aliyun-rtc-sdk 7.2.1 → 7.2.3

package/dist/plugins/audioProcessPlugin.d.ts

@@ -1027,6 +1027,11 @@ declare class AliRtcLocalStreamInfo extends EventEmitter$1<AliRtcLocalStreamList
      * @param publishStream
      */
     cloneVideoProfile(publishStream: LocalStream): Promise<void>;
+    /**
+     * 复制音频 profile 到指定 LocalStream
+     * @param publishStream
+     */
+    cloneAudioProfile(publishStream: LocalStream): Promise<void>;
     addPlugin(plugin: AliRtcPlugin): void;
     removePlugin(plugin: AliRtcPlugin): boolean;
 }

package/dist/plugins/beautyPlugin.d.ts

@@ -1028,6 +1028,11 @@ declare class AliRtcLocalStreamInfo extends EventEmitter$1<AliRtcLocalStreamList
      * @param publishStream
      */
     cloneVideoProfile(publishStream: LocalStream): Promise<void>;
+    /**
+     * 复制音频 profile 到指定 LocalStream
+     * @param publishStream
+     */
+    cloneAudioProfile(publishStream: LocalStream): Promise<void>;
     addPlugin(plugin: AliRtcPlugin): void;
     removePlugin(plugin: AliRtcPlugin): boolean;
 }

package/dist/types/index.d.ts

@@ -267,6 +267,7 @@ declare class AliRtcEngine_2 extends default_2<AliRtcEngineEventListener> {
     private _localCheckAuthInfoTimestamp;
     private _proxyAiTid;
     private _agentParam;
+    sessionId?: string;
     constructor(config?: AliRtcEngineConfig | undefined);
     /**
      * 获取当前频道号，已经入会成功返回频道号，否则返回undefined
@@ -1021,10 +1022,17 @@ declare class AliRtcEngine_2 extends default_2<AliRtcEngineEventListener> {
      * @param dataChannelMsg 伴奏控制消息
      */
     sendDataChannelMessage(dataChannelMsg: AliRtcDataChannelMsg): void;
+    /**
+     * @brief 不依赖入会，直接订阅远程流
+     * @param config 订阅配置对象
+     * @returns RemoteStreamHandler 远端流处理器实例
+     */
+    subscribeRemoteStream(config: SubscribeRemoteStreamConfig): RemoteStreamHandler;
 }
 
 declare interface AliRtcEngineConfig {
     env?: AliRtcEnv;
+    globalEnv?: 'DEFAULT' | 'SEA';
     webTrack?: boolean | AliRtcWebTrackConfig;
     maxSignalingReconnectDuration?: number;
     parameter?: Parameter;
@@ -1324,6 +1332,21 @@ declare interface AliRtcEngineEventListener {
      * @param result 结果
      */
     AIAgentResult: (code: number, action: string, result: string) => void;
+    /**
+     * @brief 网络质量回调
+     * @param event 网络质量信息
+     * @param event.subscribe 下行网络质量
+     * @param event.subscribe.rtt 下行网络往返延时（ms）
+     * @param event.subscribe.loss 下行网络丢包率
+     * @param event.subscribe.quality 下行网络质量类型
+     */
+    networkQuality: (event: {
+        subscribe: {
+            rtt: number;
+            loss: number;
+            quality: AliRtcNetworkQuality;
+        };
+    }) => void;
 }
 
 export declare enum AliRtcEngineLocalDeviceExceptionType {
@@ -1902,6 +1925,11 @@ export declare class AliRtcLocalStreamInfo extends default_2<AliRtcLocalStreamLi
      * @param publishStream
      */
     cloneVideoProfile(publishStream: LocalStream): Promise<void>;
+    /**
+     * 复制音频 profile 到指定 LocalStream
+     * @param publishStream
+     */
+    cloneAudioProfile(publishStream: LocalStream): Promise<void>;
     addPlugin(plugin: AliRtcPlugin): void;
     removePlugin(plugin: AliRtcPlugin): boolean;
 }
@@ -1928,6 +1956,26 @@ export declare enum AliRtcLogLevel {
     NONE = 5
 }
 
+/**
+ * @brief 网络质量类型
+ */
+export declare enum AliRtcNetworkQuality {
+    /** 网络极好 */
+    AliRtcNetworkExcellent = 0,
+    /** 网络好 */
+    AliRtcNetworkGood = 1,
+    /** 网络不好 */
+    AliRtcNetworkPoor = 2,
+    /** 网络差 */
+    AliRtcNetworkBad = 3,
+    /** 网络极差 */
+    AliRtcNetworkVeryBad = 4,
+    /** 网络断开 */
+    AliRtcNetworkDisconnected = 5,
+    /** 网络未知 */
+    AliRtcNetworkUnknown = 6
+}
+
 /**
  * @brief OnBye类型枚举
  */
@@ -2310,6 +2358,7 @@ declare class BizControl extends default_2<BizControlListener> {
     private parameter;
     private audio3AConfig;
     private dcReadyHelper?;
+    private networkQualityTimer?;
     constructor(config: BizControlConfig);
     private addSignalingManagerListener;
     private addRTSListener;
@@ -2480,6 +2529,13 @@ declare interface BizControlListener {
     occurError: (error: AliRtcError, uid?: string) => void;
     remoteDataChannelMessage: (uid: string, message: AliRtcDataChannelMsg) => void;
     remoteUserSubscribedDataChannel: (uid: string) => void;
+    networkQuality: (event: {
+        subscribe: {
+            rtt: number;
+            loss: number;
+            quality: AliRtcNetworkQuality;
+        };
+    }) => void;
 }
 
 declare enum ClientEventType {
@@ -3005,9 +3061,7 @@ declare interface LocalUserConfig {
     signalingManager: SignalingManager;
     pluginManager: PluginManager;
     audioVolumeIndicationInterval: number;
-    parameter: {
-        [key: string]: any;
-    };
+    parameter: Parameter;
 }
 
 /**
@@ -3019,14 +3073,18 @@ declare class LogClient {
     protected param?: any;
     protected msgCacheArr: any[];
     protected index: number;
-    protected stsOpt?: any;
-    protected slsToken?: SLSSTSToken;
-    protected ossToken?: OSSSTSToken;
     protected ntpClock: NTPClient;
-    constructor();
+    private clientId;
+    private stsData?;
+    private stsExpiration;
+    private updateTaskTimer?;
+    private globalEnv;
+    constructor(globalEnv?: 'DEFAULT' | 'SEA');
     private createTracker;
+    private startSTSUpdateTask;
+    private refreshSTSToken;
+    private requestSTSToken;
     start(info: AliRtcAuthInfo): void;
-    updateToken(slsToken: SLSSTSToken, ossToken?: OSSSTSToken): void;
     /**
      * 断开连接
      */
@@ -3099,24 +3157,6 @@ declare class NTPClient {
     now(): number;
 }
 
-declare interface OSSSTSToken {
-    access_key_id: string;
-    access_key_secret: string;
-    security_token: string;
-    region_endpoint: string;
-    bucket_name: string;
-    expiration: number;
-}
-
-declare interface OSSToken {
-    access_key_id: string;
-    access_key_secret: string;
-    security_token: string;
-    region_endpoint: string;
-    bucket_name: string;
-    expiration: number;
-}
-
 declare interface PackageAdaptInfo {
     os_name: string;
     device_name: string;
@@ -3175,9 +3215,86 @@ declare class PackageCreator {
     protected createBaseSendPackage(messageType: MessageType): ISendPackage;
 }
 
-declare type Parameter = {
+/**
+ * SDK 参数配置
+ */
+declare interface Parameter {
+    /** 是否在推流信息中报告 L1 IP */
+    reportL1IP?: boolean;
+    /** 数据通道配置 */
+    data?: ParameterData;
+    /** 是否使用 AudioContext 进行音量控制 */
+    enableAudioContextPlayoutVolume?: boolean;
+    /** 网络相关配置 */
+    net?: ParameterNet;
+    /** 加入房间模式，如 'Conference'、'AIUser' 等 */
+    joinMode?: string;
+    /** 是否启用空白音频（用于保持音频轨道活跃） */
+    enableBlankAudio?: boolean;
+    /** 音频 3A 处理配置 */
+    audio?: ParameterAudio;
+    /** 支持其他自定义参数 */
     [key: string]: any;
-};
+}
+
+/**
+ * 音频 3A 处理配置参数
+ */
+declare interface ParameterAudio {
+    /** 是否使用服务端配置的 3A 设置 */
+    useAudio3AConfigFromServer?: boolean;
+    /** 是否启用系统回声消除 */
+    useSysEchoCancellation?: boolean;
+    /** 是否启用系统噪声抑制 */
+    useSysNoiseSuppression?: boolean;
+    /** 是否启用系统自动增益控制 */
+    useSysAutoGainControl?: boolean;
+    /** 是否启用智能 AEC */
+    useAIAEC?: boolean;
+    /** 是否启用软件 3A 处理 */
+    enableSoft3A?: boolean;
+    /** 是否启用旁路软 3A 检查 */
+    enableSoft3ACheck?: boolean;
+    /** 是否仅使用硬件 3A */
+    useHardOnly?: boolean;
+    /** 是否导出音频数据 */
+    dumpAudioData?: boolean;
+}
+
+/**
+ * 数据通道配置参数
+ */
+declare interface ParameterData {
+    /** 是否启用发布端数据通道 */
+    enablePubDataChannel?: boolean;
+    /** 是否启用订阅端数据通道 */
+    enableSubDataChannel?: boolean;
+}
+
+/**
+ * 网络相关配置参数
+ */
+declare interface ParameterNet {
+    /** 是否使用 TCP 传输 */
+    useTCP?: boolean;
+    /** 是否启用累积延迟 */
+    cumuDelay?: boolean;
+    /** 是否禁用心跳检测 */
+    disableHeartbeat?: boolean;
+    /** 心跳探活参数 */
+    heartbeat?: {
+        /** 心跳间隔，毫秒数，最小 200，默认 2500 */
+        interval?: number;
+        /** 超时时间，毫秒数 */
+        timeout?: number;
+        /** 是否启用心跳日志（监听 hebtSent/hebtReceived 事件） */
+        enableLog?: boolean;
+    };
+    /** 数据通道配置 */
+    datachannelInitOptions?: RTCDataChannelInit;
+    /** 媒体超时时间(毫秒)，有效范围 1000-100000 */
+    mediaTimeout?: number;
+}
 
 declare class PluginManager extends default_2<PluginManagerListener> {
     private plugins;
@@ -3301,6 +3418,46 @@ declare interface RemoteMediaTrackInfo extends MediaTrackInfo {
     subscribeState?: RemoteTrackSubscribeState;
 }
 
+/**
+ * 远端流订阅失败原因
+ */
+declare enum RemoteStreamErrorReason {
+    /** 无法获取播放地址 */
+    FETCH_STREAM_URL_FAILED = "FETCH_STREAM_URL_FAILED",
+    /** 目标用户没有可用的流 */
+    NO_STREAM_AVAILABLE = "NO_STREAM_AVAILABLE",
+    /** 无效的播放元素 */
+    INVALID_VIEW_ELEMENT = "INVALID_VIEW_ELEMENT",
+    /** 订阅失败 */
+    SUBSCRIBE_FAILED = "SUBSCRIBE_FAILED"
+}
+
+declare class RemoteStreamHandler extends default_2<RemoteStreamHandlerEvents> {
+    private config;
+    private static logName;
+    private readonly roomServerUrl;
+    private readonly label;
+    private rtsInstance?;
+    private currentStatus;
+    private latestStreamInfo?;
+    constructor(config: SubscribeRemoteStreamConfig);
+    /**
+     * 更新连接状态，只有当状态发生变化时才触发事件
+     */
+    private updateConnectionStatus;
+    /**
+     * 处理错误，使实例进入不可恢复状态，业务侧应销毁实例
+     */
+    private handleError;
+    private fetchStreamUrlAndPlay;
+    dispose(): void;
+}
+
+declare interface RemoteStreamHandlerEvents {
+    connectionStatusChange: (status: AliRtcConnectionStatus) => void;
+    error: (reason: RemoteStreamErrorReason, detail: string) => void;
+}
+
 declare class RemoteStreamInfo extends default_2<StreamListener> {
     /**
      * @ignore
@@ -3428,6 +3585,7 @@ declare class RemoteUser extends User {
      * @ignore
      */
     static logName: string;
+    get logName(): string;
     remoteCallId: string;
     remoteUserInfo: AliRtcRemoteUserInfo;
     protected localUser?: LocalUser;
@@ -3480,7 +3638,9 @@ declare class RemoteUser extends User {
     setPlayoutVolume(value: number): void;
     getAudioMuted(): boolean;
     get hasAudioTrack(): boolean;
+    get audioSSRC(): string;
     get hasVideoTrack(): boolean;
+    get videoSSRC(): string;
     get hasVideoLargeTrack(): boolean;
     get hasVideoSmallTrack(): boolean;
     get hasScreenTrack(): boolean;
@@ -3639,9 +3799,7 @@ declare interface RemoteUserConfig {
     localUser?: LocalUser;
     audioVolumeIndicationInterval: number;
     playoutVolume: number;
-    parameter: {
-        [key: string]: any;
-    };
+    parameter: Parameter;
 }
 
 declare interface RemoteUserMap {
@@ -3958,12 +4116,16 @@ declare class RtsManager extends default_2<RtsManagerEventListener> {
     private dcResolve?;
     private dcReject?;
     private dcConnectingPromise?;
+    private _prevDownlinkStat?;
     private _publishingTracks;
     private parameter;
     preferredPubL1Ip?: string;
     preferredSubL1Ip?: string;
+    private httpPublishUrlMap;
+    private httpSubscribeUrlMap;
     private traceIdMap;
     constructor(localStreamManager: LocalStreamManager, slsReporter: SLSReporter, parameter: Parameter);
+    private createRtsInstance;
     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
@@ -3983,6 +4145,8 @@ declare class RtsManager extends default_2<RtsManagerEventListener> {
     private setConnected;
     get isConnecting(): boolean;
     get isConnected(): boolean;
+    private suffixHttpUrl;
+    private getSuffixedUrl;
     get publishingTracks(): TrackInfo[];
     private getPubMsid;
     private updatePublishingTracks;
@@ -3997,6 +4161,11 @@ declare class RtsManager extends default_2<RtsManagerEventListener> {
     publishAddDataChannel(streamUrl: string): Promise<any>;
     publishStopDataChannel(streamUrl: string, datachannel: any): Promise<void>;
     unpublish(): Promise<void>;
+    getSubscribeNetworkQuality(): Promise<{
+        rtt: number;
+        loss: number;
+        quality: AliRtcNetworkQuality;
+    }>;
     /**
      * 获取 sub/subAdd config
      * @param {ISubscribeConfig | ISubConfigItem} options
@@ -4005,6 +4174,16 @@ declare class RtsManager extends default_2<RtsManagerEventListener> {
     private getSubConfig;
     private handleHTTPConfig;
     private httpSubscribe;
+    /**
+     * 订阅时支持 preferredSubL1Ip 超时降级
+     * 如果存在 preferredSubL1Ip，则设置 3 秒超时，超时后清除 preferredSubL1Ip，降级到 DNS 解析的 L1 节点
+     */
+    private subscribeWithPreferredL1Fallback;
+    /**
+     * 推流时支持 preferredPubL1Ip 超时降级
+     * 如果存在 preferredPubL1Ip，则设置 3 秒超时，超时后清除 preferredPubL1Ip，降级到 DNS 解析的 L1 节点
+     */
+    private publishWithPreferredL1Fallback;
     private subscribeAdd;
     subscribeDelete(subscribeOptions: RemoteSubscribeOptions): Promise<ISubDeleteResult | undefined>;
     subscribeAddDataChannel(streamUrl: string): Promise<any>;
@@ -4023,6 +4202,10 @@ declare class RtsManager extends default_2<RtsManagerEventListener> {
         msid?: string;
     }): any;
     getPCStats(): Promise<any>;
+    /**
+     * 获取当前使用的 candidate 协议类型（udp/tcp）
+     */
+    private logCandidateProtocol;
     sendSEI(streamUrl: string, data: ArrayBuffer, repeatCount: number, payloadType: number): Promise<void>;
 }
 
@@ -4063,7 +4246,6 @@ declare class SignalingManager extends default_2<RoomServerListener> {
     protected clientRole: AliRtcSdkClientRole;
     protected env: AliRtcEnv;
     protected maxSignalingReconnectDuration?: number;
-    stsManager: StsManager;
     private slsReporter;
     constructor(channelProfile: AliRtcSdkChannelProfile, clientRole: AliRtcSdkClientRole, slsReporter: SLSReporter, env?: AliRtcEnv);
     reset(): void;
@@ -4190,10 +4372,7 @@ declare class SLSReporter {
     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);
+    constructor(engine: WrappedAliRtcEngine, globalEnv?: 'DEFAULT' | 'SEA');
     private customFields;
     /**
      * 设置通用字段
@@ -4213,7 +4392,6 @@ declare class SLSReporter {
      * @param authInfo
      */
     start(info: AliRtcAuthInfo): void;
-    updateToken(token: SLSSTSToken, ossToken?: OSSSTSToken): void;
     /**
      * 断开连接
      */
@@ -4223,6 +4401,8 @@ declare class SLSReporter {
     reportSubscribeMonitor(callId: string, remoteId: string, traceId: string, msid: string, stats: any[]): void;
     reportNetworkMonitor(candidates: any[]): void;
     reportLoopAudioDelay(ssrc: string, result: any): void;
+    reportCumuAudioDelay(audioSSRC: string, result: any): void;
+    reportCumuVideoDelay(videoSSRC: string, result: any): void;
     /**
      * 加入房间成功埋点
      * @param {number} joinTime
@@ -4365,28 +4545,6 @@ declare class SLSReporter {
     protected data2String(data: any): string;
 }
 
-declare interface SLSSTSToken {
-    access_key_id: string;
-    access_key_secret: string;
-    expiration: number;
-    log_store_debug: string;
-    log_store_stats: string;
-    project: string;
-    region_endpoint: string;
-    security_token: string;
-}
-
-declare interface SLSToken {
-    access_key_id: string;
-    access_key_secret: string;
-    security_token: string;
-    region_endpoint: string;
-    project: string;
-    log_store_stats: string;
-    log_store_debug: string;
-    expiration: number;
-}
-
 /**
  * Websocket类
  */
@@ -4495,25 +4653,6 @@ declare interface StreamOptions {
     data?: boolean;
 }
 
-declare class StsManager extends default_2<StsManagerListener> {
-    static logName: string;
-    private env;
-    private authInfo?;
-    private timer?;
-    private slsReporter;
-    private expiration;
-    private skipTime;
-    constructor(slsReporter: SLSReporter, env: AliRtcEnv);
-    refreshAuthInfo(authInfo: AliRtcAuthInfo): void;
-    private requestToken;
-    startSTSUpdate(authInfo: AliRtcAuthInfo): void;
-    clear(): void;
-}
-
-declare interface StsManagerListener {
-    onTokenUpdate: (ossToken: OSSToken, slsToken: SLSToken) => void;
-}
-
 declare interface SubConfig {
     isAudioSubscribing: boolean;
     isVideoSubscribing: boolean;
@@ -4538,6 +4677,18 @@ declare enum SubscribeReason {
     Reconnect = "reconnect"
 }
 
+declare interface SubscribeRemoteStreamConfig {
+    env?: AliRtcEnv;
+    appId: string;
+    currentUserId: string;
+    targetChannelId: string;
+    targetUserId: string;
+    token: string;
+    nonce?: string;
+    timestamp: number;
+    view?: string | HTMLVideoElement;
+}
+
 declare interface TimeRecorder {
     start: number;
     [key: string]: number;

package/package.json

@@ -1,6 +1,6 @@
 {
   "name": "aliyun-rtc-sdk",
-  "version": "7.2.1",
+  "version": "7.2.3",
   "type": "module",
   "description": "rtc web sdk of aliyun",
   "main": "dist/aliyun-rtc-sdk.umd.js",
@@ -10,7 +10,7 @@
     "@aliyun-sls/web-sts-plugin": "^0.3.5",
     "@aliyun-sls/web-track-browser": "^0.3.5",
     "aliyun-queen-engine": "^6.3.14",
-    "aliyun-rts-sdk": "2.13.7",
+    "aliyun-rts-sdk": "2.14.4",
     "crypto-js": "^4.1.1",
     "dateformat": "^5.0.3",
     "eventemitter3": "^5.0.1",