d8d-admin-mobile-starter-public/client/share/aliyun-rtc-sdk.d.ts

/// <reference types="https://esm.d8d.fun/@types/dom-mediacapture-transform@0.1.11/index.d.ts" />

import { default as default_2 } from 'https://esm.d8d.fun/eventemitter3@5.0.1/index.d.ts';
import { default as default_3 } from 'https://esm.d8d.fun/@aliyun-sls/web-track-browser@0.3.9/dist/web-track-browser.d.ts';
import { IAudioConstraints } from 'https://esm.d8d.fun/media-device@1.3.0/dist/index.d.ts';
import { IProfile } from 'https://esm.d8d.fun/aliyun-rts-sdk@2.12.3/dist/index.d.ts';
import { IStreamConfig } from 'https://esm.d8d.fun/aliyun-rts-sdk@2.12.3/dist/index.d.ts';
import { IVideoConstraints } from 'https://esm.d8d.fun/media-device@1.3.0/dist/index.d.ts';
import { LocalStream } from 'https://esm.d8d.fun/aliyun-rts-sdk@2.12.3/dist/index.d.ts';
import { RemoteStream } from 'https://esm.d8d.fun/aliyun-rts-sdk@2.12.3/dist/index.d.ts';

declare interface AliEngineCameraCapturerConfiguration {
    width?: number;
    height?: number;
    frameRate?: number;
    maxSendFrameRate?: number;
    bitrate?: number;
    cameraDirection?: AliRtcCameraDirection;
    deviceId?: string;
}

declare enum AliRtcAudioTrack {
    /** 无麦克风 */
    AliRtcAudioTrackNo = 0,
    /** 麦克风 */
    AliRtcAudioTrackMic = 1,
    /** 第二音频流 */
    AliRtcAudioTrackDual = 2
}

export declare interface AliRtcAuthInfo {
    /** 频道ID */
    channelId: string;
    /** 用户ID */
    userId: string;
    /** 应用ID */
    appId: string;
    /** 随机串 */
    nonce?: string;
    /** 时间戳 */
    timestamp: number;
    /** 令牌 */
    token: string;
    /** 会议id */
    sessionId?: string;
    /** 角色鉴权值 */
    role?: AliRtcSdkTokenRole;
}

declare interface AliRtcBeautyConfig {
    whiteningLevel: number;
    /*! 磨皮等级[0-1.0] */
    smoothnessLevel: number;
}

declare enum AliRtcCameraDirection {
    /** 无效 */
    CAMERA_INVALID = -1,
    /** 后置 */
    CAMERA_REAR = 0,
    /** 前置 */
    CAMERA_FRONT = 1
}

export declare interface AliRtcCheckResult {
    support: boolean;
    detail: {
        isWebRTCSupported: boolean;
        isH264EncodeSupported: boolean;
        isH264DecodeSupported: boolean;
        isBrowserSupported: boolean;
        isScreenShareSupported: boolean;
        isSendMediaExtensionMsgSupported: boolean;
    };
}

export declare enum AliRtcConnectionStatus {
    /** 初始化完成 */
    AliRtcConnectionStatusInit = 0,
    /** 网络连接断开 */
    AliRtcConnectionStatusDisconnected = 1,
    /** 建立网络连接中 */
    AliRtcConnectionStatusConnecting = 2,
    /** 网络已连接 */
    AliRtcConnectionStatusConnected = 3,
    /** 重新建立网络连接中 */
    AliRtcConnectionStatusReconnecting = 4,
    /** 网络连接失败 */
    AliRtcConnectionStatusFailed = 5
}

export declare enum AliRtcConnectionStatusChangeReason {
    /** 未知原因 */
    AliRtcConnectionChangedDummyReason = 0,
    /** 媒体通道变更 */
    AliRtcConnectionMediaPathChanged = 1,
    /** 心跳超时 */
    AliRtcConnectionSignalingHeartbeatTimeout = 2,
    /** 心跳恢复 */
    AliRtcConnectionSignalingHeartbeatAlive = 3,
    /** DNS解析成功 */
    AliRtcConnectionSignalingHttpDnsResolved = 4,
    /** DNS解析失败 */
    AliRtcConnectionSignalingHttpDnsFailure = 5,
    /** GSLB请求失败 */
    AliRtcConnectionSignalingGslbFailure = 6,
    /** GSLB请求成功 */
    AliRtcConnectionSignalingGslbSuccess = 7,
    /** 加入频道失败 */
    AliRtcConnectionSignalingJoinRoomFailure = 8,
    /** 加入频道成功 */
    AliRtcConnectionSignalingJoinRoomSuccess = 9,
    /** 离开频道 */
    AliRtcConnectionSignalingLeaveRoom = 10,
    /** 信令建立连接 */
    AliRtcConnectionSignalingConnecting = 11,
    /** 网络连接中断 */
    AliRtcConnectionChangedNetworkInterrupted = 12
}

export declare class AliRtcDataChannelMsg {
    data: ArrayBuffer;
    networkTime: number;
    futureOrProgress: number;
    type: AliRtcDataMsgType;
    constructor(data: ArrayBuffer, type?: AliRtcDataMsgType, networkTime?: number, progress?: number);
    toBuffer(): ArrayBuffer;
    static parse(buffer: ArrayBuffer): AliRtcDataChannelMsg | undefined;
}

export declare enum AliRtcDataMsgType {
    AliEngineDataMsgNone = 0,
    AliEngineDataMsgMusicProgress = 1,
    AliEngineDataMsgCustom = 2
}

declare type AliRtcDualVideoPluginOptions = VideoScalerOptions;

declare class AliRtcEngine_2 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
     * @returns {Promise<AliRtcCheckResult>}
     */
    static isSupported: (direction?: "sendonly" | "recvonly" | "sendrecv" | undefined) => Promise<AliRtcCheckResult>;
    /**
     * 检查浏览器是否支持开启大小流
     * @returns {boolean}
     */
    static isDualVideoStreamSupported: typeof VideoScalerFactory.isSupport;
    /**
     * 获取摄像头列表
     * @returns {Promise<string[]>}
     */
    static getCameraList: () => Promise<MediaDeviceInfo[]>;
    static getMicrophoneList: () => Promise<MediaDeviceInfo[]>;
    static getSpeakerList: () => Promise<MediaDeviceInfo[]>;
    static getSdkVersion(): string;
    /**
     * 设置日志等级
     * @param level
     */
    static setLogLevel(level: AliRtcLogLevel): void;
    private static _instance?;
    /**
     * @brief 获取AliRtcEngine实例
     * @param key key 可选参与，用于创建多个实例
     * @details 该方法为同步调用，以单例的形式提供对象创建。
     */
    static getInstance(config?: AliRtcEngineConfig): AliRtcEngine_2;
    static AliRtcSdkChannelProfile: typeof AliRtcSdkChannelProfile;
    static AliRtcSdkClientRole: typeof AliRtcSdkClientRole;
    static AliRtcVideoStreamType: typeof AliRtcVideoStreamType;
    static AliRtcVideoTrack: typeof AliRtcVideoTrack;
    static AliRtcAudioTrack: typeof AliRtcAudioTrack;
    static AliRtcPublishState: typeof AliRtcPublishState;
    static AliRtcSubscribeState: typeof AliRtcSubscribeState;
    static AliRtcRawDataStreamType: typeof AliRtcRawDataStreamType;
    static AliRtcLogLevel: typeof AliRtcLogLevel;
    static AliRtcEngineLocalDeviceType: typeof AliRtcEngineLocalDeviceType;
    static AliRtcEngineLocalDeviceExceptionType: typeof AliRtcEngineLocalDeviceExceptionType;
    static AliRtcConnectionStatus: typeof AliRtcConnectionStatus;
    static AliRtcConnectionStatusChangeReason: typeof AliRtcConnectionStatusChangeReason;
    static AliRtcPlugin: typeof AliRtcPlugin;
    static AliRtcPluginTrackType: typeof AliRtcPluginTrackType;
    static AliRtcLiveTranscodingParam: typeof AliRtcLiveTranscodingParam;
    static AliRtcLiveTranscodingEncodeParam: typeof AliRtcLiveTranscodingEncodeParam;
    static AliRtcLiveTranscodingMixParam: typeof AliRtcLiveTranscodingMixParam;
    static AliRtcLiveTranscodingSingleParam: typeof AliRtcLiveTranscodingSingleParam;
    static AliRtcTranscodingClockWidget: typeof AliRtcTranscodingClockWidget;
    static AliRtcTranscodingImage: typeof AliRtcTranscodingImage;
    static AliRtcTranscodingText: typeof AliRtcTranscodingText;
    static AliRtcTranscodingUser: typeof AliRtcTranscodingUser;
    static AliRtcLiveTranscodingMixMode: typeof AliRtcLiveTranscodingMixMode;
    static AliRtcLiveTranscodingStreamType: typeof AliRtcLiveTranscodingStreamType;
    static AliRtcLiveTranscodingSourceType: typeof AliRtcLiveTranscodingSourceType;
    static AliRtcLiveTranscodingTaskProfile: typeof AliRtcLiveTranscodingTaskProfile;
    static AliRtcLiveTranscodingAudioSampleRate: typeof AliRtcLiveTranscodingAudioSampleRate;
    static AliRtcLiveTranscodingSegmentType: typeof AliRtcLiveTranscodingSegmentType;
    static AliRtcLiveTranscodingFontType: typeof AliRtcLiveTranscodingFontType;
    static AliRtcLiveTranscodingCropMode: typeof AliRtcLiveTranscodingCropMode;
    static AliRtcLiveTranscodingMediaProcessMode: typeof AliRtcLiveTranscodingMediaProcessMode;
    static AliRtcLiveTranscodingState: typeof AliRtcLiveTranscodingState;
    static AliRtcLiveTranscodingErrorCode: typeof AliRtcLiveTranscodingErrorCode;
    static AliRtcTrascodingPublishTaskStatus: typeof AliRtcTrascodingPublishTaskStatus;
    static AliRtcDataChannelMsg: typeof AliRtcDataChannelMsg;
    static AliRtcDataMsgType: typeof AliRtcDataMsgType;
    /**
     * @brief 销毁 {@link AliRtcEngine}实例
     * @details 调用该方法将会释放所有内部使用的资源,当完成音视频通信之后都建议调用该方法释放实例.调用该方法后，你将不能再使用 {@link AliRtcEngine} 的其他方法和任何回调，如需要再次使用需要重新调用 {@link AliRtcEngine.getInstance} 方法创建一个新的实例。
     * @note 该方法为同步调用,需要等待内部资源释放之后才能执行其他方法
     */
    destroy(): Promise<void>;
    protected channelProfile: AliRtcSdkChannelProfile;
    protected clientRole: AliRtcSdkClientRole;
    protected signalingManager: SignalingManager;
    protected bizControl: BizControl;
    protected liveTranscoding: AliRtcLiveTranscoding;
    protected maxSignalingConnectRetryCount?: number;
    protected slsReporter: SLSReporter;
    protected connectState: AliRtcConnectionStatus;
    protected networkError: boolean;
    private _audioOnlyMode;
    private _authInfoWillExpireTimer;
    private _authInfoExpiredTimer;
    private _serverCheckAuthInfoTimestamp;
    private _localCheckAuthInfoTimestamp;
    constructor(config?: AliRtcEngineConfig);
    /**
     * 获取当前频道号，已经入会成功返回频道号，否则返回undefined
     */
    get channel(): string | undefined;
    /**
     * 获取当前用户的 userId，已经入会成功返回自己的 userId，否则返回 undefined
     */
    get userId(): string | undefined;
    get remoteUsers(): RemoteUser[];
    get publisher(): LocalUser | undefined;
    get publishAudioStream(): LocalStream | undefined;
    get isPublishing(): boolean;
    private get currentRoleCanPublish();
    /**
     * 监听 Publisher 事件
     */
    private addPublisherListeners;
    private occurError;
    /**
     * 监听 UM 事件
     */
    private addBizControlListeners;
    /**
     * 添加信令处理
     * @returns {void}
     */
    private addSignalingManagerListeners;
    private backToUnpublishStatus;
    /**
     * 处理入会远程的用户
     * @param {JoinInfo[]} users
     * @returns {void}
     */
    private handleUsersJoin;
    /**
     * 本地用户设备状态变化，通过信令发送给其他人
     * @param {string} status
     * @returns {void}
     */
    protected onLocalDeviceStatusChange(status: string): void;
    /**
     * 网络恢复后，重推；
     */
    private resumePublishForNetworkRecovery;
    /**
     * 处理房间信令重连成功/心跳恢复
     * @returns {void}
     */
    protected onNetworkRecovery(users?: JoinInfo[]): Promise<void>;
    private resumeSubList;
    /**
     * 重新订阅远端用户
     * @param {string} userId
     * @returns {Promise<void>}
     */
    retrySubscribe(userId: string): Promise<void>;
    /**
     * 处理网络异常
     * @param {AliRtcError} err
     * @returns {void}
     */
    protected onNetworkError(): void;
    /**
     * 处理错误
     * @param {AliRtcError} err
     * @returns {void}
     */
    protected onError(err: AliRtcError): void;
    /**
     * 是否开启大小流
     * @param {boolean} enabled
     * @param {AliRtcDualVideoPluginOptions} [options]
     */
    setEnableDualVideoStream(enabled: boolean, options?: AliRtcDualVideoPluginOptions): void;
    /**
     * 对本地流的操作预检查
     */
    private hadPushStream;
    /**
     * 获取当前使用的麦克风 deviceId
     * @returns {string | undefined}
     */
    getCurrentMicDeviceId(): string | undefined;
    /**
     * 设置频道模式
     * @param {AliRtcSdkChannelProfile} channelProfile 频道类型，默认值为 AliEngineCommunication
     * @note 该接口只可以在加入频道之前调用，会议中不可以重新设置，离开频道后可以重新设置
     */
    setChannelProfile(channelProfile: AliRtcSdkChannelProfile): void;
    private startAndPublishDefaultDevices;
    /**
     * 设置用户角色
     * @param {AliRtcSdkClientRole} clientRole 用户角色类型，默认值为 {@link AliRtcSdkClientRole.AliRtcSdkLive}（观众角色）
     * @note 非通信模式下角色类型才有效
     */
    setClientRole(clientRole: AliRtcSdkClientRole): Promise<void>;
    /**
     * 获取用户角色
     * @returns {AliRtcSdkClientRole}
     */
    getCurrentClientRole(): AliRtcSdkClientRole;
    /**
     * 是否开启纯音频模式
     * @param {boolean} audioOnly
     * - true：只有音频推流和拉流
     * - false（默认值）：音视频都支持
     * @note 只可以在 {@link joinChannel} 之前设置
     */
    setAudioOnlyMode(audioOnly: boolean): void;
    setAudioRedEnabled(audioRedEnabled: boolean): void;
    /**
     * 设定鉴权信息过期事件定时器
     * @param expireTimestamp 鉴权信息过期时间
     * @param serverTimestamp 入会时服务端时间
     */
    private setCheckAuthInfoTimer;
    /**
     * 清除鉴权信息过期事件定时器
     */
    private clearCheckAuthInfoTimer;
    /**
     * @brief 加入频道
     * @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 {AliRtcAuthInfo | string} authInfoOrToken (多参数/单参数)   认证信息，从App Server获取。
     * @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(authInfoOrToken: AliRtcAuthInfo | string, userName?: string): Promise<void>;
    /**
     * 离开频道
     * @note 如果当前不在频道内，调用 leaveChannel 不会对实例产生任何影响，但会产生消息，通知频道内其他用户
     */
    leaveChannel(): Promise<void>;
    /**
     * 检查当前是否在频道中
     * @returns {boolean}
     * - true：在频道中
     * - false：不在频道中
     */
    isInCall(): boolean;
    /**
     * 刷新鉴权信息
     * @param {AliRtcRefreshAuthInfo | string} authInfoOrToken
     */
    refreshAuthInfo(authInfoOrToken: AliRtcRefreshAuthInfo | string): Promise<void>;
    /**
     * 检查当前是否纯音频模式
     * @returns {boolean}
     * - true：纯音频模式
     * - false：音视频模式
     */
    isAudioOnly(): boolean;
    /**
     * @brief 是否推送音频流
     * @param enabled 是否开启/关闭本地音频流推送
     *  - true: 开启音频流推送
     *  - false: 关闭音频流推送
     * @note SDK默认设置推送音频流，在加入频道前也可以调用此接口修改默认值，并在加入频道成功时生效
     * @note 加入频道后调用不会停止本地音频采集
     */
    publishLocalAudioStream(enabled: boolean): Promise<void>;
    /**
     * @brief 查询是否设置推送音频流
     * @return
     *  - true: 已设置推送
     *  - false: 未设置推送
     * @note SDK默认设置推送音频流，如果需要取消推送音频流，可以通过 {@link publishLocalAudioStream}方法设置为不推送
     */
    isLocalAudioStreamPublished(): boolean;
    /**
     * 设置是否默认接收音频流
     * @param sub
     * - true: 接收用户的音频流
     * - false: 停止接收用户的音频流
     * @note 入会前入会后均可调用。如果在加入频道后调用 setDefaultSubscribeAllRemoteAudioStreams(false)，会接收不到设置后加入频道的用户的音频流。
     * @note 停止接收音频流后，如果想要恢复接收，请调用 {@link subscribeRemoteAudioStream}，并指定你想要接收的远端用户ID；
     * @note 如果想恢复接收多个用户的音频流，则需要多次调用 {@link subscribeRemoteAudioStream}。setDefaultSubscribeAllRemoteAudioStreams(true) 只能恢复接收后面加入频道的用户的音频流。
     */
    setDefaultSubscribeAllRemoteAudioStreams(sub: boolean): void;
    /**
     * 停止/恢复接收所有远端音频流
     * @param sub
     * - true: 接收所有用户的音频流
     * - false: 停止接收所有用户的音频流
     * @note 该接口作为订阅远端音频流的总开关，如果设置为false，则不仅当前会议中所有远端音频流都会停止订阅，后续入会的新用户也将不再订阅（即使设置了setDefaultSubscribeAllRemoteAudioStreams(true))
     */
    subscribeAllRemoteAudioStreams(sub: boolean): void;
    /**
     * 手动拉音频流
     * @param uid uid
     * @param sub，默认为 true
     * - true: 接收指定用户的音频流
     * - false: 停止接收指定用户的音频流
     * @note 如果之前有调用过 subscribeAllRemoteAudioStreams(false) 对所有远端音频进行静音，在调用本 API 之前请确保你已调用 subscribeAllRemoteAudioStreams(true)。
     * @note {@link subscribeAllRemoteAudioStreams} 是全局控制，{@link subscribeRemoteAudioStream} 是精细控制。
     */
    subscribeRemoteAudioStream(uid: string, sub: boolean): void;
    /**
     * 手动恢复远端用户音频流
     * @param uid
     * 用于在自动播放失败等场景恢复播放
     */
    resumeRemoteAudioStream(uid: string): void;
    stopRemoteAudioPlay(uid: string): void;
    /**
     * @brief 是否推送本地视频流
     * @param enabled 是否开启/关闭本地视频流推送
     * - true: 开启视频流推送
     * - false: 关闭视频流推送
     * @note SDK默认设置推送视频流，在加入频道前也可以调用此接口修改默认值，并在加入频道成功时生效
     * @note 加入频道后调用不会停止本地视频采集
     */
    publishLocalVideoStream(enabled: boolean): Promise<void>;
    /**
     * @brief 查询是否设置推送视频流
     * @return
     *  - true: 已设置推送
     *  - false: 未设置推送
     * @note SDK默认设置推送视频流，如果需要取消推送视频流，可以通过 {@link publishLocalVideoStream} 方法设置为不推送
     */
    isLocalVideoStreamPublished(): boolean;
    /**
     * 设置默认订阅的相机流格式，大流或小流，会覆盖单人的大小流设置
     * @param {AliRtcVideoStreamType} streamType 流类型
     * @note 仅支持传入 1 （默认大流）、2（默认小流），传入其他无效
     * @note 当无默认类型的相机流，但有另一个，将拉取另一个
     */
    setRemoteDefaultVideoStreamType(streamType: AliRtcVideoStreamType): void;
    /**
     * 设置某一用户默认订阅的相机流格式，大流或小流
     * @param userId 远端用户ID
     * @param {AliRtcVideoStreamType} streamType 流类型
     * @note 仅支持传入 1 （默认大流）、2（默认小流），传入其他无效
     * @note 当无默认类型的相机流，但有另一个，将拉取另一个
     */
    setRemoteUserVideoStreamType(userId: string, streamType: AliRtcVideoStreamType): void;
    /**
     * 设置是否默认接收视频流
     * @param sub
     * - true: 接收用户的视频流
     * - false: 不接收用户的视频流
     * @note 入会前入会后均可调用。如果在加入频道后调用 setDefaultSubscribeAllRemoteVideoStreams(false)，会接收不到设置后加入频道的用户的视频流。
     * @note 停止接收视频流后，如果想要恢复接收，请调用 {@link subscribeRemoteVideoStream}，并指定你想要接收的远端用户ID；
     * @note 如果想恢复接收多个用户的视频流，则需要多次调用 {@link subscribeRemoteVideoStream}；setDefaultSubscribeAllRemoteVideoStreams(true) 只能恢复接收后面加入频道的用户的视频流。
     */
    setDefaultSubscribeAllRemoteVideoStreams(sub: boolean): void;
    /**
     * 停止/恢复接收所有远端视频流
     * @param sub
     * - true: 接收所有用户的视频流
     * - false: 停止接收所有用户的视频流
     * @note 该接口作为订阅远端视频流的总开关，如果设置为false，则不仅当前会议中所有远端视频流都会停止订阅，后续入会的新用户也将不再订阅（即使设置了setDefaultSubscribeAllRemoteVideoStreams(true)）
     */
    subscribeAllRemoteVideoStreams(sub: boolean): void;
    /**
     * 合并订阅远端音视频流
     * @param uid uid
     * @param videoTrack 视频流类型
     * @param subVideo 是否订阅视频
     * @param subAudio 是否订阅音频
     * @return
     */
    subscribeRemoteMediaStream(uid: string, videoTrack: AliRtcVideoTrack, subVideo: boolean, subAudio: boolean): void;
    /**
     * @brief 手动恢复远端用户视频流
     * @param uid
     * @param videoTrack
     * 用于在自动播放失败等场景恢复播放
     */
    resumeRemoteMediaStream(uid: string, videoTrack: AliRtcVideoTrack): void;
    /**
     * @brief 订阅目标频道，指定用户的流
     * @param channelId 目标频道
     * @param uid 用户ID，从App server分配的唯一标示符
     * @param track 订阅的视频流类型
     * @param sub_audio 是否订阅远端用户的音频流
     * - true:订阅指定用户的音频流
     * - false:停止订阅指定用户的音频流
     * @param sub 是否订阅远端用户的流
     * - true:订阅指定用户的流
     * - false:停止订阅指定用户的流
     */
    /**
     * @brief 获取远端在线用户列表
     * @return 当前频道内所有在线用户userid列表
     * @note 若当前频道ChannelProfile设置为 {@link AliRtcSdkChannelProfile.AliRtcSdkInteractiveLive} 模式，只有设置角色为 {@link AliRtcSdkClientRole.AliRtcSdkInteractive} 的用户为在线用户，本方法不返回角色为 {@link AliRtcSdkClientRole.AliRtcSdkLive} 的用户
     */
    getOnlineRemoteUsers(): string[];
    /**
     * @brief 查询远端用户状态
     * @details 通过 uid 查询指定远端用户的状态信息.
     * @param uid 用户ID。从App server分配的唯一标示符
     * @return 返回一个远端用户对象。详见{@link RemoteUser}
     */
    getUserInfo(uid: string): AliRtcRemoteUserInfo | undefined;
    /**
     * @brief 查询用户是否在线
     * @param uid 用户ID，从App server分配的唯一标示符。不可为NULL
     * @return
     *  - true: 用户在线
     *  - false: 用户不在线
     * @note 根据特定的UserID，查询该用户是否在线。
     */
    isUserOnline(uid: string): boolean;
    /**
     * @brief 开始屏幕共享
     * @param config 开启屏幕共享的配置
     * @note 可以通过 config 选择是否开启音频或使用外部自定义 MediaStream
     * @note 自定义 MediaStream 仅支持一个视频流和一个音频流
     */
    startScreenShare(config?: AliRtcScreenShareStartConfig): Promise<void>;
    /**
     * @brief 停止屏幕共享
     */
    stopScreenShare(): Promise<void>;
    /**
     * @brief 设置屏幕流视频编码属性
     * @details 该方法用于设置屏幕流视频编码属性对应的视频参数，如分辨率、帧率、码率、等 所有设置的参数都有相应的范围限制，如果设置的参数不在有效范围内，SDK会自动调节
     * @param config 预定义的屏幕共享编码属性，
     * @note
     * - 该方法在入会前和入会后都可以调用，如果每次入会只需要设置一次屏幕流视频编码属性，建议在入会前调用
     */
    setScreenShareConfiguration(config: AliRtcScreenShareConfiguration): Promise<void>;
    /**
     * @breif 设置屏幕流采集内容提示
     * @param contentHint 设置屏幕流内容提示，用于提升在不同内容场景下的视频编码质量 [参考文档](https://developer.mozilla.org/en-US/docs/Web/API/MediaStreamTrack/contentHint)
     */
    setScreenShareContentHint(contentHint: string): Promise<void>;
    /**
     * @brief 是否推送屏幕分享
     * @param enabled 是否开启/关闭本地屏幕分享推送
     *  - true: 开启屏幕分享推送
     *  - false: 关闭屏幕分享推送
     * @param config 开启屏幕共享的配置
     * @note SDK默认设置推送屏幕分享，在加入频道前也可以调用此接口修改默认值，并在加入频道成功时生效
     * @note 加入频道后调用不会停止本地屏幕采集
     */
    publishLocalScreenShareStream(enabled: boolean, config?: AliRtcScreenShareStartConfig): Promise<void>;
    /**
     * @brief 查询是否设置推送屏幕分享
     * @return
     - true: 已设置推送
     - false: 未设置推送
     * @note SDK默认设置不推送屏幕分享，如果需要推送屏幕共享，可以通过{@link startScreenShare}方法开启屏幕共享推送
     */
    isScreenSharePublished(): boolean;
    /**
     * @brief 设置共享音频的音量大小
     * @param volume 共享音频音量大小,取值范围[0, 100]
     */
    setAudioShareVolume(volume: number): void;
    /**
     * 开启音量检测功能
     * @param {number} interval 检测间隔，单位: ms
     * @note 建议设置为300ms~500ms,取值 ≤0 表示不启用音量检测功能
     */
    enableAudioVolumeIndication(interval: number): void;
    /**
     * 停止/恢复本地音频数据发送
     * @param {boolean} mute 停止或恢复本地音频数据发送
     * - true：静音本地音频
     * - false（默认值）：取消静音
     */
    muteLocalMic(mute?: boolean): void;
    /**
     * 停止/恢复远端指定用户的音频流的播放
     * @param {string} uid 用户ID, 从 App server 分配的唯一标示符
     * @param {boolean} mute 停止或恢复远端的音频播放
     * - true：停止播放
     * - false（默认值）：恢复播放
     * @note 拉流和解码不受影响，支持 {@link joinChannel} 之前和之后设置
     */
    muteRemoteAudioPlaying(uid: string, mute?: boolean): void;
    /**
     * 停止/恢复远端的所有音频流的播放
     * @param {boolean} mute 停止或恢复远端的音频播放
     * - true：停止播放
     * - false（默认值）：恢复播放
     * @note 拉流和解码不受影响，支持 {@link joinChannel} 之前和之后设置
     */
    muteAllRemoteAudioPlaying(mute?: boolean): void;
    /**
     * @brief 开启音频采集
     * @details 此接口可以控制提前打开音频采集，如果不设置，则SDK会自动控制音频采集设备：
     * - 如果打开推送音频流{@link publishLocalAudioStream}，SDK在入会后会打开音频采集
     * - 如果关闭推送音频流{@link publishLocalAudioStream}，SDK在入会后不会打开音频采集
     * @param {IAudioConstraints} audioConstraints 音频采集参数
     */
    startAudioCapture(audioConstraints?: IAudioConstraints): Promise<void>;
    /**
     * @brief 设置音频采集参数
     * @param constraints 音频采集参数
     */
    setAudioConstraints(constraints: IAudioConstraints): Promise<void>;
    /**
     * 关闭音频采集
     * @note 调用此接口后，入会后、离会再入会，采集设备保持关闭状态
     */
    stopAudioCapture(): void;
    /**
     * 切换麦克风
     * @param {string} deviceId 麦克风设备ID
     * @param {MediaStreamAudioTrack} audioTrack 音频轨道
     * @note 传入 audioTrack 则会使用传入的音频轨道
     */
    switchMicrophone(deviceId: string, audioTrack?: MediaStreamAudioTrack): Promise<void>;
    /**
     * 设置录音音量
     * @param {VolumeSettings} volumeSettings 录音音量，取值范围[0,100]
     * - 0：静音
     * - <100：减小音量
     * - >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
     *
     * @param {AudioProfileKey} profileKey
     * @return {Promise<void>}
     */
    setAudioProfile(profileKey: AudioProfileKey): Promise<void>;
    /**
     * @brief 获取当前用户或远端用户的音频轨道
     * @param {string} userId 远端用户 userId
     * @return {MediaStreamTrack} 音频轨道
     *
     */
    getAudioTrack(userId?: string): Promise<MediaStreamTrack | undefined>;
    /**
     * 为本地预览设置渲染元素和视频流
     * @param {AliRtcLocalView} view 渲染元素，null 则为停止显示
     * @param {AliRtcVideoTrack} track 视频流类型
     */
    setLocalViewConfig(view: AliRtcLocalView, track: AliRtcVideoTrack): Promise<void>;
    /**
     * @brief 采集偏好设置
     * @param cameraCapturerConfiguration 偏好设置
     * - 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>;
    /**
     * @breif 设置摄像头采集内容提示
     * @param contentHint 设置视频内容提示，用于提升在不同内容场景下的视频编码质量 [参考文档](https://developer.mozilla.org/en-US/docs/Web/API/MediaStreamTrack/contentHint)
     */
    setCameraCapturerContentHint(contentHint: string): Promise<void>;
    /**
     * 禁用或启用摄像头采集
     * @param enabled
     * - true : 启用摄像头采集
     * - false : 禁用摄像头采集
     */
    enableLocalVideo(enabled?: boolean): Promise<void>;
    /**
     * 是否将停止摄像头流数据发送
     * @param mute
     * - true：停止发送摄像头流数据
     * - false：表示恢复正常
     */
    muteLocalCamera(mute: boolean): Promise<void>;
    /**
     * 是否将停止屏幕共享流数据发送
     * @param mute
     * - true：停止发送屏幕共享流数据
     * - false：表示恢复正常
     */
    muteLocalScreen(mute: boolean): Promise<void>;
    /**
     * 设置远端视频显示视图
     * @param {AliRtcLocalView} view 渲染元素
     * @param {AliRtcVideoTrack} track 视频流类型
     * @param {string} uid 用户ID, 从App server分配的唯一标示符
     * @note 支持joinChannel之前和之后切换窗口。如果 view 传入 null，则停止显示
     */
    setRemoteViewConfig(view: AliRtcLocalView, uid: string, track: AliRtcVideoTrack): void;
    /**
     * 检查camera是否打开
     * @return {boolean}
     * - true: 摄像头已打开
     * - false: 摄像头没有打开
     * @note 该接口用于检查当前camera的状态
     */
    isCameraOn(): boolean;
    /**
     * 切换摄像头
     * @param {string} deviceId 摄像头设备ID
     * @param {MediaStreamVideoTrack} videoTrack 视频流
     * @note 若不传 deviceId 则前后摄像头切换，传入 videoTrack 则会使用传入的 videoTrack
     */
    switchCamera(deviceId?: string, videoTrack?: MediaStreamVideoTrack): Promise<void>;
    /**
     * 获取摄像头设备ID
     * @return {string} 摄像头设备ID
     */
    getCurrentCameraDeviceId(): string | undefined;
    /**
     * @brief 开启摄像头预览
     * @details 开启摄像头预览并自动打开摄像头
     * @note 如果没有调用{@link setLocalViewConfig}设置显示的view，则无法预览，但不影响推流。
     * 可以在{@link joinChannel}之前就开启预览会自动打开摄像头。
     */
    startPreview(): Promise<void>;
    /**
     * @brief 开启屏幕共享预览
     * @details 开启屏幕共享预览并自动打开屏幕共享
     * @param config 开启屏幕共享的配置
     * @note 如果没有调用{@link setLocalViewConfig}设置显示的view，则无法预览，但不影响推流。
     * 可以在{@link joinChannel}之前就开启预览会自动打开屏幕共享。
     */
    startPreviewScreen(config?: AliRtcScreenShareStartConfig): Promise<void>;
    /**
     * @brief 停止本地视频预览
     */
    stopPreview(): Promise<void>;
    /**
     * @brief 停止本地屏幕共享预览
     */
    stopPreviewScreen(): Promise<void>;
    /**
     * @brief 获取当前用户或远端用户的视频轨道
     * @param {AliRtcVideoTrackConfig} config 视频轨道配置
     * @return {MediaStreamTrack} 视频轨道
     *
     */
    getVideoTrack(config: AliRtcVideoTrackConfig): Promise<MediaStreamTrack | undefined>;
    /**
     * @brief 开启旁路直播
     * @details SDK提供了开启旁路直播的功能，可以把实时音视频流经过混流、混音处理转为标准直播流并推送至指定CDN供应商
     * @param streamUrl 推流地址
     * @param transcoding 推流所需参数, 详细定义见{@link AliRtcLiveTranscodingParam}
     */
    startPublishLiveStream(streamUrl: string, transcoding: AliRtcLiveTranscodingParam): Promise<any>;
    /**
     * @brief 更新旁路直播相关参数
     * @details SDK提供了更新旁路直播的功能，用于实时更新旁路直播参数
     * @param streamUrl 推流地址
     * @param transcoding 推流所需参数, 详细定义见{@link AliRtcLiveTranscodingParam}
     */
    updatePublishLiveStream(streamUrl: string, transcoding: AliRtcLiveTranscodingParam): Promise<any>;
    /**
     * @brief 更新旁路直播相关参数
     * @details SDK提供了更新旁路直播的功能，用于实时更新旁路直播参数
     * @param taskId 任务Id
     * @param transcoding 推流所需参数, 详细定义见{@link AliRtcLiveTranscodingParam}
     * @return
     * - 0 : 成功
     * - 非0 : 失败
     *   - ERR_INNER(-1) : SDK内部错误，可能的情况为SDK未初始化或者SDK销毁后调用
     *   - 其他 : 返回值参考官网错误码
     */
    updatePublishLiveStreamWithTaskId(taskId: string, transcoding: AliRtcLiveTranscodingParam): Promise<any>;
    /**
     * @brief 停止旁路直播
     * @details SDK提供了停止旁路直播的功能
     * @param streamUrl 推流地址
     */
    stopPublishLiveStream(streamUrl: string): Promise<any>;
    /**
     * @brief 停止旁路直播
     * @details SDK提供了停止旁路直播的功能
     * @param taskId 任务Id
     */
    stopPublishLiveStreamWithTaskId(taskId: string): Promise<any>;
    /**
     * 获取旁路直播状态
     * @param streamUrl 推流地址
     * @return 路直播状态
     */
    /**
     * @brief 设置是否启用媒体扩展信息
     * @param enabled 是否启用媒体扩展，需要在加入房间之前调用
     */
    setEnableMediaExtensionMsg(enabled: boolean): void;
    /**
     * @brief 发送媒体扩展信息
     * @details SDK提供了发送和接收媒体扩展信息的功能，接收端参考 {@link AliRtcEngineEventListener.mediaExtensionMsgReceived}，使用场景：
     * - 使用媒体扩展信息传递时间戳，计算端到端的网络延迟，或者跟自身其他业务做数据同步
     * - 使用媒体扩展信息传递描述信息。目前可以传递最多 4k Byte数据，可以用来传输少量数据，建议使用Json或者纯字符串；
     * - 内部使用SEI扩展协议来实现；
     *
     * @param message 媒体扩展信息，长度限制为最大4K字节
     * @param repeatCount 重复次数，代表消息冗余度，用于防止网络丢包导致的消息丢失，-1为无限传输
     *
     * @note 启用此能力会增加客户端性能开销，如有需求需要在加入房间之前调用 setEnableMediaExtensionMsg
     * @note 使用媒体扩展信息时需要复用音视频数据通道，因此必须控制自定义消息的发送频率和消息数据长度，使用限制如下：
     * - 每秒最多发送profile条消息，因为sei信息是放到h264/h265流中传输，有视频帧编码才能附加扩展信息；
     * - 为了不影响媒体数据的传输质量，自定义消息体长度限制为4k Byte，可以用来传输少量信息；
     * - sendMediaExtensionMsg函数中repeatCount参数为自定义消息冗余度，若大于1，则会发送多次，
     * - 防止网络丢包导致的消息丢失，此时房间里的其他人也会收到多次相同的消息，需要去重
     * - 发送的自定义消息，在旁路直播时，房间里的订阅者也一样会收到
     * - 同一时间只有一条MediaExtensionMsg在传输，多次调用sendMediaExtensionMsg，新调用的数据会覆盖前一次的数据；
     */
    sendMediaExtensionMsg(message: ArrayBuffer, repeatCount: number): Promise<void>;
    /**
     * @brief 发送媒体扩展信息
     * @details SDK提供了发送和接收媒体扩展信息的功能，接收端参考 {@link AliRtcEngineEventListener.mediaExtensionMsgReceived}，使用场景：
     * - 使用媒体扩展信息传递时间戳，计算端到端的网络延迟，或者跟自身其他业务做数据同步
     * - 使用媒体扩展信息传递描述信息。目前可以传递最多 4k Byte数据，可以用来传输少量数据，建议使用Json或者纯字符串；
     * - 内部使用SEI扩展协议来实现；
     *
     * @param message 媒体扩展信息，长度限制为最大4K字节
     * @param repeatCount 重复次数，代表消息冗余度，用于防止网络丢包导致的消息丢失，-1为无限传输
     * @param payloadType [5, 100..254]范围 payloadType=5等于使用sendMediaExtensionMsg 接口
     *
     * @note 启用此能力会增加客户端性能开销，如有需求需要在加入房间之前调用 setEnableMediaExtensionMsg
     * @note 使用媒体扩展信息时需要复用音视频数据通道，因此必须控制自定义消息的发送频率和消息数据长度，使用限制如下：
     * - 每秒最多发送profile条消息，因为sei信息是放到h264/h265流中传输，有视频帧编码才能附加扩展信息；
     * - 为了不影响媒体数据的传输质量，自定义消息体长度限制为4k Byte，可以用来传输少量信息；
     * - sendMediaExtensionMsg函数中repeatCount参数为自定义消息冗余度，若大于1，则会发送多次，
     * - 防止网络丢包导致的消息丢失，此时房间里的其他人也会收到多次相同的消息，需要去重
     * - 发送的自定义消息，在旁路直播时，房间里的订阅者也一样会收到
     * - 同一时间只有一条MediaExtensionMsg在传输，多次调用sendMediaExtensionMsg，新调用的数据会覆盖前一次的数据；
     */
    sendMediaExtensionMsgEx(message: ArrayBuffer, repeatCount: number, payloadType: number): Promise<void>;
    static Plugin: typeof AliRtcPlugin;
    /**
     * @brief 开启音视频处理算法插件
     * @details RTC采用插件件形式提供音视频增强及特效能力，可通过[官网组件文档](https://help.aliyun.com/document_detail/261183.html) 进行组件下载，参考插件集成文档集成进RTC引擎；
     * @param plugin 插件库
     * @param option 插件初始化参数值，请阅读插件库接入文档
     * @note
     * - 入会前入会后均可调用；
     * - 如果重复调用会报错；
     */
    enablePlugin(plugin: AliRtcPlugin, options?: any): void;
    /**
     * @brief 关闭音视频处理算法插件
     * @param name 插件名称，通过 plugin.name 获取
     * @note
     *  - 入会前入会后均可调用；
     *  - 重复调用返回错误；
     */
    removePlugin(name: string): void;
    /**
     * @brief 根据插件ID设置插件
     * @param name 插件名称，通过 plugin.name 获取
     * @param option 插件初始化参数值，请阅读插件库接入文档
     */
    setPluginOption(name: string, option: any): void;
    /**
     * @brief 根据插件ID获取插件内部信息
     * @param name 插件名称，通过 plugin.name 获取
     */
    getPluginOption(name: string): any;
    /**
     * 美颜功能是否有效
     * @return
     * - true: 美颜功能有效
     * - false: 无效
     */
    isBeautyAvailable(): boolean;
    /**
     * 美颜功能是否开启
     * @return
     * - true: 美颜功能已开启
     * - false: 已关闭
     */
    isBeautyEnabled(): boolean;
    /**
     * @brief 开启或关闭美颜功能, 并设置美颜效果参数
     * @param enable
     * - true: 开启美颜功能
     * - false: 关闭美颜功能
     * @param config 美颜效果参数
     */
    setBeautyEffect(enable: boolean, config: AliRtcBeautyConfig): void;
    /**
     * @brief 开启/关闭虚拟背景替换功能
     * @details 需要集成美颜插件；
     * @param enable 是否开启；
     * @param path 虚拟背景本地图片路径，支持JPG，PNG 格式；
     *
     * @note
     * - 与背景虚化为替换关系，如开启虚拟背景开关后再次开启背景虚化，则只有背景虚化效果；
     */
    enableBackgroundExchange(enable: boolean, path: string): void;
    /**
     * @brief 开启/关闭虚拟背景虚化功能
     * @details 需要集成美颜插件；
     *  需要集成美颜插件，背景替换与背景虚化不可同时使用，如开启虚拟背景开关后再次开启背景虚化，则只有背景虚化效果；
     * @param enable 是否开启；
     *
     * @note
     * - 与虚拟背景为替换关系，如开启背景虚化开关后再次开启虚拟背景，则只有虚拟背景效果；
     */
    enableBackgroundBlur(enable: boolean): void;
    setParameter(parameter: string): void;
    getParameter(): string;
    /**
     * @brief 发送 DataChannel 消息
     * @param dataChannelMsg 伴奏控制消息
     */
    sendDataChannelMessage(dataChannelMsg: AliRtcDataChannelMsg): void;
}

declare interface AliRtcEngineConfig {
    env?: AliRtcEnv;
    webTrack?: boolean | AliRtcWebTrackConfig;
    maxSignalingConnectRetryCount?: number;
}

declare interface AliRtcEngineEventListener {
    /**
     * @brief 网络连接状态改变
     * @param status 连接状态
     * @param reason 连接状态变化原因
     */
    connectionStatusChange: (status: AliRtcConnectionStatus, reason: AliRtcConnectionStatusChangeReason) => void;
    /**
     * @brief 本地设备异常回调
     * @param deviceType 设备类型, 参考{@link AliRtcEngine.AliRtcEngineLocalDeviceType}
     * @param exceptionType 设备异常类型, 参考{@link AliRtcEngine.AliRtcEngineLocalDeviceExceptionType}
     * @param msg 异常时携带的信息
     * @note 此回调标识了内部无法恢复了设备异常，收到此回调时用户需要检查设备是否可用
     */
    localDeviceException: (localDeviceType: AliRtcEngineLocalDeviceType, localDeviceExceptionType: AliRtcEngineLocalDeviceExceptionType, description: string) => void;
    /**
     * @brief 用户鉴权信息即将过期通知，收到后30秒鉴权过期
     * @note 该回调在鉴权信息30秒前触发，收到该回调后应该及时更新鉴权信息，参考 {@link AliRtcEngine.refreshAuthInfo}
     */
    /**
     * 被服务器踢出/会议结束频道的消息
     * @param code 原因参考{@link AliRtcEngine.AliRtcOnByeType}
     */
    bye: (code: AliRtcOnByeType) => void;
    /**
     * @brief 远端用户（通信模式）/（互动模式，主播角色）加入频道回调
     * @details 该回调在以下场景会被触发
     * - 通信模式：远端用户加入频道会触发该回调，如果当前用户在加入频道前已有其他用户在频道中，当前用户加入频道后也会收到已加入频道用户的回调
     * - 互动模式：
     *   - 远端主播角色用户加入频道会触发该回调，如果当前用户在加入频道前已有其他主播在频道中，当前用户加入频道后也会收到已加入频道主播的回调
     *   - 远端观众角色调用 {@link AliRtcEngine.setClientRole} 切换为主播角色 {@link AliRtcEngine.AliRtcSdkClientRole.AliRtcSdkInteractive}，同时设置了推流时，会触发该回调
     *
     * @param uid 用户ID，从App server分配的唯一标示符
     * @param elapsed 用户加入频道时的耗时
     * @note 互动模式下回调行为
     * - 主播间可以互相收到加入频道回调
     * - 观众可以收到主播加入频道回调
     * - 主播无法收到观众加入频道回调
     */
    remoteUserOnLineNotify: (uid: string, elapsed: number) => void;
    /**
     * @brief 远端用户（通信模式）/（互动模式，主播角色）离开频道回调
     * @details 该回调在以下场景会被触发
     * - 通信模式：远端用户离开频道会触发该回调
     * - 互动模式：
     *   - 远端主播角色 {@link AliRtcEngine.AliRtcSdkClientRole.AliRtcSdkInteractive} 离开频道
     *   - 远端主播切换调用 {@link AliRtcEngine.setClientRole} 切换为观众角色 {@link AliRtcEngine.AliRtcSdkClientRole.AliRtcSdkLive}，会触发该回调
     * - 通信模式和互动模式主播角色情况下，当长时间收不到远端用户数据，超时掉线时，会触发该回调
     *
     * @param uid 用户ID，从App server分配的唯一标示符
     * @param reason 用户离线的原因，详见 {@link AliRtcEngine.AliRtcUserOfflineReason}
     */
    remoteUserOffLineNotify: (uid: string, aliRtcUserOfflineReason: AliRtcUserOfflineReason) => void;
    /**
     * @brief 远端用户的音视频流发生变化回调
     * @details 该回调在以下场景会被触发
     * - 当远端用户从未推流变更为推流（包括音频和视频）
     * - 当远端用户从已推流变更为未推流（包括音频和视频）
     * - 互动模式下，调用 {@link AliRtcEngine.setClientRole} 切换为主播角色 {@link AliRtcEngine.AliRtcSdkClientRole.AliRtcSdkInteractive}，同时设置了推流时，会触发该回调
     *
     * @param uid 用户ID，从App server分配的唯一标示符
     * @param audioTrack 音频流类型，详见 {@link AliRtcEngine.AliRtcAudioTrack}
     * @param videoTrack 视频流类型，详见 {@link AliRtcEngine.AliRtcVideoTrack}
     * @note 该回调仅在通信模式用户和互动模式下的主播角色才会触发
     */
    remoteTrackAvailableNotify: (uid: string, aliRtcAudioTrack: AliRtcAudioTrack, aliRtcVideoTrack: AliRtcVideoTrack) => void;
    /**
     * @brief 远端用户静音/取消静音回调
     * @param uid 远端用户ID
     * @param isMute 该用户是否静音
     * - true: 静音
     * - false: 取消静音
     */
    userAudioMuted: (uid: string, isMute: boolean) => void;
    /**
     * @brief 对端用户发送视频黑帧数据发送通知
     * @param uid 执行muteVideo的用户ID
     * @param isMute
     * - true: 推流黑帧
     * - false: 正常推流
     * @note 该接口用于对端用户发送视频黑帧数据时的回调
     */
    userVideoMuted: (uid: string, isMute: boolean) => void;
    /**
     * @brief 对端用户发送屏幕黑帧数据发送通知
     * @param uid 执行muteVideo的用户ID
     * @param isMute
     * - true: 推流黑帧
     * - false: 正常推流
     * @note 该接口用于对端用户发送屏幕黑帧数据时的回调
     */
    userScreenMuted: (uid: string, isMute: boolean) => void;
    /**
     * 音量反馈
     * @param {Array} speakers 用户音量对象数组
     * @param {string} speaker.userId 用户ID
     * @param {number} speaker.volume 音量
     * @note userId 为空字符串时代表是本地麦克风采集的音量
     */
    audioVolume: (speakers: {
        userId: string;
        volume: number;
    }[]) => void;
    /**
     * 发布视频流的数据统计
     * @param aliRtcStats 统计信息
     */
    aliRtcStats: (aliRtcStats: any) => void;
    /**
     * 音频推流变更回调
     * @param track 流标识
     * @param oldState 之前的推流状态
     * @param newState 当前的推流状态
     * @param elapseSinceLastState 状态变更时间间隔(毫秒)
     * @param channel 当前频道
     */
    audioPublishStateChanged: (oldState: AliRtcPublishState, newState: AliRtcPublishState, elapseSinceLastState: number, channel: string) => void;
    /**
     * 音频订阅情况变更回调
     * @param uid 用户ID
     * @param oldState 之前的订阅状态
     * @param newState 当前的订阅状态
     * @param elapseSinceLastState 状态变更时间间隔(毫秒)
     * @param channel 当前频道
     */
    audioSubscribeStateChanged: (uid: string, oldState: AliRtcSubscribeState, newState: AliRtcSubscribeState, elapseSinceLastState: number, channel: string) => void;
    /**
     * 视频推流变更回调
     * @param oldState 之前的推流状态
     * @param newState 当前的推流状态
     * @param elapseSinceLastState 状态变更时间间隔(毫秒)
     * @param channel 当前频道
     */
    videoPublishStateChanged: (oldState: AliRtcPublishState, newState: AliRtcPublishState, elapseSinceLastState: number, channel: string) => void;
    /**
     * 次要流推流变更回调
     * @param oldState 之前的推流状态
     * @param newState 当前的推流状态
     * @param elapseSinceLastState 状态变更时间间隔(毫秒)
     * @param channel 当前频道
     */
    dualStreamPublishStateChanged: (oldState: AliRtcPublishState, newState: AliRtcPublishState, elapseSinceLastState: number, channel: string) => void;
    /**
     * 屏幕分享推流变更回调
     * @param oldState 之前的推流状态
     * @param newState 当前的推流状态
     * @param elapseSinceLastState 状态变更时间间隔(毫秒)
     * @param channel 当前频道
     */
    screenSharePublishStateChanged: (oldState: AliRtcPublishState, newState: AliRtcPublishState, elapseSinceLastState: number, channel: string) => void;
    /**
     * DataChannel 推流变更回调
     * @param oldState 之前的推流状态
     * @param newState 当前的推流状态
     * @param elapseSinceLastState 状态变更时间间隔(毫秒)
     * @param channel 当前频道
     */
    dataPublishStateChanged: (oldState: AliRtcPublishState, newState: AliRtcPublishState, elapseSinceLastState: number, channel: string) => void;
    /**
     * 相机流订阅情况变更回调
     * @param uid 用户ID
     * @param oldState 之前的订阅状态
     * @param newState 当前的订阅状态
     * @param elapseSinceLastState 状态变更时间间隔(毫秒)
     * @param channel 当前频道
     */
    videoSubscribeStateChanged: (uid: string, oldState: AliRtcSubscribeState, newState: AliRtcSubscribeState, elapseSinceLastState: number, channel: string) => void;
    /**
     * 大小流订阅情况变更回调
     * @param uid 用户ID
     * @param oldStreamType 之前的订阅状态
     * @param newStreamType 当前的订阅状态
     * @param elapseSinceLastState 状态变更时间间隔(毫秒)
     * @param channel 当前频道
     */
    subscribeStreamTypeChanged: (uid: string, oldStreamType: AliRtcVideoStreamType, newStreamType: AliRtcVideoStreamType, elapseSinceLastState: number, channel: string) => void;
    /**
     * 屏幕分享流订阅情况变更回调
     * @param uid 用户ID
     * @param oldState 之前的订阅状态
     * @param newState 当前的订阅状态
     * @param elapseSinceLastState 状态变更时间间隔(毫秒)
     * @param channel 当前频道
     */
    screenShareSubscribeStateChanged: (uid: string, oldState: AliRtcSubscribeState, newState: AliRtcSubscribeState, elapseSinceLastState: number, channel: string) => void;
    /**
     * DataChannel 订阅情况变更回调
     * @param uid 用户ID
     * @param oldState 之前的订阅状态
     * @param newState 当前的订阅状态
     * @param elapseSinceLastState 状态变更时间间隔(毫秒)
     * @param channel 当前频道
     */
    dataSubscribeStateChanged: (uid: string, oldState: AliRtcSubscribeState, newState: AliRtcSubscribeState, elapseSinceLastState: number, channel: string) => void;
    /**
     * 收到媒体扩展信息回调
     * @param uid 用户ID，媒体信息发送用户的ID
     * @param message 媒体扩展信息
     * @param payloadType payload类型，sendMediaExtensionMsg返回5，sendMediaExtensionMsgEx发送的会返回具体类型
     * @note 当一端通过 {@link AliRtcEngine.sendMediaExtensionMsg} 发送信息后，其他端通过该回调接收数据
     */
    mediaExtensionMsgReceived: (uid: string, message: ArrayBuffer, payloadType: number) => void;
    /**
     * @brief 用户鉴权信息即将过期通知，收到后30秒鉴权过期
     * @note 该回调在鉴权信息30秒前触发，收到该回调后应该及时更新鉴权信息，参考 {@link AliRtcEngine.refreshAuthInfo:}
     */
    authInfoWillExpire: () => void;
    /**
     * @brief 用户调用需要鉴权的接口，服务端返回信息过期
     * @note 该回调触发代表鉴权信息已过期，想要继续在会中，需要重新入会，参考 {@link AliRtcEngine.joinChannel:name:onResultWithUserId:}
     */
    authInfoExpired: () => {};
    /**
     * 远程用户的音频自动播放失败
     * @param uid
     * @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
     */
    remoteAudioAutoPlayFail: (uid: string) => void;
    /**
     * 远程用户的视频自动播放失败
     * @param uid
     * @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;
    /**
     * 远程用户的音频播放出错
     * @param uid 用户ID
     * @param reason 出错原因，可选
     */
    remoteAudioPlayError: (uid: string, reason: string | undefined) => void;
    /**
     * 远程用户的视频播放出错
     * @param uid 用户ID
     * @param reason 出错原因，可选
     */
    remoteVideoPlayError: (uid: string, reason: string | undefined) => 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
     */
    rtcLocalVideoStats: (stats: AliRtcLocalVideoStats[]) => void;
    /**
     * 发布音频流的数据统计
     * @param aliRtcStats 统计信息
     */
    rtcLocalAudioStats: (stats: AliRtcLocalAudioStats[]) => void;
    /**
     * 订阅视频流的数据统计
     * @param aliRtcStats 统计信息
     */
    rtcRemoteVideoStats: (stats: AliRtcRemoteVideoStats[]) => void;
    /**
     * 订阅远端音频流的数据统计
     * @param aliRtcStats 统计信息
     */
    rtcRemoteAudioStats: (stats: AliRtcRemoteAudioStats[]) => void;
    /**
     * @brief 错误通知
     * @deprecated 请使用 occurError
     * @details 如果engine出现error，通过这个回调通知app
     * @param error  错误类型，参考 {@link AliRtcError}
     * @param uid 用户 Id，不一定存在
     */
    onOccurError: (error: AliRtcError, uid?: string) => void;
    /**
     * @brief 错误通知
     * @details 如果engine出现error，通过这个回调通知app
     * @param error  错误类型，参考 {@link AliRtcError}
     * @param uid 用户 Id，不一定存在
     */
    occurError: (error: AliRtcError, uid?: string) => void;
    /**
     * @brief 收到数据通道消息
     * @param uid 用户ID
     * @param message 消息
     */
    dataChannelMsg: (uid: string, message: AliRtcDataChannelMsg) => void;
}

export declare enum AliRtcEngineLocalDeviceExceptionType {
    /** 未知异常类型 */
    AliEngineLocalDeviceExceptionTypeUnknown = 0,
    /** 麦克风打开失败 */
    AliEngineLocalDeviceExceptionTypeMicOpenFail = 1,
    /** 麦克风被打断 */
    AliEngineLocalDeviceExceptionTypeMicInterrupt = 2,
    /** 麦克风无权限 */
    AliEngineLocalDeviceExceptionTypeMicAuthFail = 3,
    /** 无可用麦克风 */
    AliEngineLocalDeviceExceptionTypeMicNotAvailable = 4,
    /** 扬声器打开失败 */
    AliEngineLocalDeviceExceptionTypeSpeakerOpenFail = 5,
    /** 扬声器被打断 */
    AliEngineLocalDeviceExceptionTypeSpeakerInterrupt = 6,
    /** 无可用扬声器 */
    AliEngineLocalDeviceExceptionTypeSpeakerNotAvailable = 7,
    /** 音频设备异常 */
    AliEngineLocalDeviceExceptionTypeAudioDeviceException = 8,
    /** 摄像头打开失败 */
    AliEngineLocalDeviceExceptionTypeCameraOpenFail = 9,
    /** 摄像头被打断 */
    AliEngineLocalDeviceExceptionTypeCameraInterrupt = 10,
    /** 摄像头无权限 */
    AliEngineLocalDeviceExceptionTypeCameraAuthFail = 11,
    /** 显示设备异常 */
    AliEngineLocalDeviceExceptionTypeDisplayException = 12,
    /** 视频设备异常 */
    AliEngineLocalDeviceExceptionTypeVideoDeviceException = 13
}

export declare enum AliRtcEngineLocalDeviceType {
    /** 未知设备类型 */
    AliEngineLocalDeviceTypeUnknown = 0,
    /** 麦克风设备 */
    AliEngineLocalDeviceTypeMic = 1,
    /** 扬声器设备 */
    AliEngineLocalDeviceTypeSpeaker = 2,
    /** 音频设备 */
    AliEngineLocalDeviceTypeAudioDevice = 3,
    /** 摄像头设备 */
    AliEngineLocalDeviceTypeCamera = 4,
    /** 显示设备 */
    AliEngineLocalDeviceTypeDisplay = 5,
    /** 视频设备 */
    AliEngineLocalDeviceTypeVideoDevice = 6
}

declare enum AliRtcEnv {
    pre = "pre",
    prod = "prod"
}

declare class AliRtcError extends Error {
    _code: number;
    _extra?: any;
    static fromRtsError(error: any): AliRtcError;
    static fromError(error: any, code?: number, fallbackMessage?: string): AliRtcError;
    constructor(code: number, message: string, extra?: any);
    get code(): number;
    set code(code: number);
    get reason(): string;
    get extra(): any;
}

export declare enum AliRtcErrorCode {
    /****************************************************
     * 入会错误码
     ****************************************************/
    /** 已经加入频道 */
    ERR_JOIN_ALREADY_JOINED = 16843521,
    /** AppId不存在，请在控制台创建应用 */
    ERR_JOIN_BAD_APPID = 33620481,
    /** AppId已失效，请在控制台重新启用应用 */
    ERR_JOIN_INVALID_APPID = 33620482,
    /**
     * 频道不存在，请尝试重新加入频道
     * 如会议发起者入会后又很快离会，第二个人随后入会，发现会议已经不存在了。
     */
    ERR_JOIN_BAD_CHANNEL = 33620484,
    /** 频道已经失效（目前频道有效期为48小时），请重新生成频道鉴权令牌（Token） */
    ERR_JOIN_INVALID_CHANNEL = 33620483,
    /** Token无效。 请重新生成频道鉴权令牌（Token） */
    ERR_JOIN_BAD_TOKEN = 33620485,
    /** 加入频道超时，请检查网络连接是否正常 */
    ERR_JOIN_TIMEOUT = 16908804,
    /** 参数错误，{@link AliRtcAuthInfo}中的字段为空或者timestamp<=0，请检查字段是否完整 */
    ERR_JOIN_ERR_JOIN_BAD_PARAMBAD = 16974081,
    /** 加入频道失败 */
    ERR_JOIN_CHANNEL_FAILED = 16974338,
    /****************************************************
     * 日志模块错误码
     ****************************************************/
    /** 日志模块已经初始化 */
    ERR_LOG_ALREADY_INIT = 16974085,
    /** 日志模块根路径错误，请检查路径是否存在或路径是否具有写入权限 */
    ERR_LOG_ROOT_PATH_ERROR = 16974086,
    /****************************************************
     * 音频设备错误码
     ****************************************************/
    /**
     * 采集设备初始化失败、采集设备媒体服务异常、音频设备被占用。
     * - Mac端和Windows端查看系统设置中音频设备是否有能量条
     * - iOS端和Android端确认是否被其他应用占用
     * - 如果都正常，请重启设备
     */
    ERR_MIC_OPEN_FAIL = 17040388,
    /**
     * 播放设备初始化失败、播放设备媒体服务异常、音频设备被占用。
     * - Mac端和Windows端查看系统设置中音频设备是否有能量条
     * - iOS端和Android端确认是否被其他应用占用
     * - 如果都正常，请重启设备
     */
    ERR_SPEAKER_OPEN_FAIL = 17040389,
    /** 系统性能不足、采集设备媒体服务异常、音频设备被占用。建议重启应用或重启设备 */
    ERR_MIC_INTERRUPT = 17040390,
    /** 系统性能不足，播放设备媒体服务异常，音频设备被占用。建议重启应用或重启设备 */
    ERR_SPEAKER_INTERRUPT = 17040391,
    /** 麦克风设备未授权，建议去系统设置中，打开麦克风授权，并且重新加入频道 */
    ERR_MIC_AUTH_FAIL = 17040392,
    /** 无可用的音频采集设备，检查是否接入了可用的麦克风，并检查系统设置中，该设备是否被禁用 */
    ERR_MIC_NOT_AVAILABLE = 17040393,
    /** 无可用的音频播放设备，检查是否接入了可用的扬声器或耳机，并检查系统设置中，该设备是否被禁用 */
    ERR_SPEAKER_NOT_AVAILABLE = 17040400,
    /****************************************************
     * 视频设备错误码
     ****************************************************/
    /** 采集设备初始化失败，检查系统相机是否可用，移动端App是否占用了相机 */
    ERR_CAMERA_OPEN_FAIL = 17039620,
    /** 采集设备未授权，建议去系统设置中，打开摄像头授权，并且重新加入频道 */
    ERR_CAMERA_AUTH_FAIL = 17039621,
    /** 采集过程中出现异常，Mac、Winodows电脑相机采集灯熄灭，设备被强制中断。建议重新加入频道或者重启App */
    ERR_CAMERA_INTERRUPT = 17039622,
    /** 无可用的视频采集设备，检查是否接入了可用的摄像头，并检查系统设置中，该设备是否被禁用 */
    ERR_CAMERA_NOT_AVAILABLE = 17039623,
    /** 渲染设备初始化失败，检查Windows端显卡驱动是否需要升级，移动端手机型号或者Android平板盒子等外设是否支持OpenGL */
    ERR_VIDEO_DISPLAY_OPEN_FAIL = 17039873,
    /** 渲染过程中出现异常，系统异常错误导致渲染引擎报错，重新加入频道或者重启App */
    ERR_VIDEO_DISPLAY_INTERRUPT = 1064992,
    /****************************************************
     * 网络错误码
     ****************************************************/
    /** 媒体通道建立失败，检查网络连接是否正常 */
    ERR_ICE_CONNECTION_CONNECT_FAIL = 17105409,
    /** 媒体通道重连失败，检查网络连接是否正常 */
    ERR_ICE_CONNECTION_RECONNECT_FAI = 17105410,
    /** 媒体通道mobility失败，应用层需要重走入会流程 */
    ERR_ICE_CONNECTION_MOBILITY_FAIL = 17105411,
    /** 信令心跳超时，检查网络连接是否正常 */
    ERR_ICE_CONNECTION_HEARTBEAT_TIMEOUT = 16908812,
    /**
     * 低延时互动直播模式相关错误码
     */
    /** 低延时互动直播信令失败 */
    ERR_LATENCY_LIVE_COMMUNICATION_FAILED = 17235971,
    /** 低延时互动直播媒体设置错误 */
    ERR_LATENCY_LIVE_MEIDASETTING_FAILED = 17235972,
    /** 低延时互动直播当前房间没有主播推流 */
    ERR_LATENCY_LIVE_NONE_BROADCASTER = 17235973,
    /** 低延时互动直播错误，应用层需要重新创建新的SDK实例 */
    ERR_LATENCY_LIVE_NEED_RESTART = 17235974,
    /** 低延时互动直播错误，应用层需要稍等一定时间（自定义间隔）重新创建新的SDK实例 */
    ERR_LATENCY_LIVE_NEED_RESTART_AGAIN_LATER = 17235975,
    /** 低延迟互动直播服务不可用或者服务降级，客户端应该选择其它方式拉超大方会流，即不从cdn sfu拉流 */
    ERR_LATENCY_LIVE_SERVICE_UNAVAILABLE = 17235976,
    /** 低延迟互动直播网络中断错误，销毁当前sdk实例重新创建sdk后重新走订阅流程 */
    ERR_LATENCY_LIVE_MEDIACONNECTION_INTERRUPTION = 17235977,
    /****************************************************
     * 推流相关错误码
     ****************************************************/
    /** 推流无效 */
    ERR_SDK_PUBLISH_INVAILD = 16974597,
    /** 未进入频道推流失败 */
    ERR_SDK_PUBLISH_NOT_JOIN_CHANNEL = 16843782,
    /** 推送音频流失败 */
    ERR_SDK_PUBLISH_AUDIO_STREAM_FAILED = 16843856,
    /** 推送视频流失败 */
    ERR_SDK_PUBLISH_VIDEO_STREAM_FAILED = 16843857,
    /** 推送小流失败 */
    ERR_SDK_PUBLISH_DUAL_STREAM_FAILED = 16843858,
    /** 推送屏幕共享失败 */
    ERR_SDK_PUBLISH_SCEEN_SHARE_FAILED = 16843859,
    /** 屏幕共享配置错误 */
    ERR_SDK_PUBLISH_SCREEN_SHARE_CONFIG_ERROR = 16843860,
    /****************************************************
     * 订阅相关错误码
     ****************************************************/
    /** 订阅无效 */
    ERR_SDK_SUBSCRIBE_INVAILD = 16974852,
    /** 未进入频道订阅错误 */
    ERR_SDK_SUBSCRIBE_NOT_JOIN_CHANNEL = 16844112,
    /** 订阅音频流失败 */
    ERR_SDK_SUBSCRIBE_AUDIO_STREAM_FAILED = 16844113,
    /** 订阅视频流失败 */
    ERR_SDK_SUBSCRIBE_VIDEO_STREAM_FAILED = 16844114,
    /** 订阅小流失败 */
    ERR_SDK_SUBSCRIBE_DUAL_STREAM_FAILED = 16844115,
    /** 订阅屏幕共享失败 */
    ERR_SDK_SUBSCRIBE_SCREEN_SHARE_FAILED = 16844116,
    /** 订阅 DataChannel 失败 */
    ERR_SDK_SUBSCRIBE_DATA_FAILED = 16844117,
    /** 订阅 DataChannel 过程中报错 */
    ERR_SDK_SUBSCRIBE_DATA_ERROR = 16844118,
    /****************************************************
     * 其他错误码
     ****************************************************/
    /** SDK状态错误，建议销毁SDK重新创建实例。 */
    ERR_SDK_INVALID_STATE = 16974340,
    /** 参数不匹配 */
    ERR_INVALID_ARGUMENTS = 16974083,
    /** Session已经被移除，建议销毁实例之后重新创建实例并加入频道。 */
    ERR_SESSION_REMOVED = 33620229,
    /** 互动模式下设置角色错误 */
    ERR_SDK_UPDATE_ROLE_CHANNEL = 16908801,
    /** SDK内部错误 */
    ERR_INNER = -1,
    /************************************************************************************/
    /******************************** 旁路直播错误码 start ********************************/
    /************************************************************************************/
    /**
     * MPU任务发布成功
     */
    ERR_SDK_MPU_TASK_PUBLISH_OK = 0,
    /**
     * MPU任务流未找到
     */
    ERR_SDK_MPU_TASK_STREAM_NOT_FOUND = 17825793,
    /**
     * MPU任务流已存在
     */
    ERR_SDK_MPU_TASK_STREAM_ALREADY_EXIST = 17825794,
    /**
     * MPU任务无效参数
     */
    ERR_SDK_MPU_TASK_INVALID_PARAM = 17825795,
    /**
     * MPU任务内部错误
     */
    ERR_SDK_MPU_TASK_INTERNAL_ERROR = 17825796,
    /**
     * MPU任务rtmp服务错误
     */
    ERR_SDK_MPU_TASK_RTMP_SERVER_ERROR = 17825797,
    /**
     * MPU任务rtmp流地址错误
     */
    ERR_SDK_MPU_TASK_RTMP_STREAM_URL_ERROR = 17825798,
    /**
     * MPU任务发布超时
     */
    ERR_SDK_MPU_TASK_PUBLISH_TIMEOUT = 17825799,
    /**
     * MPU任务发布成功
     */
    ERR_SDK_MPU_TASK_NOT_AUTHORIZED = 17825800,
    /************************************************************************************/
    /******************************** 旁路直播错误码 end ********************************/
    /************************************************************************************/
    /************************************************************************************/
    /******************************** 跨频道转推错误码 start ******************************/
    /************************************************************************************/
    /**
     * 检测到本地状态不对，没入会，角色错误，频道模式错误
     */
    ERR_SDK_CHANNEL_RELAY_STATE_ERROR = 18874369,
    /**
     * 服务端返回436，源通道模式不匹配
     */
    ERR_SDK_CHANNEL_RELAY_SRC_NOT_ALLOW = 18874370,
    /**
     * 服务端返回437，目标通道不存在或者模式不匹配
     */
    ERR_SDK_CHANNEL_RELAY_JOIN_DEST_FAILED = 18874371,
    /**
     * 服务端返回438，token无效
     */
    ERR_SDK_CHANNEL_RELAY_TOKEN_INVALID = 18874372,
    /**
     * 服务端返651，session不是主播
     */
    ERR_SDK_CHANNEL_RELAY_ROLE_ERROR = 18874373,
    /**
     * 服务端400，参数错误
     */
    ERR_SDK_CHANNEL_RELAY_INVALID_PARAM = 18874374,
    /**
     * 服务端返回440，不能转推到本频道
     */
    ERR_SDK_CHANNEL_RELAY_TO_SELF_ERROR = 18874375
}

declare class AliRtcLiveTranscoding {
    authInfo?: AliRtcAuthInfo;
    private slsReporter;
    constructor(slsReporter: SLSReporter);
    private doRequest;
    setAuthInfo(authInfo: AliRtcAuthInfo): void;
    clear(): void;
    startPublishLiveStream(streamUrl: string, transcodingParam: AliRtcLiveTranscodingParam): Promise<any>;
    startPublishLiveStreamWithTaskId(taskId: string, transcodingParam: AliRtcLiveTranscodingParam): Promise<any>;
    updatePublishLiveStream(streamUrl: string, transcodingParam: AliRtcLiveTranscodingParam): Promise<any>;
    updatePublishLiveStreamWithTaskId(taskId: string, transcodingParam: AliRtcLiveTranscodingParam): Promise<any>;
    stopPublishLiveStream(streamUrl: string): Promise<any>;
    stopPublishLiveStreamWithTaskId(taskId: string): Promise<any>;
    refreshAuthInfo(authInfo: AliRtcRefreshAuthInfo): void;
}

export declare enum AliRtcLiveTranscodingAudioSampleRate {
    LiveTranscoding_HZ_48000 = 48000,
    LiveTranscoding_HZ_44100 = 44100,
    LiveTranscoding_HZ_32000 = 32000,
    LiveTranscoding_HZ_16000 = 16000,
    LiveTranscoding_HZ_8000 = 8000
}

export declare enum AliRtcLiveTranscodingCropMode {
    /*! 缩放模式 */
    LiveTranscodingOrigin = 0,
    /*! 剪裁 */
    LiveTranscodingCrop = 1,
    /*! 填充 */
    LiveTranscodingFill = 2
}

export declare class AliRtcLiveTranscodingEncodeParam {
    videoWidth: number;
    videoHeight: number;
    videoFramerate: number;
    videoBitrate: number;
    videoGop: number;
    audioSamplerate: AliRtcLiveTranscodingAudioSampleRate;
    audioBitrate: number;
    audioChannels: number;
    constructor(videoWidth: number, videoHeight: number, videoFramerate?: number, videoBitrate?: number, videoGop?: number, audioSamplerate?: AliRtcLiveTranscodingAudioSampleRate, audioBitrate?: number, audioChannels?: number);
}

export declare enum AliRtcLiveTranscodingErrorCode {
    LiveTranscodingErrorPublishOk = 0,
    LiveTranscodingErrorStreamNotFound = 17825793,
    LiveTranscodingErrorStreamAlreadyExist = 17825794,
    LiveTranscodingErrorInvalidParam = 17825795,
    LiveTranscodingErrorInternalError = 17825796,
    LiveTranscodingErrorRtmpServerError = 17825797,
    LiveTranscodingErrorRtmpStreamUrlError = 17825798,
    LiveTranscodingErrorPublishTimeout = 17825799,
    LiveTranscodingErrorNotAuthorized = 17825800
}

export declare enum AliRtcLiveTranscodingFontType {
    NOTO_SERIF_CJKSC_REGULAR = 0,
    ALIBABA_PUHUITI_REGULAR = 1,
    ALIBABA_PUHUITI_BOLD = 2,
    ALIBABA_PUHUITI_Heavy = 3,
    ALIBABA_PUHUITI_LIGHT = 4,
    ALIBABA_PUHUITI_MEDIUM = 5
}

export declare enum AliRtcLiveTranscodingMediaProcessMode {
    /*! 通用模式 */
    LiveTranscodingNormal = 0,
    /*! 虚拟背景模式 */
    LiveTranscodingVirtualBackground = 1
}

export declare enum AliRtcLiveTranscodingMixMode {
    /*! 单路模式 */
    LiveTranscodingSingle = 0,
    /*! 混流模式 */
    LiveTranscodingMix = 1
}

export declare class AliRtcLiveTranscodingMixParam {
    taskProfile: AliRtcLiveTranscodingTaskProfile;
    encodeParam?: AliRtcLiveTranscodingEncodeParam;
    users: AliRtcTranscodingUser[];
    backgroundColor: number;
    backgrounds: AliRtcTranscodingImage[];
    watermarks: AliRtcTranscodingImage[];
    clockWidgets: AliRtcTranscodingClockWidget[];
    cropMode?: AliRtcLiveTranscodingCropMode;
    mediaProcessMode?: AliRtcLiveTranscodingMediaProcessMode;
    constructor(taskProfile: AliRtcLiveTranscodingTaskProfile);
}

export declare class AliRtcLiveTranscodingParam {
    /*! 旁路模式 */
    mixMode: AliRtcLiveTranscodingMixMode;
    /*! 路模式单路参数 */
    singleParam?: AliRtcLiveTranscodingSingleParam;
    /*! 旁路模式混流参数 */
    mixParam?: AliRtcLiveTranscodingMixParam;
    constructor(mixMode?: AliRtcLiveTranscodingMixMode, singleParam?: AliRtcLiveTranscodingSingleParam, mixParam?: AliRtcLiveTranscodingMixParam);
    toJson(type: MPU_TASK_TRANSACTION_TYPE, authInfo: AliRtcAuthInfo, taskId: string, streamUrl?: string, messageId?: string): {
        [key: string]: any;
    };
}

export declare enum AliRtcLiveTranscodingSegmentType {
    /*! 无人像分割 */
    LiveTranscodingNoBody = 0,
    /*! 人像分割 */
    LiveTranscodingBody = 1
}

export declare class AliRtcLiveTranscodingSingleParam {
    userId?: string;
    streamType?: AliRtcLiveTranscodingStreamType;
    sourceType?: AliRtcLiveTranscodingSourceType;
    constructor(uid?: string, streamType?: AliRtcLiveTranscodingStreamType, sourceType?: AliRtcLiveTranscodingSourceType);
}

export declare enum AliRtcLiveTranscodingSourceType {
    /*! 相机流 */
    LiveTranscodingCamera = 0,
    /*! 屏幕流 */
    LiveTranscodingShareScreen = 1
}

export declare enum AliRtcLiveTranscodingState {
    LiveTranscodingState_IDLE = 0,
    LiveTranscodingState_CONNNECT = 1,
    LiveTranscodingState_RUNNING = 2,
    LiveTranscodingState_RECOVERING = 3,
    LiveTranscodingState_FAILURE = 4,
    LiveTranscodingState_END = 5
}

export declare enum AliRtcLiveTranscodingStreamType {
    /*! 原始流 */
    LiveTranscodingOrigin = 0,
    /*! 音频 */
    LiveTranscodingAudio = 1,
    /*! 视频 */
    LiveTranscodingVideo = 2
}

export declare enum AliRtcLiveTranscodingTaskProfile {
    LiveTranscoding_Profile_1IN_1080P = 0,
    LiveTranscoding_Profile_1IN_720P = 1,
    LiveTranscoding_Profile_1IN_360P = 2,
    LiveTranscoding_Profile_2IN_1080P = 3,
    LiveTranscoding_Profile_2IN_720P = 4,
    LiveTranscoding_Profile_2IN_360P = 5,
    LiveTranscoding_Profile_4IN_1080P = 6,
    LiveTranscoding_Profile_4IN_720P = 7,
    LiveTranscoding_Profile_4IN_360P = 8,
    LiveTranscoding_Profile_9IN_1080P = 9,
    LiveTranscoding_Profile_9IN_720P = 10,
    LiveTranscoding_Profile_9IN_360P = 11,
    LiveTranscoding_Profile_12IN_1080P = 12,
    LiveTranscoding_Profile_12IN_720P = 13,
    LiveTranscoding_Profile_12IN_360P = 14,
    LiveTranscoding_Profile_16IN_1080P = 15,
    LiveTranscoding_Profile_16IN_720P = 16,
    LiveTranscoding_Profile_16IN_360P = 17,
    LiveTranscoding_Profile_Mixed = 9999
}

declare interface AliRtcLocalAudioStats {
    track: AliRtcAudioTrack;
    stats: any;
}

export declare class AliRtcLocalStreamInfo extends default_2<AliRtcLocalStreamListener> {
    type: AliRtcRawDataStreamType;
    originVideoTrack?: MediaStreamTrack;
    videoSource?: VideoStreamSource;
    private _videoMuted;
    originAudioTrack?: MediaStreamTrack;
    audioSource?: AudioStreamSource;
    private _audioMuted;
    private _targetVideoTrack?;
    private _targetDualVideoTrack?;
    private _targetAudioTrack?;
    private _publishVideoStream?;
    private _publishDualVideoStream?;
    private _publishAudioStream?;
    private _previewStream?;
    plugins: AliRtcPlugin[];
    private _profileManager?;
    cameraVideoConstraints?: IVideoConstraints;
    micAudioConstraints?: IAudioConstraints;
    private get profileManager();
    get audioProfile(): AudioProfileKey | undefined;
    constructor(type: AliRtcRawDataStreamType);
    private onVideoTrackEnded;
    private onAudioTrackEnded;
    updateSource(newStream: LocalStream, newVideoSource?: VideoStreamSource, newAudioSource?: AudioStreamSource): Promise<void>;
    get currentProfile(): string | undefined;
    get currentVideoTrack(): MediaStreamTrack | undefined;
    get currentDualVideoTrack(): MediaStreamTrack | undefined;
    get publishVideoStream(): LocalStream | undefined;
    get publishDualVideoStream(): LocalStream | undefined;
    get currentAudioTrack(): MediaStreamTrack | undefined;
    get publishAudioStream(): LocalStream | undefined;
    get previewStream(): LocalStream | undefined;
    /**
     * 更新 VideoTrack
     * @param videoTrack
     */
    updateVideoTrack(videoTrack?: MediaStreamTrack, force?: boolean): Promise<void>;
    /**
     * 更新 DualVideoTrack
     * @param videoTrack
     * @param force
     */
    updateDualVideoTrack(videoTrack?: MediaStreamTrack, force?: boolean): Promise<void>;
    /**
     * 更新 AudioTrack
     * @param audioTrack
     */
    updateAudioTrack(audioTrack?: MediaStreamTrack, force?: boolean): Promise<void>;
    /**
     * 设置音频流静音
     * @param muted
     */
    setAudioMuted(muted: boolean): void;
    get isAudioMuted(): boolean;
    /**
     * 设置视频流静音
     * @param muted
     */
    setVideoMuted(muted: boolean): void;
    get isVideoMuted(): boolean;
    process(localStreamInfos: AliRtcLocalStreamInfo[]): Promise<void>;
    /**
     * 停止视频流
     */
    stopVideo(): void;
    /**
     * 停止音频流
     */
    stopAudio(): void;
    /**
     * 停止视频流和音频流
     */
    stop(): void;
    /**
     * 应该在执行完 plugin 后调用
     * @param profileName
     * @param profileValue
     */
    updateVideoProfile(profileName?: string, profileValue?: Partial<IProfile>, skipConstraints?: boolean): Promise<void>;
    get videoProfile(): string | undefined;
    setVideoContentHint(hint?: string): void;
    /**
     * 应该在执行完 plugin 后调用
     * @param profileName
     */
    updateAudioProfile(profileKey: AudioProfileKey): Promise<void>;
    /**
     * 复制视频 profile 到指定 LocalStream
     * @param publishStream
     */
    cloneVideoProfile(publishStream: LocalStream): Promise<void>;
    addPlugin(plugin: AliRtcPlugin): void;
    removePlugin(plugin: AliRtcPlugin): boolean;
}

declare interface AliRtcLocalStreamListener {
    videotrackended: () => void;
    audiotrackended: () => void;
}

declare interface AliRtcLocalVideoStats {
    track: AliRtcVideoTrack;
    streamType?: AliRtcVideoStreamType;
    stats: any;
}

declare type AliRtcLocalView = string | HTMLVideoElement | string[] | HTMLVideoElement[] | null;

export declare enum AliRtcLogLevel {
    DEBUG = 0,
    INFO = 1,
    API = 2,
    WARNING = 3,
    ERROR = 4,
    NONE = 5
}

/**
 * @brief OnBye类型枚举
 */
declare enum AliRtcOnByeType {
    /** 当前user被踢出channel */
    AliRtcOnByeBeKickedOut = 1,
    /** channel已结束，需要离开会议 */
    AliRtcOnByeChannelTerminated = 2,
    /** 相同userID在其他设备joinChannel，当前设备被下线 */
    AliRtcOnByeUserReplaced = 3
}

export declare abstract class AliRtcPlugin extends default_2<AliRtcPluginListener> {
    name: string;
    options: any;
    type: AliRtcPluginType;
    streamType: AliRtcRawDataStreamType;
    trackType: AliRtcPluginTrackType;
    zIndex: number;
    private _isEnable;
    lastInputAudioTrack?: MediaStreamTrack;
    lastOutputAudioTrack?: MediaStreamTrack;
    lastInputVideoTrack?: MediaStreamTrack;
    lastOutputVideoTrack?: MediaStreamTrack;
    private _hasCleared;
    constructor(name: string, streamType?: AliRtcRawDataStreamType, trackType?: AliRtcPluginTrackType);
    get initOptions(): {};
    getOptions(): any;
    abstract setOptions(options: any): void;
    abstract isSupported(version: string): boolean;
    init(): Promise<void>;
    enable(): void;
    disable(): void;
    get isEnable(): boolean;
    protected audioUpdated(streamInfo: AliRtcLocalStreamInfo): boolean;
    protected videoUpdated(streamInfo: AliRtcLocalStreamInfo): boolean;
    execute(streamInfo: AliRtcLocalStreamInfo, streamInfos: AliRtcLocalStreamInfo[]): Promise<void>;
    abstract shouldUpdate(streamInfo: AliRtcLocalStreamInfo, streamInfos: AliRtcLocalStreamInfo[]): boolean;
    abstract process(streamInfo: AliRtcLocalStreamInfo, streamInfos: AliRtcLocalStreamInfo[]): Promise<void>;
    clear(_streamInfo?: AliRtcLocalStreamInfo): void;
}

declare interface AliRtcPluginListener {
    enabled: () => void;
    disabled: () => void;
    updated: () => void;
    ready: () => void;
    overload: (info: any) => void;
    error: (error: any) => void;
    unsupported: () => void;
}

export declare enum AliRtcPluginTrackType {
    AUDIO = 0,
    VIDEO = 1,
    BOTH = 2
}

export declare enum AliRtcPluginType {
    PRE_PROCESSOR = 0,
    POST_PROCESSOR = 1
}

export declare enum AliRtcPublishState {
    /** 0: SDK初始推流状态 */
    AliRtcStatePublishIdle = 0,
    /** 1: 未推流状态，可能是因为:
     * - 已设置停止推送媒体流
     * - 推送媒体流失败
     */
    AliRtcStateNoPublish = 1,
    /** 2: 推流链接建立过程中 */
    AliRtcStatePublishing = 2,
    /** 3: 推流成功 */
    AliRtcStatePublished = 3
}

export declare enum AliRtcRawDataStreamType {
    /** 相机流 */
    AliRtcSdkStreamTypeCapture = 0,
    /** 屏幕共享流 */
    AliRtcSdkStreamTypeScreen = 1
}

export declare type AliRtcRefreshAuthInfo = Pick<AliRtcAuthInfo, 'userId' | 'nonce' | 'token' | 'timestamp' | 'role'>;

declare interface AliRtcRemoteAudioStats {
    uid: string;
    track: AliRtcAudioTrack;
    stats: any;
}

export declare class AliRtcRemoteUserInfo {
    private remoteUser;
    constructor(remoteUser: RemoteUser);
    get userId(): string;
    get displayName(): string;
    get isMuteAudioPlaying(): boolean;
    get hasAudio(): boolean;
    get hasCamera(): boolean;
    get hasCameraLarge(): boolean;
    get hasCameraSmall(): boolean;
    get hasScreenShare(): boolean;
    get isAudioSubscribing(): boolean;
    get isVideoSubscribing(): boolean;
    get isVideoLargeSubscribing(): boolean;
    get isVideoSmallSubscribing(): boolean;
    get isScreenSubscribing(): boolean;
    get audioElement(): HTMLAudioElement | undefined;
}

declare interface AliRtcRemoteVideoStats {
    uid: string;
    track: AliRtcVideoTrack;
    streamType?: AliRtcVideoStreamType;
    stats: any;
}

declare interface AliRtcScreenShareConfiguration {
    frameRate?: number;
    bitrate?: number;
}

declare interface AliRtcScreenShareStartConfig {
    audio?: boolean;
    videoTrack?: MediaStreamVideoTrack;
    audioTrack?: MediaStreamAudioTrack;
}

export declare enum AliRtcSdkChannelProfile {
    /** 普通通信模式 */
    AliRtcSdkCommunication = "communication",
    /** 直播模式(大方会模式) */
    AliRtcSdkInteractiveLive = "interactive_live",
    /** 低延迟互动直播模式 */
    AliRtcSdkInteractiveWithLowLatencyLive = "cdn_live"
}

export declare enum AliRtcSdkClientRole {
    /** 互动角色 */
    AliRtcSdkInteractive = "interactive",
    /** 观众角色 */
    AliRtcSdkLive = "live"
}

/** 角色鉴权 主播pub｜观众sub｜转推流鉴权（跨房间转推中使用）relay */
declare type AliRtcSdkTokenRole = 'pub' | 'sub' | 'relay';

export declare enum AliRtcSubscribeState {
    /** 初始状态 */
    AliRtcStateSubscribeIdle = 0,
    /** 未订阅 */
    AliRtcStateNoSubscribe = 1,
    /** 订阅中 */
    AliRtcStateSubscribing = 2,
    /** 已订阅 */
    AliRtcStateSubscribed = 3
}

export declare class AliRtcTranscodingClockWidget {
    x: number;
    y: number;
    fontSize: number;
    zOrder: number;
    fontColor: number;
    fontType: AliRtcLiveTranscodingFontType;
    constructor(x: number, y: number, fontSize: number);
    constructor(x: number, y: number, fontSize: number, zOrder: number);
    constructor(x: number, y: number, fontSize: number, zOrder: number, fontColor: number);
    constructor(x: number, y: number, fontSize: number, zOrder: number, fontColor: number, fontType: AliRtcLiveTranscodingFontType);
    toJson(videoWidth: number, videoHeight: number): {
        x: number;
        y: number;
        fonttype: AliRtcLiveTranscodingFontType;
        fontsize: number;
        fontcolor: number;
        zorder: number;
    };
}

export declare class AliRtcTranscodingImage {
    static DisplayType: typeof DisplayType;
    url: string;
    x: number;
    y: number;
    width: number;
    height: number;
    alpha: number;
    display: DisplayType;
    zOrder: number;
    constructor(url: string, x: number, y: number, width: number, height: number);
    constructor(url: string, x: number, y: number, width: number, height: number, zOrder: number);
    constructor(url: string, x: number, y: number, width: number, height: number, zOrder: number, alpha: number);
    constructor(url: string, x: number, y: number, width: number, height: number, zOrder: number, alpha: number, display: DisplayType);
    toJson(videoWidth: number, videoHeight: number): {
        url: string;
        alpha: number;
        display: DisplayType;
        x: number;
        y: number;
        width: number;
        height: number;
        zorder: number;
    };
}

export declare class AliRtcTranscodingText {
    text: string;
    x: number;
    y: number;
    fontSize: number;
    zOrder: number;
    fontColor: number;
    fontType: AliRtcLiveTranscodingFontType;
    constructor(text: string, x: number, y: number, fontSize: number);
    constructor(text: string, x: number, y: number, fontSize: number, zOrder: number);
    constructor(text: string, x: number, y: number, fontSize: number, zOrder: number, fontColor: number);
    constructor(text: string, x: number, y: number, fontSize: number, zOrder: number, fontColor: number, fontType: AliRtcLiveTranscodingFontType);
    toJson(videoWidth: number, videoHeight: number): {
        text: string;
        x: number;
        y: number;
        fonttype: AliRtcLiveTranscodingFontType;
        fontsize: number;
        fontcolor: number;
        zorder: number;
    };
}

export declare class AliRtcTranscodingUser {
    userId: string;
    x: number;
    y: number;
    width: number;
    height: number;
    zOrder?: number;
    sourceType?: AliRtcLiveTranscodingSourceType;
    segmentType: AliRtcLiveTranscodingSegmentType;
    images: AliRtcTranscodingImage[];
    texts: AliRtcTranscodingText[];
    constructor(userId: string, x: number, y: number, width: number, height: number);
    constructor(userId: string, x: number, y: number, width: number, height: number, zOrder: number);
    constructor(userId: string, x: number, y: number, width: number, height: number, zOrder: number, sourceType: AliRtcLiveTranscodingSourceType);
    toJson(index: number, videoWidth: number, videoHeight: number, mediaProcessMode?: AliRtcLiveTranscodingMediaProcessMode): {
        [key: string]: any;
    };
}

export declare enum AliRtcTrascodingPublishTaskStatus {
    /*! 任务开始 */
    TrascodingPublishTaskStatusStart = 0,
    /*! 任务更新 */
    TrascodingPublishTaskStatusUpdate = 1,
    /*! 任务已停止 */
    TrascodingPublishTaskStatusStop = 2
}

declare enum AliRtcUserOfflineReason {
    /** 用户主动离开 */
    AliRtcUserOfflineQuit = 0,
    /** 因过长时间收不到对方数据包，超时掉线 */
    AliRtcUserOfflineDropped = 1,
    /** 用户身份从主播切换为观众时触发 */
    AliRtcUserOfflineBecomeAudience = 2
}

export declare enum AliRtcVideoStreamType {
    /** 无，在OnSubscribeStreamTypeChanged回调表示当前未订阅 */
    AliRtcVideoStreamTypeNone = 0,
    /** 高码率，高分辨率流（大流） */
    AliRtcVideoStreamTypeHigh = 1,
    /** 低码率，低分辨率流（小流） */
    AliRtcVideoStreamTypeLow = 2
}

export declare enum AliRtcVideoTrack {
    /** 无视频流 **/
    AliRtcVideoTrackNo = 0,
    /** 相机流 **/
    AliRtcVideoTrackCamera = 1,
    /** 屏幕共享流 **/
    AliRtcVideoTrackScreen = 2,
    /** 相机流和屏幕共享流 **/
    AliRtcVideoTrackBoth = 3
}

declare interface AliRtcVideoTrackConfig {
    userId?: string;
    streamType: AliRtcRawDataStreamType;
}

declare interface AliRtcWebTrackConfig {
    biz: string;
    extraParams?: string;
}

declare class AudioLevelMonitor extends default_2<AudioLevelMonitorListener> {
    private audioContext;
    private stream;
    private sourceNode?;
    private analyser?;
    constructor();
    getLevel(): number;
    /**
     * 开启监听 audio level 的变化
     *
     * @param {MediaStreamTrack} track 音频轨
     * @param {number} [interval=1000] 检测间隔，单位毫秒，默认 1000ms
     */
    start(track: MediaStreamTrack, interval?: number): void;
    stop(): void;
    dispose(): void;
}

declare interface AudioLevelMonitorListener {
    audioLevel: (level: number) => void;
}

declare interface AudioMixerPluginOptions {
    audio?: number;
    screen?: number;
}

/**
 * ------------------    模式    ----- 采样率  -----  声道  -----  码率（kbps）  -----
 *
 * standard:           标准音质         48000       单声道          64
 *
 * high:                 高音质         48000       单声道         128
 *
 * standard-stereo:   立体声音质         48000       双声道         80
 *
 * high-stereo:       立体声高音质       48000       双声道         192
 */
declare type AudioProfileKey = 'standard' | 'high' | 'standard-stereo' | 'high-stereo';

declare enum AudioStreamSource {
    Microphone = 0,
    Screen = 1,
    Mixed = 2
}

declare class BizControl extends default_2<BizControlListener> {
    /**
     * @ignore
     */
    static logName: string;
    protected self: LocalUser;
    protected userMap: RemoteUserMap;
    private usersViewMap;
    rtsManager: RtsManager;
    protected localStreamManager: LocalStreamManager;
    protected authInfo?: AliRtcAuthInfo;
    protected signalingManager: SignalingManager;
    pluginManager: PluginManager;
    private audioMixerPlugin?;
    private dualVideoPlugin?;
    private blankAudioPlugin?;
    protected slsReporter: SLSReporter;
    private statsMonitorId?;
    private autoSubAudio;
    private autoSubVideo;
    private autoSubScreen;
    private wantSubAudio;
    private wantSubVideo;
    private wantSubScreen;
    private defaultStreamType;
    private audioMuted;
    private playoutVolume;
    private refreshUrlTimer?;
    private audioVolumeIndicationInterval;
    private indicationTimer;
    private parameter;
    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;
    /** 定期更新所有人的推拉流地址，便于重连
     * @param pushStreamUrl 已有的推流地址，根据地址中的过期时间来设置定时器
     * @private
     */
    private setTimeoutToUpdateAllStreamUrls;
    /**
     * 登陆成功后初始化用户管理，创建自己
     * @param {AliRtcAuthInfo} authInfo
     * @param {JoinResult} data
     * @returns {Promise<void>}
     */
    init(authInfo: AliRtcAuthInfo, data: JoinResult): void;
    leave(): Promise<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;
    setEnableDualVideoStream(enabled: boolean, options?: AliRtcDualVideoPluginOptions): void;
    setAudioVolume(options: AudioMixerPluginOptions): void;
    enableAudioVolumeIndication(interval: number): void;
    setPlayoutVolume(volume: number): void;
    refreshAuthInfo(authInfo: AliRtcRefreshAuthInfo): void;
    setEnableMediaExtensionMsg(enable: boolean): void;
    setAudioRedEnabled(enable: boolean): void;
    setParameter(parameter: string): void;
    getParameter(): string;
    sendDataChannelMessage(message: AliRtcDataChannelMsg): Promise<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;
    dataSubscribeStateChanged: (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;
    remoteAudioPlayError: (uid: string, reason?: string) => void;
    remoteVideoPlayError: (uid: string, reason?: string) => void;
    remoteSubscribeError: (error: AliRtcError, uid?: string) => void;
    occurError: (error: AliRtcError, uid?: string) => void;
    remoteDataChannelMessage: (uid: string, message: AliRtcDataChannelMsg) => void;
}

declare enum ClientEventType {
    PullStreamStats = "remoteStat",
    RtsSubscribeExceeds = "rtsSubscribeExceeds",
    UserJoined = "userJoined",
    UserLeft = "userLeft",
    Bye = "bye",
    MuteAudio = "muteAudio",
    UnmuteAudio = "unmuteAudio",
    MuteVideo = "muteVideo",
    UnmuteVideo = "unmuteVideo",
    MuteScreen = "muteScreen",
    UnmuteScreen = "unmuteScreen",
    Error = "error"
}

declare enum CmdType {
    Add = 0,
    Del = 1,
    Update = 2,
    None = 100
}

declare enum CodecType {
    OPUS = "opus",
    H264 = "H264",
    UNKNOWN = ""
}

declare enum ConnectionLatencyStage {
    CREATE_ENGINE = "create_engine",
    JOIN = "join",
    WANT_INSERT_PUB_TASK = "wantInsertPubTask",
    INSERT_PUB_TASK = "InsertPubTask",
    HANDLE_PUB_TASK = "HandlePubTask",
    RECV_NOTIFY_PUBLISH = "RecvNotifyPublish",
    INSERT_SUB_TASK = "InsertSubTask",
    HANDLE_SUB_TASK = "HandleSubTask",
    FIRST_REMOTE = "first_remote"
}

declare enum ConnectionLatencyStatus {
    START = "start",
    SEND_SIG = "send_sig",
    RECV_RST = "recv_rst",
    END = "end",
    RECEIVED = "received",
    DECODED = "decoded",
    PLAYED = "played"
}

/**
 * websocket链接状态
 */
declare enum CONNECTSTATUS {
    DISCONNECTED = 0,
    CONNECTING = 1,
    CONNECTED = 2
}

declare interface DeviceStatus {
    micOpen: boolean;
    micInterrupted: boolean;
    cameraOpen: boolean;
    screenShare: boolean;
    audioDisable: boolean;
    videoDisable: boolean;
    screenDisable: boolean;
    backgroundMode: boolean;
    accompanyPlaying: boolean;
}

declare enum DeviceStatusChange {
    Local = "local-device-status-change",
    Remote = "remote-device-status-change"
}

declare type DeviceStatusChangeType = ClientEventType.MuteAudio | ClientEventType.UnmuteAudio | ClientEventType.MuteScreen | ClientEventType.UnmuteScreen | ClientEventType.MuteVideo | ClientEventType.UnmuteVideo;

declare enum DisplayType {
    NOT_DISPLAY = 0,
    ALWAYS = 1,
    WHEN_NO_VIDEO = 2
}

declare interface IResponse {
    version: number;
    code: number;
    msgtype: MessageType;
    tid: string;
    isproxy?: boolean;
}

declare interface ISendPackage {
    version: number;
    msgtype: MessageType;
    tid: string;
    header: {
        appid: string;
        channelid: string;
        sessionid: string;
        userid: string;
        displayName?: string;
        clientversion?: string;
        platform?: string;
    };
    data: {
        seq: number;
        l1ip?: string;
        status?: string;
        clientrole?: string;
        configure?: {
            clientrole?: AliRtcSdkClientRole;
            channelprofile: AliRtcSdkChannelProfile;
        };
        auth?: SendPackageAuthInfo;
        sessionid?: string;
        pushstreamurl?: string;
        users?: any[];
    };
}

/**
 * 服务端通知的远端用户信息
 */
declare interface JoinInfo {
    displayname: string;
    session: string;
    userid: string;
    callid: string;
    clientrole?: string;
    subscribe?: {
        pullstreamurl: string;
    };
    tracks?: TrackInfo[];
}

declare interface JoinResult {
    pushstreamurl: string;
    tid: string;
    users: JoinInfo[];
    timestamp: number;
}

declare interface LeaveInfo {
    userid: string;
    leavecode: AliRtcUserOfflineReason;
    reason: string;
}

declare class LocalProfileManager {
    private defaultProfile;
    profile?: string;
    profileMap: Map<string, IProfile>;
    audioProfile?: AudioProfileKey;
    contentHint?: string;
    constructor(defaultProfile?: IProfile);
    updateProfile(profileKey: string, profileValue?: Partial<IProfile>): void;
    get videoProfile(): string | undefined;
    updateAudioProfile(profileKey: AudioProfileKey): void;
    setContentHint(videoTrack: MediaStreamTrack | undefined, contentHint?: string): void;
    /**
     * 更新源流 Constraints
     * 更新 PeerConnectiong Sender 相关配置
     * @param stream
     */
    setVideoProfile(originVideoTrack: MediaStreamTrack | undefined, target: LocalStream | undefined, skipConstraints?: boolean): Promise<void>;
}

declare type LocalStreamConfig = Omit<_LocalStreamConfig, 'custom'>;

declare interface _LocalStreamConfig extends IStreamConfig {
    videoProfile?: string;
    screenProfile?: string;
}

declare class LocalStreamManager extends default_2<LocalStreamManagerListener> {
    /**
     * @ignore
     */
    static logName: string;
    private pluginManager;
    private slsReporter;
    private publishingCameraVideoStream;
    private publishingScreenVideoStream;
    cameraStreamInfo: AliRtcLocalStreamInfo;
    screenStreamInfo: AliRtcLocalStreamInfo;
    private streamInfos;
    private _audioCaptureDisabled;
    private _cameraCaptureDisabled;
    private _publishLocalAudioStreamEnabled;
    private _publishLocalVideoStreamEnabled;
    private _publishLocalScreenStreamEnabled;
    constructor(pluginManager: PluginManager, slsReporter: SLSReporter);
    get hasCamera(): boolean;
    get audioCaptureDisabled(): boolean;
    set audioCaptureDisabled(closed: boolean);
    get cameraCaptureDisabled(): boolean;
    set cameraCaptureDisabled(closed: boolean);
    setAudioMuted(muted: boolean): void;
    get isAudioMuted(): boolean;
    setCameraMuted(muted: boolean): void;
    get isCameraMuted(): boolean;
    setScreenMuted(muted: boolean): void;
    get isScreenMuted(): boolean;
    get publishLocalAudioStreamEnabled(): boolean;
    set publishLocalAudioStreamEnabled(enable: boolean);
    get publishLocalVideoStreamEnabled(): boolean;
    set publishLocalVideoStreamEnabled(enable: boolean);
    get publishLocalScreenStreamEnabled(): boolean;
    set publishLocalScreenStreamEnabled(enable: boolean);
    updateStreams(): Promise<void>;
    private setAvailableCameraDeviceId;
    private setAvailableMicrophoneDeviceId;
    /**
     * 创建 LocalStream
     * @param {LocalStreamConfig} params 创建 LocalStream 的配置
     * @returns {LocalStream} 当前创建的 LocalStream
     */
    createLocalStream(params: LocalStreamConfig): Promise<LocalStream | undefined>;
    /**
     * 停止推流，底层调用 rts publishStop，不会导致 DC 马上关闭。适合暂时停推的场景
     * @param {StopPublishOptions} options
     * @returns {Promise<PublishStreamInfo>} 如果返回了 audio，则表示需要更新音频流
     */
    stopLocalStream(options: StopPublishOptions): Promise<void>;
    get publishStreams(): PublishStreamInfo;
    createHTTPPublishStream(): Promise<{
        publishStream: LocalStream | undefined;
        videoSmallStream: LocalStream | undefined;
        screenStream: LocalStream | undefined;
        vMsid: string | undefined;
    }>;
    /**
     * 停止所有流
     */
    stop(): void;
    /**
     * 设置摄像头流 profile
     * @param {LocalStream | null} stream
     * @param {string} profileKey
     * @param {IProfile} profile
     */
    setVideoProfile(profileKey?: string, profile?: Partial<IProfile>, skipConstraints?: boolean): Promise<void>;
    /**
     * 设置摄像头流 content hint
     * @param {string} hint
     * @note 不传则尝试复用之前的 hint
     */
    setVideoContentHint(hint?: string): void;
    /**
     * 设置音频流 profile
     * @param {string} profileKey
     */
    setAudioProfile(profileKey: AudioProfileKey): Promise<void>;
    /**
     * 设置屏幕共享流 profile
     * @param {LocalStream} stream
     * @param {string} profileKey
     * @param {IProfile} profile
     */
    setScreenProfile(profileKey?: string, profile?: Partial<IProfile>): Promise<void>;
    setScreenContentHint(hint?: string): void;
    /**
     * 设置 cameraVideoConstraints
     * @param {IVideoConstraints} cameraVideoConstraints
     */
    setCameraVideoConstraints(cameraVideoConstraints: IVideoConstraints): void;
    /**
     * 设置 micAudioConstraints
     * @param {IAudioConstraints} micAudioConstraints
     */
    setMicAudioConstraints(micAudioConstraints: IAudioConstraints): void;
}

declare interface LocalStreamManagerListener {
    trackended: (type: TrackEvent_2) => {};
    publishupdate: () => {};
}

declare class LocalUser extends User {
    /**
     * @ignore
     */
    static logName: string;
    private _url;
    private joinTime;
    private _isPublishing;
    private publishPromise;
    private previewElements;
    private screenPreviewElements;
    private streamManager;
    private dc;
    protected streamTracks?: TrackInfo[];
    protected publishId: string;
    protected signalingManager: SignalingManager;
    traceId: string;
    private monitorTimerId?;
    private monitorDataCache;
    private trackTraceIdMap;
    private candidateDataCache;
    private pcMediaStatsMap;
    protected slsReporter: SLSReporter;
    private audioPubState;
    private videoPubState;
    private videoSmallPubState;
    private screenPubState;
    private dataPubState;
    private parameter;
    constructor(config: LocalUserConfig);
    get url(): string;
    set url(value: string);
    get joined(): boolean;
    get isPublishing(): boolean;
    get publishStreams(): PublishStreamInfo;
    get cameraTrack(): MediaStreamTrack | undefined;
    get screenTrack(): MediaStreamTrack | undefined;
    get audioTrack(): MediaStreamTrack | undefined;
    get hasCamera(): boolean;
    get hasMicrophone(): boolean;
    get hasScreenShare(): boolean;
    get hasCameraTrackInMS(): boolean;
    get hasCameraDualTrackInMS(): boolean;
    get hasScreenTrackInMS(): boolean;
    get hasAudioTrackInMS(): boolean;
    get cameraDirection(): string | undefined;
    get audioCaptureDisabled(): boolean;
    get cameraCaptureDisabled(): boolean;
    get publishLocalVideoStreamEnabled(): boolean;
    get publishLocalAudioStreamEnabled(): boolean;
    get publishLocalScreenStreamEnabled(): boolean;
    setPublishLocalAudioStreamEnabled(enable: boolean): void;
    setPublishLocalVideoStreamEnabled(enable: boolean): void;
    setPublishLocalScreenStreamEnabled(enable: boolean): void;
    initInfo(authInfo: AliRtcAuthInfo, pushStreamUrl: string): void;
    setVideoProfile(profile?: VideoProfileWithSendFramerate): Promise<void>;
    setVideoContentHint(hint: string): void;
    setScreenContentHint(hint: string): void;
    setAudioProfile(profileKey: AudioProfileKey): Promise<void>;
    setScreenProfile(profile?: Partial<IProfile>): Promise<void>;
    setCameraDirection(direction: AliRtcCameraDirection): void;
    setCameraDeviceId(deviceId: string): void;
    muteLocalCamera(mute: boolean): void;
    muteLocalMic(mute: boolean): void;
    muteLocalScreen(mute: boolean): void;
    setMicrophoneDeviceId(deviceId: string): void;
    setMicrophoneConstraints(constraints: MediaTrackConstraints): void;
    updatePubId(): void;
    clearPublishState(): void;
    leave(): Promise<void>;
    /**
     * 销毁，停止推流
     * @returns {Promise<void>}
     */
    clear(): Promise<void>;
    resetElement(element: HTMLVideoElement): void;
    setPreviewElement(previewParams: PreviewConfig): void;
    startDefaultCamera(audioOnly: boolean): Promise<void>;
    startDefaultAudio(): Promise<void>;
    /**
     * 为本地预览设置渲染元素和视频流
     * @param {AliRtcLocalView} view 渲染元素，null 则为停止显示
     * @param {AliRtcVideoTrack} track 视频流类型
     */
    setViewConfig(elements: null | HTMLVideoElement[], track: AliRtcVideoTrack): void;
    /**
     * 开启摄像头采集
     * @param params
     * @returns
     */
    startCameraCapture(): Promise<LocalStream | undefined>;
    /**
     * 关闭摄像头采集
     * @note 调用此接口后，入会后、离会再入会，采集设备保持关闭状态
     */
    stopCameraCapture(): Promise<void>;
    /**
     * 开启麦克风采集
     * @param params
     * @returns
     */
    startAudioCapture(): Promise<LocalStream | undefined>;
    /**
     * 关闭音频采集
     * @note 调用此接口后，入会后、离会再入会，采集设备保持关闭状态
     */
    stopAudioCapture(): Promise<void>;
    /**
     * 预览本地流
     * @param {PreviewConfig} previewParams
     * @returns {Promise<MediaStream>}
     */
    preview(previewParams: PreviewConfig): Promise<MediaStream | undefined>;
    /**
     * 在所有预览容器中预览当前流
     * @param {boolean} screen 是否指定预览辅流
     */
    previewAll(previewAllParams?: PreviewAllConfig): Promise<void>;
    /**
     * 停止预览本地流
     */
    stopPreview(previewParams?: StopPreviewConfig): void;
    /**
     * 恢复推流
     * @returns {Promise<void>}
     */
    resumePublish(): Promise<void>;
    createLocalStream(params: LocalStreamConfig): Promise<LocalStream | undefined>;
    stopLocalStream(options: PublishOptions): Promise<void>;
    private compareAndPublish;
    private reportAudioProfile;
    private reportVideoProfile;
    private reportScreenProfile;
    private updateDataChannel;
    private publishDataChannel;
    /**
     * 开始推流
     * @param isResume 是否是恢复推流
     * @returns
     */
    startPublish(isResume?: boolean, reason?: PublishReason): Promise<void>;
    /**
     * 重新执行 progress 并触发 publishupdate
     */
    updateStreams(): 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
     */
    publishUpdate(reason?: PublishReason): Promise<PublishInfo | undefined>;
    private syncPublishToRoomServer;
    private refreshPushStreamUrl;
    /**
     * 推流，用于 rts subscribe/publish 被调用后，此时 DataChannel 已存在
     * @param {LocalStream} stream
     * @returns {Promise<PublishInfo>}
     */
    publishAdd(stream: LocalStream, extra: PublishExtraParams, isResume?: boolean, retryTimes?: number): Promise<PublishInfo>;
    /**
     * 替换推流 Track（仅限于替换已存在的 Track 类型）
     * @param {LocalStream} stream
     * @returns {Promise<PublishInfo>}
     */
    publishReplace(stream: LocalStream, extra?: PublishExtraParams): Promise<PublishInfo>;
    /**
     * 停止推某些轨，不销毁 DataChannel 和 LocalStream，用于非销毁场景
     * @param {PublishOptions} options
     * @returns {Promise<PublishInfo>}
     */
    publishDeleteTracks(options: PublishOptions): Promise<PublishInfo>;
    /**
     * 停止推流，不销毁  LocalStream，用于非销毁场景
     * @returns {Promise<PublishInfo>}
     */
    publishStop(): Promise<PublishInfo>;
    /**
     * 停止推流，销毁 DataChannel，用于销毁的场景
     * @returns {Promise<PublishInfo>}
     */
    unPublish(): Promise<PublishInfo>;
    /**
     * 事件通知本地的设备使用状态
     */
    reportSelfDeviceStatus(): void;
    /**
     * 从localStream 中获取当前的设备状态
     * @returns {string}
     */
    private getDeviceStatus;
    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
     */
    private stopStreamMonitor;
    /**
     * 清除本地流
     * @returns {void}
     */
    clearStream(): void;
    /**
     * 获取当前使用的摄像头 deviceId
     * @returns {string | undefined}
     */
    getCurrentCameraDeviceId(): string | undefined;
    /**
     * 获取当前使用的摄像头 deviceId
     * @returns {string | undefined}
     */
    getCurrentMicDeviceId(): string | undefined;
    enableAudioVolumeIndication(interval: number): void;
    sendSEI(message: ArrayBuffer, repeatCount: number, payloadType: number): Promise<void>;
    sendDataChannelMessage(message: AliRtcDataChannelMsg): Promise<void>;
    getStats(): Promise<{
        audio: AliRtcLocalAudioStats[];
        video: AliRtcLocalVideoStats[];
        candidate: any;
    }>;
}

declare interface LocalUserConfig {
    rtsManager: RtsManager;
    localStreamManager: LocalStreamManager;
    slsReporter: SLSReporter;
    signalingManager: SignalingManager;
    pluginManager: PluginManager;
    audioVolumeIndicationInterval: number;
    parameter: {
        [key: string]: any;
    };
}

/**
 * 日志埋点类，调用LogReporter进行日志埋点
 * 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 tracker?: default_3;
    protected param?: any;
    protected msgCacheArr: any[];
    protected index: number;
    protected stsOpt?: any;
    protected slsToken?: SLSSTSToken;
    protected ossToken?: OSSSTSToken;
    protected ntpClock: NTPClient;
    constructor();
    private createTracker;
    start(info: AliRtcAuthInfo): void;
    updateToken(slsToken: SLSSTSToken, ossToken?: OSSSTSToken): void;
    /**
     * 断开连接
     */
    stop(): void;
    /**
     * 发送一条日志
     * @param {any} log
     */
    sendReport(log: any): void;
    /**
     * 初始化公共参数
     */
    protected initParam(info: AliRtcAuthInfo): void;
}

declare interface LogInfo {
    [key: string]: string | number;
}

declare interface MediaTrackInfo {
    ssrc: string;
    msid: string;
    red: number;
    codec: CodecType;
    sample?: number;
    pt: number;
    type?: CmdType;
}

declare enum MessageType {
    KEEPALIVE = "keepalive",
    JOINCHANNEL = "joinchannel",
    NOTIFYJOIN = "notifyjoin",
    NOTIFYSTATUS = "notifystatus",
    STATUSREPORT = "statusreport",
    LEAVECHANNEL = "leavechannel",
    NOTIFYLEAVE = "notifyleave",
    NOTIFYPUBLISH = "notifypublish",
    BYE = "bye",
    RECONNECT = "reconnect",
    PUBLISH = "publish",
    REFRESHURL = "refreshurl",
    UNSUBSCRIBE = "unsubscribe",
    ROLEUPDATE = "roleupdate"
}

declare enum MPU_TASK_TRANSACTION_TYPE {
    MPU_TASK_TRANSACTION_START = 0,
    MPU_TASK_TRANSACTION_UPDATE = 1,
    MPU_TASK_TRANSACTION_STOP = 2,
    MPU_TASK_TRANSACTION_LIST = 3,
    MPU_TASK_TRANSACTION_POLLING = 4,
    MPU_TASK_TRANSACTION_ACKNOWLEDGE = 5,
    MPU_TASK_TRANSACTION_END = 6
}

declare enum MsidType {
    Audio = "sophon_audio",
    VideoLarge = "sophon_video_camera_large",
    VideoSmall = "sophon_video_camera_small",
    Screen = "sophon_video_screen_share",
    Data = "sophon_data"
}

declare class NTPClient {
    private static instance;
    private ntpClock;
    private constructor();
    static getInstance(): NTPClient;
    now(): number;
}

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 PackageCreater {
    protected authInfo: AliRtcAuthInfo;
    protected userName: string;
    protected seqIndex: number;
    constructor(authInfo: AliRtcAuthInfo, userName: string);
    get roleAuth(): SendPackageAuthInfo | undefined;
    refreshAuthInfo(authInfo: AliRtcAuthInfo): void;
    getSeqIndex(): number;
    /**
     * 创建一个心跳包
     * @returns
     */
    createKeepAlivePackage(): ISendPackage;
    /**
     * 创建一个用于发送入会消息的Package
     * @returns
     */
    createJoinPackage(): ISendPackage;
    /**
     * 创建一个用于发送离会消息的Package
     * @returns
     */
    crateLeavePackage(): ISendPackage;
    createStatusPackage(status: string): ISendPackage;
    /**
     * 创建一个用于发送推流消息的Package
     * @returns
     */
    createPublishPackage(): ISendPackage;
    createRoleUpdatePackage(): ISendPackage;
    createReconnectPackage(data?: ReconnectData): ISendPackage;
    createRefreshUrlPackage(): ISendPackage;
    /**
     * 创建一个Response包
     * @param {string} tid            Notify消息的tid
     * @param {MessageType} msgType   消息类型
     * @returns {IResponse}
     */
    createResponsePackage(tid: string, msgType: MessageType): IResponse;
    /**
     *
     * @param messageType
     * @returns
     */
    protected createBaseSendPackage(messageType: MessageType): ISendPackage;
}

declare type Parameter = {
    [key: string]: any;
};

declare class PluginManager extends default_2<PluginManagerListener> {
    private plugins;
    add(plugin: AliRtcPlugin, options?: any): void;
    remove(name: string): void;
    removeAll(): void;
    get(name: string): AliRtcPlugin | undefined;
    has(name: string): boolean;
    getAll(): AliRtcPlugin[];
}

declare interface PluginManagerListener {
    added: (plugin: AliRtcPlugin) => void;
    removed: (plugin: AliRtcPlugin) => void;
    updated: (plugin: AliRtcPlugin) => void;
}

declare interface PreviewAllConfig {
    primary?: boolean;
    screen?: boolean;
}

declare interface PreviewConfig {
    videoElement: HTMLVideoElement;
    screen?: boolean;
}

declare interface PublishExtraParams {
    vMsid?: string;
    aMsid?: string;
    isVideoSmall?: boolean;
    isScreen?: boolean;
    isAudio?: boolean;
}

declare interface PublishInfo {
    type: PublishType;
    userid: string;
    sessionid: string;
    displayname: string;
    pubid?: string;
    resume?: boolean;
    subscribe: {
        signal: string;
        pullstreamurl: string;
    };
    l1ip?: string;
    callid: string;
    tracks: TrackInfo[];
    traceid?: string;
    signaltid?: string;
}

declare type PublishOptions = StreamOptions & {
    vMsid?: string;
};

declare enum PublishReason {
    MANUAL = 0,
    JOIN_CHANNEL = 1,
    ROLE_CHANGE = 2,
    RECONNECT = 3
}

declare interface PublishStreamInfo {
    audio?: LocalStream;
    video?: LocalStream;
    videoSmall?: LocalStream;
    screen?: LocalStream;
    videoProfile?: LocalProfileManager;
    screenProfile?: LocalProfileManager;
}

declare enum PublishType {
    Add = "add",
    Delete = "delete",
    Replace = "replace",
    Stop = "stop",
    UnPublish = "unpublish"
}

declare interface PublishUser {
    callid: string;
    clientrole?: string;
    displayname: string;
    subscribe?: {
        pullstreamurl: string;
    };
    pubid?: string;
    tracks?: TrackInfo[];
    userid: string;
    sessionid?: string;
}

declare interface ReconnectData {
    role?: RoleForReconnectData;
    auth?: SendPackageAuthInfo;
    publish?: {
        users: [PublishInfo];
    };
}

declare interface RefreshInfo {
    pushstreamurl?: string;
    users?: RefreshUser[];
}

declare interface RefreshUrlResult extends SignalingResult {
    pushstreamurl?: string;
    users?: RefreshUser[];
}

declare interface RefreshUser {
    userid: string;
    subscribe: {
        pullstreamurl: string;
    };
}

declare interface RemoteMediaTrackInfo extends MediaTrackInfo {
    subscribeState?: RemoteTrackSubscribeState;
}

declare class RemoteStreamInfo extends default_2<StreamListener> {
    /**
     * @ignore
     */
    static logName: string;
    protected audioInfo: RemoteMediaTrackInfo;
    protected videoLargeInfo: RemoteMediaTrackInfo;
    protected videoSmallInfo: RemoteMediaTrackInfo;
    protected screenInfo: RemoteMediaTrackInfo;
    protected dataInfo: RemoteMediaTrackInfo;
    protected audioVolume: number;
    protected audioMuted: boolean;
    protected userStatus?: DeviceStatus;
    constructor();
    /**
     * 远端流音频轨道信息
     * @returns
     */
    getAudioInfo(): RemoteMediaTrackInfo;
    /**
     * 远端流视频轨道信息
     * @returns
     */
    getVideoSmallInfo(): RemoteMediaTrackInfo;
    /**
     * 远端流视频轨道信息
     * @returns
     */
    getVideoLargeInfo(): RemoteMediaTrackInfo;
    /**
     * 远端流屏幕共享轨道信息
     * @returns
     */
    getScreenInfo(): RemoteMediaTrackInfo;
    getDataInfo(): RemoteMediaTrackInfo;
    get subscribingInfos(): RemoteMediaTrackInfo[];
    get hasTrack(): boolean;
    /**
     * 远端流是否包含音频轨道
     * @returns
     */
    get hasAudio(): boolean;
    /**
     * 远端流是否包含视频轨道
     * @returns
     */
    get hasVideo(): boolean;
    /**
     * 远端流是否包含视频大流轨道
     * @returns
     */
    get hasVideoLarge(): boolean;
    /**
     * 远端流是否包含视频小流轨道
     * @returns
     */
    get hasVideoSmall(): boolean;
    /**
     * 远端流是否包含屏幕共享轨道
     * @returns
     */
    get hasScreen(): boolean;
    get hasData(): boolean;
    /**
     * 是否订阅远端流音频轨道
     * @returns
     */
    get isAudioSubscribed(): boolean;
    get isAudioSubscribing(): boolean;
    /**
     * 是否订阅远端流视频轨道
     * @returns
     */
    get isVideoSubscribed(): boolean;
    get isVideoSubscribing(): boolean;
    /**
     * 是否订阅远端流视频大流轨道
     * @returns
     */
    get isVideoLargeSubscribed(): boolean;
    get isVideoLargeSubscribing(): boolean;
    /**
     * 是否订阅远端流视频小流轨道
     * @returns
     */
    get isVideoSmallSubscribed(): boolean;
    get isVideoSmallSubscribing(): boolean;
    /**
     * 是否订阅远端流屏幕共享轨道
     * @returns
     */
    get isScreenSubscribed(): boolean;
    get isScreenSubscribing(): boolean;
    get isDataSubscribed(): boolean;
    get isDataSubscribing(): boolean;
    get audioCodec(): CodecType;
    get videoCodec(): CodecType;
    get audioTrackInfo(): AliRtcAudioTrack;
    get videoTrackInfo(): AliRtcVideoTrack;
    updateTracks(tracks?: TrackInfo[]): UpdateTracksResult;
    protected updateRemoteTrackSsrc(audioSsrc: string, videoLargeSsrc: string, videoSmallSsrc: string, screenSsrc: string, dataSsrc: string): void;
    markRemoteTrackSubscribing(option: SubscribeOptions): any;
    updateRemoteTrackSubState(option: SubscribeOptions): any;
    /**
     * 更新远端用户设备状态
     * @param status
     * @returns {boolean} 状态改变返回true，否则返回false
     */
    updateRemoteUserDeviceStatus(status: string): boolean;
}

declare interface RemoteSubscribeOptions {
    remoteUser: RemoteUser;
    options: SubscribeOptions;
}

declare enum RemoteTrackSubscribeState {
    NotSubscribe = 0,
    Subscribing = 1,
    Subscribed = 2
}

declare class RemoteUser extends User {
    /**
     * @ignore
     */
    static logName: string;
    remoteCallid: string;
    remoteUserInfo: AliRtcRemoteUserInfo;
    protected localUser?: LocalUser;
    protected lastPubId: string;
    protected stream?: RemoteStream;
    protected screenStream?: RemoteStream;
    private viewMap;
    private audioTrack?;
    private audioElement?;
    protected subscribed: boolean;
    protected mediaStream?: MediaStream;
    protected secondaryMediaStream?: MediaStream;
    protected prevSubConfig?: SubConfig;
    protected signalingManager: SignalingManager;
    protected slsReporter: SLSReporter;
    private wantSubAudio;
    private wantSubVideo;
    private wantSubScreen;
    private defaultVideoStreamType;
    private audioMuted;
    private playoutVolume;
    private playoutGainNode?;
    private audioSubState;
    private videoSubState;
    private videoLargeSubState;
    private videoSmallSubState;
    private screenSubState;
    private dataSubState;
    private dc;
    private dcMsgWaitingPieces;
    streamInfo: RemoteStreamInfo;
    private _streamUrl;
    private parameter;
    private monitorTimerId?;
    private monitorDataCache;
    private trackTraceIdMap;
    constructor(config: RemoteUserConfig);
    get streamUrl(): string;
    set streamUrl(value: string);
    isWantSubAudio(): boolean;
    setWantSubAudio(value: boolean): void;
    isWantSubVideo(): boolean;
    setWantSubVideo(value: boolean): void;
    isWantSubScreen(): boolean;
    setWantSubScreen(value: boolean): void;
    setRemoteDefaultVideoStreamType(type: AliRtcVideoStreamType): void;
    setAudioMuted(value: boolean): void;
    getAudioElement(): HTMLAudioElement | undefined;
    private handleAudioContextSuspended;
    setPlayoutVolume(value: number): void;
    getAudioMuted(): boolean;
    get hasAudioTrack(): boolean;
    get hasVideoTrack(): boolean;
    get hasVideoLargeTrack(): boolean;
    get hasVideoSmallTrack(): boolean;
    get hasScreenTrack(): boolean;
    get isAudioSubscribing(): boolean;
    get isVideoSubscribing(): boolean;
    get isVideoLargeSubscribing(): boolean;
    get isVideoSmallSubscribing(): boolean;
    get isScreenSubscribing(): boolean;
    get isDataSubscribing(): boolean;
    get audioEnabled(): boolean;
    get videoEnabled(): boolean;
    get screenEnabled(): boolean;
    get getAudioTrack(): MediaStreamAudioTrack | undefined;
    get getCameraStream(): RemoteStream | undefined;
    get getScreenStream(): RemoteStream | undefined;
    setViewConfig(view: AliRtcLocalView, track: AliRtcVideoTrack): void;
    /**
     * 销毁
     * @returns {Promise<void>}
     */
    clear(needStopSub?: boolean): Promise<void>;
    /**
     * 更新远端用户的设备状态
     * @param {string} status
     * @returns {void}
     */
    updateRemoteUserDeviceStatus(status: string): void;
    /**
     * 更新远端用户的推流状态，已经停止推流的 tracks 要 delete，或者整体 stop
     * @param {string} callid
     * @param {string} pullStreamUrl
     * @param {TrackInfo[]} tracks
     * @returns {Promise<TrackChangeState>}
     */
    updateRemoteTracks(callid: string, pullStreamUrl: string, _pubid: string, tracks?: TrackInfo[]): Promise<UpdateTracksResult>;
    /**
     * 订阅远端流
     * @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, reason?: SubscribeReason, startTs?: number): Promise<void>;
    /**
     * 订阅远端流
     * @param {SubscribeOptions} options 订阅配置
     * @param {boolean} secondary 是否订阅辅流
     * @returns {Promise<SubscribeResult>}
     */
    private subscribeProxy;
    /**
     * 移除对某些轨道的订阅
     * @param {UnSubscribeOptions} options 取消订阅轨道配置
     * @returns {Promise<SubscribeResult>}
     */
    private subscribeDeleteTracks;
    /**
     * 移除对某些轨道的订阅
     * @param {Partial<ISubConfigItem>} deleteTrack 删除订阅配置
     * @returns {Promise<SubscribeResult>}
     */
    private subscribeDeleteProxy;
    /**
     * 用保存的订阅参数恢复订阅
     */
    restore(): void;
    resumeAudio(): Promise<void>;
    resumeVideo(videoTrack?: AliRtcVideoTrack): Promise<void>;
    resumePlay(): void;
    /**
     * 停止订阅远端流
     * @param {string} userId
     * @param {UnSubscribeOptions} options
     * @returns {Promise<void>}
     */
    stopSubscribe(options?: UnSubscribeOptions): Promise<void>;
    /**
     * 停止对某个拉流 URL 的订阅
     * @param {url} pullStreamUrl
     * @returns {Promise<SubscribeResult>}
     */
    private doStopSubscribe;
    private updateStream;
    private updateScreenStream;
    private clearStream;
    private clearScreenStream;
    private onDataChannelMessage;
    private onDataChannelError;
    private dataChannelConnected;
    private subscribeDataChannel;
    private subscribeStopDataChannel;
    getRetryOptions(): RemoteSubscribeOptions | undefined;
    /**
     * 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(reason: SubscribeReason): {
        addOption: SubscribeOptions;
        deleteOption: SubscribeOptions;
    };
    private handleAutoPlayFailed;
    private handleAudioPaused;
    private handleVideoPlayFailded;
    private playAudio;
    stopPlayAudio(): void;
    playAll(isScreen?: boolean, isReplace?: boolean): void;
    /**
     * 播放
     */
    private play;
    stopPlayAll(isScreen?: boolean): void;
    /**
     * 是否开启声音
     * @param {boolean} enable
     * @returns {void}
     */
    toggleAudio(enable?: boolean): void;
    /**
     * 是否开启画面
     * @param {boolean} enable
     * @returns {void}
     */
    toggleVideo(enable?: boolean): void;
    /**
     * 是否开启画面
     * @param {boolean} enable
     * @returns {void}
     */
    toggleScreen(enable?: boolean): void;
    getStats(): Promise<{
        audio: AliRtcRemoteAudioStats[];
        video: AliRtcRemoteVideoStats[];
    }>;
    /**
     * 开启音量检测
     * @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 {
    userId: string;
    displayName: string;
    rtsManager: RtsManager;
    usersViewMap: {
        [key: string]: RemoteUserViewMap;
    };
    slsReporter: SLSReporter;
    signalingManager: SignalingManager;
    localUser?: LocalUser;
    audioVolumeIndicationInterval: number;
    playoutVolume: number;
    parameter: {
        [key: string]: any;
    };
}

declare interface RemoteUserMap {
    [userId: string]: RemoteUser;
}

declare interface RemoteUserStatus {
    userid: string;
    status: string;
}

declare interface RemoteUserViewMap {
    cameraViews: HTMLVideoElement[];
    screenViews: HTMLVideoElement[];
}

declare interface RoleForReconnectData {
    channelprofile?: AliRtcSdkChannelProfile;
    clientrole?: AliRtcSdkClientRole;
}

/**
 * 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;
}

/**
 * RoomServer消息中心
 * 负责发送和接收信令数据，消息重发、回复Response
 */
declare class RoomServerMessageCenter extends default_2<RoomServerListener> {
    static logName: string;
    protected authInfo: AliRtcAuthInfo;
    protected userName: string;
    protected channelProfile: AliRtcSdkChannelProfile;
    protected clientRole: AliRtcSdkClientRole;
    private slsReporter;
    protected wsClient: Socket;
    protected packageCreater: PackageCreater;
    protected keepAliveTimer: number;
    protected packageCache: Map<string, ISendPackage>;
    protected isFirstConnect: boolean;
    protected aliveChecker: number;
    protected reconnectData?: ReconnectData;
    constructor(authInfo: AliRtcAuthInfo, userName: string, channelProfile: AliRtcSdkChannelProfile, clientRole: AliRtcSdkClientRole, slsReporter: SLSReporter, maxConnectRetryCount?: number);
    get retryCount(): number;
    /**
     * 开始连接信令
     * @param roomserverurl
     */
    start(roomserverurl: string): void;
    /**
     * 开始心跳计时器，并自动发送心跳包
     */
    startKeepAlive(): void;
    /**
     * 停止心跳计时器
     */
    stopKeepAlive(): void;
    /**
     * 发送入会、重连前所缓存信息
     */
    sendPkgCache(): void;
    /**
     * 发送离开房间
     */
    leave(): void;
    /**
     * 透明通道消息上报
     * @param status
     */
    reportStatus(status: string): void;
    /**
     * 发送推流消息
     */
    publish(_pubInfo: PublishInfo): void;
    /**
     * 更新角色
     */
    roleUpdate(role: string): void;
    /**
     * 更新推拉流地址
     * @param {RefreshInfo} info
     */
    refreshUrl(info: RefreshInfo): void;
    refreshAuthInfo(authInfo: AliRtcAuthInfo): void;
    /**
     * 关闭websocket连接，并销毁
     */
    close(): void;
    /**
     * 通过此方法发送数据到业务信令
     * 此方法会自动保存发送的package，并做重发.
     * response包不要通过此方法发送
     * @param pkg
     */
    protected sendPackage(pkg: ISendPackage): void;
    /**
     * 更新重连 payload
     * @param {ReconnectData} reconnectData
     */
    updateReconnectData(reconnectData?: ReconnectData): void;
    /**
     * 收到websocket连接成功的消息
     * ws连接建立成功后立刻发送join消息
     */
    protected onOpen(): void;
    /**
     * 收到业务信令消息
     * @param event
     */
    protected onMessage(event: MessageEvent): void;
    /**
     * 收到错误消息
     * @param event
     */
    protected onError(event: Event): void;
    /**
     * 连接失败
     */
    protected onConnectFail(): void;
    /**
     * 鉴权失败
     */
    protected onAuthInvalid(): void;
    /**
     * 网络异常
     */
    protected onNetworkError(): void;
    /**
     * 收到websocket断开消息
     * @param event
     */
    protected onClose(event: CloseEvent): void;
    protected onKeepAliveRsp(): void;
    protected onReconnectStart(): void;
    protected onReconnectFail(): void;
    /**
     * 处理 Keepalive 心跳，加速判断网络异常
     */
    protected checkAlive(): void;
    /**
     * 处理收到的业务信令
     * @param response
     */
    protected onRecvMessage(response: any): void;
    /**
     * 收到入会消息的response
     * @param response
     */
    protected onJoinResult(response: any): void;
    /**
     * 收到离开消息的response
     * @param response
     */
    protected onLeaveResult(response: any): void;
    /**
     * 收到publish消息的response
     * @param response
     */
    protected onPublishResult(response: any): void;
    /**
     * 收到roleUpdate消息的response
     * @param response
     */
    protected onRoleUpdateResult(response: any): void;
    /**
     * 收到 refreshurl 消息的response
     * @param response
     */
    protected onRefreshUrlResult(response: any): void;
    /**
     * 收到重联消息的response
     * @param response
     */
    protected onReconnectResult(response: any): void;
    /**
     * 收到业务信令下发的消息
     * @param response
     */
    protected onNotifyJoin(response: any): void;
    /**
     * 收到业务信令下发的透明通道消息
     * @param response
     */
    protected onNotifyStatus(response: any): void;
    /**
     * 收到远端用户离开的通知
     * @param response
     */
    protected onNotifyLeave(response: any): void;
    /**
     * 收到远端用户推流的通知
     * @param response
     */
    protected onNotifyPublish(response: any): void;
    /**
     * 收到statusreport的response
     * @param response
     */
    protected onStatusReport(response: any): void;
    /**
     * 收到被踢出频道的消息
     * @param {any} response
     */
    protected onBye(response: any): void;
}

declare class RoomServerSignaling extends default_2<RoomServerListener> {
    /**
     * @ignore
     */
    static logName: string;
    protected messageCenter?: RoomServerMessageCenter;
    protected authInfo?: AliRtcAuthInfo;
    protected userName?: string;
    protected channelProfile: AliRtcSdkChannelProfile;
    protected clientRole: AliRtcSdkClientRole;
    private slsReporter;
    protected reconnectData?: ReconnectData;
    protected roomServerUri: string;
    constructor(channelProfile: AliRtcSdkChannelProfile, clientRole: AliRtcSdkClientRole, slsReporter: SLSReporter, env?: AliRtcEnv);
    /**
     * 更新重连数据
     * @param {ReconnectData} reconnectData
     */
    private updateReconnectData;
    /**
     * 释放资源
     */
    reset(): void;
    /**
     * 加入频道
     * @param {AliRtcAuthInfo} authInfo
     * @returns
     */
    join(authInfo: AliRtcAuthInfo, userName: string, maxConnectRetryCount?: number): Promise<JoinResult>;
    /**
     * 离开频道
     * @returns
     */
    leave(): Promise<SignalingResult>;
    /**
     * 开始推流
     * @param pubInfo
     * @returns
     */
    publish(pubInfo: PublishInfo): Promise<SignalingResult>;
    roleUpdate(role: AliRtcSdkClientRole): Promise<SignalingResult>;
    setChannelProfile(channelProfile: AliRtcSdkChannelProfile): void;
    /**
     * 更新推流地址
     */
    refreshUrl(info: RefreshInfo): Promise<RefreshUrlResult>;
    /**
     * 上报设备状态
     * @param status
     */
    reportStatus(status: string): Promise<void>;
    protected handleSignalError(code: number, description: string, errorCode?: AliRtcErrorCode): AliRtcError;
    refreshAuthInfo(authInfo: AliRtcAuthInfo): Promise<void>;
    /**
     * 初始化MessageCenter，增加事件监听
     */
    protected initMessageCenter(): void;
}

declare class RtsManager extends default_2<RtsManagerEventListener> {
    /**
     * @ignore
     */
    static logName: string;
    private rts;
    private encodedInsertableStreams;
    private audioRedEnabled;
    private localStreamManager;
    private slsReporter;
    private _rtsReconnecting;
    private _rtsPeerConnectionType;
    private connecting;
    private connected;
    private _pcTraceId;
    private _localJoinTime;
    private connectionResolve?;
    private connectingPromise?;
    private dcResolve?;
    private dcReject?;
    private dcConnectingPromise?;
    private _publishingTracks;
    private parameter;
    constructor(localStreamManager: LocalStreamManager, slsReporter: SLSReporter, parameter: Parameter);
    private addRTSListener;
    /**
     * 更新鉴权信息，传入一个新的带鉴权的 URL，RTS 会去除鉴权信息做更新
     * @param newUrlWithAuth
     */
    updateAuth(newUrlWithAuth: string): void;
    get pcTraceId(): string;
    set localJoinTime(time: number);
    setEncodedInsertableStreams(enable: boolean): void;
    setAudioRedEnabled(enable: boolean): void;
    clear(): void;
    private startConnect;
    private setConnected;
    get isConnecting(): boolean;
    get isConnected(): boolean;
    get publishingTracks(): TrackInfo[];
    private getPubMsid;
    private updatePublishingTracks;
    private httpPublish;
    waitPublishSender(streamUrl: string, isAudio?: boolean, msid?: string): Promise<any>;
    getPublishVideoStats(streamUrl: string, msid?: string): Promise<unknown>;
    getPublishAudioStats(streamUrl: string, msid?: string): Promise<unknown>;
    publishAdd(streamUrl: string, stream: LocalStream, callId: string, extra?: PublishExtraParams, isResume?: boolean, 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>;
    publishAddDataChannel(streamUrl: string): Promise<any>;
    publishStopDataChannel(streamUrl: string, datachannel: any): Promise<void>;
    unpublish(): Promise<void>;
    /**
     * 获取 sub/subAdd config
     * @param {ISubscribeConfig | ISubConfigItem} options
     * @returns {Omit<ISignalConfig | ISubConfigItem, 'url'>}
     */
    private getSubConfig;
    private httpSubscribe;
    private subscribeAdd;
    subscribeDelete(subscribeOptions: RemoteSubscribeOptions): Promise<{
        url: string;
        stream: any;
    } | undefined>;
    subscibeStop(streamUrl: string): Promise<void>;
    subscribeAddDataChannel(streamUrl: string): Promise<any>;
    subscribeStopDatachannel(streamUrl: string, datachannel: any): Promise<any>;
    publish(streamUrl: string, callId: string, isResume?: boolean, reason?: PublishReason): Promise<{
        traceId: string;
        l1ip?: string;
    }>;
    private reportSubscribeLatency;
    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;
    getDatachannelByMsid(config: {
        url: string;
        msid?: string;
    }): any;
    getPCStats(): Promise<any>;
    sendSEI(streamUrl: string, data: ArrayBuffer, repeatCount: number, payloadType: number): Promise<void>;
}

declare interface RtsManagerEventListener {
    reconnecting: (type: RtsPeerConnectionType) => {};
    disconnected: () => {};
    connected: (isReconnected: boolean, type: RtsPeerConnectionType) => {};
    subscribeexceeds: () => {};
    seimessage: (url: string, payloadType: number, data: ArrayBuffer) => {};
}

declare enum RtsPeerConnectionType {
    PUBLISH = "publish",
    SUBSCRIBE = "subscribe"
}

declare interface SendPackageAuthInfo {
    timestamp: number;
    nonce?: string;
    token: string;
    tokenrole: AliRtcSdkTokenRole;
}

declare class SignalingManager extends default_2<RoomServerListener> {
    /**
     * @ignore
     */
    static logName: string;
    protected signaling?: RoomServerSignaling;
    protected authInfo?: AliRtcAuthInfo;
    protected userName?: string;
    protected channelProfile: AliRtcSdkChannelProfile;
    protected clientRole: AliRtcSdkClientRole;
    protected env: AliRtcEnv;
    protected maxConnectRetryCount?: number;
    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, timeRecorder?: TimeRecorder): Promise<JoinResult>;
    retryJoin(): void;
    /**
     * 向业务信令发送离会
     * @returns
     */
    leave(): Promise<SignalingResult>;
    /**
     * 推流
     * @param pubInfo
     * @returns
     */
    publish(pubInfo: PublishInfo): Promise<SignalingResult>;
    /**
     * 通过透明通道发送设备状态
     * @param {string} status
     * @returns
     */
    reportStatus(status: string): Promise<void>;
    /**
     * 切换角色
     * @param {AliRtcSdkClientRole} role
     * @returns
     */
    setClientRole(role: AliRtcSdkClientRole): Promise<SignalingResult | undefined>;
    /**
     * 切换模式
     * @param {AliRtcSdkChannelProfile} channelProfile
     */
    setChannelProfile(channelProfile: AliRtcSdkChannelProfile): void;
    /**
     * 更新推流url
     * @param {RefreshInfo} info
     * @returns
     */
    refreshUrl(info: RefreshInfo): Promise<RefreshUrlResult>;
    refreshAuthInfo(authInfo: AliRtcRefreshAuthInfo): Promise<void>;
    /**
     * 监听业务信令相关的事件
     */
    protected initSignaling(): void;
    /**
     * 收到心跳消息
     */
    protected onKeepAliveRsp(): void;
    /**
     * 收到远端用户入会消息
     * @param {JoinInfo[]} users
     */
    protected onNotifyJoin(users: JoinInfo[]): void;
    /**
     * 收到远端用户的透明通道消息
     * @param {RemoteUserStatus[]} users
     */
    protected onNotifyStatus(users: RemoteUserStatus[]): void;
    /**
     * 收到远端用户的离会消息
     * @param {LeaveInfo[]} users
     */
    protected onNotifyLeave(users: LeaveInfo[]): void;
    /**
     * 收到远端用户推流状态变化消息
     * @param {PublishInfo[]} users
     */
    protected onNotifyPublish(users: PublishUser[]): void;
    /**
     * 收到被踢掉消息
     * @param {string} reason
     * @param {string} description
     */
    protected onBye(reason: AliRtcOnByeType, description: string): void;
    /**
     * 收到连接失败消息
     * @param {AliRtcError} err
     */
    protected onConnectFail(err: AliRtcError): void;
    /**
     * 收到网络异常消息
     * @param {AliRtcError} err
     */
    protected onNetworkError(err: AliRtcError): void;
    /**
     * 收到网络异常消息
     */
    protected onAuthInvalid(): void;
    protected onReconnectStart(): void;
    /**
     * 收到重连成功消息
     * @param {JoinInfo[]} users
     */
    protected onReconnectSuccess(users: JoinInfo[]): void;
    /**
     * 收到重连失败消息
     */
    protected onReconnectFail(err: AliRtcError): void;
    /**
     * 收到错误消息
     * @param {AliRtcError} err
     */
    protected onError(err: AliRtcError): void;
}

declare interface SignalingResult {
    tid: string;
    code?: number;
    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 ntpClock;
    private static staticClient;
    private static getLogClient;
    static reportOSSUpload(sessionId: string, date: string, responseCode?: number): void;
    constructor(engine: WrappedAliRtcEngine);
    private customFields;
    /**
     * 设置通用字段
     * @param fields 字段键值对
     */
    setCustomFields(fields: {
        [key: string]: any;
    }): void;
    /**
     * 获取通用字段
     * @param key 字段键
     * @returns 字段值或undefined
     */
    getCustomField(key: string): any;
    /**
     * 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, startTs: number): void;
    reportLeaveInvoked(): void;
    reportSEIMessage(payloadType: number, length: number, repeatCount: number, delay: number, isKey: boolean): void;
    /**
     * 接通耗时事件（12001）
     * 用stage和status来区分接通过程中各个时间节点。
     * @param state
     * @param status
     * @param calid
     * @param tckid
     */
    reportConnectionLatencyEvent<Stage extends keyof ValidStatusMap>(stage: Stage, status: ValidStatusMap[Stage], calid: string, tckid: string, tm?: number, ntptm?: number, http?: 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;
}

/**
 * Websocket类
 */
declare class Socket extends default_2<SocketListener> {
    /**
     * @ignore
     */
    static logName: string;
    protected ws?: WebSocket;
    protected socketStatus: CONNECTSTATUS;
    protected wsUrl: string;
    protected lastConnectStartTime: number;
    protected maxConnectRetryCount: number;
    connectRetryCount: number;
    protected gotErr: boolean;
    protected closedByInvoke: boolean;
    protected isReconnecting: boolean;
    protected networkAvailableChecked: boolean;
    private reconnectTimeoutId;
    constructor(maxConnectRetryCount?: number);
    get connectUrl(): string;
    /**
     * 返回websocket连接的状态
     */
    get connectStatus(): CONNECTSTATUS;
    /**
     * 开始连接
     * @param wsUrl
     */
    start(wsUrl: string): void;
    /**
     * 发送消息
     * @param pkg
     */
    send(pkg: ISendPackage | IResponse): boolean;
    /**
     * 主动断开连接
     */
    close(): void;
    reconnect(): void;
    /**
     * 连接成功
     * @param event
     */
    protected onOpen(event: Event): void;
    /**
     * 收到消息
     * @param event
     */
    protected onMessage(event: MessageEvent): void;
    /**
     * 收到错误
     * @param event
     */
    protected onError(event: Event): void;
    /**
     * 连接断开
     * @param event
     */
    protected onClose(event: CloseEvent): Promise<void>;
    /**
     * 初始化websocket
     */
    protected initWebSocket(): void;
    /**
     * 反初始化websocket
     */
    protected uninitWebsocket(): void;
}

/**
 * Socket的支持的事件
 */
declare interface SocketListener {
    onOpen: (event: Event) => void;
    onMessage: (event: MessageEvent) => void;
    onError: (event: Event) => void;
    onClose: (event: CloseEvent) => void;
    onConnectFail: () => void;
    onAuthInvalid: () => void;
    onNetworkError: () => void;
    onReconnectStart: () => void;
    onReconnectFail: (event: Event) => void;
}

declare interface StopPreviewConfig {
    videoElement?: HTMLVideoElement;
    primary?: boolean;
    screen?: boolean;
}

declare type StopPublishOptions = StreamOptions;

declare interface StreamListener {
    [DeviceStatusChange.Remote]: (event: DeviceStatusChangeType) => void;
    error: (err: AliRtcError) => void;
}

declare interface StreamOptions {
    audio?: boolean;
    video?: boolean;
    screen?: boolean;
    data?: 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;
    isScreenSubscribing: boolean;
    audioEnabled: boolean;
    videoEnabled: boolean;
    screenEnabled: boolean;
}

declare type SubscribeOptions = StreamOptions & {
    aMsid?: string;
    vMsid?: string;
};

declare enum SubscribeReason {
    InRoom = "inRoom",
    Join = "join",
    Publish = "publish",
    User = "user",
    Resume = "resume",
    Retry = "retry",
    Reconnect = "reconnect"
}

declare interface TimeRecorder {
    start: number;
    [key: string]: number;
}

declare enum TrackEvent_2 {
    VideoTrackEnded = "videoTrackEnded",
    AudioTrackEnded = "audioTrackEnded",
    ScreenTrackEnded = "screenTrackEnded"
}

declare interface TrackInfo {
    ssrc: string;
    codec: string;
    msid?: MsidType;
    type: string;
    from?: string | undefined;
    userdata?: string;
    trackId?: string;
    attr?: Record<string, any>;
    pt?: number;
}

declare type UnSubscribeOptions = SubscribeOptions;

declare interface UpdateTracksResult {
    restoreSub?: boolean;
}

declare class User extends default_2<UserListener> {
    protected uid: string;
    protected name: string;
    protected rtsManager: RtsManager;
    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;
    /**
     * 获取音频 level
     */
    getAudioLevel(): number;
    enableAudioVolumeIndication(interval: number): void;
    /**
     * 销毁用户，清空数据
     */
    clear(): void;
}

declare interface UserListener {
    [DeviceStatusChange.Local]: (status: string) => void;
    [DeviceStatusChange.Remote]: (event: DeviceStatusChangeType, userId: string) => void;
    [ClientEventType.PullStreamStats]: (userId: string, stat: any, pullStreamUrl: string) => void;
    [ClientEventType.RtsSubscribeExceeds]: (userId: string, options?: SubscribeOptions) => void;
    [TrackEvent_2.VideoTrackEnded]: () => void;
    [TrackEvent_2.AudioTrackEnded]: () => void;
    [TrackEvent_2.ScreenTrackEnded]: () => void;
    remoteTrackAvailableChange: (uid: string, aliRtcAudioTrack: AliRtcAudioTrack, aliRtcVideoTrack: AliRtcVideoTrack) => void;
    audioSubscribeStateChanged: (uid: string, oldState: AliRtcSubscribeState, newState: AliRtcSubscribeState, elapseSinceLastState: number) => void;
    videoSubscribeStateChanged: (uid: string, oldState: AliRtcSubscribeState, newState: AliRtcSubscribeState, elapseSinceLastState: number) => void;
    screenSubscribeStateChanged: (uid: string, oldState: AliRtcSubscribeState, newState: AliRtcSubscribeState, elapseSinceLastState: number) => void;
    dataSubscribeStateChanged: (uid: string, oldState: AliRtcSubscribeState, newState: AliRtcSubscribeState, elapseSinceLastState: number) => void;
    audioPublishStateChanged: (oldState: AliRtcPublishState, newState: AliRtcPublishState, elapseSinceLastState: number) => void;
    videoPublishStateChanged: (oldState: AliRtcPublishState, newState: AliRtcPublishState, elapseSinceLastState: number) => void;
    dualStreamPublishStateChanged: (oldState: AliRtcPublishState, newState: AliRtcPublishState, elapseSinceLastState: number) => void;
    screenPublishStateChanged: (oldState: AliRtcPublishState, newState: AliRtcPublishState, elapseSinceLastState: number) => void;
    dataPublishStateChanged: (oldState: AliRtcPublishState, newState: AliRtcPublishState, elapseSinceLastState: number) => void;
    localDeviceException: (localDeviceType: AliRtcEngineLocalDeviceType, localDeviceExceptionType: AliRtcEngineLocalDeviceExceptionType, description: string) => void;
    publishDataError: (error: AliRtcError) => void;
    remoteAudioAutoPlayFail: (uid: string) => void;
    remoteVideoAutoPlayFail: (uid: string, track: AliRtcVideoTrack) => void;
    remoteAudioPlayError: (uid: string, reason?: string) => void;
    remoteVideoPlayError: (uid: string, reason?: string) => void;
    remoteSubscribeError: (error: AliRtcError) => void;
    occurError: (error: AliRtcError) => void;
    remoteDataChannelMessage: (uid: string, message: AliRtcDataChannelMsg) => void;
}

declare type ValidStatusMap = {
    [ConnectionLatencyStage.CREATE_ENGINE]: ConnectionLatencyStatus.START | ConnectionLatencyStatus.END;
    [ConnectionLatencyStage.JOIN]: ConnectionLatencyStatus.START | ConnectionLatencyStatus.SEND_SIG | ConnectionLatencyStatus.RECV_RST | ConnectionLatencyStatus.END;
    [ConnectionLatencyStage.FIRST_REMOTE]: ConnectionLatencyStatus.RECEIVED | ConnectionLatencyStatus.DECODED | ConnectionLatencyStatus.PLAYED;
    [ConnectionLatencyStage.RECV_NOTIFY_PUBLISH]: any;
    [ConnectionLatencyStage.HANDLE_PUB_TASK]: ConnectionLatencyStatus.START | ConnectionLatencyStatus.END;
    [ConnectionLatencyStage.HANDLE_SUB_TASK]: ConnectionLatencyStatus.START | ConnectionLatencyStatus.END;
};

declare type VideoProfileWithSendFramerate = Partial<IProfile & {
    maxSendFrameRate: number;
}>;

declare interface VideoScaler {
    getVideoTrack: () => MediaStreamTrack;
    updateOptions: (options: VideoScalerOptions) => void;
    dispose: () => void;
}

declare class VideoScalerFactory {
    static isSupport(): boolean;
    /**
     * 获取实例
     * @param {MediaStreamTrack} videoTrack
     * @param {IProfile} profile
     * @return {VideoScaler}
     */
    static getInstance(videoTrack: MediaStreamTrack, options: VideoScalerOptions): VideoScaler;
}

declare interface VideoScalerOptions {
    width: number;
    height: number;
    frameRate: number;
}

declare enum VideoStreamSource {
    Camera = 0,
    Screen = 1,
    Image = 2
}

declare const WrappedAliRtcEngine: typeof AliRtcEngine_2;

declare type WrappedAliRtcEngine = InstanceType<typeof AliRtcEngine_2>;
export { WrappedAliRtcEngine as AliRtcEngine }
export default WrappedAliRtcEngine;

export { }