@wetspace/wetrtc 1.0.0

package/LICENSE

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+MIT License
+
+Copyright (c) 2021 想要飞翔的猪
+
+Permission is hereby granted, free of charge, to any person obtaining a copy
+of this software and associated documentation files (the "Software"), to deal
+in the Software without restriction, including without limitation the rights
+to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+copies of the Software, and to permit persons to whom the Software is
+furnished to do so, subject to the following conditions:
+
+The above copyright notice and this permission notice shall be included in all
+copies or substantial portions of the Software.
+
+THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
+SOFTWARE.

package/README.md

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+# WetRTC
+> 全称 @wetspace/webrtc
+
+#### 介绍
+一个基于webRTC的库，简化了使用流程。使得webRTC开发更为友好快捷
+
+#### 核心功能
+
+1. b2b点对点视频连接功能
+2. b2b文件数据传输功能
+3. 本地视频/语言流捕获上传给中间服务器
+4. 本地视频/语言流录制和截取
+
+## 基本使用
+
+最开始，先实例化`WetRTC`
+```javascript
+import { WetRTC }  from '@wetspace/wetrtc';
+const instance = new WetRTC();
+```
+### 创建点对点连接
+#### 1. 创建一个捕获的流房间
+这是很重要的一步，有捕获了流，才能按业务需求，把流推入到连接
+
+在创建捕获流的房间的过程中，会检测可用设备，因为考虑到即便是没有音视频输入，只是捕获不到流，对方无法看到视频而已，因此，该函数接受一个回调函数，接受一个参数，参数就是当前音视频的可用输入设备,业务如何处理就交给使用者
+
+```javascript
+/**
+ * @param { Object } params 回调参数
+ * @param { MediaDeviceInfo[] } params.audioinput
+ * @param { MediaDeviceInfo[] } params.videoinput
+ **/
+instance.createStreamRoom(callback) // callback可选
+```
+### 发起连接请求
+发起通讯连接，直接使用`instance.connect(options)`即可，内置默认参数
+
+```javascript
+/**
+ * @param { Object } options 连接配置，属性均可选，默认如下
+ * @param { string | number } params.id 连接id
+ * @param { RTCConfiguration } params.configuration
+ * 建立连接发送offer回调
+ * @param { ( params: RTCSessionDescriptionInit)=>void } params.sendOffer
+ * 建立连接发送Candidate回调
+ * @param { ( params: RTCIceCandidate)=>void } params.sendCandidate 
+ * @param { HTMLVideoElement } params.remoteVideoPlay 显示远程视频流
+ * @param { HTMLAudioElement} params.remoteAudioPlay 显示远程音频流
+ **/
+{
+    configuration:{
+        iceServers:[{
+            urls: 'stun:stun.l.google.com:19302'
+        }],
+    },
+}
+```
+如果业务上有需要显示本地流视频业务，可以调用`instance.openLocalVideoOrAudio(localVideo)`,localVideo是显示本地流的视频DOM
+
+**注意：**
+对于主动发起方的远程连接是需要提供`sendCandidate `,对于应答连接则不需要，应答连接只需要在连接后立即`applyAnswer`即可
+
+完整示例代码如下:
+
+```javascript
+const connect = ( options = {
+    sendCandidate:(candidate)=>{sendMsg(candidate)},
+    sendOffer:(sdp)=>{sendMsg(sdp)},
+    remoteVideoPlay,
+})=>{
+    instance.connect(Object.assign(options))
+    // 可选，如果业务有把本地视频流展示出来的需求
+    if(localVideo){
+        instance.openLocalVideoOrAudio(localVideo)
+    }
+}
+```
+
+### 信令交换必要信息流程
+
+发起连接了，这时候就是要响应`answer`，和响应`candidate`，这里本sdk提供了2个方法`applyAnswer`,用于应答，接受一个点对点对象`RTCPeerConnection`，和用于发送`answer(sdp)`的回调函数；另一个就是`addIceCandidate`,是用来添加网络候选的，接受一个点对点对象`RTCPeerConnection`和`RTCIceCandidateInit`类型的参数
+```javascript
+import { applyAnswer, addIceCandidate} from '@wetspace/wetrtc'
+...
+switch (message.action) {
+    case "answer":
+        applyAnswer(RTCPeerConnection,message.answer);
+        break;
+    case "candidate":
+        addIceCandidate(RTCPeerConnection,message.candidate);
+        break;
+}
+...
+```
+### 关闭连接
+
+#### 1.1 关闭点对点连接
+使用`instance.disconnect()`,如果没有入参，则关闭全部连接，如果有入参id，则关闭以该id建立点对点连接
+```javascript
+instance.disconnect()
+instance.disconnect('uuid')
+```
+
+#### 1.2 关闭流房间
+
+关闭流房间，是当你不采集音视频流，或者当这个对讲功能完全关闭的时候进行关闭处理，这样就能释放设备对流的捕获。如果不关闭，换一个应用，发起了WebRTC对讲则会导致设备被占用，从而导致捕获流失败
+
+值得注意的是，关闭流房间不会导致对讲连接关闭，只是会导致一方无法获取对方的远程音视频
+
+```javascript
+instance.closeStreamRoom()
+```
+
+### 具体使用文档
+
+[文档手册](https://wetspace.gitee.io/wetrtc/)
+
+#### 补充
+欢迎大家补充英文文档（虽然我也会补充，但是本人英语是个菜逼）
+#### 特技
+
+1.  使用 Readme\_XXX.md 来支持不同的语言，例如 Readme\_en.md, Readme\_zh.md
+2.  Gitee 官方博客 [blog.gitee.com](https://blog.gitee.com)
+3.  你可以 [https://gitee.com/explore](https://gitee.com/explore) 这个地址来了解 Gitee 上的优秀开源项目
+4.  [GVP](https://gitee.com/gvp) 全称是 Gitee 最有价值开源项目，是综合评定出的优秀开源项目
+5.  Gitee 官方提供的使用手册 [https://gitee.com/help](https://gitee.com/help)
+6.  Gitee 封面人物是一档用来展示 Gitee 会员风采的栏目 [https://gitee.com/gitee-stars/](https://gitee.com/gitee-stars/)

package/package.json

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+{
+  "name": "@wetspace/wetrtc",
+  "version": "1.0.0",
+  "description": "一个基于webRTC的库，简化了使用流程。使得webRTC开发更为友好快捷",
+  "main": "src/index.ts",
+  "module": "build/wetrtc.js",
+  "scripts": {
+    "dev": "rollup -c -w",
+    "build": "rollup -c",
+    "docs:dev": "vuepress dev docs",
+    "docs:build": "vuepress build docs",
+    "test": "echo \"Error: no test specified\" && exit 1"
+  },
+  "repository": {
+    "type": "git",
+    "url": "https://gitee.com/wetspace/wetrtc"
+  },
+  "publishConfig": {
+    "registry": "https://registry.npmjs.org/",
+    "access": "public"
+  },
+  "files":["README.md",""],
+  "author": "",
+  "license": "ISC",
+  "dependencies": {
+    "@babel/core": "^7.16.0",
+    "@rollup/plugin-babel": "^5.3.0",
+    "@rollup/plugin-commonjs": "^21.0.1",
+    "@rollup/plugin-node-resolve": "^13.0.6",
+    "@rollup/plugin-replace": "^3.0.0",
+    "@rollup/plugin-typescript": "^8.3.0",
+    "@types/lodash": "^4.14.178",
+    "lodash": "^4.17.21",
+    "rollup-plugin-uglify": "^6.0.4",
+    "tslib": "^2.3.1",
+    "typescript": "^4.5.2"
+  },
+  "devDependencies": {
+    "rollup": "^2.60.1",
+    "vuepress": "^2.0.0-beta.27"
+  }
+}

package/src/index.ts

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+import { WetRTC } from './core/index';
+import { getMediaStream, concatStream,filterStream} from './core/stream';
+import { RecordVideoOrAudio } from './lib/record';
+import { getUsAbleDevices } from './core/devices';
+import { getTracks } from './core/tracks';
+import { connectRTC, createLocalPlayer} from './core/create';
+import { applyOffer,  applyAnswer, addIceCandidate } from './core/actions';
+
+export{
+    WetRTC, // rtc连接类，一个主体一个WetRTC连接，其他参与者都是wetChildRTC创建点对点
+    getMediaStream,// 获取媒体流
+    concatStream,// 合并流
+    filterStream,// 按video/audio过滤流
+    RecordVideoOrAudio,// 录制
+    getUsAbleDevices,// 获取可用设备
+    getTracks,// 获取流输入
+    connectRTC, // 创建点对点连接
+    createLocalPlayer, // 将本地视频流添加到视频
+    applyOffer, // 提供offer SDP
+    applyAnswer, // 提供answer SDP 或者应答设置
+    addIceCandidate // 给RTCPeerConnection实例设置候选者
+}