fix(frontend): clean up voice preview listeners

frontend/package-lock.json

@@ -8,6 +8,7 @@
       "name": "ai-video-admin-frontend",
       "version": "0.1.0",
       "dependencies": {
+        "@daily-co/daily-js": "^0.90.0",
         "@pipecat-ai/client-js": "^1.12.0",
         "@pipecat-ai/small-webrtc-transport": "^1.10.5",
         "@xyflow/react": "^12.11.0",

frontend/package.json

@@ -9,6 +9,7 @@
     "lint": "eslint"
   },
   "dependencies": {
+    "@daily-co/daily-js": "^0.90.0",
     "@pipecat-ai/client-js": "^1.12.0",
     "@pipecat-ai/small-webrtc-transport": "^1.10.5",
     "@xyflow/react": "^12.11.0",

frontend/src/components/workflow/WorkflowCanvas.tsx

@@ -176,6 +176,10 @@ export function WorkflowCanvas({
   const [addOpen, setAddOpen] = useState(false);
   const [addSourceId, setAddSourceId] = useState<string | null>(null);
   const [addPosition, setAddPosition] = useState<{ x: number; y: number } | null>(null);
+  // Fast Refresh 会替换模块级对象，而保留 React Flow 实例。用 state 固定本次
+  // 挂载使用的映射，避免热更新后被误判为每次渲染都在创建新对象。
+  const [flowNodeTypes] = useState(nodeTypes);
+  const [flowEdgeTypes] = useState(edgeTypes);
   const { screenToFlowPosition } = useReactFlow();
 
   // 只在可持久化的画布内容真正变化时回传。React Flow 挂载时会写入节点尺寸等
@@ -569,8 +573,8 @@ export function WorkflowCanvas({
               <ReactFlow
                 nodes={nodes}
                 edges={edges}
-                nodeTypes={nodeTypes}
-                edgeTypes={edgeTypes}
+                nodeTypes={flowNodeTypes}
+                edgeTypes={flowEdgeTypes}
                 onNodesChange={handleNodesChange}
                 onEdgesChange={onEdgesChange}
                 onConnect={onConnect}

frontend/src/hooks/use-voice-preview.ts

@@ -9,6 +9,7 @@
  */
 
 import { useCallback, useEffect, useRef, useState } from "react";
+import Daily from "@daily-co/daily-js";
 import type { PipecatClientOptions, RTVIMessage } from "@pipecat-ai/client-js";
 import { SmallWebRTCTransport } from "@pipecat-ai/small-webrtc-transport";
 
@@ -101,6 +102,21 @@ function publicVariableSnapshot(
 class AppSmallWebRTCTransport extends SmallWebRTCTransport {
   onAppMessage?: (message: AppMessage) => void;
 
+  /**
+   * SmallWebRTCTransport 的媒体管理器会复用 Daily 的全局 call object，
+   * 但 disconnect() 不会移除构造时注册的设备和 track 监听器。销毁 call
+   * object 后，下一次连接会得到干净的实例，避免监听器跨会话累积。
+   */
+  async dispose(): Promise<void> {
+    this.onAppMessage = undefined;
+    try {
+      await this.disconnect();
+    } finally {
+      const call = Daily.getCallInstance();
+      if (call && !call.isDestroyed()) await call.destroy();
+    }
+  }
+
   override handleMessage(raw: string): void {
     try {
       const message = JSON.parse(raw) as AppMessage;
@@ -262,6 +278,8 @@ export function useVoicePreview(
   const [selectedOutputDeviceId, setSelectedOutputDeviceId] = useState("");
 
   const transportRef = useRef<AppSmallWebRTCTransport | null>(null);
+  const cleanupPromiseRef = useRef<Promise<void>>(Promise.resolve());
+  const connectionGenerationRef = useRef(0);
   const startingRef = useRef(false);
   const messageSeqRef = useRef(0);
   const networkStatsRef = useRef<NetworkStatsSample | null>(null);
@@ -321,9 +339,14 @@ export function useVoicePreview(
   }, [refreshDevices]);
 
   const releaseResources = useCallback(() => {
+    connectionGenerationRef.current += 1;
     const transport = transportRef.current;
     transportRef.current = null;
-    transport?.disconnect().catch(() => {});
+    if (transport) {
+      cleanupPromiseRef.current = cleanupPromiseRef.current
+        .then(() => transport.dispose())
+        .catch(() => {});
+    }
     if (audioRef.current) audioRef.current.srcObject = null;
     startingRef.current = false;
     pendingAssistantTurnsRef.current.clear();
@@ -585,6 +608,8 @@ export function useVoicePreview(
     }
 
     startingRef.current = true;
+    const generation = connectionGenerationRef.current + 1;
+    connectionGenerationRef.current = generation;
     setStatus("connecting");
     setError(null);
     setMicWarning(null);
@@ -594,10 +619,15 @@ export function useVoicePreview(
     setCallEnded(false);
     endedByServerRef.current = false;
 
+    await cleanupPromiseRef.current;
+    if (connectionGenerationRef.current !== generation) return;
+
     const iceServers = await webrtcApi
       .iceServers()
       .then((response) => response.iceServers)
       .catch(() => [{ urls: "stun:stun.l.google.com:19302" }]);
+    if (connectionGenerationRef.current !== generation) return;
+
     const transport = new AppSmallWebRTCTransport({ iceServers });
     transportRef.current = transport;
     transport.onAppMessage = handleAppMessage;
@@ -648,9 +678,11 @@ export function useVoicePreview(
 
     try {
       await transport.initDevices();
+      if (connectionGenerationRef.current !== generation) return;
       if (selectedDeviceIdRef.current) {
         await transport.updateMic(selectedDeviceIdRef.current);
       }
+      if (connectionGenerationRef.current !== generation) return;
       const localAudio = transport.tracks().local.audio;
       const localVideo = transport.tracks().local.video;
       setLocalStream(localAudio ? new MediaStream([localAudio]) : null);
@@ -678,12 +710,17 @@ export function useVoicePreview(
           },
         },
       });
+      if (connectionGenerationRef.current !== generation) return;
       transport.sendAppMessage({ type: "client-ready" });
       setStatus("connected");
     } catch (connectionError) {
-      fail(errorMessage(connectionError, "无法连接语音服务。"));
+      if (connectionGenerationRef.current === generation) {
+        fail(errorMessage(connectionError, "无法连接语音服务。"));
+      }
     } finally {
-      startingRef.current = false;
+      if (connectionGenerationRef.current === generation) {
+        startingRef.current = false;
+      }
     }
   }, [
     assistantId,