@byteplus/rtc 4.51.1 → 4.52.0

package/index.d.ts

@@ -251,7 +251,7 @@ export interface MixOption {
 	
 	/** 
 	 * @brief Mix playback times
-	 * @notes + Play_count < = 0: Infinite loop
+	 *        + Play_count < = 0: Infinite loop
 	 *        + Play_count == 1: Play once (default)
 	 *        + Play_count > 1: Play play_count times
 	 */
@@ -672,7 +672,7 @@ declare enum ErrorCode {
 	
 	/** 
 	 * @hidden currently not available
-	 * @brief The substitling task didn't start, or call an API to update subtitles without turning on subtitles.
+	 * @brief The subtitling task didn't start, or call an API to update subtitles without turning on subtitles.
 	 */
 	SUBTITLE_NOT_TURNED_ON = "SUBTITLE_NOT_TURNED_ON",
 	
@@ -748,64 +748,7 @@ declare enum AudioMixingError {
 	 */
 	AUDIO_MIXING_ERROR_INVALID_AUDIO_TRACK = 9
 }
-declare class AudioProfileManager {
-	private _appId;
-	private _roomId?;
-	private _constraints;
-	private _profile?;
-	constructor(_appId: string);
-	setRoomId(roomId?: string): void;
-	setAudioProfile(profile: AudioProfileType): void;
-	getOpusConfigStr(conf?: string): string;
-	updateConstraints(conf: MediaTrackConstraints): void;
-	getConstraints(): MediaTrackConstraints;
-	private _getConfigByAudioProfile;
-}
-/** 
- * @brief Cloud proxy informatio
- */
-export interface CloudProxyInfo {
-	
-	/** 
-	 * @brief The cloud proxy IP of log report
-	 */
-	logProxy: string;
-	
-	/** 
-	 * @brief The cloud proxy IP of the access
-	 */
-	accessProxy: string;
-	
-	/** 
-	 * @brief The cloud proxy IP of RTC admin control
-	 */
-	configProxy: string;
-}
-export interface RTCContext {
-	id: string;
-	audioProfileManager?: AudioProfileManager;
-	serverConfig?: {
-		videoCodec?: string;
-		audioRed?: boolean;
-		muteReplaceUnsub?: boolean;
-		simulcastOnDemand?: boolean;
-	};
-	streamRTT?: {
-		[streamId: string]: {
-			audio: number;
-			video: number;
-		};
-	};
-	activeSimStreams?: boolean[];
-	subscribeFallbackOption?: SubscribeFallbackOption;
-	userPriority?: Map<string, RemoteUserPriority>;
-	avSync: boolean;
-}
 export declare type RTCExtension = any;
-/**
- * 设置到pc上的视频编码参数
- */
-export declare type VideoEncoderConfig_PC = Pick<VideoEncoderConfig, "maxKbps">;
 declare enum StreamMixingEventType {
 	
 	/** 
@@ -1458,190 +1401,6 @@ RemoteStreamStats, "userId" | "isScreen"> & {
 	 */
 	publicStreamId: string;
 };
-declare enum EngineEventsTypes {
-	
-	/** 
-	 * @brief The media track ends.
-	 */
-	onTrackEnded = "onTrackEnded",
-	
-	/** 
-	 * @brief Player event
-	 */
-	onPlayerEvent = "onPlayerEvent",
-	
-	/** 
-	 * @brief Streams fails to play automatically.
-	 */
-	onAutoplayFailed = "onAutoplayFailed",
-	
-	/** 
-	 * @brief A user joins the room.
-	 */
-	onUserJoined = "onUserJoined",
-	
-	/** 
-	 * @brief A user leaves the room.
-	 */
-	onUserLeave = "onUserLeave",
-	
-	/** 
-	 * @brief Connection state changes.
-	 */
-	onConnectionStateChanged = "onConnectionStateChanged",
-	onUserPublishStream = "onUserPublishStream",
-	onUserUnpublishStream = "onUserUnpublishStream",
-	onUserPublishScreen = "onUserPublishScreen",
-	onUserUnpublishScreen = "onUserUnpublishScreen",
-	
-	/** 
-	 * @brief Receiving a broadcasting message in the room
-	 */
-	onRoomMessageReceived = "onRoomMessageReceived",
-	
-	/** 
-	 * @brief Receiving a broadcasting binary message in the room
-	 */
-	onRoomBinaryMessageReceived = "onRoomBinaryMessageReceived",
-	
-	/** 
-	 * @brief Receiving a point-to-point text message
-	 */
-	onUserMessageReceived = "onUserMessageReceived",
-	
-	/** 
-	 * @brief Receiving a point-to-point binary message
-	 */
-	onUserBinaryMessageReceived = "onUserBinaryMessageReceived",
-	
-	/** 
-	 * @brief The first video frame has been rendered successfully.
-	 * @deprecated  since 4.47, use on onRemoteVideoFirstFrame instead.
-	 */
-	onVideoFirstFrameRendered = "onVideoFirstFrameRendered",
-	
-	/** 
-	 * @brief The first video frame has been decoded successfully.
-	 * @deprecated since 4.47, use on onRemoteVideoFirstFrame instead.
-	 */
-	onVideoFirstFrameDecoded = "onVideoFirstFrameDecoded",
-	
-	/** 
-	 * @brief The first remote video frame has been rendered successfully.
-	 */
-	onRemoteVideoFirstFrame = "onRemoteVideoFirstFrame",
-	
-	/** 
-	 * @deprecated since 4.47, use on onRemoteAudioFirstFrame instead.
-	 * @brief The first audio frame has been decoded successfully.
-	 */
-	onAudioFirstFrameDecoded = "onAudioFirstFrameDecoded",
-	
-	/** 
-	 * @brief The first audio frame has been decoded successfully.
-	 */
-	onRemoteAudioFirstFrame = "onRemoteAudioFirstFrame",
-	
-	/** 
-	 * The first video frame has been rendered successfully.
-	 */
-	onFirstPublicStreamVideoFrameRendered = "onFirstPublicStreamVideoFrameRendered",
-	
-	/** 
-	 * The first video frame has been decoded successfully.
-	 */
-	onFirstPublicStreamVideoFrameDecoded = "onFirstPublicStreamVideoFrameDecoded",
-	
-	/** 
-	 * The first audio frame has been decoded successfully.
-	 */
-	onFirstPublicStreamAudioFrameDecoded = "onFirstPublicStreamAudioFrameDecoded",
-	
-	/** 
-	 * @brief The state of the video media device changes.
-	 */
-	onVideoDeviceStateChanged = "onVideoDeviceStateChanged",
-	
-	/** 
-	 * @brief The state of the audio media device changes.
-	 */
-	onAudioDeviceStateChanged = "onAudioDeviceStateChanged",
-	
-	/** 
-	 * @brief Callback including the state of the remote media stream
-	 */
-	onRemoteStreamStats = "onRemoteStreamStats",
-	
-	/** 
-	 * Callback including the state of the remote public media stream
-	 */
-	onPublicStreamStats = "onPublicStreamStats",
-	
-	/** 
-	 * @brief Callback including the state of the local media stream
-	 */
-	onLocalStreamStats = "onLocalStreamStats",
-	
-	/** 
-	 * @brief Indication of audio volume
-	 */
-	onAudioVolumeIndication = "onAudioVolumeIndication",
-	onLocalAudioPropertiesReport = "onLocalAudioPropertiesReport",
-	onRemoteAudioPropertiesReport = "onRemoteAudioPropertiesReport",
-	onActiveSpeaker = "onActiveSpeaker",
-	
-	/** 
-	 * @brief Callback of the state changing of audio playback devices
-	 */
-	onAudioPlaybackDeviceChanged = "onAudioPlaybackDeviceChanged",
-	
-	/** 
-	 * @brief A publisher starts video capture.
-	 */
-	onUserStartVideoCapture = "onUserStartVideoCapture",
-	
-	/** 
-	 * @brief A publisher stops video capture.
-	 */
-	onUserStopVideoCapture = "onUserStopVideoCapture",
-	
-	/** 
-	 * @brief A publisher starts audio capture.
-	 */
-	onUserStartAudioCapture = "onUserStartAudioCapture",
-	
-	/** 
-	 * @brief A publisher stops audio capture.
-	 */
-	onUserStopAudioCapture = "onUserStopAudioCapture",
-	onAutoPublishResult = "onAutoPublishResult",
-	onAutoSubscribeResult = "onAutoSubscribeResult",
-	
-	/** 
-	 * @brief The state of pushing streams to CDN
-	 */
-	onLiveTranscodingResult = "onLiveTranscodingResult",
-	onStreamMixingEvent = "onStreamMixingEvent",
-	onAudioPlaybackDeviceTestVolume = "onAudioPlaybackDeviceTestVolume",
-	onSEIMessageReceived = "onSEIMessageReceived",
-	onError = "onError",
-	onAudioMixingStateChanged = "onAudioMixingStateChanged",
-	onUserMessageReceivedOutsideRoom = "onUserMessageReceivedOutsideRoom",
-	onUserBinaryMessageReceivedOutsideRoom = "onUserBinaryMessageReceivedOutsideRoom",
-	onTokenWillExpire = "onTokenWillExpire",
-	onCloudProxyConnected = "onCloudProxyConnected",
-	onPushPublicStreamResult = "onPushPublicStreamResult",
-	onPublicStreamSEIMessageReceived = "onPublicStreamSEIMessageReceived",
-	onNetworkQuality = "onNetworkQuality",
-	onSimulcastSubscribeFallback = "onSimulcastSubscribeFallback",
-	onRemoteVideoSizeChanged = "onRemoteVideoSizeChanged",
-	onVideoStreamBanned = "onVideoStreamBanned",
-	onAudioStreamBanned = "onAudioStreamBanned",
-	onLocalVideoSizeChanged = "onLocalVideoSizeChanged",
-	onSubtitleStateChanged = "onSubtitleStateChanged",
-	onSubtitleMessageReceived = "onSubtitleMessageReceived",
-	onServerParamsSetResult = "onServerParamsSetResult"
-}
 
 /** 
  * @brief Event of banning/unbanning video streaming
@@ -1951,6 +1710,192 @@ declare enum StreamMixingEventErrorCode {
 	 */
 	ERR_UNKNOWN = 9999
 }
+declare enum EngineEventsTypes {
+	
+	/** 
+	 * @brief The media track ends.
+	 */
+	onTrackEnded = "onTrackEnded",
+	onTrackMute = "onTrackMute",
+	onTrackUnmute = "onTrackUnmute",
+	
+	/** 
+	 * @brief Player event
+	 */
+	onPlayerEvent = "onPlayerEvent",
+	
+	/** 
+	 * @brief Streams fails to play automatically.
+	 */
+	onAutoplayFailed = "onAutoplayFailed",
+	
+	/** 
+	 * @brief A user joins the room.
+	 */
+	onUserJoined = "onUserJoined",
+	
+	/** 
+	 * @brief A user leaves the room.
+	 */
+	onUserLeave = "onUserLeave",
+	
+	/** 
+	 * @brief Connection state changes.
+	 */
+	onConnectionStateChanged = "onConnectionStateChanged",
+	onUserPublishStream = "onUserPublishStream",
+	onUserUnpublishStream = "onUserUnpublishStream",
+	onUserPublishScreen = "onUserPublishScreen",
+	onUserUnpublishScreen = "onUserUnpublishScreen",
+	
+	/** 
+	 * @brief Receiving a broadcasting message in the room
+	 */
+	onRoomMessageReceived = "onRoomMessageReceived",
+	
+	/** 
+	 * @brief Receiving a broadcasting binary message in the room
+	 */
+	onRoomBinaryMessageReceived = "onRoomBinaryMessageReceived",
+	
+	/** 
+	 * @brief Receiving a point-to-point text message
+	 */
+	onUserMessageReceived = "onUserMessageReceived",
+	
+	/** 
+	 * @brief Receiving a point-to-point binary message
+	 */
+	onUserBinaryMessageReceived = "onUserBinaryMessageReceived",
+	
+	/** 
+	 * @brief The first video frame has been rendered successfully.
+	 * @deprecated  since 4.47, use on onRemoteVideoFirstFrame instead.
+	 */
+	onVideoFirstFrameRendered = "onVideoFirstFrameRendered",
+	
+	/** 
+	 * @brief The first video frame has been decoded successfully.
+	 * @deprecated since 4.47, use on onRemoteVideoFirstFrame instead.
+	 */
+	onVideoFirstFrameDecoded = "onVideoFirstFrameDecoded",
+	
+	/** 
+	 * @brief The first remote video frame has been rendered successfully.
+	 */
+	onRemoteVideoFirstFrame = "onRemoteVideoFirstFrame",
+	
+	/** 
+	 * @deprecated since 4.47, use on onRemoteAudioFirstFrame instead.
+	 * @brief The first audio frame has been decoded successfully.
+	 */
+	onAudioFirstFrameDecoded = "onAudioFirstFrameDecoded",
+	
+	/** 
+	 * @brief The first audio frame has been decoded successfully.
+	 */
+	onRemoteAudioFirstFrame = "onRemoteAudioFirstFrame",
+	
+	/** 
+	 * The first video frame has been rendered successfully.
+	 */
+	onFirstPublicStreamVideoFrameRendered = "onFirstPublicStreamVideoFrameRendered",
+	
+	/** 
+	 * The first video frame has been decoded successfully.
+	 */
+	onFirstPublicStreamVideoFrameDecoded = "onFirstPublicStreamVideoFrameDecoded",
+	
+	/** 
+	 * The first audio frame has been decoded successfully.
+	 */
+	onFirstPublicStreamAudioFrameDecoded = "onFirstPublicStreamAudioFrameDecoded",
+	
+	/** 
+	 * @brief The state of the video media device changes.
+	 */
+	onVideoDeviceStateChanged = "onVideoDeviceStateChanged",
+	
+	/** 
+	 * @brief The state of the audio media device changes.
+	 */
+	onAudioDeviceStateChanged = "onAudioDeviceStateChanged",
+	
+	/** 
+	 * @brief Callback including the state of the remote media stream
+	 */
+	onRemoteStreamStats = "onRemoteStreamStats",
+	
+	/** 
+	 * Callback including the state of the remote public media stream
+	 */
+	onPublicStreamStats = "onPublicStreamStats",
+	
+	/** 
+	 * @brief Callback including the state of the local media stream
+	 */
+	onLocalStreamStats = "onLocalStreamStats",
+	
+	/** 
+	 * @brief Indication of audio volume
+	 */
+	onAudioVolumeIndication = "onAudioVolumeIndication",
+	onLocalAudioPropertiesReport = "onLocalAudioPropertiesReport",
+	onRemoteAudioPropertiesReport = "onRemoteAudioPropertiesReport",
+	onActiveSpeaker = "onActiveSpeaker",
+	
+	/** 
+	 * @brief Callback of the state changing of audio playback devices
+	 */
+	onAudioPlaybackDeviceChanged = "onAudioPlaybackDeviceChanged",
+	
+	/** 
+	 * @brief A publisher starts video capture.
+	 */
+	onUserStartVideoCapture = "onUserStartVideoCapture",
+	
+	/** 
+	 * @brief A publisher stops video capture.
+	 */
+	onUserStopVideoCapture = "onUserStopVideoCapture",
+	
+	/** 
+	 * @brief A publisher starts audio capture.
+	 */
+	onUserStartAudioCapture = "onUserStartAudioCapture",
+	
+	/** 
+	 * @brief A publisher stops audio capture.
+	 */
+	onUserStopAudioCapture = "onUserStopAudioCapture",
+	onAutoPublishResult = "onAutoPublishResult",
+	onAutoSubscribeResult = "onAutoSubscribeResult",
+	
+	/** 
+	 * @brief The state of pushing streams to CDN
+	 */
+	onLiveTranscodingResult = "onLiveTranscodingResult",
+	onStreamMixingEvent = "onStreamMixingEvent",
+	onAudioPlaybackDeviceTestVolume = "onAudioPlaybackDeviceTestVolume",
+	onSEIMessageReceived = "onSEIMessageReceived",
+	onError = "onError",
+	onAudioMixingStateChanged = "onAudioMixingStateChanged",
+	onUserMessageReceivedOutsideRoom = "onUserMessageReceivedOutsideRoom",
+	onUserBinaryMessageReceivedOutsideRoom = "onUserBinaryMessageReceivedOutsideRoom",
+	onTokenWillExpire = "onTokenWillExpire",
+	onCloudProxyConnected = "onCloudProxyConnected",
+	onPushPublicStreamResult = "onPushPublicStreamResult",
+	onPublicStreamSEIMessageReceived = "onPublicStreamSEIMessageReceived",
+	onNetworkQuality = "onNetworkQuality",
+	onSimulcastSubscribeFallback = "onSimulcastSubscribeFallback",
+	onRemoteVideoSizeChanged = "onRemoteVideoSizeChanged",
+	onVideoStreamBanned = "onVideoStreamBanned",
+	onAudioStreamBanned = "onAudioStreamBanned",
+	onLocalVideoSizeChanged = "onLocalVideoSizeChanged",
+	onSubtitleStateChanged = "onSubtitleStateChanged",
+	onSubtitleMessageReceived = "onSubtitleMessageReceived",
+	onServerParamsSetResult = "onServerParamsSetResult"
+}
 
 /** 
  * @id EngineEvents
@@ -2142,6 +2087,30 @@ export interface IEngineEvents {
 		 */
 		isScreen: boolean;
 	}) => void;
+	[EngineEventsTypes.onTrackMute]: (event: {
+		
+		/** 
+		 * @brief Type of the trac
+		 */
+		kind: string;
+		
+		/** 
+		 * @brief Whether it is a screen-sharing stream.
+		 */
+		isScreen: boolean;
+	}) => void;
+	[EngineEventsTypes.onTrackUnmute]: (event: {
+		
+		/** 
+		 * @brief Type of the trac
+		 */
+		kind: string;
+		
+		/** 
+		 * @brief Whether it is a screen-sharing stream.
+		 */
+		isScreen: boolean;
+	}) => void;
 	
 	/** 
 	 * @brief Event of receiving a broadcast text message in the room
@@ -2593,7 +2562,6 @@ export interface IEngineEvents {
 
 /** 
  * @id RTCEngine
- *     <span id="irtcengine"></span>
  * @list overview
  * @detail api
  */
@@ -2617,11 +2585,10 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @return
 	 * @notes + In the same room with the same App ID, the user ID of each user must be unique. The user will be kicked out of the room when a user with the same userID joins the room because of the error of `DUPLICATE_LOGIN`.
 	 *        + Remote users will receive [onUserJoined](104479#onuserjoined) callback when a visible user joins the room. Call [setUserVisibility](#setuservisibility) to switch the visibility of the user.
-	 *        + RTC SDK will try reconnect after losing connection to the server caused by a poor network connection. At this point, [onConnectionStateChanged](104479#onConnectionStateChanged) callback will be triggered and the SDK will automatically retry until it successfully reconnects to the server. The remote user will receive [onUserJoined](104479#onuserjoined) if a visible user reconnected successfully.
+	 *        + RTC SDK will try reconnect after losing connection to the server caused by a poor network connection. At this point, [onConnectionStateChanged](104479#onconnectionstatechanged) callback will be triggered and the SDK will automatically retry until it successfully reconnects to the server. The remote user will receive [onUserJoined](104479#onuserjoined) if a visible user reconnected successfully.
 	 * @region Room Management
 	 */
 	joinRoom: (token: string | null, roomId: string, userInfo: UserInfo, roomConfig?: RoomConfig) => Promise<void>;
-	setConnectionMode: (mode: ConnectionMode) => void;
 	
 	/** 
 	 * @brief Leave the room.
@@ -2948,10 +2915,12 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 *        When the resolution of the max stream is in [540P, 1080P], you can simulcast 3 streams; when in [270P, 450P], you can simulcast 2 streams; when in [0, 180P], you can not enable simulcast.
 	 * @notes + The default parameters are 640 × 480 px, 15 FPS, with 600 Kbps max bitrate.
 	 *        + You can call this API before and after publishing video streams. After the video stream is published, you cannot switch from single stream to multi-stream, or modify the number of published multi-stream layers.
+	 *        + When using mobile devices to publish video streams in the vertical holding state, the set resolution width and height should be the opposite of the actual sending parameters. For example, when setting the send resolution to 640 × 480 px, the actual send resolution is 480 × 640 px.
 	 *        + When simulcasting, it is recommended to cancel automatic publishing and publish audio/video streams manually by calling [publishStream](#publishstream), otherwise simulcast will not take effect.
 	 *        + If the set parameters are not supported by the browser, RTC SDK will capture and encode according to the range of parameters supported by the browser, and report the actual parameters of the capture by [onLocalVideoSizeChanged](104479#onlocalvideosizechanged).
 	 *        + You need to call [enableSimulcastMode](#enablesimulcastmode) to start simulcasting. If you set the parameters for multiple streams without calling [enableSimulcastMode](#enablesimulcastmode), only the stream with the largest resolution will be published, and the setting will take effect after simulcasting is enabled. If simulcasting is enabled, but only one stream's parameters are set, RTC SDK will automatically fill 1~2 groups of smaller stream parameters.
 	 *        + This API is applicable to the video stream captured by the camera, see [setScreenEncoderConfig](#setscreenencoderconfig) for setting parameters for screen sharing video stream.
+	 *        + If simulcasting is started, for better compatibility, it is recommended to set the ratio of the maximum level resolution to other level resolutions as an integer multiple.
 	 */
 	setVideoEncoderConfig: (descriptions: VideoEncoderConfig | VideoEncoderConfig[]) => Promise<void>;
 	
@@ -3358,10 +3327,11 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @param onAutoplayFailed Autoplay fails due to the limitation of the browser. We recommend to execute `resume` as soon as the user clicks on the UI element.
 	 * @return Resolve when the test ends.
 	 * @notes + You can call this API whether the user is in a room.
-	 * + The audio capturing will stop and begins to play the recording in 30 s after calling this API. Before that, call [stopAudioDeviceRecordAndPlayTest](#stopaudiodevicerecordandplaytest) to stop capturing and start playing the recording.
-	 * + You must stop the test before starting another test for audio devices.
-	 * + Call [stopAudioDevicePlayTest](#stopaudiodeviceplaytest) to stop the test, including capturing and playing the recording.
-	 * + This test performs locally and does not involve network connection testing.
+	 *        + You can call [setCaptureVolume]({{api}}#RTCEngine.setCaptureVolume) to adjust the capture volume during audio route testing. You cannot call [setPlaybackVolume]({{api}}#RTCEngine.setPlaybackVolume) during audio route testing.
+	 *        + The audio capturing will stop and begins to play the recording in 30 s after calling this API. Before that, call [stopAudioDeviceRecordAndPlayTest](#stopaudiodevicerecordandplaytest) to stop capturing and start playing the recording.
+	 *        + You must stop the test before starting another test for audio devices.
+	 *        + Call [stopAudioDevicePlayTest](#stopaudiodeviceplaytest) to stop the test, including capturing and playing the recording.
+	 *        + This test performs locally and does not involve network connection testing.
 	 */
 	startAudioDeviceRecordTest: (indicationInterval: number, onAutoplayFailed?: (resume: () => Promise<any>) => any) => Promise<void>;
 	
@@ -3395,10 +3365,10 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @notes + Call this API after you start the local media capture or designate a customized MediaStreamTrack.
 	 * + Callback of the MediaStreamTrack will inform you by the event of `ended` that the local media capture stops, probably caused either by an API call or the removal of devices.
 	 * + You can get the following info from the MediaStreamTrack: video resolution, device ID, and volume. You can manage the track actions by the method of `stop` or `mute`. You can also change the properties of the stream, such as audio volume.
-	 * + We recommend to call RTC APIs to enable your app with the following capabilities instead of using mediaStreamTrack.
-	 *   1. Get the video resolution or audio volume of the local streams: [onLocalStreamStats](104479#onlocalstreamstats)
-	 *   2. Get device IDs: [enumerateDevices](#enumeratedevices)
-	 *   3. stop: [stopAudioCapture](#stopaudiocapture), [stopVideoCapture](#stopvideocapture)
+	 * + We recommend to call RTC APIs to enable your app with the following capabilities instead of using mediaStreamTrack.<br/>
+	 *   &ensp; 1. Get the video resolution or audio volume of the local streams: [onLocalStreamStats](104479#onlocalstreamstats)
+	 *   &ensp; 2. Get device IDs: [enumerateDevices](#enumeratedevices)
+	 *   &ensp; 3. stop: [stopAudioCapture](#stopaudiocapture), [stopVideoCapture](#stopvideocapture)
 	 */
 	getLocalStreamTrack: (streamIndex: StreamIndex, type: "video" | "audio") => MediaStreamTrack | undefined;
 	
@@ -3411,9 +3381,9 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @notes + Call this API after the local user subscribe to the remote media stream.
 	 * + Callback of the MediaStreamTrack will inform you by the event of `ended` that the remote media stream stops, probably caused by the API call of [unsubscribeStream](#unsubscribestream) or [unsubscribeScreen](#unsubscribescreen).
 	 * + You can get video resolution and audio volume from the MediaStreamTrack. You can manage the track actions by the method of `stop` or `mute`. You can also change the properties of the stream, such as audio volume.
-	 * + We recommend to call RTC APIs to enable your app with the following capabilities instead of using mediaStreamTrack.
-	 *   1. Get the video resolution or audio volume of the remote streams: [onRemoteStreamStats](70081#onremotestreamstats)
-	 *   2. stop: [unsubscribeStream](#unsubscribestream), [unsubscribeScreen](#unsubscribescreen)
+	 * + We recommend to call RTC APIs to enable your app with the following capabilities instead of using mediaStreamTrack.<br/>
+	 *   &ensp; 1. Get the video resolution or audio volume of the remote streams: [onRemoteStreamStats]({{callback}}#onremotestreamstats)
+	 *   &ensp; 2. Unsubscribe: [unsubscribeStream](#unsubscribestream), [unsubscribeScreen](#unsubscribescreen)
 	 */
 	getRemoteStreamTrack: (userId: string, streamIndex: StreamIndex, type: "video" | "audio") => MediaStreamTrack | undefined;
 	
@@ -3446,9 +3416,10 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 * @brief Registers an extension.
 	 * @param extension The extension instance.
 	 * @return
-	 * @notes You can use the following extensions.
-	 *        • [Beauty Effects Extension](https://docs.byteplus.com/byteplus-rtc/docs/148646).
-	 *        • [AI Noise-Cancelling Extension](https://docs.byteplus.com/byteplus-rtc/docs/148647).
+	 * @notes You can use the extension to implement the following features. </br>
+	 *     &ensp; + [Beauty Effects](https://docs.byteplus.com/byteplus-rtc/docs/148646)
+	 *     &ensp; + [Virtual Background](https://docs.byteplus.com/byteplus-rtc/docs/697417)
+	 *     &ensp; + [AI Noise-Cancelling](https://docs.byteplus.com/byteplus-rtc/docs/148647)
 	 */
 	registerExtension: (extension: RTCExtension) => Promise<void>;
 	
@@ -3462,7 +3433,7 @@ export interface IRTCEngine extends EnhancedEventEmitter<IEngineEvents> {
 	 *         + `!0`: Failure.
 	 * @notes
 	 *        You must call this method after joining the room.
-	 *        You can set your source language by calling `joinRoom`  and importing a json formatted string `"SourceLanguage": "zh"` through the paramater of extraInfo. Only chinese, English and Japanese are supported.
+	 *        You can set your source language by calling `joinRoom`  and importing a json formatted string `"source_language": "zh"` through the paramater of extraInfo. Only chinese, English and Japanese are supported.
 	 *        The language of your browser will be automatically set as the source language if you don't set it.
 	 */
 	startSubtitle: (config: RTCSubtitleConfig) => Promise<void>;
@@ -3471,21 +3442,53 @@ 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.
-	 * @return  +  `0`: Success.
-	 *          + `!0`: Failure.
 	 */
 	stopSubtitle: () => void;
 	
 	/** 
 	 * @brief Set whether the remote audio and video streams are rendered synchronously.
 	 *        You can achieve ultra low latency by turning off the audio and video synchronization.
-	 * @param sync `true`: (Default) Turn on sync.
-	 *             `false`: Turn off sync.
+	 * @param sync
+	 *     &ensp; `true`: (Default) Turn on sync.
+	 *     &ensp; `false`: Turn off sync.
 	 * @return + `true`: Success.
 	 *         + `false`: Failure.
 	 * @notes You must call this API before joining the room. Calling it afterward is invalid.
 	 */
 	setRemoteStreamRenderSync: (sync: boolean) => boolean;
+	
+	/** 
+	 * @brief Adjust audio capture volume.
+	 * @param streamIndex The index of the target stream whose recording volume is to be adjusted.
+	 * @param volume The percentage of capturing volume to the original volume in range of [0, 400]. The default value is 100%.
+	 *               For better call quality, it is recommended to set the value in range of [0, 100].
+	 * @notes + It is recommended to set the capture volume after the capture has started.
+	 *        + If capture has not yet started when you call this API, the setting will take effect after it starts.
+	 *        + The capture volume adjustment controls the audio stream before audio mixing.
+	 */
+	setCaptureVolume: (streamIndex: StreamIndex, volume: number) => void;
+	
+	/** 
+	 * @brief Adjust audio playback volume of a specified remote user.
+	 * @param userId The Id of the specified remote user.
+	 * @param streamIndex The index of the target stream whose playback volume is to be adjusted.
+	 * @param volume The percentage of playback volume to the original volume in range of [0, 400]. The default value is 100%.
+	 *               For better call quality, it is recommended to set the value in range of [0, 100].
+	 * @notes + It is recommended to set the playback volume after the playing has started.
+	 *        + If the playing has not yet started when you call this API, the setting will take effect after it starts.
+	 *        + The playback volume cannot be adjusted in [Conference Mode](113547).
+	 */
+	setPlaybackVolume: (userId: string, streamIndex: StreamIndex, volume: number) => void;
+	
+	/** 
+	 * @brief Adjust audio playback volume of a public stream.
+	 * @param publicStreamId The Id of a specified public stream.
+	 * @param volume The percentage of playback volume to the original volume in range of [0, 400]. The default value is 100%.
+	 *               For better call quality, it is recommended to set the value in range of [0, 100].
+	 * @notes + It is recommended to set the playback volume after the playing has started.
+	 *        + If the playing has not yet started when you call this API, the setting will take effect after it starts.
+	 */
+	setPublicStreamVolume: (publicStreamId: string, volume: number) => void;
 }
 
 /** 
@@ -4885,10 +4888,25 @@ export interface SubtitleMessage {
 	 */
 	sequence: number;
 }
-export declare enum ConnectionMode {
-	AUTO = "auto",
-	FORCE_TCP = "forceTCP",
-	FORCE_UDP = "forceUDP"
+/** 
+ * @brief Cloud proxy informatio
+ */
+export interface CloudProxyInfo {
+	
+	/** 
+	 * @brief The cloud proxy IP of log report
+	 */
+	logProxy: string;
+	
+	/** 
+	 * @brief The cloud proxy IP of the access
+	 */
+	accessProxy: string;
+	
+	/** 
+	 * @brief The cloud proxy IP of RTC admin control
+	 */
+	configProxy: string;
 }
 
 /** 
@@ -5133,15 +5151,12 @@ declare class VERTC implements IVERTC {
 	setParameter: typeof setParameter;
 	isSupported: () => Promise<boolean>;
 	getSupportedCodecs: () => Promise<string[]>;
-	ConnectionMode: typeof ConnectionMode;
 	events: typeof EngineEventsTypes;
 	ErrorCode: typeof ErrorCode;
 	platform: string;
-	private _monitor;
-	private _rtmMonitor;
 	downloadLog: (key?: string | undefined) => void;
 	setLogConfig: ({ logLevel, LogfileSize }: {
-		logLevel?: "none" | "error" | "warning" | "info" | "trace" | undefined;
+		logLevel?: "none" | "error" | "info" | "warning" | "trace" | undefined;
 		LogfileSize?: number | undefined;
 	}) => void;
 }