aliyun-rtc-sdk 6.12.1-beta.0 → 6.12.1-beta.1

package/dist/types/index.d.ts

@@ -227,8 +227,6 @@ export declare class AliRtcEngine extends default_2<AliRtcEngineEventListener> {
     protected slsReporter: SLSReporter;
     protected connectState: AliRtcConnectionStatus;
     protected networkError: boolean;
-    private audioMixerPlugin;
-    private dualVideoPlugin;
     private _audioOnlyMode;
     private _authInfoWillExpireTimer;
     private _authInfoExpiredTimer;
@@ -295,6 +293,12 @@ export declare class AliRtcEngine extends default_2<AliRtcEngineEventListener> {
      */
     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
@@ -328,6 +332,7 @@ export declare class AliRtcEngine extends default_2<AliRtcEngineEventListener> {
      * @note 该接口只可以在加入频道之前调用，会议中不可以重新设置，离开频道后可以重新设置
      */
     setChannelProfile(channelProfile: AliRtcSdkChannelProfile): void;
+    private startAndPublishDefaultDevices;
     /**
      * 设置用户角色
      * @param {AliRtcSdkClientRole} clientRole 用户角色类型，默认值为 {@link AliRtcSdkClientRole.AliRtcSdkLive}（观众角色）
@@ -2129,6 +2134,11 @@ declare interface AudioLevelMonitorListener {
     audioLevel: (level: number) => void;
 }
 
+declare interface AudioMixerPluginOptions {
+    audio?: number;
+    screen?: number;
+}
+
 /**
  * ------------------    模式    ----- 采样率  -----  声道  -----  码率（kbps）  -----
  *
@@ -2161,6 +2171,8 @@ declare class BizControl extends default_2<BizControlListener> {
     protected authInfo?: AliRtcAuthInfo;
     protected signalingManager: SignalingManager;
     pluginManager: PluginManager;
+    private audioMixerPlugin?;
+    private dualVideoPlugin?;
     protected slsReporter: SLSReporter;
     private statsMonitorId?;
     private autoSubAudio;
@@ -2288,6 +2300,8 @@ declare class BizControl extends default_2<BizControlListener> {
     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;
@@ -2630,7 +2644,8 @@ declare class LocalUser extends User {
     clear(): Promise<void>;
     resetElement(element: HTMLVideoElement): void;
     setPreviewElement(previewParams: PreviewConfig): void;
-    startDefaultDevicesCapture(audioOnly: boolean): Promise<void>;
+    startDefaultCamera(audioOnly: boolean): Promise<void>;
+    startDefaultAudio(): Promise<void>;
     /**
      * 为本地预览设置渲染元素和视频流
      * @param {AliRtcLocalView} view 渲染元素，null 则为停止显示
@@ -2865,8 +2880,9 @@ declare interface OSSToken {
 
 declare class PluginManager extends default_2<PluginManagerListener> {
     private plugins;
-    add(plugin: AliRtcPlugin, options: any): void;
+    add(plugin: AliRtcPlugin, options?: any): void;
     remove(name: string): void;
+    removeAll(): void;
     get(name: string): AliRtcPlugin | undefined;
     has(name: string): boolean;
     getAll(): AliRtcPlugin[];
@@ -2919,7 +2935,8 @@ declare type PublishOptions = StreamOptions & {
 
 declare enum PublishReason {
     MANUAL = 0,
-    JOINCHANNEL = 1
+    JOIN_CHANNEL = 1,
+    ROLE_CHANGE = 2
 }
 
 declare interface PublishStreamInfo {
@@ -3788,7 +3805,8 @@ declare enum SubscribeReason {
     Join = "join",
     Publish = "publish",
     User = "user",
-    Resume = "resume"
+    Resume = "resume",
+    Retry = "retry"
 }
 
 declare interface TimeRecorder {

package/package.json

@@ -1,6 +1,6 @@
 {
   "name": "aliyun-rtc-sdk",
-  "version": "6.12.1-beta.0",
+  "version": "6.12.1-beta.1",
   "type": "module",
   "description": "rtc web sdk of aliyun",
   "main": "dist/aliyun-rtc-sdk.js",