "video" | "audio") => MediaStreamTrack | undefined;
 	
@@ -4105,7 +4171,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 *          &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
+	 * @order 25
 	 */
 	getRemoteStreamTrack: (userId: string, streamIndex: StreamIndex, type: "video" | "audio") => MediaStreamTrack | undefined;
 	
@@ -4117,14 +4183,15 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @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).
 	 * @group Custom Stream Processing
-	 * @order 5
+	 * @order 30
 	 */
 	getPublicStreamTrack: (id: string, type: "video" | "audio") => MediaStreamTrack | undefined;
 	
 	/** 
 	 * @brief Take a snapshot of the local video.
 	 * @param streamIndex The specified video stream.
-	 * @notes + The snapshot is taken with all video effects on, like rotation, and mirroring.
+	 * @param playerId Player ID. An empty value indicates the default player. If the default player does not exist by then, the call will not take effect.
+	 * @notes + The snapshot is taken with all video effects on, including rotation, mirroring, virtual background.
 	 *        + You can take the snapshot either using SDK internal video capture or customized capture.
 	 * @return `INVOKED_BEFORE_CAPTURE`: No such stream. Please make sure that the specified stream has been published.
 	 * @group Video Snapshot
@@ -4136,6 +4203,7 @@ 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.
+	 * @param playerId Player ID. An empty value indicates the default player. If the default player does not exist by then, the call will not take effect.
 	 * @return `STREAM_NOT_EXIST`: The target stream does not exist. Please enter the correct ID or make sure that the target stream is in the room.
 	 * @group Video Snapshot
 	 * @order 1
@@ -4144,7 +4212,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	
 	/** 
 	 * @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).
+	 *        You can use the extension to implement features such as [Beauty Effects](https://docs.byteplus.com/en/docs/byteplus-rtc/docs-148646), [Virtual Background](https://docs.byteplus.com/en/docs/byteplus-rtc/docs-148647).
 	 * @param extension The extension instance.
 	 * @return
 	 * @group Extensions
@@ -4155,8 +4223,8 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	/** 
 	 * @hidden currently not available
 	 * @brief Turns on subtitles.
-	 *        After calling this method, you can choose between the recognition and translation modes based on your needs. In the recognition mode, the speech of the speaker in the room will be recognized and converted into captions. In the translation mode, the transcribed text will be translated.You will receive the [onSubtitleMessageReceived](104479#onsubtitlemessagereceived) callback to informs you of related information about subtitles.
-	 *        After calling this method, you will receive the [onSubtitleStateChanged](104479#onsubtitlestatechanged) to inform you of whether subtitles are on.
+	 *        After calling this method, you can choose between the recognition and translation modes based on your needs. In the recognition mode, the speech of the speaker in the room will be recognized and converted into captions. In the translation mode, the transcribed text will be translated.You will receive the [onSubtitleMessageReceived](docs-104479#onsubtitlemessagereceived) callback to informs you of related information about subtitles.
+	 *        After calling this method, you will receive the [onSubtitleStateChanged](docs-104479#onsubtitlestatechanged) to inform you of whether subtitles are on.
 	 * @param config Subtitle configrations.
 	 * @return `SUBTITLE_ALREADY_ON`: Subtitle is already turned on.
 	 * @notes + You must call this method after joining the room.
@@ -4170,7 +4238,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @hidden currently not available
 	 * @brief Update the source language after turning on subtitles.
 	 * @param config Subtitle configurations.
-	 * @return After updating the configurations, you will receive the [onSubtitleStateChanged](104479#onsubtitlestatechanged) to inform you.
+	 * @return After updating the configurations, you will receive the [onSubtitleStateChanged](docs-104479#onsubtitlestatechanged) to inform you.
 	 * @notes + You must call this method after turning on the subtitles via [startSubtitle]({{api}}#RTCEngine.startSubtitle).
 	 *        + 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"`, to Korean by importing  `"source_language": "ko"`, 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
@@ -4181,7 +4249,7 @@ 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.
+	 *        After calling this method, you will receive the [onSubtitleStateChanged](docs-104479#onsubtitlestatechanged) to inform you of whether subtitles are off.
 	 * @group Live Caption and Translation
 	 * @order 1
 	 */
@@ -4197,7 +4265,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 *         + `false`: Failure.
 	 * @notes You must call this API before joining the room. Calling it afterward is invalid.
 	 * @group Custom Stream Processing
-	 * @order 10
+	 * @order 35
 	 */
 	setRemoteStreamRenderSync: (sync: boolean) => boolean;
 	
@@ -4232,7 +4300,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 *               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.
-	 *        + The playback volume cannot be adjusted in [Conference Mode](113547).
+	 *        + The playback volume cannot be adjusted in [Conference Mode](docs-113547).
 	 * @group Audio Management
 	 * @order 26
 	 */
@@ -4251,7 +4319,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	setPublicStreamVolume: (publicStreamId: string, volume: number) => void;
 	
 	/** 
-	 * @brief Start forwarding media stream across rooms. See [Forward media stream across rooms](https://docs.byteplus.com/en/byteplus-rtc/docs/104398) for details.
+	 * @brief Start forwarding media stream across rooms. See [Forward media stream across rooms](https://docs.byteplus.com/en/docs/byteplus-rtc/docs-104398) for details.
 	 *        After calling [joinRoom]({{api}}#RTCEngine.joinRoom), you can call this API to publish the media stream to multiple rooms. This feature is suitable for scenarios such as co-host PK and online karaoke.
 	 * @param forwardStreamInfos Information of the target rooms where the stream will be forwarded.
 	 * @return Status information for forwarding media stream across rooms.
@@ -4322,7 +4390,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	
 	/** 
 	 * @brief Set the priority of the local audio stream to be published.
-	 * @param audio_selection_priority The priority of the local audio stream which defaults to be subscribable only up to the result of the Audio Selection.
+	 * @param audioSelectionPriority The priority of the local audio stream which defaults to be subscribable only up to the result of the Audio Selection.
 	 * @note + You must enable Audio Selection in the RTC console before using this API. You can call this API whether the user has joined a room. Refer to [Audio Selection](https://docs.byteplus.com/byteplus-rtc/docs/113547).<br>
 	 * + The setting is independent in each room that the user joins.
 	 * @group Room Management
@@ -4359,7 +4427,7 @@ export interface IVERTC {
 	createBLWEngine: (appId: string, config?: EngineConfig) => IRTCEngine;
 	
 	/** 
-	 * @brief Call this API to destroy the IRTCEngine instance created by [createEngine](104478#createengine).
+	 * @brief Call this API to destroy the IRTCEngine instance created by [createEngine](docs-104478#createengine).
 	 * @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.
@@ -4378,9 +4446,9 @@ export interface IVERTC {
 	
 	/** 
 	 * @brief Call this API to enumerate media devices and their IDs, such as microphones, cameras, and headsets.
-	 *        Call this API to get deviceId used to specify the device when calling [setAudioPlaybackDevice](104478#setaudioplaybackdevice), [startVideoCapture](104478#startvideocapture), and [startAudioCapture](104478#startaudiocapture).
+	 *        Call this API to get deviceId used to specify the device when calling [setAudioPlaybackDevice](docs-104478#setaudioplaybackdevice), [startVideoCapture](docs-104478#startvideocapture), and [startAudioCapture](docs-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.
+	 * @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 via [enableDevices](docs-104478#vertc-enabledevices) before calling this API.
 	 * @group Device Management
 	 * @order 0
 	 */
@@ -4416,9 +4484,9 @@ export interface IVERTC {
 	
 	/** 
 	 * @brief Call this API to enumerate microphone devices.
-	 *        Call this API to get Device IDs used to specify the device when calling [startAudioCapture](104478#startaudiocapture) and [setAudioCaptureDevice](104478#setaudiocapturedevice).
+	 *        Call this API to get Device IDs used to specify the device when calling [startAudioCapture](docs-104478#startaudiocapture) and [setAudioCaptureDevice](docs-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.
+	 * @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 [enableDevices](docs-104478#vertc-enabledevices) before calling this API.
 	 * @group Device Management
 	 * @order 10
 	 */
@@ -4426,9 +4494,9 @@ export interface IVERTC {
 	
 	/** 
 	 * @brief Call this API to enumerate audio and video devices and their IDs, such as microphones, cameras, and headsets.
-	 *        Call this API to get Device IDs used to specify the device when calling [startVideoCapture](104478#startvideocapture) and [setVideoCaptureDevice](104478#setvideocapturedevice).
+	 *        Call this API to get Device IDs used to specify the device when calling [startVideoCapture](docs-104478#startvideocapture) and [setVideoCaptureDevice](docs-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.
+	 * @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 [enableDevices](docs-104478#vertc-enabledevices) before calling this API.
 	 * @group Device Management
 	 * @order 11
 	 */
@@ -4436,9 +4504,9 @@ export interface IVERTC {
 	
 	/** 
 	 * @brief Call this API to enumerate audio playback devices and their IDs, such as speakers and headsets.
-	 *        Call this API to get Device IDs used to specify the device when calling [setAudioPlaybackDevice](104478#setaudioplaybackdevice).
+	 *        Call this API to get Device IDs used to specify the device when calling [setAudioPlaybackDevice](docs-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.
+	 * @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 [enableDevices](docs-104478#vertc-enabledevices) before calling this API.
 	 * @group Device Management
 	 * @order 12
 	 */
@@ -4495,6 +4563,32 @@ export interface IVERTC {
 	downloadLog: (key?: string) => void;
 }
 
+/** 
+ * @brief Initial publish state
+ */
+export interface InitPublishState {
+	
+	/** 
+	 * @brief Microphone audio
+	 */
+	audio: boolean;
+	
+	/** 
+	 * @brief Video stream captured by cameras
+	 */
+	video: boolean;
+	
+	/** 
+	 * @brief Computer audio
+	 */
+	screenAudio: boolean;
+	
+	/** 
+	 * @brief Screen recording
+	 */
+	screenVideo: boolean;
+}
+
 /** 
  * @brief Audio transcoding configurations. With invalid or empty input, the configurations will be set as the default values.
  */
@@ -4952,6 +5046,11 @@ export interface PlayerEvent {
 	 * @brief Whether it is a screen-sharing stream
 	 */
 	isScreen?: boolean;
+	
+	/** 
+	 * @brief Player ID. An empty value indicates the default player.
+	 */
+	playerId?: string;
 }
 
 /** 
@@ -5453,21 +5552,21 @@ export interface RoomConfig {
 	/** 
 	 * @brief Enable/disable auto-subscribe audio. Enable by default.
 	 *        This does not include the screen stream.
-	 *        Call [publishStream](104478#publishstream) to publish media streams if you disable `isAutoPublish`.
+	 *        Call [publishStream](docs-104478#publishstream) to publish media streams if you disable `isAutoPublish`.
 	 */
 	isAutoPublish?: boolean;
 	
 	/** 
 	 * @brief Enable/disable auto-subscribe audio. Enable by default.
 	 *        This does not include the screen stream.
-	 *        You can call [subscribeStream](104478#subscribestream) and [unsubscribeStream](104478#unsubscribestream) to revise the subscription after entering the room.
+	 *        You can call [subscribeStream](docs-104478#subscribestream) and [unsubscribeStream](docs-104478#unsubscribestream) to revise the subscription after entering the room.
 	 */
 	isAutoSubscribeAudio?: boolean;
 	
 	/** 
 	 * @brief Enable/disable auto-subscribe video. Enable by default.
 	 *        This does not include the screen stream.
-	 *        You can call [subscribeStream](104478#subscribestream) and [unsubscribeStream](104478#unsubscribestream) to revise the subscription after entering the room.
+	 *        You can call [subscribeStream](docs-104478#subscribestream) and [unsubscribeStream](docs-104478#unsubscribestream) to revise the subscription after entering the room.
 	 */
 	isAutoSubscribeVideo?: boolean;
 	
@@ -5476,7 +5575,7 @@ export interface RoomConfig {
 	 * @deprecated since 4.51, use setAudioProfile instead.
 	 * @brief  Room profile
 	 *        Choose an appropriate room-profile option to meet the requirement of the scenario for media encoding algorithm, video profiles, and network configurations.
-	 *        Call [setAudioProfile](104478#setaudioprofile) to choose an audio quality option.
+	 *        Call [setAudioProfile](docs-104478#setaudioprofile) to choose an audio quality option.
 	 */
 	roomProfileType?: RoomProfileType;
 	
@@ -5497,6 +5596,40 @@ export interface ScreenConfig {
 	 *        • On macOS, you can only share audio from a tab through sharing a Chrome tab.
 	 */
 	enableAudio?: boolean;
+	
+	/** 
+	 * @brief The type of display surface (window, browser, or monitor)
+	 * For Chrome 107, Edge 107, and later versions
+	 * See [displaySurface](https://developer.chrome.com/docs/web-platform/screen-sharing-controls/#displaySurface).
+	 */
+	displaySurface?: "monitor" | "browser" | "window";
+	
+	/** 
+	 * @brief Whether to include the system audio
+	 * `include`：(Default) Include.
+	 * `exclude`：Exclude.
+	 * For Chrome 107, Edge 107, and later versions on Windows
+	 * See [systemAudio](https://developer.chrome.com/docs/web-platform/screen-sharing-controls/#systemAudio).
+	 */
+	systemAudio?: "include" | "exclude";
+	
+	/** 
+	 * @brief Whether to allow the user to dynamically switch the shared tab during screen-sharing.
+	 * `include`: (Default) Allow.
+	 * `exclude`: Disallow.
+	 * For Chrome 107, Edge 107, and later versions
+	 * See [surfaceSwitching](https://developer.chrome.com/docs/web-platform/screen-sharing-controls/#surfaceSwitching).
+	 */
+	surfaceSwitching?: "include" | "exclude";
+	
+	/** 
+	 * @brief Whether to allow the user to select the current tab for capture.
+	 * `include`: (Default) Allow.
+	 * `exclude`: Disallow.
+	 * For Chrome 107, Edge 107, and later versions
+	 * See [selfBrowserSurface](https://developer.chrome.com/docs/web-platform/screen-sharing-controls/#selfBrowserSurface).
+	 */
+	selfBrowserSurface?: "include" | "exclude";
 }
 
 /** 
@@ -5563,6 +5696,8 @@ export interface SetParamInterface {
 	FALLBACK_ENCODE_CODEC: string;
 	SEI_TIME_OUT: number;
 	SEI_COUNT_FPS: number;
+	/** 是否开启提前建连 */
+	PRE_ICE: boolean;
 }
 
 /** 
@@ -5732,6 +5867,11 @@ export interface UserConnectionEvent {
 	 * @brief User information.
 	 */
 	userInfo: UserInfo;
+	
+	/** 
+	 * @brief The publish state of the media stream.
+	 */
+	publishState: InitPublishState;
 }
 
 /** 
@@ -5761,7 +5901,7 @@ export interface UserInfo {
 	userId: string;
 	
 	/** 
-	 * @brief The additional information of the user, with a maximum length of 200 bytes. Remote users will receive the [onUserJoined](104479#onuserjoined).
+	 * @brief The additional information of the user, with a maximum length of 200 bytes. Remote users will receive the [onUserJoined](docs-104479#onuserjoined).
 	 * @notes  The supported character types are as follows:
 	 *      + Uppercase letters: A~ Z
 	 *      + Lowercase letters: a~ z
@@ -5849,14 +5989,21 @@ export interface VideoPlayerOption {
 	 *      + 26 lowercase letters a~ z
 	 *      + 10 numbers 0~ 9
 	 *      + underscore "_", at character "@", minus, number "-"
+	 *     This property does not apply to the public stream.
 	 */
 	userId: string;
 	
 	/** 
-	 * @brief The clockwise rotation angle of the video frame, which can be set to 0, 90, 180, 270. The default value is 0, which means no rotation is applied.<br>
-	 *        This parameter only applies to remote video. Setting this parameter will not take effect when calling [setLocalVideoPlayer](104478#setlocalvideoplayer).
+	 * @brief The clockwise rotation angle of the video frame, which can be set to 0, 90, 180, 270. The default value is 0, which means no rotation is applied.
+	 *  This parameter only applies to remote video. Setting this parameter will not take effect when calling [setLocalVideoPlayer](docs-104478#rtcengine-setlocalvideoplayer).
 	 */
 	rotation?: number;
+	
+	/** 
+	 * @brief Player ID. An empty value indicates the default player.
+	 *        This property does not apply to the public stream.
+	 */
+	playerId?: string;
 }
 
 /** 
@@ -5909,7 +6056,7 @@ export interface VideoStreamDescription {
 export interface onAudioFirstFrameDecodedEvent {
 	
 	/** 
-	 * @brief userId of the user published the video stream
+	 * @brief userId of the user published the audio stream
 	 */
 	userId: string;
 	
@@ -6035,6 +6182,11 @@ export interface onUserJoinedEvent {
 	 * @brief userId of the user who is joining the room
 	 */
 	userInfo: UserInfo;
+	
+	/** 
+	 * @brief The publish state of the media stream.
+	 */
+	publishState: InitPublishState;
 }
 
 /** 
@@ -6098,6 +6250,12 @@ export interface onVideoFirstFrameRenderedEvent {
 	 * @brief If this event comes from a screen video stream
 	 */
 	isScreen: boolean;
+	
+	/** 
+	 * @brief Player ID. An empty value indicates the default player.
+	 *        This property does not apply to the public stream.
+	 */
+	playerId?: string;
 }
 export type CoreSetParamterInterface = {
 	[P in keyof CoreConfigType]: CoreConfigType[P] extends number ? number | string : CoreConfigType[P];