~

public/config.js

@@ -33,3 +33,5 @@ function isWhich() {
   }
 }
 window.GLOBAL_CONFIG = isWhich()
+
+window.RTC_USE = false

src/components/rtc-dialog/index.vue

@@ -13,6 +13,8 @@ import { showConfirmDialog, showSuccessToast, showFailToast } from 'vant';
 let offsetTimer: NodeJS.Timeout
 let offsetTime = $ref<number>(0)
 
+const RTC_USE = window.RTC_USE
+
 const emits = defineEmits<{
   (event: 'update-info', info: Partial<type_dyaw_xlfw_zxhd_log>, isUpdate?: boolean): void;
 }>()
@@ -215,6 +217,7 @@ type TRtcOptions = {
 let rtcOptions: TRtcOptions;
 
 function handleSwitchUse(v: boolean) {
+  if (!RTC_USE) return
   // console.info('[in handleSwitchUse] start ');
   try {
     audioInUse = v
@@ -232,6 +235,7 @@ function handleSwitchUse(v: boolean) {
 }
 
 function initRtcClient() {
+  if (!RTC_USE) return
   console.info('[in initRtcClient] start')
 
   if (rtcInstance.client) {
@@ -253,6 +257,7 @@ function initRtcClient() {
 }
 
 async function getRtcOption() {
+  if (!RTC_USE) return
   console.info('[in getRtcOption] start')
   try {
     await request({
@@ -309,23 +314,24 @@ async function handleAudioChatStart() {
       }, 5000)
       return
     };
-
-    console.info('验证相关权限')
-    // 手机版跳过验证
-    try {
-      console.info('验证麦克风权限')
-      const microInfo = await AgoraRTC.getMicrophones()
-      console.info('麦克风权限验证结束 : ', microInfo)
-      if (mode === 'video') {
-        console.info('验证摄像头权限')
-        const camerasInfo = await AgoraRTC.getCameras()
-        console.info('摄像头权限验证结束 : ', camerasInfo)
+    if (RTC_USE) {
+
+      console.info('验证相关权限')
+      // 手机版跳过验证
+      try {
+        console.info('验证麦克风权限')
+        const microInfo = await AgoraRTC.getMicrophones()
+        console.info('麦克风权限验证结束 : ', microInfo)
+        if (mode === 'video') {
+          console.info('验证摄像头权限')
+          const camerasInfo = await AgoraRTC.getCameras()
+          console.info('摄像头权限验证结束 : ', camerasInfo)
+        }
+      } catch (error) {
+        console.error(error)
       }
-    } catch (error) {
-      console.error(error)
+      console.info('相关权限验证结束')
     }
-    console.info('相关权限验证结束')
-
     initRtcClient()
 
     await getRtcOption()
@@ -335,17 +341,20 @@ async function handleAudioChatStart() {
     console.info('发送拨号信息结束')
 
     await nextTick(async () => {
-      console.info('初始化rtc频道')
-      await rtcInstance.client!.join(rtcOptions.appId, rtcOptions.channel, rtcOptions?.token, /*rtcOptions.uid*/ user.user_id);
-      console.info('初始化rtc频道结束，开始创建本地音视频轨道')
-      rtcInstance.localAudioTrack = await AgoraRTC.createMicrophoneAudioTrack();
-      // await rtcInstance.client!.publish(rtcInstance.localAudioTrack);
-      if (mode === 'video') {
-        console.info('创建本地视频轨道')
-        rtcInstance.localVideoTrack = await AgoraRTC.createCameraVideoTrack();
-        console.info('创建本地视频轨道结束，开始播放本地视频')
-        rtcInstance.localVideoTrack.play(LocalPlayerContainerRef as HTMLElement);
-        // await rtcInstance.client!.publish(rtcInstance.localVideoTrack);
+      if (RTC_USE) {
+
+        console.info('初始化rtc频道')
+        await rtcInstance.client!.join(rtcOptions.appId, rtcOptions.channel, rtcOptions?.token, /*rtcOptions.uid*/ user.user_id);
+        console.info('初始化rtc频道结束，开始创建本地音视频轨道')
+        rtcInstance.localAudioTrack = await AgoraRTC.createMicrophoneAudioTrack();
+        // await rtcInstance.client!.publish(rtcInstance.localAudioTrack);
+        if (mode === 'video') {
+          console.info('创建本地视频轨道')
+          rtcInstance.localVideoTrack = await AgoraRTC.createCameraVideoTrack();
+          console.info('创建本地视频轨道结束，开始播放本地视频')
+          rtcInstance.localVideoTrack.play(LocalPlayerContainerRef as HTMLElement);
+          // await rtcInstance.client!.publish(rtcInstance.localVideoTrack);
+        }
       }
       console.info('socket发送拨号信息')
       socketSend(ws2, {
@@ -377,7 +386,8 @@ async function handleAudioChatCancel() {
       fullSendData
     })
     isOpen = false
-    rtcInstance.client?.leave();
+    if (RTC_USE)
+      rtcInstance.client?.leave();
     handleSwitchUse(false)
     console.info('[success]handleAudioChatCancel end')
   } catch (error) {
@@ -394,25 +404,27 @@ async function handleAudioChatAccept() {
     const fullSendData = await handleInfoEdit({ dxzl_status: '3' }, '通话中')
     console.info('发送接听信息结束')
     // ...
-    console.info('初始化rtc频道')
-    initRtcClient()
+    if (RTC_USE) {
+      console.info('初始化rtc频道')
+      initRtcClient()
 
-    await getRtcOption()
-    console.info('初始化rtc频道')
-    await rtcInstance.client!.join(rtcOptions.appId, rtcOptions.channel, rtcOptions?.token, /*rtcOptions.uid*/ user.user_id);
-    console.info('初始化rtc频道结束，开始创建本地音视频轨道')
-    rtcInstance.localAudioTrack = await AgoraRTC.createMicrophoneAudioTrack();
-    console.info('创建本地音频轨道结束，发布本地音频')
-    await rtcInstance.client!.publish(rtcInstance.localAudioTrack);
-    console.info('发布本地音频结束')
-    if (mode === 'video') {
-      console.info('创建本地视频轨道')
-      rtcInstance.localVideoTrack = await AgoraRTC.createCameraVideoTrack();
-      console.info('创建本地视频轨道结束，开始播放本地视频')
-      rtcInstance.localVideoTrack.play(LocalPlayerContainerRef as HTMLElement);
-      console.info('发布本地视频')
-      await rtcInstance.client!.publish(rtcInstance.localVideoTrack);
-      console.info('发布本地视频结束')
+      await getRtcOption()
+      console.info('初始化rtc频道')
+      await rtcInstance.client!.join(rtcOptions.appId, rtcOptions.channel, rtcOptions?.token, /*rtcOptions.uid*/ user.user_id);
+      console.info('初始化rtc频道结束，开始创建本地音视频轨道')
+      rtcInstance.localAudioTrack = await AgoraRTC.createMicrophoneAudioTrack();
+      console.info('创建本地音频轨道结束，发布本地音频')
+      await rtcInstance.client!.publish(rtcInstance.localAudioTrack);
+      console.info('发布本地音频结束')
+      if (mode === 'video') {
+        console.info('创建本地视频轨道')
+        rtcInstance.localVideoTrack = await AgoraRTC.createCameraVideoTrack();
+        console.info('创建本地视频轨道结束，开始播放本地视频')
+        rtcInstance.localVideoTrack.play(LocalPlayerContainerRef as HTMLElement);
+        console.info('发布本地视频')
+        await rtcInstance.client!.publish(rtcInstance.localVideoTrack);
+        console.info('发布本地视频结束')
+      }
     }
     //
     console.info('socket发送已接听信息')
@@ -455,8 +467,10 @@ async function handleAudioChatEnd() {
   try {
     const fullSendData = await handleInfoEdit({ dxzl_status: '4' }, /*'已结束'*/formatOffsetSec(offsetTime))
     offsetTimer && clearInterval(offsetTimer)
-    await rtcInstance.client!.leave();
-    handleSwitchUse(false)
+    if (RTC_USE) {
+      await rtcInstance.client!.leave();
+      handleSwitchUse(false)
+    }
 
     socketSend(ws2, {
       dxzl_stu_user_id: dyaw_xlfw_zxhd!.dxz_stu_user_id,
@@ -476,6 +490,8 @@ let audioInUse = $ref(true)
 let videoInUse = $ref(true)
 
 function handleSwitchAudio(v?: boolean) {
+  if (!RTC_USE) return
+
   audioInUse = v ?? !audioInUse
   try {
     rtcInstance.localAudioTrack!.setEnabled(audioInUse)
@@ -485,6 +501,8 @@ function handleSwitchAudio(v?: boolean) {
 
 }
 function handleSwitchVideo(v?: boolean) {
+  if (!RTC_USE) return
+
   videoInUse = v ?? !videoInUse
   try {
     rtcInstance.localVideoTrack!.setEnabled(videoInUse)

src/vite-env.d.ts

@@ -7,4 +7,5 @@ interface Window {
     api: string;
     oss: string;
   },
+  RTC_USE: boolean;
 }