aliyun-rtc-sdk 6.11.5 → 6.11.6-beta.2

package/dist/types/index.d.ts

@@ -1,12 +1,12 @@
 /// <reference types="dom-mediacapture-transform" />
 
 import { default as default_2 } from 'eventemitter3';
+import { default as default_3 } from '@aliyun-sls/web-track-browser';
 import { IAudioConstraints } from 'media-device';
 import { IProfile } from 'aliyun-rts-sdk';
 import { IStreamConfig } from 'aliyun-rts-sdk';
 import { IVideoConstraints } from 'media-device';
 import { LocalStream } from 'aliyun-rts-sdk';
-import { LogReporter } from 'browser-log-reporter';
 import { RemoteStream } from 'aliyun-rts-sdk';
 
 declare interface AliEngineCameraCapturerConfiguration {
@@ -41,7 +41,7 @@ export declare interface AliRtcAuthInfo {
     /** 令牌 */
     token: string;
     /** 会议id */
-    callid?: string;
+    sessionId?: string;
     /** 角色鉴权值 */
     role?: AliRtcSdkTokenRole;
 }
@@ -124,6 +124,10 @@ export declare class AliRtcEngine extends default_2<AliRtcEngineEventListener> {
      * @ignore
      */
     static logName: string;
+    /**
+     * @ignore
+     */
+    static logError: boolean;
     /**
      * 检查浏览器是否支持 WebRTC
      * @param {string} direction
@@ -199,11 +203,10 @@ export declare class AliRtcEngine extends default_2<AliRtcEngineEventListener> {
     protected channelProfile: AliRtcSdkChannelProfile;
     protected clientRole: AliRtcSdkClientRole;
     protected signalingManager: SignalingManager;
-    protected userManager: UserManager;
+    protected bizControl: BizControl;
     protected liveTranscoding: AliRtcLiveTranscoding;
     protected maxSignalingConnectRetryCount?: number;
-    protected reporter: Reporter;
-    protected webTrackingClient: WebTrackingClient;
+    protected slsReporter: SLSReporter;
     protected connectState: AliRtcConnectionStatus;
     protected networkError: boolean;
     private audioMixerPlugin;
@@ -245,7 +248,7 @@ export declare class AliRtcEngine extends default_2<AliRtcEngineEventListener> {
     /**
      * 监听 UM 事件
      */
-    private addUserManagerListeners;
+    private addBizControlListeners;
     /**
      * 添加信令处理
      * @returns {void}
@@ -339,7 +342,29 @@ export declare class AliRtcEngine extends default_2<AliRtcEngineEventListener> {
     private clearCheckAuthInfoTimer;
     private startDefaultDevicesCapture;
     /**
-     * @brief 加入频道
+     * @brief 加入频道(单参数)
+     * @details 该方法让用户加入频道，一个频道内可以加入多个用户进行互相通话。
+     * @details 一般在调用该方法前会根据场景先选择频道模式，通过调用 {@link setChannelProfile} 实现，默认频道模式为通信模式 {@link AliRtcSdkChannelProfile.AliRtcSdkCommunication}
+     * ``` javascript
+     * // 设置频道为互动模式
+     * engine.setChannelProfile(AliRtcSdkChannelProfile.AliRtcInteractivelive);
+     * // 设置角色为主播角色
+     * engine.setClientRole(AliRtcSdkClientRole.AliRtcSdkInteractive);
+     * ```
+     * @param {string} token    单参数认证信息，从App Server获取。
+     * @param {string} channelId    频道ID
+     * @param {string} userId    用户ID
+     * @param {string} userName    任意用于显示的用户名称。不是User ID
+     * @note
+     *    - 异步接口，方法执行成功则为成功入会
+     *    - 正常情况一个Appid对应一个App，只有同一个AppId的应用可进行互相童话,不同AppId和channelID不同都不能互通
+     *    - 当入会成功后，会触发远端用户收到 {@link AliRtcEngineEventListener.remoteUserOnLineNotify} 回调
+     *    - 如果已在会中想重新入会，需要先调用 {@link leaveChannel} 离会，且确保方法执行成功
+     *    - 用户加入频道后，默认订阅频道内所有其他用户的音视频流，默认推送音视频流到远端，因此将会产生计费，如果想要取消默认订阅可以 {@link joinChannel} 前调用 {@link setDefaultSubscribeAllRemoteAudioStreams} 和 {@link setDefaultSubscribeAllRemoteVideoStreams}
+     */
+    joinChannel(token: string, channelId: string, userId: string, userName: string): Promise<void>;
+    /**
+     * @brief 加入频道(多参数)
      * @details 该方法让用户加入频道，一个频道内可以加入多个用户进行互相通话。
      * @details 一般在调用该方法前会根据场景先选择频道模式，通过调用 {@link setChannelProfile} 实现，默认频道模式为通信模式 {@link AliRtcSdkChannelProfile.AliRtcSdkCommunication}
      * ``` javascript
@@ -636,6 +661,19 @@ export declare class AliRtcEngine extends default_2<AliRtcEngineEventListener> {
      * - >100：放大音量
      */
     setRecordingVolume(volume: number): void;
+    /**
+     * @brief 设置播放音量
+     * @param volume 播放音量，取值范围[0,100], 0：静音；100：原始音量
+     *   @note 注意此方法与 {@link setRemoteAudioVolume} 相互影响，以后调用的一个为准
+     */
+    setPlayoutVolume(volume: number): void;
+    /**
+     * @brief 设置播放音量
+     * @param userId 用户ID
+     * @param volume 播放音量，取值范围[0,100], 0：静音；100：原始音量
+     * @note 注意此方法与 {@link setPlayoutVolume} 相互影响，以后调用的一个为准
+     */
+    setRemoteAudioVolume(userId: string, volume: number): void;
     /**
      * 设置麦克风 profile
      *
@@ -659,9 +697,12 @@ export declare class AliRtcEngine extends default_2<AliRtcEngineEventListener> {
     /**
      * @brief 采集偏好设置
      * @param cameraCapturerConfiguration 偏好设置
-     *      - videoProfile: videoProfileKey 或 IProfile
-     *      - cameraDirection: 设置采集方向，前置或后置摄像头
-     *      - deviceId: 设置采集相机ID
+     * - width: 设置采集分辨率宽度 默认 1280
+     * - height: 设置采集分辨率高度 默认 720
+     * - frameRate: 设置采集帧率 默认 30
+     * - bitrate: 设置采集码率 默认 2000
+     * - cameraDirection: 设置采集方向，前置或后置摄像头
+     * - deviceId: 设置采集相机ID
      */
     setCameraCapturerConfiguration(cameraCapturerConfiguration: AliEngineCameraCapturerConfiguration): Promise<void>;
     /**
@@ -1124,6 +1165,11 @@ declare interface AliRtcEngineEventListener {
      * @note 应该在收到该回调时弹出弹窗引导用户点击页面进行播放，用户点击页面后 SDK 会自动尝试播放
      */
     remoteVideoAutoPlayFail: (uid: string, videoTrack: AliRtcVideoTrack) => void;
+    /**
+     * rtc 传输数据统计
+     * @param stats 统计信息
+     */
+    rtcStats: (stats: any) => void;
     /**
      * 发布视频流的数据统计
      * @param aliRtcStats 统计信息
@@ -1437,6 +1483,8 @@ export declare enum AliRtcErrorCode {
 
 declare class AliRtcLiveTranscoding {
     authInfo?: AliRtcAuthInfo;
+    private slsReporter;
+    constructor(slsReporter: SLSReporter);
     private doRequest;
     setAuthInfo(authInfo: AliRtcAuthInfo): void;
     clear(): void;
@@ -1747,6 +1795,7 @@ export declare abstract class AliRtcPlugin extends default_2<AliRtcPluginListene
     lastOutputAudioTrack?: MediaStreamTrack;
     lastInputVideoTrack?: MediaStreamTrack;
     lastOutputVideoTrack?: MediaStreamTrack;
+    private _hasCleared;
     constructor(name: string, streamType?: AliRtcRawDataStreamType, trackType?: AliRtcPluginTrackType);
     get initOptions(): {};
     getOptions(): any;
@@ -2058,6 +2107,182 @@ declare enum AudioStreamSource {
     Mixed = 2
 }
 
+declare class BizControl extends default_2<BizControlListener> {
+    /**
+     * @ignore
+     */
+    static logName: string;
+    protected self: LocalUser | undefined;
+    protected userMap: RemoteUserMap;
+    private usersViewMap;
+    rtsManager: RtsManager;
+    protected localStreamManager: LocalStreamManager;
+    protected authInfo?: AliRtcAuthInfo;
+    protected signalingManager: SignalingManager;
+    pluginManager: PluginManager;
+    protected slsReporter: SLSReporter;
+    private statsMonitorId?;
+    private autoSubAudio;
+    private autoSubVideo;
+    private autoSubScreen;
+    private wantSubAudio;
+    private wantSubVideo;
+    private wantSubScreen;
+    private defaultStreamType;
+    private audioMuted;
+    private playoutVolume;
+    private audioVolumeIndicationInterval;
+    private indicationTimer;
+    constructor(config: BizControlConfig);
+    private addSignalingManagerListener;
+    isAutoSubAudio(): boolean;
+    setAutoSubAudio(value: boolean): void;
+    isAutoSubVideo(): boolean;
+    setAutoSubVideo(value: boolean): void;
+    isAutoSubScreen(): boolean;
+    setAutoSubScreen(value: boolean): void;
+    isWantSubAudio(): boolean;
+    setWantSubAudio(value: boolean): void;
+    isWantSubVideo(): boolean;
+    setWantSubVideo(value: boolean): void;
+    isWantSubScreen(): boolean;
+    setWantSubScreen(value: boolean): void;
+    setRemoteDefaultVideoStreamType(type: AliRtcVideoStreamType): void;
+    setRemoteUserVideoStreamType(userId: string, type: AliRtcVideoStreamType): void;
+    setAudioMuted(mute: boolean): void;
+    setLocalViewConfig(view: AliRtcLocalView, track: AliRtcVideoTrack): void;
+    setRemoteViewConfig(uid: string, view: AliRtcLocalView, track: AliRtcVideoTrack): void;
+    get channel(): string | undefined;
+    get userId(): string | undefined;
+    get remoteUsers(): RemoteUser[];
+    /**
+     * 获取推流对象，只能是自己
+     */
+    get publisher(): LocalUser | undefined;
+    get isSubscribing(): boolean;
+    /**
+     * 处理远端用户推流状态变化
+     * @param {string} userId
+     * @param {string} displayname
+     * @param {string} callid
+     * @param {string} pullStreamUrl
+     * @param {TrackInfo[]} tracks
+     * @returns {void}
+     */
+    onRemotePublishStateChange(publishInfo: PublishUser): void;
+    /**
+     * 登陆成功后初始化用户管理，创建自己
+     * @param {AliRtcAuthInfo} authInfo
+     * @param {string} pushStreamUrl
+     * @returns {Promise<void>}
+     */
+    init(authInfo: AliRtcAuthInfo, data: JoinResult): void;
+    /**
+     * 离会后清除所有数据
+     * @returns {Promise<void>}
+     */
+    clear(): Promise<void>;
+    private startStatsMonitor;
+    private stopStatsMonitor;
+    /**
+     * 添加远端用户
+     * @param {string} userId
+     * @param {string} displayName
+     * @returns {void}
+     */
+    addUser(userId: string, displayName: string, reason: SubscribeReason): RemoteUser;
+    /**
+     * 执行添加远端用户
+     * @param {string} userId
+     * @param {string} displayName
+     * @returns {RemoteUser}
+     */
+    private executeAddUser;
+    /**
+     * 远端用户设备状态变化
+     * @param {DeviceStatusChangeType} event
+     * @param {string} userId
+     */
+    private onRemoteDeviceStatusChange;
+    /**
+     * 远端流数据采集上报
+     * @param {any} stat
+     */
+    private onRemoteStatReport;
+    /**
+     * rts 单 PC 订阅超限
+     */
+    private onSubscribeExceeds;
+    /**
+     * 移除远端用户
+     * @param {LeaveInfo} user
+     * @returns {Promise<void>}
+     */
+    removeLeftUser(user: LeaveInfo): Promise<void>;
+    /**
+     * 更新远端用户数据
+     * @param {JoinInfo[]} users
+     * @returns {{joinList: JoinInfo[], leaveList: LeaveInfo[], presentList: JoinInfo[]}}
+     */
+    updateUsers(users: JoinInfo[]): {
+        joinList: JoinInfo[];
+        leaveList: LeaveInfo[];
+        presentList: JoinInfo[];
+    };
+    /**
+     * 根据 userId 获取 RemoteUser
+     * @param {string} userId
+     * @returns {RemoteUser | undefined}
+     */
+    getRemoteUser(userId: string): RemoteUser | undefined;
+    /**
+     * 更新远端用户设备信息
+     * @param {RemoteUserStatus} user
+     * @returns {boolean} true: 用户状态改变  false: 用户状态没有改变
+     */
+    updateRemoteUserDeviceStatus(user: RemoteUserStatus): void;
+    updateAudioWant(uid: string, sub: boolean): void;
+    updateVideoWant(uid: string, sub: boolean): void;
+    updateScreenWant(uid: string, sub: boolean): void;
+    private stopIndication;
+    private startIndication;
+    enableAudioVolumeIndication(interval: number): void;
+    setPlayoutVolume(volume: number): void;
+    refreshAuthInfo(authInfo: AliRtcRefreshAuthInfo): void;
+    setEnableMediaExtensionMsg(enable: boolean): void;
+    setAudioRedEnabled(enable: boolean): void;
+}
+
+declare interface BizControlConfig {
+    slsReporter: SLSReporter;
+    signalingManager: SignalingManager;
+}
+
+declare interface BizControlListener {
+    [ClientEventType.RtsSubscribeExceeds]: (userId: string, options?: SubscribeOptions) => void;
+    [DeviceStatusChange.Remote]: (event: DeviceStatusChangeType, userId: string) => void;
+    [ClientEventType.PullStreamStats]: (userId: string, stat: any, pullStreamUrl: string) => void;
+    remoteTrackAvailableChange: (uid: string, aliRtcAudioTrack: AliRtcAudioTrack, aliRtcVideoTrack: AliRtcVideoTrack) => void;
+    audioSubscribeStateChanged: (uid: string, oldState: AliRtcSubscribeState, newState: AliRtcSubscribeState, elapseSinceLastState: number, channel: string) => void;
+    videoSubscribeStateChanged: (uid: string, oldState: AliRtcSubscribeState, newState: AliRtcSubscribeState, elapseSinceLastState: number, channel: string) => void;
+    screenSubscribeStateChanged: (uid: string, oldState: AliRtcSubscribeState, newState: AliRtcSubscribeState, elapseSinceLastState: number, channel: string) => void;
+    userAudioMuted: (uid: string, isMute: boolean) => void;
+    userVideoMuted: (uid: string, isMute: boolean) => void;
+    userScreenMuted: (userId: string, isMute: boolean) => void;
+    audioVolume: (speakers: {
+        userId: string;
+        volume: number;
+    }[]) => void;
+    rtcLocalVideoStats: (stats: AliRtcLocalVideoStats[]) => void;
+    rtcLocalAudioStats: (stats: AliRtcLocalAudioStats[]) => void;
+    rtcRemoteVideoStats: (stats: AliRtcRemoteVideoStats[]) => void;
+    rtcRemoteAudioStats: (stats: AliRtcRemoteAudioStats[]) => void;
+    rtcStats: (stats: any) => void;
+    remoteAudioAutoPlayFail: (uid: string) => void;
+    remoteVideoAutoPlayFail: (uid: string, track: AliRtcVideoTrack) => void;
+    remoteSubscribeError: (error: AliRtcError) => void;
+}
+
 declare enum ClientEventType {
     PullStreamStats = "remoteStat",
     RtsSubscribeExceeds = "rtsSubscribeExceeds",
@@ -2114,16 +2339,6 @@ declare enum DisplayType {
     WHEN_NO_VIDEO = 2
 }
 
-declare enum EventId {
-    JoinChannel = 70012,
-    JoinChannelRes = 70013,
-    LeaveChannel = 70015,
-    EnvCheck = 100001,
-    Push = 10030,
-    Pull = 10050,
-    SubscribeStats = 10080
-}
-
 declare interface ISignalingInterface {
     join(authInfo: AliRtcAuthInfo, userName: string, connectRetryCount?: number): Promise<JoinResult>;
     leave(): Promise<SignalingResult>;
@@ -2195,6 +2410,7 @@ declare class LocalStreamManager extends default_2<LocalStreamManagerListener> {
      */
     static logName: string;
     private pluginManager;
+    private slsReporter;
     private publishingCameraVideoStream;
     private publishingScreenVideoStream;
     cameraStreamInfo: AliRtcLocalStreamInfo;
@@ -2205,7 +2421,7 @@ declare class LocalStreamManager extends default_2<LocalStreamManagerListener> {
     private _publishLocalAudioStreamEnabled;
     private _publishLocalVideoStreamEnabled;
     private _publishLocalScreenStreamEnabled;
-    constructor(pluginManager: PluginManager);
+    constructor(pluginManager: PluginManager, slsReporter: SLSReporter);
     get hasCamera(): boolean;
     get audioCaptureDisabled(): boolean;
     set audioCaptureDisabled(closed: boolean);
@@ -2298,6 +2514,7 @@ declare class LocalUser extends User {
      */
     static logName: string;
     protected url: string;
+    private joinTime;
     private _isPublishing;
     private publishPromise;
     private previewElements;
@@ -2308,8 +2525,11 @@ declare class LocalUser extends User {
     protected signalingManager: SignalingManager;
     traceId: string;
     private monitorTimerId?;
-    protected reporter: Reporter;
-    protected webTrackingClient: WebTrackingClient;
+    private monitorDataCache;
+    private trackTraceIdMap;
+    private candidateDataCache;
+    private pcMediaStatsMap;
+    protected slsReporter: SLSReporter;
     private audioPubState;
     private videoPubState;
     private videoSmallPubState;
@@ -2403,12 +2623,15 @@ declare class LocalUser extends User {
     createLocalStream(params: LocalStreamConfig): Promise<LocalStream | undefined>;
     stopLocalStream(options: PublishOptions): Promise<void>;
     private compareAndPublish;
+    private reportAudioProfile;
+    private reportVideoProfile;
+    private reportScreenProfile;
     /**
      * 开始推流
      * @param isResume 是否是恢复推流
      * @returns
      */
-    startPublish(isResume?: boolean): Promise<void>;
+    startPublish(isResume?: boolean, reason?: PublishReason): Promise<void>;
     /**
      * 对比本地流和正在推流，进行推流更新
      * @param isResume 是否是恢复推流
@@ -2456,6 +2679,10 @@ declare class LocalUser extends User {
     private onTrackEnd;
     private resetPreviewElements;
     private resetSecondayPreviewElements;
+    getMediaStats(id: string): any;
+    private reportPublishMonitor;
+    private reportNetwork;
+    private startStreamMonitor;
     /**
      * 停止本地流数据采集
      */
@@ -2480,14 +2707,14 @@ declare class LocalUser extends User {
     getStats(): Promise<{
         audio: AliRtcLocalAudioStats[];
         video: AliRtcLocalVideoStats[];
+        transport: any;
     }>;
 }
 
 declare interface LocalUserConfig {
     rtsManager: RtsManager;
     localStreamManager: LocalStreamManager;
-    reporter: Reporter;
-    webTrackingClient: WebTrackingClient;
+    slsReporter: SLSReporter;
     signalingManager: SignalingManager;
     pluginManager: PluginManager;
     audioVolumeIndicationInterval: number;
@@ -2498,12 +2725,16 @@ declare interface LocalUserConfig {
  * 在入会之前，会将埋点缓存，入会后以每200ms一条的速度将缓存的埋点按顺序上报
  */
 declare class LogClient {
-    protected reporter?: LogReporter;
+    protected tracker?: default_3;
     protected param?: any;
     protected msgCacheArr: any[];
     protected index: number;
-    constructor();
+    protected stsOpt?: any;
+    protected slsToken?: SLSSTSToken;
+    protected ossToken?: OSSSTSToken;
+    private createTracker;
     start(info: AliRtcAuthInfo): void;
+    updateToken(slsToken: SLSSTSToken, ossToken?: OSSSTSToken): void;
     /**
      * 断开连接
      */
@@ -2517,11 +2748,10 @@ declare class LogClient {
      * 初始化公共参数
      */
     protected initParam(info: AliRtcAuthInfo): void;
-    /**
-     * 通过LogReporter发送日志
-     * @param log
-     */
-    protected realSendReport(log: any): void;
+}
+
+declare interface LogInfo {
+    [key: string]: string | number;
 }
 
 declare interface MediaTrackInfo {
@@ -2551,6 +2781,24 @@ declare enum MsidType {
     Screen = "sophon_video_screen_share"
 }
 
+declare interface OSSSTSToken {
+    access_key_id: string;
+    access_key_secret: string;
+    security_token: string;
+    region_endpoint: string;
+    bucket_name: string;
+    expiration: number;
+}
+
+declare interface OSSToken {
+    access_key_id: string;
+    access_key_secret: string;
+    security_token: string;
+    region_endpoint: string;
+    bucket_name: string;
+    expiration: number;
+}
+
 declare class PluginManager extends default_2<PluginManagerListener> {
     private plugins;
     add(plugin: AliRtcPlugin, options: any): void;
@@ -2585,6 +2833,7 @@ declare interface PublishExtraParams {
 }
 
 declare interface PublishInfo {
+    type: PublishType;
     userid: string;
     sessionid: string;
     displayname: string;
@@ -2604,6 +2853,11 @@ declare type PublishOptions = StreamOptions & {
     vMsid?: string;
 };
 
+declare enum PublishReason {
+    MANUAL = 0,
+    JOINCHANNEL = 1
+}
+
 declare interface PublishStreamInfo {
     audio?: LocalStream;
     video?: LocalStream;
@@ -2613,6 +2867,14 @@ declare interface PublishStreamInfo {
     screenProfile?: LocalProfileManager;
 }
 
+declare enum PublishType {
+    Add = "add",
+    Delete = "delete",
+    Replace = "replace",
+    Stop = "stop",
+    UnPublish = "unpublish"
+}
+
 declare interface PublishUser {
     callid: string;
     clientrole?: string;
@@ -2679,6 +2941,7 @@ declare class RemoteStreamInfo extends default_2<StreamListener> {
      * @returns
      */
     getScreenInfo(): RemoteMediaTrackInfo;
+    get subscribingInfos(): RemoteMediaTrackInfo[];
     /**
      * 远端流是否包含音频轨道
      * @returns
@@ -2756,7 +3019,7 @@ declare class RemoteUser extends User {
     static logName: string;
     remoteCallid: string;
     remoteUserInfo: AliRtcRemoteUserInfo;
-    protected localUserId: string;
+    protected localUser?: LocalUser;
     protected lastPubId: string;
     protected stream?: RemoteStream;
     protected screenStream?: RemoteStream;
@@ -2768,13 +3031,13 @@ declare class RemoteUser extends User {
     protected secondaryMediaStream?: MediaStream;
     protected prevSubConfig?: SubConfig;
     protected signalingManager: SignalingManager;
-    protected reporter: Reporter;
-    protected webTrackingClient: WebTrackingClient;
+    protected slsReporter: SLSReporter;
     private wantSubAudio;
     private wantSubVideo;
     private wantSubScreen;
     private defaultVideoStreamType;
     private audioMuted;
+    private playoutVolume;
     private audioSubState;
     private videoSubState;
     private videoLargeSubState;
@@ -2782,6 +3045,9 @@ declare class RemoteUser extends User {
     private screenSubState;
     streamInfo: RemoteStreamInfo;
     streamUrl: string;
+    private monitorTimerId?;
+    private monitorDataCache;
+    private trackTraceIdMap;
     constructor(config: RemoteUserConfig);
     isWantSubAudio(): boolean;
     setWantSubAudio(value: boolean): void;
@@ -2791,6 +3057,7 @@ declare class RemoteUser extends User {
     setWantSubScreen(value: boolean): void;
     setRemoteDefaultVideoStreamType(type: AliRtcVideoStreamType): void;
     setAudioMuted(value: boolean): void;
+    setPlayoutVolume(value: number): void;
     getAudioMuted(): boolean;
     get hasAudioTrack(): boolean;
     get hasVideoTrack(): boolean;
@@ -2833,7 +3100,7 @@ declare class RemoteUser extends User {
      * @param {SubscribeOptions} options
      * @returns {Promise<void>} 返回可播放的 MediaStream，可能为空。如果同时订阅了主流和辅流，只会返回主流对应的 MediaStream
      */
-    subscribe(options?: SubscribeOptions): Promise<void>;
+    subscribe(options?: SubscribeOptions, reason?: SubscribeReason, startTs?: number): Promise<void>;
     /**
      * 订阅远端流
      * @param {SubscribeOptions} options 订阅配置
@@ -2881,7 +3148,7 @@ declare class RemoteUser extends User {
      * 尝试更新订阅
      * 1. 需要新增订阅的情况：想要订阅 & 没有正在订阅 & 远端有流
      */
-    updateSubscribe(): {
+    updateSubscribe(reason: SubscribeReason): {
         addOption: SubscribeOptions;
         deleteOption: SubscribeOptions;
     };
@@ -2922,6 +3189,12 @@ declare class RemoteUser extends User {
      * @param {number} interval 时间间隔
      */
     enableAudioVolumeIndication(interval: number): void;
+    private reportSubscribeMonitor;
+    private startStreamMonitor;
+    /**
+     * 停止本地流数据采集
+     */
+    private stopStreamMonitor;
 }
 
 declare interface RemoteUserConfig {
@@ -2931,11 +3204,11 @@ declare interface RemoteUserConfig {
     usersViewMap: {
         [key: string]: RemoteUserViewMap;
     };
-    reporter: Reporter;
-    webTrackingClient: WebTrackingClient;
+    slsReporter: SLSReporter;
     signalingManager: SignalingManager;
-    localUserId: string;
+    localUser?: LocalUser;
     audioVolumeIndicationInterval: number;
+    playoutVolume: number;
 }
 
 declare interface RemoteUserMap {
@@ -2953,222 +3226,52 @@ declare interface RemoteUserViewMap {
 }
 
 /**
- * 日志埋点类
- * 每个埋点暴露对应的接口
+ * RoomServerMessageCenter支持的事件
  */
-declare class Reporter {
-    protected logClient: LogClient;
-    constructor();
-    /**
-     * 开始建立连接
-     * @param authInfo
-     */
-    start(info: AliRtcAuthInfo): void;
-    /**
-     * 断开连接
-     */
-    stop(): void;
-    /**
-     * 加入房间成功埋点
-     * @param {number} joinTime
-     * @param {number} result
-     * @param {string} tid
-     */
-    reportJoin(joinTime: number, result: number, tid: string): void;
+declare interface RoomServerListener {
+    onJoinChannelResult: (data: any) => void;
+    onPublishResult: (code: number, tid: string, description: string) => void;
+    onNotifyJoin: (users: JoinInfo[]) => void;
+    onNotifyStatus: (users: RemoteUserStatus[]) => void;
+    onLeaveResult: (code: number, tid: string) => void;
+    onRoleUpdateResult: (code: number, tid: string, description: string) => void;
+    onRefreshUrlResult: (data: any) => void;
+    onNotifyLeave: (users: LeaveInfo[]) => void;
+    onNotifyPublish: (users: PublishUser[]) => void;
+    onStatusReport: (code: number, tid: string, description: string) => void;
+    onBye: (reason: AliRtcOnByeType, description: string) => void;
+    onKeepAliveRsp: () => void;
+    onConnectFail: (err: AliRtcError) => void;
+    onNetworkError: (err: AliRtcError) => void;
+    onAuthInvalid: () => void;
+    onReconnectStart: () => void;
+    onReconnectFail: (err: AliRtcError) => void;
+    onReconnectSuccess: (users: JoinInfo[]) => void;
+    onError: (err: AliRtcError) => void;
+}
+
+declare class RtsManager extends default_2<RtsManagerEventListener> {
     /**
-     * 调用 client.join 入会上报事件及用户等信息
-     * @param {string} usrid 用户id
-     * @param {string} usrn 用户名
-     * @param {string} channelid 频道id
-     */
-    reportJoinInvoked(usrid: string, usrn: string, channelid: string): void;
-    /**
-     * 上报ice状态变化
-     * @param {string} calid
-     * @param {string} icestate
-     * @param {number} evt_seq
-     */
-    reportIceState(calid: string, icestate: string, evt_seq: number): void;
-    /**
-     * 用户离开频道事件埋点
-     * @param {number} leaveTime
-     * @param {number} result
-     * @param {string} tid
-     */
-    reportLeave(leaveTime: number, result: number, tid: string): void;
-    /**
-     * 调用 client.leave 离会上报事件
-     */
-    reportLeaveInvoked(): void;
-    /**
-     * 推流事件上报
-     * @param {string} callid
-     * @param {boolean} isLargeVideo
-     * @param {boolean} isSmallVideo
-     * @param {boolean} isScreenShare
-     * @param {boolean} isAudio
-     * @param {number} result
-     * @param {string} tid
-     * @param {number} pubTime
-     */
-    reportPublish(callid: string, isLargeVideo: boolean, isSmallVideo: boolean, isScreenShare: boolean, isAudio: boolean, result: number | undefined, tid: string, pubTime: number): void;
-    /**
-     * 调用 client.publish 推流上报事件及流轨道等信息
-     * @param {boolean} [isLargeVideo] 是否推视频大流
-     * @param {boolean} [isSmallVideo] 是否推视频小流
-     * @param {boolean} [isScreenShare] 是否推屏幕流
-     * @param {boolean} [isAudio] 是否推音频流
-     */
-    reportPublishInvoked(isLargeVideo?: boolean, isSmallVideo?: boolean, isScreenShare?: boolean, isAudio?: boolean): void;
-    /**
-     * republish事件上报
-     * @param callid
-     * @param islarge       是否推大流
-     * @param isScreenShare 是否推屏幕流
-     * @param isAudio       是否推音频流
-     * @param result        推流结果(0成功，非0失败)
-     * @param tid           tid
-     * @param repubTime     repub耗时
-     * @param msg           详情描述
-     */
-    reportRepublish(callid: string, isLarge: boolean, isScreen: boolean, isAudio: boolean, result: number, tid: string, repubTime: number, msg: string): void;
-    /**
-     * 用户调用LocalStream.addTrack 或者 LocalStream.removeTrack
-     * @param callid
-     * @param trackType
-     * @param state       -1: remove 0: replace 1: add
-     */
-    reportRepublishInvoked(callid: string, trackType: string, state: 0 | 1 | -1): void;
-    /**
-     * 停止推流事件上报
-     * @param {string} callid
-     * @param {number} unpubTime
-     * @param {number} result
-     * @param {string} tid
-     */
-    reportUnpublish(callid: string, unpubTime: number, result: number, tid: string): void;
-    /**
-     * 调用 client.unpublish 停止推流上报事件
-     */
-    reportUnpublishInvoked(): void;
-    /**
-     * 订阅事件埋点
-     * @param {string} callid
-     * @param {string} remoteid
-     * @param {boolean} isLargeVideo
-     * @param {boolean} isSmallVideo
-     * @param {boolean} isScreenShare
-     * @param {boolean} isAudio
-     * @param {number} result
-     * @param {string} tid
-     * @param {number} subTime
-     */
-    reportSubscribe(callid: string, remoteid: string, isLargeVideo: boolean, isSmallVideo: boolean, isScreenShare: boolean, isAudio: boolean, result: number, tid: string, subTime: number, usrid: string): void;
-    /**
-     * 调用 client.subscribe 拉流上报事件及用户信息、流轨道等信息
-     * @param {string} rmtid 远程流所属的用户id
-     * @param {boolean} [isLargeVideo] 是否推视频大流
-     * @param {boolean} [isSmallVideo] 是否推视频小流
-     * @param {boolean} [isScreenShare] 是否推屏幕流
-     * @param {boolean} [isAudio] 是否推音频流
-     */
-    reportSubscribeInvoked(rmtid?: string, isLargeVideo?: boolean, isSmallVideo?: boolean, isScreenShare?: boolean, isAudio?: boolean): void;
-    /**
-     * 停止订阅事件埋点
-     * @param {string} callid
-     * @param {string} remoteid
-     * @param {number} unsubTime
-     * @param {number} result
-     * @param {string} tid
-     */
-    reportUnsubscribe(callid: string, remoteid: string, unsubTime: number, result: number, tid: string): void;
-    /**
-     * 调用 client.unsubscribe 停止拉流上报事件
-     * @param {string} rmtid 远程流所属的用户id
-     */
-    reportUnsubscribeInvoked(rmtid?: string): void;
-    /**
-     * 更新角色埋点
-     * @param {string|undefined} or 旧的角色
-     * @param {string} nr 新的角色
-     * @param {number} ctm 耗时
-     * @param {number} rslt 是否成功
-     */
-    reportRoleUpdate(or: string | undefined, nr: string, ctm: number, rslt: number): void;
-    /**
-     * 调用 client.setClientRole 更新角色上报事件
-     * @param {string} nr 新角色值
-     */
-    reportRoleUpdateInvoked(nr: string): void;
-    /**
-     * 设置camera videoprofile
-     */
-    reportVideoProfile(profile: string): void;
-    /**
-     * 设置screenshare videoprofile
-     */
-    reportScreenProfile(profile: string): void;
-    /**
-     * 错误事件埋点
-     * @param err
-     */
-    reportError(err: any): void;
-    /**
-     * 日志埋点
-     * 在埋点中增加tm字段，表示发生埋点的客户端本地时间
-     * @param {any} param 埋点内容
-     */
-    protected log(param: any): void;
-    /**
-     * 将Object转换成string
-     * @param {any} data
-     * @returns {string}
-     */
-    protected data2String(data: any): string;
-}
-
-/**
- * RoomServerMessageCenter支持的事件
- */
-declare interface RoomServerListener {
-    onJoinChannelResult: (data: any) => void;
-    onPublishResult: (code: number, tid: string, description: string) => void;
-    onNotifyJoin: (users: JoinInfo[]) => void;
-    onNotifyStatus: (users: RemoteUserStatus[]) => void;
-    onLeaveResult: (code: number, tid: string) => void;
-    onRoleUpdateResult: (code: number, tid: string, description: string) => void;
-    onRefreshUrlResult: (data: any) => void;
-    onNotifyLeave: (users: LeaveInfo[]) => void;
-    onNotifyPublish: (users: PublishUser[]) => void;
-    onStatusReport: (code: number, tid: string, description: string) => void;
-    onBye: (reason: AliRtcOnByeType, description: string) => void;
-    onKeepAliveRsp: () => void;
-    onConnectFail: (err: AliRtcError) => void;
-    onNetworkError: (err: AliRtcError) => void;
-    onAuthInvalid: () => void;
-    onReconnectStart: () => void;
-    onReconnectFail: (err: AliRtcError) => void;
-    onReconnectSuccess: (users: JoinInfo[]) => void;
-    onError: (err: AliRtcError) => void;
-}
-
-declare class RtsManager extends default_2<RtsManagerEventListener> {
-    /**
-     * @ignore
+     * @ignore
      */
     static logName: string;
     private rts;
     private encodedInsertableStreams;
     private audioRedEnabled;
     private localStreamManager;
+    private slsReporter;
     private connecting;
     private connected;
+    private _pcTraceId;
+    private _localJoinTime;
     private connectionResolve?;
     connectingPromise?: Promise<unknown>;
     private dcResolve?;
     dcConnectingPromise?: Promise<unknown>;
     private _publishingTracks;
-    constructor(localStreamManager: LocalStreamManager);
+    constructor(localStreamManager: LocalStreamManager, slsReporter: SLSReporter);
+    get pcTraceId(): string;
+    set localJoinTime(time: number);
     setEncodedInsertableStreams(enable: boolean): void;
     setAudioRedEnabled(enable: boolean): void;
     destroy(): void;
@@ -3181,10 +3284,9 @@ declare class RtsManager extends default_2<RtsManagerEventListener> {
     private updatePublishingTracks;
     private httpPublish;
     private waitPublishSender;
-    private checkSubscribeStats;
     getPublishVideoStats(streamUrl: string, msid?: string): Promise<unknown>;
     getPublishAudioStats(streamUrl: string, msid?: string): Promise<unknown>;
-    publishAdd(streamUrl: string, stream: LocalStream, extra?: PublishExtraParams, isResume?: boolean, retryTimes?: number, waitSender?: boolean): Promise<string>;
+    publishAdd(streamUrl: string, stream: LocalStream, callId: string, extra?: PublishExtraParams, isResume?: boolean, retryTimes?: number, waitSender?: boolean, reason?: PublishReason): Promise<string>;
     publishDelete(streamUrl: string, options: PublishOptions): Promise<string>;
     publishReplace(streamUrl: string, stream: LocalStream, extra?: PublishExtraParams): Promise<string>;
     publishStop(streamUrl: string): Promise<string>;
@@ -3202,17 +3304,12 @@ declare class RtsManager extends default_2<RtsManagerEventListener> {
         stream: any;
     } | undefined>;
     subscibeStop(streamUrl: string): Promise<void>;
-    publish(streamUrl: string, isResume?: boolean): Promise<string>;
-    subscribe(remoteSubscribeOptions: RemoteSubscribeOptions): Promise<{
-        traceId: string;
-        stream: RemoteStream;
-    } | {
-        url: string;
-        stream: RemoteStream;
-        traceId?: string | undefined;
-    } | null>;
+    publish(streamUrl: string, callId: string, isResume?: boolean, reason?: PublishReason): Promise<string>;
+    subscribe(remoteSubscribeOptions: RemoteSubscribeOptions, reason: SubscribeReason, callId: string, remoteCallId: string, startTs: number): Promise<any>;
     getSubscribeVideoStats(streamUrl: string, msid?: string): Promise<unknown>;
     getSubscribeAudioStats(streamUrl: string, msid?: string): Promise<unknown>;
+    getStreamByMsid(config: any): LocalStream | RemoteStream;
+    getPCStats(): Promise<any>;
     sendSEI(streamUrl: string, data: ArrayBuffer, repeatCount: number, payloadType: number): Promise<void>;
 }
 
@@ -3232,14 +3329,16 @@ declare class SignalingManager extends default_2<RoomServerListener> {
     protected clientRole: AliRtcSdkClientRole;
     protected env: AliRtcEnv;
     protected maxConnectRetryCount?: number;
-    constructor(channelProfile: AliRtcSdkChannelProfile, clientRole: AliRtcSdkClientRole, env?: AliRtcEnv);
+    stsManager: StsManager;
+    private slsReporter;
+    constructor(channelProfile: AliRtcSdkChannelProfile, clientRole: AliRtcSdkClientRole, slsReporter: SLSReporter, env?: AliRtcEnv);
     reset(): void;
     /**
      * 向业务信令发送入会
      * @param {AliRtcAuthInfo} authInfo
      * @returns
      */
-    join(authInfo: AliRtcAuthInfo, userName?: string, maxConnectRetryCount?: number): Promise<JoinResult>;
+    join(authInfo: AliRtcAuthInfo, userName?: string, maxConnectRetryCount?: number, timeRecorder?: TimeRecorder): Promise<JoinResult>;
     retryJoin(): void;
     /**
      * 向业务信令发送离会
@@ -3348,6 +3447,186 @@ declare interface SignalingResult {
     message?: string;
 }
 
+/**
+ * 日志埋点类
+ * 每个埋点暴露对应的接口
+ */
+declare class SLSReporter {
+    private engine;
+    protected logClient: LogClient;
+    protected authInfo?: AliRtcAuthInfo;
+    private static staticClient;
+    private static getLogClient;
+    static reportOSSUpload(sessionId: string, date: string, responseCode?: number): void;
+    constructor(engine: AliRtcEngine);
+    /**
+     * 开始建立连接
+     * @param authInfo
+     */
+    start(info: AliRtcAuthInfo): void;
+    updateToken(token: SLSSTSToken, ossToken?: OSSSTSToken): void;
+    /**
+     * 断开连接
+     */
+    stop(): void;
+    reportPublishMonitor(callId: string, traceId: string, msid: string, track: MediaStreamTrack | undefined, stats: any[]): void;
+    reportSubscribeMonitor(callId: string, remoteId: string, traceId: string, msid: string, stats: any[]): void;
+    reportNetworkMonitor(candidates: any[]): void;
+    /**
+     * 加入房间成功埋点
+     * @param {number} joinTime
+     * @param {number} result
+     * @param {string} tid
+     */
+    reportJoin(result: number, timeRecord: TimeRecorder, tid?: string): void;
+    /**
+     * Join 链接细节
+     */
+    reportJoinConnection(logInfo: LogInfo): void;
+    /**
+     * 用户离开频道事件埋点
+     * @param {number} leaveTime
+     * @param {number} result
+     * @param {string} tid
+     */
+    reportLeave(leaveTime: number, result: number, tid: string): void;
+    /**
+     * 推流事件上报
+     * @param {string} callid
+     * @param {boolean} isLargeVideo
+     * @param {boolean} isSmallVideo
+     * @param {boolean} isScreenShare
+     * @param {boolean} isAudio
+     * @param {number} result
+     * @param {string} tid
+     * @param {number} pubTime
+     */
+    reportPublish(isRepublish: boolean, url: string, callid: string, isLargeVideo: boolean, isSmallVideo: boolean, isScreenShare: boolean, isAudio: boolean, result: number | undefined, pcTraceId: string, traceId: string, startTime: number): void;
+    /**
+     * 首包上报
+     * @param {string} callid
+     * @param {number} startTime
+     * @param {number} publishTime
+     * @param {number} joinTime
+     * @param {string} tckid
+     * @param {string} pcTraceId
+     * @param {string} traceId
+     * @param {PublishReason} reason
+     */
+    reportPublishFirstPacket(isRepublish: boolean, callid: string, startTime: number, publishTime: number, joinTime: number, tckid: string, pcTraceId: string, traceId: string, reason: PublishReason): void;
+    /**
+     * 停止推流事件上报
+     * @param {string} callid
+     * @param {number} unpubTime
+     * @param {number} result
+     * @param {string} tid
+     */
+    reportUnpublish(callid: string, unpubTime: number, result: number, tid: string): void;
+    /**
+     * 订阅事件埋点
+     * @param {string} callid
+     * @param {string} remoteid
+     * @param {boolean} isLargeVideo
+     * @param {boolean} isSmallVideo
+     * @param {boolean} isScreenShare
+     * @param {boolean} isAudio
+     * @param {number} result
+     * @param {string} tid
+     * @param {number} subTime
+     */
+    reportSubscribe(callid: string, remoteid: string, isLargeVideo: boolean, isSmallVideo: boolean, isScreenShare: boolean, isAudio: boolean, result: number, tid: string, subTime: number, logInfo: LogInfo): void;
+    reportSubscribeFirstPacket(isResume: boolean, callid: string, remoteid: string, trackId: string, logInfo: LogInfo): void;
+    reportSubscribeFirstFrame(isResume: boolean, callid: string, remoteid: string, trackId: string, logInfo: LogInfo): void;
+    /**
+     * 停止订阅事件埋点
+     * @param {string} callid
+     * @param {string} remoteid
+     * @param {number} unsubTime
+     * @param {number} result
+     * @param {string} tid
+     */
+    reportUnsubscribe(callid: string, remoteid: string, unsubTime: number, result: number, tid: string): void;
+    reportMute(callid: string, enable: boolean, type: 'audio' | 'video' | 'screen'): void;
+    /**
+     * 更新角色埋点
+     * @param {string|undefined} or 旧的角色
+     * @param {string} nr 新的角色
+     */
+    reportRoleUpdate(or: string | undefined, nr: string): void;
+    /**
+     * 更新角色埋点
+     * @param {string|undefined} or 旧的角色
+     * @param {string} nr 新的角色
+     */
+    reportRoleUpdateResult(or: string | undefined, nr: string, ctm: number, result: number): void;
+    /**
+     * 设置camera videoprofile
+     */
+    reportVideoProfile(profile: string): void;
+    /**
+     * 设置screenshare videoprofile
+     */
+    reportScreenProfile(profile: string): void;
+    /**
+     * 错误事件埋点
+     * @param err
+     */
+    reportError(err: any): void;
+    /**
+     * 错误事件埋点
+     * @param err
+     */
+    reportWarning(msg: string): void;
+    reportRoomServerEvent(isRequest: boolean, type: string, requestId: string, content: any): void;
+    reportTranscodingError(result: number, taskid: string): void;
+    reportBye(result: number): void;
+    reportAudio(error: any, cost: number, deviceName: string): void;
+    reportScreen(error: any, cost: number): void;
+    reportScreenStop(error: any): void;
+    reportSwitchCamera(deviceId: string): void;
+    reportPublishProfile(trackId: string, logInfo: LogInfo): void;
+    reportSDKCreated(startTs: number): void;
+    reportJoinInvoked(authInfo: AliRtcAuthInfo): void;
+    reportJoinResult(error: any): void;
+    reportSTSResult(code: number): void;
+    reportLeaveInvoked(): void;
+    reportSEIMessage(payloadType: number, length: number, repeatCount: number, delay: number, isKey: boolean): void;
+    /**
+     * 日志埋点
+     * 在埋点中增加tm字段，表示发生埋点的客户端本地时间
+     * @param {any} param 埋点内容
+     */
+    protected log(param: any): void;
+    /**
+     * 将Object转换成string
+     * @param {any} data
+     * @returns {string}
+     */
+    protected data2String(data: any): string;
+}
+
+declare interface SLSSTSToken {
+    access_key_id: string;
+    access_key_secret: string;
+    expiration: number;
+    log_store_debug: string;
+    log_store_stats: string;
+    project: string;
+    region_endpoint: string;
+    security_token: string;
+}
+
+declare interface SLSToken {
+    access_key_id: string;
+    access_key_secret: string;
+    security_token: string;
+    region_endpoint: string;
+    project: string;
+    log_store_stats: string;
+    log_store_debug: string;
+    expiration: number;
+}
+
 declare interface StopPreviewConfig {
     videoElement?: HTMLVideoElement;
     primary?: boolean;
@@ -3367,6 +3646,25 @@ declare interface StreamOptions {
     screen?: boolean;
 }
 
+declare class StsManager extends default_2<StsManagerListener> {
+    static logName: string;
+    private env;
+    private authInfo?;
+    private timer?;
+    private slsReporter;
+    private expiration;
+    private skipTime;
+    constructor(slsReporter: SLSReporter, env: AliRtcEnv);
+    refreshAuthInfo(authInfo: AliRtcAuthInfo): void;
+    private requestToken;
+    startSTSUpdate(authInfo: AliRtcAuthInfo): void;
+    clear(): void;
+}
+
+declare interface StsManagerListener {
+    onTokenUpdate: (ossToken: OSSToken, slsToken: SLSToken) => void;
+}
+
 declare interface SubConfig {
     isAudioSubscribing: boolean;
     isVideoSubscribing: boolean;
@@ -3380,6 +3678,19 @@ declare type SubscribeOptions = StreamOptions & {
     vMsid?: string;
 };
 
+declare enum SubscribeReason {
+    InRoom = "inRoom",
+    Join = "join",
+    Publish = "publish",
+    User = "user",
+    Resume = "resume"
+}
+
+declare interface TimeRecorder {
+    start: number;
+    [key: string]: number;
+}
+
 declare enum TrackEvent_2 {
     VideoTrackEnded = "videoTrackEnded",
     AudioTrackEnded = "audioTrackEnded",
@@ -3408,12 +3719,13 @@ declare class User extends default_2<UserListener> {
     protected uid: string;
     protected name: string;
     protected rtsManager: RtsManager;
-    callid: string;
+    callId: string;
     protected audioLevelMonitor: AudioLevelMonitor;
     protected audioVolumeIndicationInterval: number;
     constructor(userId: string, displayname: string, rtsManager: RtsManager, interval: number);
     get userId(): string;
     get displayname(): string;
+    protected createCallId(): void;
     setUserInfo(uid: string, displayName: string): void;
     refreshUserInfo(uid: string): void;
     /**
@@ -3449,188 +3761,6 @@ declare interface UserListener {
     remoteSubscribeError: (error: AliRtcError) => void;
 }
 
-declare class UserManager extends default_2<UserManagerListener> {
-    /**
-     * @ignore
-     */
-    static logName: string;
-    protected self: LocalUser | undefined;
-    protected userMap: RemoteUserMap;
-    private usersViewMap;
-    rtsManager: RtsManager;
-    protected localStreamManager: LocalStreamManager;
-    protected authInfo?: AliRtcAuthInfo;
-    protected signalingManager: SignalingManager;
-    pluginManager: PluginManager;
-    protected reporter: Reporter;
-    protected webTrackingClient: WebTrackingClient;
-    private statsMonitorId?;
-    private autoSubAudio;
-    private autoSubVideo;
-    private autoSubScreen;
-    private wantSubAudio;
-    private wantSubVideo;
-    private wantSubScreen;
-    private defaultStreamType;
-    private audioMuted;
-    private audioVolumeIndicationInterval;
-    private indicationTimer;
-    constructor(config: UserManagerConfig);
-    private addSignalingManagerListener;
-    isAutoSubAudio(): boolean;
-    setAutoSubAudio(value: boolean): void;
-    isAutoSubVideo(): boolean;
-    setAutoSubVideo(value: boolean): void;
-    isAutoSubScreen(): boolean;
-    setAutoSubScreen(value: boolean): void;
-    isWantSubAudio(): boolean;
-    setWantSubAudio(value: boolean): void;
-    isWantSubVideo(): boolean;
-    setWantSubVideo(value: boolean): void;
-    isWantSubScreen(): boolean;
-    setWantSubScreen(value: boolean): void;
-    setRemoteDefaultVideoStreamType(type: AliRtcVideoStreamType): void;
-    setRemoteUserVideoStreamType(userId: string, type: AliRtcVideoStreamType): void;
-    setAudioMuted(mute: boolean): void;
-    setLocalViewConfig(view: AliRtcLocalView, track: AliRtcVideoTrack): void;
-    setRemoteViewConfig(uid: string, view: AliRtcLocalView, track: AliRtcVideoTrack): void;
-    get channel(): string | undefined;
-    get userId(): string | undefined;
-    get remoteUsers(): RemoteUser[];
-    /**
-     * 获取推流对象，只能是自己
-     */
-    get publisher(): LocalUser | undefined;
-    get isSubscribing(): boolean;
-    /**
-     * 处理远端用户推流状态变化
-     * @param {string} userId
-     * @param {string} displayname
-     * @param {string} callid
-     * @param {string} pullStreamUrl
-     * @param {TrackInfo[]} tracks
-     * @returns {void}
-     */
-    onRemotePublishStateChange(publishInfo: PublishUser): void;
-    /**
-     * 登陆成功后初始化用户管理，创建自己
-     * @param {AliRtcAuthInfo} authInfo
-     * @param {string} pushStreamUrl
-     * @returns {Promise<void>}
-     */
-    init(authInfo: AliRtcAuthInfo, data: JoinResult): void;
-    /**
-     * 离会后清除所有数据
-     * @returns {Promise<void>}
-     */
-    clear(): Promise<void>;
-    private startStatsMonitor;
-    private stopStatsMonitor;
-    /**
-     * 根据userId获取RemoteUser
-     * @param {string} userId
-     * @param {string} displayname
-     * @returns {RemoteUser}
-     */
-    getUser(userId: string, displayname: string): RemoteUser;
-    /**
-     * 添加远端用户
-     * @param {string} userId
-     * @param {string} displayName
-     * @returns {void}
-     */
-    addUser(userId: string, displayName: string): RemoteUser;
-    /**
-     * 执行添加远端用户
-     * @param {string} userId
-     * @param {string} displayName
-     * @returns {RemoteUser}
-     */
-    private executeAddUser;
-    /**
-     * 远端用户设备状态变化
-     * @param {DeviceStatusChangeType} event
-     * @param {string} userId
-     */
-    private onRemoteDeviceStatusChange;
-    /**
-     * 远端流数据采集上报
-     * @param {any} stat
-     */
-    private onRemoteStatReport;
-    /**
-     * rts 单 PC 订阅超限
-     */
-    private onSubscribeExceeds;
-    /**
-     * 移除远端用户
-     * @param {LeaveInfo} user
-     * @returns {Promise<void>}
-     */
-    removeLeftUser(user: LeaveInfo): Promise<void>;
-    /**
-     * 更新远端用户数据
-     * @param {JoinInfo[]} users
-     * @returns {{joinList: JoinInfo[], leaveList: LeaveInfo[], presentList: JoinInfo[]}}
-     */
-    updateUsers(users: JoinInfo[]): {
-        joinList: JoinInfo[];
-        leaveList: LeaveInfo[];
-        presentList: JoinInfo[];
-    };
-    /**
-     * 根据 userId 获取 RemoteUser
-     * @param {string} userId
-     * @returns {RemoteUser | undefined}
-     */
-    getRemoteUser(userId: string): RemoteUser | undefined;
-    /**
-     * 更新远端用户设备信息
-     * @param {RemoteUserStatus} user
-     * @returns {boolean} true: 用户状态改变  false: 用户状态没有改变
-     */
-    updateRemoteUserDeviceStatus(user: RemoteUserStatus): void;
-    updateAudioWant(uid: string, sub: boolean): void;
-    updateVideoWant(uid: string, sub: boolean): void;
-    updateScreenWant(uid: string, sub: boolean): void;
-    private stopIndication;
-    private startIndication;
-    enableAudioVolumeIndication(interval: number): void;
-    refreshAuthInfo(authInfo: AliRtcRefreshAuthInfo): void;
-    setEnableMediaExtensionMsg(enable: boolean): void;
-    setAudioRedEnabled(enable: boolean): void;
-}
-
-declare interface UserManagerConfig {
-    reporter: Reporter;
-    webTrackingClient: WebTrackingClient;
-    signalingManager: SignalingManager;
-}
-
-declare interface UserManagerListener {
-    [ClientEventType.RtsSubscribeExceeds]: (userId: string, options?: SubscribeOptions) => void;
-    [DeviceStatusChange.Remote]: (event: DeviceStatusChangeType, userId: string) => void;
-    [ClientEventType.PullStreamStats]: (userId: string, stat: any, pullStreamUrl: string) => void;
-    remoteTrackAvailableChange: (uid: string, aliRtcAudioTrack: AliRtcAudioTrack, aliRtcVideoTrack: AliRtcVideoTrack) => void;
-    audioSubscribeStateChanged: (uid: string, oldState: AliRtcSubscribeState, newState: AliRtcSubscribeState, elapseSinceLastState: number, channel: string) => void;
-    videoSubscribeStateChanged: (uid: string, oldState: AliRtcSubscribeState, newState: AliRtcSubscribeState, elapseSinceLastState: number, channel: string) => void;
-    screenSubscribeStateChanged: (uid: string, oldState: AliRtcSubscribeState, newState: AliRtcSubscribeState, elapseSinceLastState: number, channel: string) => void;
-    userAudioMuted: (uid: string, isMute: boolean) => void;
-    userVideoMuted: (uid: string, isMute: boolean) => void;
-    userScreenMuted: (userId: string, isMute: boolean) => void;
-    audioVolume: (speakers: {
-        userId: string;
-        volume: number;
-    }[]) => void;
-    rtcLocalVideoStats: (stats: AliRtcLocalVideoStats[]) => void;
-    rtcLocalAudioStats: (stats: AliRtcLocalAudioStats[]) => void;
-    rtcRemoteVideoStats: (stats: AliRtcRemoteVideoStats[]) => void;
-    rtcRemoteAudioStats: (stats: AliRtcRemoteAudioStats[]) => void;
-    remoteAudioAutoPlayFail: (uid: string) => void;
-    remoteVideoAutoPlayFail: (uid: string, track: AliRtcVideoTrack) => void;
-    remoteSubscribeError: (error: AliRtcError) => void;
-}
-
 declare interface VideoScaler {
     getVideoTrack: () => MediaStreamTrack;
     updateOptions: (options: VideoScalerOptions) => void;
@@ -3660,90 +3790,4 @@ declare enum VideoStreamSource {
     Image = 2
 }
 
-declare interface WebTrackConfig {
-    biz: string;
-    extraParams?: string;
-}
-
-declare class WebTrackingClient {
-    protected index: number;
-    private tracker?;
-    private enable;
-    private commonParams;
-    constructor(config?: WebTrackConfig | boolean);
-    updateAuthInfo(info: AliRtcAuthInfo): void;
-    /**
-     * 调用加入房间API事件
-     * @param {AliRtcAuthInfo} authInfo 鉴权信息
-     */
-    reportJoinChannel(authInfo: AliRtcAuthInfo): void;
-    /**
-     * @method: 调用加入房间结果
-     * @param { rslt: 执行结果 code，0为成功，错误返回其他错误码 }
-         * @param { message: 执行结果，成功返回 success，失败返回失败 message }
-             * @param { ct: 耗时 }
-                 * @return {*}
-                 */
-             reportJoinChannelRes(rslt: number, message: string, ct: number): void;
-             /**
-              * 离会调用通知
-              */
-             reportLeaveChannel(): void;
-             /**
-              * 推流成功
-              * @param {string} calid（失败时为空）
-              * @param {boolean} isvl 是否推大流
-              * @param {boolean} isvs 是否推小流
-              * @param {boolean} isvss 是否推屏幕共享流
-              * @param {boolean} isaudio 是否推音频流
-              * @param {number} rslt 推流结果：成功0，失败错误码
-              * @param {string} tid 房间信令 tid
-              * @param {number} pubtm 执行推流到推流结束信令通知结束的耗时
-              * @param {number} est 开始时间
-              * @param {string} dctid 使用 datachannel 的 traceId
-              * @param {string} httptid 使用 http 建连的 traceId
-              * @param {string} url 推流地址
-              */
-             reportPush(calid: string, isvl: boolean, isvs: boolean, isvss: boolean, isaudio: boolean, rslt: number, tid: string, pubtm: number, est: number, dctid: string, httptid: string, url: string): void;
-             /**
-              * 拉流成功
-              * @param {string} calid 鉴权 callid (失败为空)
-              * @param {string} rmtid 远端用户 userId
-              * @param {boolean} isvl 是否拉视频大流
-              * @param {boolean} isvs 是否拉视频小流
-              * @param {boolean} isvss 是否拉屏幕分享流
-              * @param {boolean} isaudio 是否拉音频流
-              * @param {number} rslt 拉流结果：成功0，失败错误码
-              * @param {number} subtm 订阅耗时
-              * @param {string} dctid 使用 datachannel 的 traceId
-              * @param {string} httptid 使用 http 建连的 traceId
-              * @param {string} url 拉流地址
-              * @param {number} est 开始时间
-              */
-             reportPull(calid: string, rmtid: string, isvl: boolean, isvs: boolean, isvss: boolean, isaudio: boolean, rslt: number, subtm: number, dctid: string, httptid: string, url: string, est: number): void;
-             /**
-              * 上报订阅流的音频数据
-              * @param {string} rmtid 远端用户 userId
-              * @param {any} stats 流数据
-              */
-             reportSubscribeAudioStats(rmtid: string, stats: any): void;
-             /**
-              * 上报订阅流的视频数据
-              * @param {string} rmtid 远端用户 userId
-              * @param {any} stats 流数据
-              */
-             reportSubscribeVideoStats(rmtid: string, stats: any): void;
-             /**
-              * webTracking 埋点上报
-              * @param {string | number} msgId 事件标识
-              * @param {Record<string, any>} args 事件携带参数
-              * @param {Record<string, any>} params 单个打点携带的其他参数
-              */
-             protected report(msgId: string | number | EventId, args?: Record<string, any>, params?: Record<string, any>): void;
-             /**
-              * 上报环境监测的结果
-              */
-             static reportCheck(result: any, biz?: string): void;
-         }
-
-         export { }
+export { }