aliyun-rtc-sdk 7.1.8 → 7.2.0-beta.0

package/dist/plugins/audioProcessPlugin.d.ts

@@ -702,15 +702,6 @@ var grammarExports = grammar$1.exports;
 	}; 
 } (parser));
 
-interface ISubDeleteResult {
-    url: string;
-    stream?: any;
-    aStream?: any;
-    vStream?: any;
-    aMsid?: string;
-    vMsid?: string;
-}
-
 interface IListenerMap {
     [s: string]: any;
 }
@@ -853,34 +844,6 @@ declare class LocalStream extends Stream {
     protected handleAudioTrackEnded(): void;
 }
 
-declare class RemoteStream extends Stream {
-    protected recvCanPlay: boolean;
-    protected autoplay: boolean;
-    constructor();
-    set muted(muted: boolean);
-    get muted(): boolean;
-    get audioTrack(): MediaStreamTrack | undefined;
-    get videoTrack(): MediaStreamTrack | undefined;
-    get hasVideo(): boolean;
-    get hasAudio(): boolean;
-    get mediaStream(): MediaStream | undefined;
-    updateMediaStream(value: MediaStream): void;
-    getMediaStream(): MediaStream | undefined;
-    mergeStream(newStream: MediaStream): void;
-    play(element: HTMLMediaElement, config?: {
-        autoplay?: boolean;
-    }): void;
-    disableVideo(): void;
-    enableVideo(): void;
-    disableAudio(): void;
-    enableAudio(): void;
-    stop(): void;
-    onCanPlay(): void;
-    protected release(): void;
-    protected bindMediaStream(mediastream: MediaStream): void;
-    protected tryToPlay(): void;
-}
-
 interface IStreamConfig {
     audio?: MediaTrackConstraints | boolean;
     video?: MediaTrackConstraints | boolean;
@@ -938,4428 +901,35 @@ declare global {
     }
 }
 
-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 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;
-}
-
-type DeviceStatusChangeType = ClientEventType.MuteAudio | ClientEventType.UnmuteAudio | ClientEventType.MuteScreen | ClientEventType.UnmuteScreen | ClientEventType.MuteVideo | ClientEventType.UnmuteVideo;
-declare enum CmdType {
-    Add = 0,
-    Del = 1,
-    Update = 2,
-    None = 100
-}
-declare enum CodecType {
-    OPUS = "opus",
-    H264 = "H264",
-    H265 = "H265",
-    AV1 = "AV1",
-    UNKNOWN = ""
-}
-interface MediaTrackInfo {
-    ssrc: string;
-    msid: string;
-    red: number;
-    codec: CodecType;
-    sample?: number;
-    pt: number;
-    type?: CmdType;
-}
-declare enum RemoteTrackSubscribeState {
-    NotSubscribe = 0,
-    Subscribing = 1,
-    Subscribed = 2
-}
-interface RemoteMediaTrackInfo extends MediaTrackInfo {
-    subscribeState?: RemoteTrackSubscribeState;
-}
-interface StreamListener {
-    [DeviceStatusChange.Remote]: (event: DeviceStatusChangeType) => void;
-    error: (err: AliRtcError) => void;
-}
-declare enum TrackEvent {
-    VideoTrackEnded = "videoTrackEnded",
-    AudioTrackEnded = "audioTrackEnded",
-    ScreenTrackEnded = "screenTrackEnded"
-}
-
-/** 角色鉴权 主播pub｜观众sub｜转推流鉴权（跨房间转推中使用）relay */
-type AliRtcSdkTokenRole = 'pub' | 'sub' | 'relay';
-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;
-}
-type AliRtcRefreshAuthInfo = Pick<AliRtcAuthInfo, 'userId' | 'nonce' | 'token' | 'timestamp' | 'role'>;
-
-declare enum AliRtcEnv {
-    pre = "pre",
-    prod = "prod"
-}
-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"
-}
-declare enum AliRtcSdkClientRole {
-    /** 互动角色 */
-    AliRtcSdkInteractive = "interactive",
-    /** 观众角色 */
-    AliRtcSdkLive = "live"
-}
-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 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
-}
-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
-}
-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
-}
-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
-}
-declare enum AliRtcAudioTrack {
-    /** 无麦克风 */
-    AliRtcAudioTrackNo = 0,
-    /** 麦克风 */
-    AliRtcAudioTrackMic = 1,
-    /** 第二音频流 */
-    AliRtcAudioTrackDual = 2
-}
-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 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
-}
-declare enum AliRtcSubscribeState {
-    /** 初始状态 */
-    AliRtcStateSubscribeIdle = 0,
-    /** 未订阅 */
-    AliRtcStateNoSubscribe = 1,
-    /** 订阅中 */
-    AliRtcStateSubscribing = 2,
-    /** 已订阅 */
-    AliRtcStateSubscribed = 3
-}
-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
-}
-/**
- * @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
-}
-declare enum AliRtcCameraDirection {
-    /** 无效 */
-    CAMERA_INVALID = -1,
-    /** 后置 */
-    CAMERA_REAR = 0,
-    /** 前置 */
-    CAMERA_FRONT = 1
-}
-
-interface SendPackageAuthInfo {
-    timestamp: number;
-    nonce?: string;
-    token: string;
-    tokenrole: AliRtcSdkTokenRole;
-}
-interface PackageAdaptInfo {
-    os_name: string;
-    device_name: string;
-    brand_name: string;
-    device_key: string;
-    os_version: string;
-    sdk_version: string;
-    version: string;
-}
-interface JoinResult {
-    l1ip?: string;
-    l1subip?: string;
-    pushstreamurl: string;
-    tid: string;
-    users: JoinInfo[];
-    timestamp: number;
-    config: any;
-}
-interface RefreshUrlResult extends SignalingResult {
-    pushstreamurl?: string;
-    users?: RefreshUser[];
-}
-interface SignalingResult {
-    tid: string;
-    code?: number;
-    message?: string;
-}
-interface RoleForReconnectData {
-    channelprofile?: AliRtcSdkChannelProfile;
-    clientrole?: AliRtcSdkClientRole;
-}
-interface ReconnectData {
-    role?: RoleForReconnectData;
-    auth?: SendPackageAuthInfo;
-    publish?: {
-        users: [PublishInfo];
-    };
-}
-
-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"
-}
-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;
-        adapt?: PackageAdaptInfo;
-        sessionid?: string;
-        pushstreamurl?: string;
-        users?: any[];
-    };
-}
-interface IResponse {
-    version: number;
-    code: number;
-    msgtype: MessageType;
-    tid: string;
-    isproxy?: boolean;
-}
-declare class PackageCreater {
-    protected authInfo: AliRtcAuthInfo;
-    protected userName: string;
-    protected seqIndex: number;
-    constructor(authInfo: AliRtcAuthInfo, userName: string);
-    get roleAuth(): SendPackageAuthInfo | undefined;
-    get adaptInfo(): PackageAdaptInfo | 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(extraData?: any): 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;
-}
-
-/**
- * websocket链接状态
- */
-declare enum CONNECTSTATUS {
-    DISCONNECTED = 0,
-    CONNECTING = 1,
-    CONNECTED = 2
-}
-/**
- * Socket的支持的事件
- */
-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;
-}
-/**
- * Websocket类
- */
-declare class Socket extends EventEmitter$1<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;
-}
-
-interface WebTrackerOptions {
-    host: string;
-    project: string;
-    logstore: string;
-    /**
-     * 发送时间阈值, default 10s
-     */
-    time?: number;
-    /**
-     * 发送条数阈值, default 10
-     */
-    count?: number;
-    /**
-     * 日志主题
-     */
-    topic?: string;
-    /**
-     * 日志来源
-     */
-    source?: string;
-    /**
-     * 日志标签
-     */
-    tags?: Record<string, any>;
-    installUnloadHook?: (hook: () => void) => void;
-    sendPayload: (url: string, payload: string) => void;
-}
-type WebTrackerBrowserOptions = Omit<WebTrackerOptions, 'installUnloadHook' | 'sendPayload'>;
-interface StsPlugin {
-    transString: (obj: Record<string, any>) => Record<string, any>;
-    process: (url: string, payload: string) => Promise<{
-        data: any;
-        header: {
-            [k: string]: string;
-        };
-    }>;
-}
-
-declare class WebTracker {
-    private timer;
-    private time;
-    private count;
-    private arr;
-    private url;
-    private opt;
-    constructor(opt: WebTrackerOptions);
-    private assemblePayload;
-    private platformSend;
-    private transString;
-    private sendImmediateInner;
-    private sendInner;
-    send(originLog: Record<string, any>): void;
-    sendImmediate(originLog: Record<string, any>): void;
-    sendBatchLogs(originLogs: Array<Record<string, any>>): void;
-    sendBatchLogsImmediate(originLogs: Array<Record<string, any>>): void;
-    overwriteTransString(stsPlugin: StsPlugin): void;
-    getOpt(): WebTrackerOptions;
-}
-
-declare class WebTrackerBrowser extends WebTracker {
-    constructor(opt: WebTrackerBrowserOptions);
-    useStsPlugin(stsPlugin: StsPlugin): void;
-}
-
-declare class NTPClient {
-    private static instance;
-    private ntpClock;
-    private constructor();
-    static getInstance(): NTPClient;
-    now(): number;
+declare enum AliRtcRawDataStreamType {
+    /** 相机流 */
+    AliRtcSdkStreamTypeCapture = 0,
+    /** 屏幕共享流 */
+    AliRtcSdkStreamTypeScreen = 1
 }
 
-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 enum VideoStreamSource {
+    Camera = 0,
+    Screen = 1,
+    Image = 2
 }
-interface OSSSTSToken {
-    access_key_id: string;
-    access_key_secret: string;
-    security_token: string;
-    region_endpoint: string;
-    bucket_name: string;
-    expiration: number;
+declare enum AudioStreamSource {
+    Microphone = 0,
+    Screen = 1,
+    Mixed = 2
 }
 /**
- * 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?: WebTrackerBrowser;
-    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;
-}
-
-interface TimeRecorder {
-    start: number;
-    [key: string]: number;
-}
-interface LogInfo {
-    [key: string]: string | number;
-}
-
-interface VideoScalerOptions {
-    width: number;
-    height: number;
-    frameRate: number;
-}
-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;
-}
-
-interface AudioLevelMonitorListener {
-    audioLevel: (level: number) => void;
-}
-declare class AudioLevelMonitor extends EventEmitter$1<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 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>;
-}
-
-interface PublishStreamInfo {
-    audio?: LocalStream;
-    video?: LocalStream;
-    videoSmall?: LocalStream;
-    screen?: LocalStream;
-    videoProfile?: LocalProfileManager;
-    screenProfile?: LocalProfileManager;
-}
-
-interface PluginManagerListener {
-    added: (plugin: AliRtcPlugin) => void;
-    removed: (plugin: AliRtcPlugin) => void;
-    updated: (plugin: AliRtcPlugin) => void;
-}
-declare class PluginManager extends EventEmitter$1<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[];
-}
-
-interface LocalStreamManagerListener {
-    trackended: (type: TrackEvent) => {};
-    publishupdate: () => {};
-}
-declare class LocalStreamManager extends EventEmitter$1<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: MediaTrackConstraints): void;
-    /**
-     * 设置 micAudioConstraints
-     * @param {IAudioConstraints} micAudioConstraints
-     */
-    setMicAudioConstraints(micAudioConstraints: MediaTrackConstraints): void;
-}
-
-interface DeviceStatus {
-    micOpen: boolean;
-    micInterrupted: boolean;
-    audioDisable: boolean;
-    cameraOpen: boolean;
-    videoDisable: boolean;
-    screenShare: boolean;
-    screenDisable: boolean;
-    backgroundMode: boolean;
-    accompanyPlaying: boolean;
-}
-
-interface UpdateTracksResult {
-    restoreSub?: boolean;
-}
-declare class RemoteStreamInfo extends EventEmitter$1<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;
-}
-
-type AliRtcLocalView = string | HTMLVideoElement | string[] | HTMLVideoElement[] | null;
-
-interface AliRtcRemoteAudioStats {
-    uid: string;
-    track: AliRtcAudioTrack;
-    stats: any;
-}
-
-interface AliRtcRemoteVideoStats {
-    uid: string;
-    track: AliRtcVideoTrack;
-    streamType?: AliRtcVideoStreamType;
-    stats: any;
-}
-
-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;
-}
-
-interface RemoteUserViewMap {
-    cameraViews: HTMLVideoElement[];
-    screenViews: HTMLVideoElement[];
-}
-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;
-}
-
-interface RemoteSubscribeOptions {
-    remoteUser: RemoteUser;
-    options: SubscribeOptions;
-}
-declare enum RtsPeerConnectionType {
-    PUBLISH = "publish",
-    SUBSCRIBE = "subscribe"
-}
-interface RtsManagerEventListener {
-    reconnecting: (type: RtsPeerConnectionType) => {};
-    disconnected: () => {};
-    connected: (isReconnected: boolean, type: RtsPeerConnectionType) => {};
-    subscribeexceeds: () => {};
-    seimessage: (url: string, payloadType: number, data: ArrayBuffer) => {};
-}
-declare class RtsManager extends EventEmitter$1<RtsManagerEventListener> {
-    /**
-     * @ignore
-     */
-    static logName: string;
-    private rts;
-    private encodedInsertableStreams;
-    private audioRedEnabled;
-    private localStreamManager;
-    private slsReporter;
-    private _rtsPeerConnectionType;
-    private connecting;
-    private connected;
-    private rtsDisconnected;
-    private _pcTraceId;
-    private _localJoinTime;
-    private connectionResolve?;
-    private connectingPromise?;
-    private dcResolve?;
-    private dcReject?;
-    private dcConnectingPromise?;
-    private _publishingTracks;
-    private parameter;
-    preferredPubL1Ip?: string;
-    preferredSubL1Ip?: string;
-    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;
-    setReceiverPreferredVideoCodecH265(): void;
-    setSenderPreferredVideoCodecH265(): void;
-    supportVideoCodec(codec: CodecType): boolean | undefined;
-    changeVideoCodec(codec: CodecType): 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 handleHTTPConfig;
-    private httpSubscribe;
-    private subscribeAdd;
-    subscribeDelete(subscribeOptions: RemoteSubscribeOptions): Promise<ISubDeleteResult | undefined>;
-    subscribeStop(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 class User extends EventEmitter$1<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;
-}
-
-interface AliRtcLocalAudioStats {
-    track: AliRtcAudioTrack;
-    stats: any;
-}
-
-interface AliRtcLocalVideoStats {
-    track: AliRtcVideoTrack;
-    streamType?: AliRtcVideoStreamType;
-    stats: any;
-}
-
-declare enum AliRtcDataMsgType {
-    AliEngineDataMsgNone = 0,
-    AliEngineDataMsgMusicProgress = 1,
-    AliEngineDataMsgCustom = 2
-}
-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;
-}
-
-interface AliEngineCameraCapturerConfiguration {
-    width?: number;
-    height?: number;
-    frameRate?: number;
-    maxSendFrameRate?: number;
-    bitrate?: number;
-    cameraDirection?: AliRtcCameraDirection;
-    deviceId?: string;
-}
-type VideoProfileWithSendFramerate = Partial<IProfile & {
-    maxSendFrameRate: number;
-}>;
-
-declare class LocalUser extends User {
-    /**
-     * @ignore
-     */
-    static logName: string;
-    private _url;
-    private joinTime;
-    private publishPromise;
-    private previewElements;
-    private screenPreviewElements;
-    private streamManager;
-    private dc;
-    protected streamTracks?: TrackInfo[];
-    protected publishId: string;
-    protected signalingManager: SignalingManager;
-    private shouldPublish;
-    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;
-    startDefaultCameraAndMic(audioOnly?: boolean): 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>;
-    startCameraAndMic(): Promise<LocalStream | undefined>;
-    /**
-     * 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): void;
-    getStats(): Promise<{
-        audio: AliRtcLocalAudioStats[];
-        video: AliRtcLocalVideoStats[];
-        candidate: any;
-    }>;
-}
-
-interface AudioMixerPluginOptions {
-    audio?: number;
-    screen?: number;
-}
-
-type AliRtcDualVideoPluginOptions = VideoScalerOptions;
-
-declare enum AliRtcCapabilityProfile {
-    Default = 0,
-    Human = 1,
-    Robot = 2
-}
-type AliRtcChannelParam = {
-    userName?: string;
-    capabilityProfile?: AliRtcCapabilityProfile;
-};
-
-interface RemoteUserMap {
-    [userId: string]: RemoteUser;
-}
-declare class BizControl extends EventEmitter$1<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?;
-    private audioProcessorPlugin?;
-    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;
-    private audio3AConfig;
-    private dcReadyHelper?;
-    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, updateSubscribe?: boolean): 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, channelParam: AliRtcChannelParam): 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 {JoinInfo} userJoinInfo
-     * @param {SubscribeReason} reason
-     * @returns {void}
-     */
-    addUser(userJoinInfo: JoinInfo, 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[]}}
-     */
-    diffUsers(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;
-    setVideoCodec(codec: CodecType): void;
-    setParameter(parameter: string): void;
-    private setupAudioProcessingPluginIfNeeded;
-    getParameter(): string;
-    sendDataChannelMessage(message: AliRtcDataChannelMsg): void;
-}
-
-declare enum AliRtcLiveTranscodingMixMode {
-    /*! 单路模式 */
-    LiveTranscodingSingle = 0,
-    /*! 混流模式 */
-    LiveTranscodingMix = 1
-}
-declare enum AliRtcLiveTranscodingStreamType {
-    /*! 原始流 */
-    LiveTranscodingOrigin = 0,
-    /*! 音频 */
-    LiveTranscodingAudio = 1,
-    /*! 视频 */
-    LiveTranscodingVideo = 2
-}
-declare enum AliRtcLiveTranscodingSourceType {
-    /*! 相机流 */
-    LiveTranscodingCamera = 0,
-    /*! 屏幕流 */
-    LiveTranscodingShareScreen = 1
-}
-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 enum AliRtcLiveTranscodingAudioSampleRate {
-    LiveTranscoding_HZ_48000 = 48000,
-    LiveTranscoding_HZ_44100 = 44100,
-    LiveTranscoding_HZ_32000 = 32000,
-    LiveTranscoding_HZ_16000 = 16000,
-    LiveTranscoding_HZ_8000 = 8000
-}
-declare enum AliRtcLiveTranscodingSegmentType {
-    /*! 无人像分割 */
-    LiveTranscodingNoBody = 0,
-    /*! 人像分割 */
-    LiveTranscodingBody = 1
-}
-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
-}
-declare enum AliRtcLiveTranscodingCropMode {
-    /*! 缩放模式 */
-    LiveTranscodingOrigin = 0,
-    /*! 剪裁 */
-    LiveTranscodingCrop = 1,
-    /*! 填充 */
-    LiveTranscodingFill = 2
-}
-declare enum AliRtcLiveTranscodingMediaProcessMode {
-    /*! 通用模式 */
-    LiveTranscodingNormal = 0,
-    /*! 虚拟背景模式 */
-    LiveTranscodingVirtualBackground = 1
-}
-declare enum AliRtcLiveTranscodingState {
-    LiveTranscodingState_IDLE = 0,
-    LiveTranscodingState_CONNNECT = 1,
-    LiveTranscodingState_RUNNING = 2,
-    LiveTranscodingState_RECOVERING = 3,
-    LiveTranscodingState_FAILURE = 4,
-    LiveTranscodingState_END = 5
-}
-declare enum AliRtcLiveTranscodingErrorCode {
-    LiveTranscodingErrorPublishOk = 0,
-    LiveTranscodingErrorStreamNotFound = 17825793,
-    LiveTranscodingErrorStreamAlreadyExist = 17825794,
-    LiveTranscodingErrorInvalidParam = 17825795,
-    LiveTranscodingErrorInternalError = 17825796,
-    LiveTranscodingErrorRtmpServerError = 17825797,
-    LiveTranscodingErrorRtmpStreamUrlError = 17825798,
-    LiveTranscodingErrorPublishTimeout = 17825799,
-    LiveTranscodingErrorNotAuthorized = 17825800
-}
-declare enum AliRtcTrascodingPublishTaskStatus {
-    /*! 任务开始 */
-    TrascodingPublishTaskStatusStart = 0,
-    /*! 任务更新 */
-    TrascodingPublishTaskStatusUpdate = 1,
-    /*! 任务已停止 */
-    TrascodingPublishTaskStatusStop = 2
-}
-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 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);
-}
-
-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;
-    };
-}
-
-declare enum DisplayType {
-    NOT_DISPLAY = 0,
-    ALWAYS = 1,
-    WHEN_NO_VIDEO = 2
-}
-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;
-    };
-}
-
-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;
-    };
-}
-
-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;
-    };
-}
-
-declare class LiveTranscodingMixParam {
-    taskProfile: AliRtcLiveTranscodingTaskProfile;
-    encodeParam?: AliRtcLiveTranscodingEncodeParam;
-    users: AliRtcTranscodingUser[];
-    backgroundColor: number;
-    backgrounds: AliRtcTranscodingImage[];
-    watermarks: AliRtcTranscodingImage[];
-    clockWidgets: AliRtcTranscodingClockWidget[];
-    cropMode?: AliRtcLiveTranscodingCropMode;
-    mediaProcessMode?: AliRtcLiveTranscodingMediaProcessMode;
-    constructor(taskProfile: AliRtcLiveTranscodingTaskProfile);
-}
-
-declare class AliRtcLiveTranscodingSingleParam {
-    userId?: string;
-    streamType?: AliRtcLiveTranscodingStreamType;
-    sourceType?: AliRtcLiveTranscodingSourceType;
-    constructor(uid?: string, streamType?: AliRtcLiveTranscodingStreamType, sourceType?: AliRtcLiveTranscodingSourceType);
-}
-
-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?: LiveTranscodingMixParam;
-    constructor(mixMode?: AliRtcLiveTranscodingMixMode, singleParam?: AliRtcLiveTranscodingSingleParam, mixParam?: LiveTranscodingMixParam);
-    toJson(type: MPU_TASK_TRANSACTION_TYPE, authInfo: AliRtcAuthInfo, taskId: string, streamUrl?: string, messageId?: string): {
-        [key: string]: any;
-    };
-}
-
-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;
-}
-
-declare enum AliRtcLogLevel {
-    DEBUG = 0,
-    INFO = 1,
-    API = 2,
-    WARNING = 3,
-    ERROR = 4,
-    NONE = 5
-}
-
-interface AliRtcBeautyConfig {
-    whiteningLevel: number;
-    /*! 磨皮等级[0-1.0] */
-    smoothnessLevel: number;
-}
-
-interface AliRtcWebTrackConfig {
-    biz: string;
-    extraParams?: string;
-}
-interface AliRtcEngineConfig {
-    env?: AliRtcEnv;
-    webTrack?: boolean | AliRtcWebTrackConfig;
-    maxSignalingConnectRetryCount?: number;
-}
-interface AliRtcAIAgentParam {
-    /*! ai agent id */
-    agentId: string;
-    /*! ai agent的区域 */
-    region: string;
-    /*! 启动ai agent的action */
-    action: string;
-    /*! 启动ai agent的参数 */
-    params: string;
-}
-
-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;
-    /**
-     * 远程用户订阅数据通道
-     * @param uid 用户ID
-     */
-    remoteUserSubscribedDataChannel: (uid: string) => 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;
-    /**
-     * @brief AI agent处理结果
-     * @param code 返回code码
-     * @param action 结果对应的action
-     * @param result 结果
-     */
-    AIAgentResult: (code: number, action: string, result: string) => void;
-}
-
-declare enum AliRtcRawDataStreamType {
-    /** 相机流 */
-    AliRtcSdkStreamTypeCapture = 0,
-    /** 屏幕共享流 */
-    AliRtcSdkStreamTypeScreen = 1
-}
-
-interface AliRtcScreenShareConfiguration {
-    width?: number;
-    height?: number;
-    frameRate?: number;
-    bitrate?: number;
-}
-
-interface AliRtcScreenShareStartConfig {
-    audio?: boolean;
-    videoTrack?: MediaStreamVideoTrack;
-    audioTrack?: MediaStreamAudioTrack;
-}
-
-interface AliRtcVideoTrackConfig {
-    userId?: string;
-    streamType: AliRtcRawDataStreamType;
-}
-
-interface ImageHolderPluginOptions {
-    imageUrl: string;
-}
-declare class AliRtcImageHolderPlugin extends AliRtcPlugin {
-    private url;
-    private image?;
-    private canvas;
-    private mediaStream?;
-    private updateableManager;
-    private updateable;
-    private hasVideoTrack;
-    constructor(url: string, fps?: number);
-    get initOptions(): {
-        imageUrl: string;
-    };
-    setOptions(options: ImageHolderPluginOptions): void;
-    private drawImageToCanvas;
-    start(): Promise<MediaStreamTrack>;
-    stop(): void;
-    isSupported(): boolean;
-    shouldUpdate(_streamInfo: AliRtcLocalStreamInfo): boolean;
-    process(streamInfo: AliRtcLocalStreamInfo): Promise<void>;
-}
-
-declare class AliRtcEngine extends EventEmitter$1<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;
-    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 AliRtcImageHolderPlugin: typeof AliRtcImageHolderPlugin;
-    static AliRtcLiveTranscodingParam: typeof AliRtcLiveTranscodingParam;
-    static AliRtcLiveTranscodingEncodeParam: typeof AliRtcLiveTranscodingEncodeParam;
-    static AliRtcLiveTranscodingMixParam: typeof LiveTranscodingMixParam;
-    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;
-    private _proxyAiTid;
-    private _agentParam;
-    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[]} userJoinInfo
-     * @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 startDefaultDevices;
-    private startPublish;
-    /**
-     * 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;
-    setVideoCodec(videoCodec: CodecType): 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
-     * @param {any} extraData      入会自定义数据
-     * @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, userNameOrParam?: string | AliRtcChannelParam, extraData?: any): Promise<void>;
-    /**
-     * @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 token          认证信息，从App Server获取。
-     * @param channelParam   入会参数，详见{@link AliRtcChannelParam}
-     * @param agentParam     入会参数，详见{@link AliRtcAIAgentParam}
-     * @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的应用可进行互相通话,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}
-     */
-    joinAIChannel(token: string, channelParam: AliRtcChannelParam, agentParam: AliRtcAIAgentParam): 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?: MediaTrackConstraints): Promise<void>;
-    /**
-     * @brief 设置音频采集参数
-     * @param constraints 音频采集参数
-     */
-    setAudioConstraints(constraints: MediaTrackConstraints): 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;
-    getPluginByName(name: string): any;
-    /**
-     * @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 const WrappedAliRtcEngine: typeof AliRtcEngine;
-type WrappedAliRtcEngine = InstanceType<typeof AliRtcEngine>;
-
-interface AliRtcCheckResult {
-    support: boolean;
-    detail: {
-        isWebRTCSupported: boolean;
-        isH264EncodeSupported: boolean;
-        isH264DecodeSupported: boolean;
-        isBrowserSupported: boolean;
-        isScreenShareSupported: boolean;
-        isSendMediaExtensionMsgSupported: boolean;
-    };
-}
-
-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"
-}
-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;
-};
-
-/**
- * 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;
-    reportPublishAudio3A(callId: string, msid: string, traceId: string, apmJson: string): 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;
-    reportLoopAudioDelay(ssrc: string, result: 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;
-}
-
-/**
- * RoomServerMessageCenter支持的事件
- */
-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;
-    onUnknownMessage: (response: any) => 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 EventEmitter$1<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;
-    private joinChannelExtraData?;
-    constructor(authInfo: AliRtcAuthInfo, userName: string, channelProfile: AliRtcSdkChannelProfile, clientRole: AliRtcSdkClientRole, slsReporter: SLSReporter, maxConnectRetryCount?: number);
-    get retryCount(): number;
-    /**
-     * 开始连接信令
-     * @param roomserverurl
-     * @param joinChannelExtraData 发送 joinChannel 信令时额外携带的数据
-     */
-    start(roomserverurl: string, joinChannelExtraData?: any): 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;
-    protected onUnknownMessage(response: any): void;
-}
-
-declare class RoomServerSignaling extends EventEmitter$1<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, joinChannelExtraData?: any): 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;
-}
-
-interface StsManagerListener {
-    onTokenUpdate: (ossToken: OSSToken, slsToken: SLSToken) => void;
-}
-interface OSSToken {
-    access_key_id: string;
-    access_key_secret: string;
-    security_token: string;
-    region_endpoint: string;
-    bucket_name: string;
-    expiration: number;
-}
-interface SLSToken {
-    access_key_id: string;
-    access_key_secret: string;
-    security_token: string;
-    region_endpoint: string;
-    project: string;
-    log_store_stats: string;
-    log_store_debug: string;
-    expiration: number;
-}
-declare class StsManager extends EventEmitter$1<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 class SignalingManager extends EventEmitter$1<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, joinChannelExtraData?: any): 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;
-    protected onUnknownMessage(message: any): void;
-}
-
-type Parameter = {
-    [key: string]: any;
-};
-interface BizControlConfig {
-    slsReporter: SLSReporter;
-    signalingManager: SignalingManager;
-}
-interface LocalUserConfig {
-    rtsManager: RtsManager;
-    localStreamManager: LocalStreamManager;
-    slsReporter: SLSReporter;
-    signalingManager: SignalingManager;
-    pluginManager: PluginManager;
-    audioVolumeIndicationInterval: number;
-    parameter: {
-        [key: string]: any;
-    };
-}
-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 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 DeviceStatusChange {
-    Local = "local-device-status-change",
-    Remote = "remote-device-status-change"
-}
-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.VideoTrackEnded]: () => void;
-    [TrackEvent.AudioTrackEnded]: () => void;
-    [TrackEvent.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;
-}
-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;
-    remoteUserJoin: (uid: string, elapseSinceLastJoin: 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;
-    remoteUserSubscribedDataChannel: (uid: string) => void;
-}
-/**
- * 服务端通知的远端用户信息
- */
-interface JoinInfo {
-    displayname: string;
-    session: string;
-    userid: string;
-    callid: string;
-    clientrole?: string;
-    subscribe?: {
-        pullstreamurl: string;
-    };
-    tracks?: TrackInfo[];
-}
-interface LeaveInfo {
-    userid: string;
-    leavecode: AliRtcUserOfflineReason;
-    reason: string;
-}
-interface PublishUser {
-    callid: string;
-    clientrole?: string;
-    displayname: string;
-    subscribe?: {
-        pullstreamurl: string;
-    };
-    pubid?: string;
-    tracks?: TrackInfo[];
-    userid: string;
-    sessionid?: string;
-}
-interface RemoteUserStatus {
-    userid: string;
-    status: string;
-}
-declare enum MsidType {
-    Audio = "sophon_audio",
-    VideoLarge = "sophon_video_camera_large",
-    VideoSmall = "sophon_video_camera_small",
-    Screen = "sophon_video_screen_share",
-    Data = "sophon_data"
-}
-interface TrackInfo {
-    ssrc: string;
-    codec: string;
-    msid?: MsidType;
-    type: string;
-    from?: string | undefined;
-    userdata?: string;
-    trackId?: string;
-    attr?: Record<string, any>;
-    pt?: number;
-}
-declare enum PublishType {
-    Add = "add",
-    Delete = "delete",
-    Replace = "replace",
-    Stop = "stop",
-    UnPublish = "unpublish"
-}
-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;
-}
-interface RefreshUser {
-    userid: string;
-    subscribe: {
-        pullstreamurl: string;
-    };
-}
-interface RefreshInfo {
-    pushstreamurl?: string;
-    users?: RefreshUser[];
-}
-declare enum VideoStreamSource {
-    Camera = 0,
-    Screen = 1,
-    Image = 2
-}
-declare enum AudioStreamSource {
-    Microphone = 0,
-    Screen = 1,
-    Mixed = 2
-}
-interface PublishExtraParams {
-    vMsid?: string;
-    aMsid?: string;
-    isVideoSmall?: boolean;
-    isScreen?: boolean;
-    isAudio?: boolean;
-}
-interface _LocalStreamConfig extends IStreamConfig {
-    videoProfile?: string;
-    screenProfile?: string;
-}
-type LocalStreamConfig = Omit<_LocalStreamConfig, 'custom'>;
-interface PreviewConfig {
-    videoElement: HTMLVideoElement;
-    screen?: boolean;
-}
-interface PreviewAllConfig {
-    primary?: boolean;
-    screen?: boolean;
-}
-interface StopPreviewConfig {
-    videoElement?: HTMLVideoElement;
-    primary?: boolean;
-    screen?: boolean;
-}
-interface SubConfig {
-    isAudioSubscribing: boolean;
-    isVideoSubscribing: boolean;
-    isScreenSubscribing: boolean;
-    audioEnabled: boolean;
-    videoEnabled: boolean;
-    screenEnabled: boolean;
-}
-interface StreamOptions {
-    audio?: boolean;
-    video?: boolean;
-    screen?: boolean;
-    data?: boolean;
-}
-type PublishOptions = StreamOptions & {
-    vMsid?: string;
-};
-type StopPublishOptions = StreamOptions;
-type SubscribeOptions = StreamOptions & {
-    aMsid?: string;
-    vMsid?: string;
-};
-type UnSubscribeOptions = SubscribeOptions;
-/**
- * ------------------    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
+ * ------------------    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
  */
 type AudioProfileKey = 'standard' | 'high' | 'standard-stereo' | 'high-stereo';
-declare enum SubscribeReason {
-    InRoom = "inRoom",
-    Join = "join",
-    Publish = "publish",
-    User = "user",
-    Resume = "resume",
-    Retry = "retry",
-    Reconnect = "reconnect"
-}
-declare enum PublishReason {
-    MANUAL = 0,
-    JOIN_CHANNEL = 1,
-    ROLE_CHANGE = 2,
-    RECONNECT = 3
-}
 
 interface AliRtcLocalStreamListener {
     videotrackended: () => void;
@@ -5514,7 +1084,7 @@ type AudioPluginOptions = {
     dumpAudioData?: boolean;
     aiaecOn?: boolean;
     farMediaStream?: MediaStream;
-    slsReporter?: SLSReporter;
+    slsReporter?: any;
     enableSoft3ACheck: boolean;
     useHardOnly: boolean;
 };