trtc-cloud-js-sdk 1.0.13 → 2.0.0

package/package.json

@@ -1,8 +1,7 @@
 {
   "name": "trtc-cloud-js-sdk",
-  "version": "1.0.13",
-  "description": "Tencent Cloud RTC SDK for Web And Electron",
-  "main": "trtc-cloud-js-sdk.js",
+  "version": "2.0.0",
+  "description": "Tencent Cloud RTC SDK for Web",
   "keywords": [
     "实时音视频",
     "视频通话",
@@ -12,13 +11,64 @@
   ],
   "repository": {
     "type": "git",
-    "url": "git@github.com:tencentyun/TRTCSDK.git"
+    "url": ""
   },
-  "author": "Tencent Cloud Client R&D Center",
+  "husky": {
+    "hooks": {
+      "pre-commit": "yarn run lint"
+    }
+  },
+  "scripts": {
+    "test": "karma start",
+    "lint": "eslint src --ext .ts",
+    "dev": "rollup -c build/rollup.config.dev.js -w",
+    "docs": "tsc && yarn run docs:clean && yarn run docs:build:zh-cn",
+    "docs:build:zh-cn": "./node_modules/.bin/jsdoc -c build/jsdoc/jsdoc.json -r && node ./build/jsdoc/fix-doc.js",
+    "docs:clean": "node ./build/jsdoc/clean-doc.js",
+    "tsc": "rm -fr ./build/ts2js && tsc",
+    "prod": "rm -fr ./dist/**/*.js* && rollup -c build/rollup.config.prod.js && node ./build/package-bundle.js ",
+    "script": "browserify ./dist/trtc-cloud.js -o ./dist/scriptJs/trtc-cloud-js-sdk.js && cp ./dist/scriptJs/trtc-cloud-js-sdk.js ./examples/jsExample/dist/trtc-cloud-js-sdk.js && rm ./dist/trtc-cloud.js",
+    "zip": "cd ./cos && tar zcvf apiExample.zip ./apiExample && tar zcvf jsExample.zip ./jsExample && cd ../dist && tar zcvf npm-package.zip ./npm-package",
+    "publish": "yarn run prod && yarn run script && yarn run docs && node ./scripts/publish.js && yarn run zip"
+  },
+  "author": "",
   "license": "ISC",
+  "devDependencies": {
+    "@babel/core": "^7.21.4",
+    "@babel/preset-env": "^7.21.4",
+    "@typescript-eslint/parser": "^5.59.0",
+    "docdash-blue": "^1.1.9",
+    "eslint": "^8.38.0",
+    "eslint-config-tencent": "^1.0.4",
+    "husky": "^8.0.3",
+    "jsdoc": "^4.0.2",
+    "jsdoc-plugin-typescript": "^2.2.1",
+    "karma": "^6.4.1",
+    "karma-chrome-launcher": "^3.1.1",
+    "karma-coverage": "^2.2.0",
+    "karma-coverage-istanbul-reporter": "^3.0.3",
+    "karma-jasmine": "^5.1.0",
+    "karma-rollup-preprocessor": "^7.0.8",
+    "lint-staged": "^13.2.1",
+    "replace": "^1.2.2",
+    "rollup-plugin-babel": "^4.4.0",
+    "rollup-plugin-commonjs": "^10.1.0",
+    "rollup-plugin-filesize": "^10.0.0",
+    "rollup-plugin-json": "^4.0.0",
+    "rollup-plugin-livereload": "^2.0.5",
+    "rollup-plugin-node-globals": "^1.4.0",
+    "rollup-plugin-node-resolve": "^5.2.0",
+    "rollup-plugin-progress": "^1.1.2",
+    "rollup-plugin-replace": "^2.2.0",
+    "rollup-plugin-serve": "^2.0.2",
+    "rollup-plugin-terser": "^7.0.2",
+    "rollup-plugin-typescript": "^1.0.1"
+  },
   "dependencies": {
     "axios": "^0.21.1",
-    "rtc-beauty-plugin": "^0.0.5",
-    "trtc-js-sdk": "4.15.8"
+    "rtc-beauty-plugin": "^0.0.6",
+    "trtc-sdk-v5": "^5.0.0",
+    "tslib": "^2.5.3",
+    "typescript": "^4.6.2"
   }
 }

package/scripts/publish.js

@@ -0,0 +1,86 @@
+const fs = require('fs');
+const path = require('path');
+const excludesPath = [
+  'node_modules',
+].map(excludePath => path.resolve(path.join('./examples/apiExample', excludePath)));
+
+function copyDir(fromPath, toPath) {
+  fs.mkdirSync(toPath);
+  const dirsPath = fs.readdirSync(fromPath);
+  dirsPath.forEach((dirPath) => {
+    const from = path.join(fromPath, dirPath);
+    const to = path.join(toPath, dirPath);
+    if (excludesPath.indexOf(from) >= 0) {
+      return;
+    }
+    if (fs.statSync(from).isDirectory()) {
+      copyDir(from, to);
+    } else {
+      copyFile(from, to);
+    }
+  });
+}
+
+function copyFile(fromPath, toPath) {
+  fs.copyFileSync(fromPath, toPath);
+}
+
+function removeDir(originPath) {
+  let stats;
+  try {
+    stats = fs.statSync(originPath);
+  } catch (error) {
+    return;
+  }
+  if (stats.isFile()) {
+    fs.unlinkSync(originPath);
+  } else if (stats.isDirectory()) {
+    const dirs = fs.readdirSync(originPath);
+    const dirsPath = dirs.map(dir => path.join(originPath, dir));
+    dirsPath.forEach((dirPath) => {
+      removeDir(dirPath);
+    });
+    fs.rmdirSync(originPath);
+  }
+}
+
+// 【1】apiExample 发布到 cos
+const apiExampleName = 'apiExample';
+const apiExternalName = 'cos';
+const apiFromPath = path.resolve(`./examples/${apiExampleName}`);
+const apiToPath = path.resolve(`./${apiExternalName}/${apiExampleName}`);
+// 清除原有的文件
+removeDir(`./${apiExternalName}/${apiExampleName}`);
+// 将 TRTC-UniApp 拷贝到目标文件夹
+copyDir(apiFromPath, apiToPath);
+// 脱敏操作
+const apiModifyFile = path.resolve(`./${apiExternalName}/${apiExampleName}/src/config/gen-test-user-sig.js`);
+
+fs.readFile(apiModifyFile, 'utf-8', (err, data) => {
+  if (err) throw err;
+  let newValue = data.replace(/const SDKAPPID = (\d{10});/, 'const SDKAPPID = 0;'); // 清除 sdkAppId
+  newValue = newValue.replace(/const SECRETKEY = ('\w{64}');/, 'const SECRETKEY = \'\';'); // 清除 secretKey
+  fs.writeFile(apiModifyFile, newValue, 'utf-8', (err) => {
+    if (err) throw err;
+  });
+});
+
+// 【2】jsExample 发布到 cos
+const jsExampleName = 'jsExample';
+const jsExternalName = 'cos';
+const jsFromPath = path.resolve(`./examples/${jsExampleName}`);
+const jsToPath = path.resolve(`./${jsExternalName}/${jsExampleName}`);
+// 清除原有的文件
+removeDir(`./${jsExternalName}/${jsExampleName}`);
+// 将 TRTC-UniApp 拷贝到目标文件夹
+copyDir(jsFromPath, jsToPath);
+// 脱敏操作
+const jsModifyFile = path.resolve(`./${jsExternalName}/${jsExampleName}/rtc/js/index.js`);
+fs.readFile(jsModifyFile, 'utf-8', (err, data) => {
+  if (err) throw err;
+  let newValue = data.replace(/const sdkAppId = (\d{10});/, 'const sdkAppId = 0;'); // 清除 sdkAppId
+  // newValue = newValue.replace(/const userSig = ('(\w*){196}}');/, 'const userSig = \'\';'); // 清除 userSig
+  fs.writeFile(jsModifyFile, newValue, 'utf-8', (err) => {
+    if (err) throw err;
+  });
+});

package/src/Camera.ts

@@ -0,0 +1,80 @@
+import { TRTCDeviceInfo, TRTCDeviceType, TRTCDeviceState } from './common/trtc_define';
+import { BaseCommon } from './BaseCommon';
+import { MixinsClass, hasVideoTrack } from './utils/utils';
+import { NAME } from './common/constants';
+
+/**
+ * 摄像头相关接口<br>
+ * @memberof TRTCClouds
+ */
+class Camera extends MixinsClass(BaseCommon) {
+  /**
+   * 获取摄像头设备列表<br>
+   * @returns {Promise(Array<TRTCDeviceInfo>)} 摄像头管理器列表
+   * @memberof TRTCCloud
+   * @example
+   * var cameralist = await trtcCloud.getCameraDevicesList();
+   * for (i = 0; i < cameralist.length; i++) {
+   *    var camera = cameralist[i];
+   *    console.info("camera deviceName: " + camera.deviceName + " deviceId:" + camera.deviceId);
+   * }
+   */
+  async getCameraDevicesList(): Promise<TRTCDeviceInfo[]> {
+    try {
+      const cameraList = await this.TRTC.getCameras();
+      this.cameraList = cameraList.map((item: TRTCDeviceInfo) => ({
+        ...item,
+        deviceName: item.label, // 就是 deviceName
+      }));
+      return Promise.resolve(this.cameraList);
+    } catch (error: any) {
+      this.callFunctionErrorManage(error, 'getCameraDevicesList');
+      this.cameraList = [];
+      return Promise.resolve([]);
+    }
+  }
+
+  /**
+   * 设置要使用的摄像头<br>
+   * @param {String} deviceId - 从 getCameraDevicesList 中得到的设备 ID
+   * @returns {Promise}
+   * @memberof TRTCCloud
+   */
+  async setCurrentCameraDevice(deviceId: string): Promise<boolean> {
+    try {
+      if (!deviceId) {
+        return Promise.resolve(false);
+      }
+      if (this.currentCameraId === deviceId) {
+        return Promise.resolve(true);
+      }
+      this.currentCameraId = deviceId || '';
+      if (hasVideoTrack(this.localStream)) {
+        await this.localStream.switchDevice(NAME.VIDEO, deviceId);
+      }
+      if (hasVideoTrack(this.localPlayStream)) {
+        await this.localPlayStream.switchDevice(NAME.VIDEO, deviceId);
+      }
+      if (hasVideoTrack(this.testCameraStream)) {
+        await this.testCameraStream.switchDevice('video', deviceId);
+      }
+      this.emitOnDeviceChange(deviceId, TRTCDeviceType.TRTCDeviceTypeCamera, TRTCDeviceState.TRTCDeviceStateActive);
+      return Promise.resolve(true);
+    } catch (error: any) {
+      this.callFunctionErrorManage(error, 'setCurrentCameraDevice');
+      this.cameraList = [];
+      return Promise.resolve(false);
+    }
+  }
+
+  /**
+   * 获取当前使用的摄像头<br>
+   * @memberof TRTCCloud
+   * @returns {TRTCDeviceInfo} 设备信息，能获取设备 ID 和设备名称
+   */
+  getCurrentCameraDevice(): TRTCDeviceInfo {
+    return (this.cameraList || []).filter((obj: any) => obj.deviceId === this.currentCameraId)[0] || {};
+  }
+}
+
+export { Camera };

package/src/Mic.ts

@@ -0,0 +1,145 @@
+import { TRTCDeviceInfo, TRTCDeviceType, TRTCDeviceState } from './common/trtc_define';
+import { BaseCommon } from './BaseCommon';
+import { MixinsClass, hasAudioTrack } from './utils/utils';
+import { NAME } from './common/constants';
+
+/**
+ * 音频设备相关接口<br>
+ * @memberof TRTCClouds
+ */
+class Mic extends MixinsClass(BaseCommon) {
+  /**
+   * 获取麦克风设备列表<br>
+   * @returns {Promise(Array<TRTCDeviceInfo>)} 麦克风管理器列表
+   * @memberof TRTCCloud
+   * @example
+   *   var micList = await trtcCloud.getMicDevicesList();
+   *   for (i = 0; i < micList.length; i++) {
+   *     var mic = micList[i];
+   *     console.info("mic deviceName: " + mic.deviceName + " deviceId:" + mic.deviceId);
+   *   }
+   */
+  async getMicDevicesList(): Promise<TRTCDeviceInfo[]> {
+    try {
+      const micList = await this.TRTC.getMicrophones();
+      this.micList = micList.map((item: TRTCDeviceInfo) => ({
+        ...item,
+        deviceName: item.label, // 就是 deviceName
+      }));
+      return Promise.resolve(this.micList);
+    } catch (error: any) {
+      this.callFunctionErrorManage(error, 'getMicDevicesList');
+      return Promise.resolve([]);
+    }
+  }
+
+  /**
+   * 获取当前选择的麦克风<br>
+   * @memberof TRTCCloud
+   * @return {TRTCDeviceInfo} 设备信息，能获取设备 ID 和设备名称
+   */
+  getCurrentMicDevice(): TRTCDeviceInfo {
+    if (this.currentMicId) {
+      return (this.micList || []).filter((obj: any) => obj.deviceId === this.currentMicId)[0] || {};
+    }
+    return new TRTCDeviceInfo();
+  }
+
+  /**
+   * 设置要使用的麦克风<br>
+   * 选择指定的麦克风作为录音设备，不调用该接口时，默认选择索引为0的麦克风
+   *
+   * @param {String} micId - 从 getMicDevicesList 中得到的设备 ID
+   * @memberof TRTCCloud
+   * @returns {Promise}
+   */
+  async setCurrentMicDevice(micId: string): Promise<boolean> {
+    try {
+      if (!micId) {
+        return Promise.resolve(false);
+      }
+      if (this.currentMicId === micId) {
+        return Promise.resolve(true);
+      }
+      this.currentMicId = micId || '';
+      if (hasAudioTrack(this.localStream)) {
+        await this.localStream.switchDevice(NAME.AUDIO, micId);
+      }
+      if (hasAudioTrack(this.localPlayStream)) {
+        await this.localPlayStream.switchDevice(NAME.AUDIO, micId);
+      }
+      this.emitOnDeviceChange(micId, TRTCDeviceType.TRTCDeviceTypeMic, TRTCDeviceState.TRTCDeviceStateActive);
+      return Promise.resolve(true);
+    } catch (error: any) {
+      this.callFunctionErrorManage(error, 'setCurrentMicDevice');
+      return Promise.resolve(false);
+    }
+  }
+
+  /**
+   * 获取扬声器设备列表<br>
+   * - 出于安全的考虑，在用户未授权摄像头或麦克风访问权限前，deviceId 字段可能都是空的。因此建议在用户授权访问后 再调用该接口获取设备详情
+   * - 移动端不支持获取扬声器列表
+   * @return {Promise(Array<TRTCDeviceInfo>)} 扬声器管理器列表
+   * @memberof TRTCCloud
+   * @example
+   *   var speakerList = await trtcCloud.getSpeakerDevicesList();
+   *   for (i = 0; i < speakerList.length; i++) {
+   *     var speaker = speakerList[i];
+   *     console.info("mic deviceName: " + speaker.deviceName + " deviceId:" + speaker.deviceId);
+   *   }
+   */
+  async getSpeakerDevicesList(): Promise<TRTCDeviceInfo[]> {
+    try {
+      const speakerList = await this.TRTC.getSpeakers();
+      this.speakerList = speakerList.map((item: TRTCDeviceInfo) => ({
+        ...item,
+        deviceName: item.label, // 就是 deviceName
+      }));
+      return Promise.resolve(this.speakerList);
+    } catch (error: any) {
+      this.callFunctionErrorManage(error, 'getSpeakerDevicesList');
+      return Promise.resolve([]);
+    }
+  }
+
+  /**
+   * 获取当前的扬声器设备<br>
+   * @memberof TRTCCloud
+   * @return {TRTCDeviceInfo} 设备信息，能获取设备 ID 和设备名称
+   */
+  getCurrentSpeakerDevice(): TRTCDeviceInfo {
+    if (this.currentSpeakerId) {
+      return (this.speakerList || []).filter((obj: any) => obj.deviceId === this.currentSpeakerId)[0] || {};
+    }
+    return new TRTCDeviceInfo();
+  }
+
+  /**
+   * 设置要使用的扬声器<br>
+   * - 移动端不支持设置扬声器。
+   * @param {String} speakerId - 从 getSpeakerDevicesList 中得到的设备 ID
+   * @returns {Promise}
+   * @memberof TRTCCloud
+   */
+  async setCurrentSpeakerDevice(speakerId: string): Promise<boolean> {
+    try {
+      if (!speakerId) {
+        return Promise.resolve(false);
+      }
+      this.currentSpeakerId = speakerId || '';
+      if (this.remoteStreamMap.size > 0) {
+        const remoteStreamList = this.getRemoteStreamList();
+        // TODO: 如果某个 remoteStream 出现切换失败呢？
+        // eslint-disable-next-line max-len
+        await Promise.all(remoteStreamList.map(async (stream: any) => await stream.setAudioOutput(this.currentSpeakerId)));
+      }
+      return Promise.resolve(true);
+    } catch (error: any) {
+      this.callFunctionErrorManage(error, 'setCurrentSpeakerDevice');
+      return Promise.resolve(false);
+    }
+  }
+}
+
+export { Mic };

package/src/common/IError.ts

@@ -0,0 +1,6 @@
+interface IError {
+  code: number;
+  message: string;
+}
+
+export { IError };

package/src/common/ITRTCCloud.ts

@@ -0,0 +1,68 @@
+import { NAME } from './constants';
+
+interface ITRTCCloudConfig {
+  frameWorkType?: number;
+  component?: number;
+}
+interface TurnServerOptions {
+  url: string;
+  username?: string;
+  credential?: string;
+  credentialType?: 'password';
+}
+
+export interface ProxyServer {
+  websocketProxy?: string;
+  loggerProxy?: string;
+  turnServer?: TurnServerOptions | TurnServerOptions[];
+  webtransportProxy?: string;
+}
+
+export const enum Scene {
+  'LIVE' = 'live',
+  'RTC' = 'rtc',
+};
+
+interface IEnterRoomParams {
+  sdkAppId: number;
+  userId: string;
+  userSig: string;
+  roomId: number | string;
+  role?: string; // webRtc 为 "anchor" | "audience", TrtcCloud 为 TRTCRoleAnchor - 20; TRTCRoleAudience - 21
+  receiveMode?: number; // webRtc 为 MANUAL = 0; AUTO = 1; TrtcCloud 详见 trtc_define.ts 中说明
+  privateMapKey?: string;
+  businessInfo?: string;
+  enableAutoPlayDialog?: boolean; // 默认为 true
+  proxy?: ProxyServer | string;
+  scene?: string; // webRtc 为 "rtc" | "live", TrtcCloud 参考 trtc_define.ts 中 TRTCAppScene 说明
+  useStringRoomId?: boolean;
+  streamId?: string;
+  userDefineRecordId?: string;
+  // 下面两个是 TrtcCloud 自己额外的参数
+  frameWorkType?: number;
+  component?: number;
+}
+
+interface IVideoProfile {
+  width?: number;
+  height?: number;
+  frameRate?: number;
+  bitrate?: number;
+}
+interface IVideoPlayOption {
+  mirror?: boolean;
+  fillMode?: VideoFillMode;
+}
+
+export const enum VideoFillMode {
+  'CONTAIN' = 'contain',
+  'COVER' = 'cover',
+  'FILL' = 'fill',
+}
+
+export {
+  ITRTCCloudConfig,
+  IEnterRoomParams,
+  IVideoProfile,
+  IVideoPlayOption,
+};

package/src/common/constants.ts

@@ -0,0 +1,116 @@
+// --------------------------------常量--------------------------------
+export const EnterRoomFailure: number = -1;
+export const NAME = {
+  DEFAULT: 'default',
+  VIDEO: 'video',
+  AUDIO: 'audio',
+  ON_ERROR: 'onError',
+  MAIN: 'main',
+  // 场景：音视频、直播
+  RTC: 'rtc',
+  LIVE: 'live',
+  // 用户角色
+  ANCHOR: 'anchor',
+  AUDIENCE: 'audience',
+  // 音频 profile
+  AUDIO_STANDARD: 'standard',
+  AUDIO_HIGH: 'high',
+  AUDIO_STANDARD_STEREO: 'standard_stereo',
+  AUDIO_HIGH_STEREO: 'high_stereo',
+  // 音频 profile
+  SMALL: 'small',
+  BIG: 'big',
+  AUXILIARY: 'auxiliary',
+  // 屏幕分享用户名的前缀
+  SCREEN_SHARE_USER_ID_PREFIX: 'share_',
+  NUMBER: 'number',
+  STRING: 'string',
+};
+
+// 网络类型与仪表盘传参的对应关系
+export const NETWORK_TYPE = {
+  unknown: 0,
+  wifi: 1,
+  '3g': 2,
+  '2g': 3,
+  '4g': 4,
+  wired: 5,
+};
+interface IRoleMap {
+  [key: number]: string;
+}
+
+// webRTC 仅在 'live' 模式下有意义, 仅支持以下两种角色. TRTCRoleAnchor = 20, TRTCRoleAudience = 21,
+export const RoleMap: IRoleMap = {
+  20: NAME.ANCHOR,
+  21: NAME.AUDIENCE,
+};
+
+// TRTCAppSceneVideoCall = 0, 视频通话场景
+// TRTCAppSceneLIVE = 1, 视频互动直播
+// TRTCAppSceneAudioCall = 2, 语音通话场景
+// TRTCAppSceneVoiceChatRoom = 3, 语音互动直播
+// webRTC 仅在 'live' 模式下有意义, 仅支持以下两种角色
+export const SceneMap = {
+  0: NAME.RTC,
+  1: NAME.LIVE,
+  2: NAME.RTC,
+  3: NAME.LIVE,
+};
+
+// webRTC 音频 https://web.sdk.qcloud.com/trtc/webrtc/doc/zh-cn/LocalStream.html#setAudioProfile
+export const audioQualityMap = {
+  0: NAME.AUDIO_STANDARD,
+  1: NAME.AUDIO_HIGH,
+  2: NAME.AUDIO_STANDARD_STEREO,
+  3: NAME.AUDIO_HIGH_STEREO,
+};
+
+// webRTC 视频
+export const streamTypeMap = {
+  0: NAME.BIG,
+  1: NAME.SMALL,
+  2: NAME.AUXILIARY,
+};
+
+//  * WebRTC 视频分辨率和 TRTCCloud 的映射关系
+//  *  export enum TRTCVideoResolution {
+//       TRTCVideoResolution_120_120 = 1,
+//       TRTCVideoResolution_160_160 = 3,
+//       TRTCVideoResolution_270_270 = 5,
+//       TRTCVideoResolution_480_480 = 7,
+//       TRTCVideoResolution_160_120 = 50,
+//       TRTCVideoResolution_240_180 = 52,
+//       TRTCVideoResolution_280_210 = 54,
+//       TRTCVideoResolution_320_240 = 56,
+//       TRTCVideoResolution_400_300 = 58,
+//       TRTCVideoResolution_480_360 = 60,
+//       TRTCVideoResolution_640_480 = 62,
+//       TRTCVideoResolution_960_720 = 64,
+//       TRTCVideoResolution_160_90 = 100,
+//       TRTCVideoResolution_256_144 = 102,
+//       TRTCVideoResolution_320_180 = 104,
+//       TRTCVideoResolution_480_270 = 106,
+//       TRTCVideoResolution_640_360 = 108,
+//       TRTCVideoResolution_960_540 = 110,
+//       TRTCVideoResolution_1280_720 = 112,
+//       TRTCVideoResolution_1920_1080 = 114
+//     }
+//   * | 视频Profile | 分辨率（宽 x 高）| 帧率（fps）| 码率（kbps）|
+//   * | :---       | :---           | :---      | :---      |
+//   * | 120p       | 160 x 120      | 15        | 200        |
+//   * | 180p       | 320 x 180      | 15        | 350        |
+//   * | 240p       | 320 x 240      | 15        | 400       |
+//   * | 360p       | 640 x 360      | 15        | 800       |
+//   * | 480p       | 640 x 480      | 15        | 900       |
+//   * | 720p       | 1280 x 720     | 15        | 1500      |
+//   * | 1080p      | 1920 x 1080    | 15        | 2000      |
+//   * | 1440p      | 2560 x 1440    | 30        | 4860      |
+//   * | 4K         | 3840 x 2160    | 30        | 9000      |
+export const videoResolutionMap = {};
+export const videoProfileMap = {};
+
+export enum ExitRoomCode {
+  'ActiveExitRoom' = 0,
+  'KickedExitRoom' = 1,
+}

package/src/common/trtc-code.ts

@@ -0,0 +1,43 @@
+import { IError } from './IError';
+
+
+export const jsExecuteError: IError = {
+  code: -1000,
+  message: 'js 执行报错',
+};
+export const ParametersError: IError = {
+  code: -1001,
+  message: '参数错误, 请检查参数格式、是否非空等',
+};
+export const enterRoomError: IError = {
+  code: -1002,
+  message: '进房失败：进房中或已进房',
+};
+export const exitRoomError: IError = {
+  code: -1003,
+  message: '退房失败：未进房或退房中',
+};
+export const conditionsError: IError = {
+  code: -1004,
+  message: '调用该函数所需要条件不满足',
+};
+export const NotSupportError: IError = {
+  code: -1005,
+  message: 'not supported',
+};
+export const NotSupportBeforeJoinRoomError: IError = {
+  code: -1006,
+  message: 'not supported before join room',
+};
+export const EnterRoomUserIdError: IError = {
+  code: -1007,
+  message: '进房 userId 不能包含前缀 share_',
+};
+export const ScreenShareUserIdError: IError = {
+  code: -1008,
+  message: '屏幕分享 userId 必须包含前缀 share_',
+};
+export const ScreenShareTypeError: IError = {
+  code: -1009,
+  message: '不支持小流推屏幕分享',
+};