aliyun-rtc-sdk 6.12.1-beta.1 → 6.12.1-beta.3

package/dist/types/index.d.ts

@@ -135,7 +135,7 @@ export declare enum AliRtcDataMsgType {
 
 declare type AliRtcDualVideoPluginOptions = VideoScalerOptions;
 
-export declare class AliRtcEngine extends default_2<AliRtcEngineEventListener> {
+declare class AliRtcEngine_2 extends default_2<AliRtcEngineEventListener> {
     /**
      * @ignore
      */
@@ -174,7 +174,7 @@ export declare class AliRtcEngine extends default_2<AliRtcEngineEventListener> {
      * @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 getInstance(config?: AliRtcEngineConfig): AliRtcEngine_2;
     static AliRtcSdkChannelProfile: typeof AliRtcSdkChannelProfile;
     static AliRtcSdkClientRole: typeof AliRtcSdkClientRole;
     static AliRtcVideoStreamType: typeof AliRtcVideoStreamType;
@@ -261,6 +261,7 @@ export declare class AliRtcEngine extends default_2<AliRtcEngineEventListener> {
      * 监听 Publisher 事件
      */
     private addPublisherListeners;
+    private occurError;
     /**
      * 监听 UM 事件
      */
@@ -1229,11 +1230,19 @@ declare interface AliRtcEngineEventListener {
     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 org::webrtc::alirtcInterface::ErrorCodeEnum}
+     * @param error  错误类型，参考 {@link AliRtcError}
      * @param message 错误消息
      */
     onOccurError: (error: AliRtcError) => 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 message 错误消息
+     */
+    occurError: (error: AliRtcError) => void;
     /**
      * @brief 收到数据通道消息
      * @param uid 用户ID
@@ -1365,14 +1374,18 @@ export declare enum AliRtcErrorCode {
     /** 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_NO_AVAILABLE = 17040400,
+    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 */
@@ -2374,9 +2387,6 @@ declare enum CodecType {
     UNKNOWN = ""
 }
 
-declare const _default: typeof AliRtcEngine;
-export default _default;
-
 declare interface DeviceStatus {
     micOpen: boolean;
     micInterrupted: boolean;
@@ -3383,6 +3393,7 @@ declare class RtsManager extends default_2<RtsManagerEventListener> {
     dcConnectingPromise?: Promise<unknown>;
     private _publishingTracks;
     constructor(localStreamManager: LocalStreamManager, slsReporter: SLSReporter);
+    private addRTSListener;
     get pcTraceId(): string;
     set localJoinTime(time: number);
     setEncodedInsertableStreams(enable: boolean): void;
@@ -3401,7 +3412,7 @@ declare class RtsManager extends default_2<RtsManagerEventListener> {
     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<any>;
+    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>;
@@ -3579,7 +3590,7 @@ declare class SLSReporter {
     private static staticClient;
     private static getLogClient;
     static reportOSSUpload(sessionId: string, date: string, responseCode?: number): void;
-    constructor(engine: AliRtcEngine);
+    constructor(engine: WrappedAliRtcEngine);
     /**
      * 开始建立连接
      * @param authInfo
@@ -3797,6 +3808,7 @@ declare interface SubConfig {
 }
 
 declare type SubscribeOptions = StreamOptions & {
+    aMsid?: string;
     vMsid?: string;
 };
 
@@ -3919,4 +3931,10 @@ declare enum VideoStreamSource {
     Image = 2
 }
 
+declare const WrappedAliRtcEngine: typeof AliRtcEngine_2;
+
+declare type WrappedAliRtcEngine = InstanceType<typeof AliRtcEngine_2>;
+export { WrappedAliRtcEngine as AliRtcEngine }
+export default WrappedAliRtcEngine;
+
 export { }

package/package.json

@@ -1,6 +1,6 @@
 {
   "name": "aliyun-rtc-sdk",
-  "version": "6.12.1-beta.1",
+  "version": "6.12.1-beta.3",
   "type": "module",
   "description": "rtc web sdk of aliyun",
   "main": "dist/aliyun-rtc-sdk.js",
@@ -9,7 +9,7 @@
     "@aliyun-sls/web-sts-plugin": "^0.3.5",
     "@aliyun-sls/web-track-browser": "^0.3.5",
     "aliyun-queen-engine": "^6.3.3",
-    "aliyun-rts-sdk": "2.9.1-beta.2",
+    "aliyun-rts-sdk": "2.9.1",
     "crypto-js": "^4.1.1",
     "dateformat": "^5.0.3",
     "eventemitter3": "^5.0.1",