WebSocket+Vue+SpringBoot实现语音通话的使用示例

<template>
  <div class="play-audio">
    <el-button @click="startCall" ref="start">开始对讲</el-button>
    <el-button @click="stopCall" ref="stop">结束对讲</el-button>
  </div>
</template>

<script>
export default {
  data() {
    return {
      ws: null,
      mediaStack: null,
      audioCtx: null,
      scriptNode: null,
      source: null,
      play: true
    }
  },

  methods: {
    initWs1() {
		//设置好友ID
	  let recipientId=localStorage.getItem('userId')=="2"?"1":"2";
      this.ws = new WebSocket('ws://192.168.206.204:8081/video/'+localStorage.getItem('userId')+"/"+recipientId)
      this.ws.onopen = () => {
        console.log('socket 已连接')
      }
      this.ws.binaryType = 'arraybuffer'
      this.ws.onmessage = ({ data }) => {
	 console.log("接收到的数据--》"+ data)
	
		
        // 将接收的数据转换成与传输过来的数据相同的Float32Array
        const buffer = new Float32Array(data)
        // 创建一个空白的AudioBuffer对象，这里的4096跟发送方保持一致，48000是采样率
        const myArrayBuffer = this.audioCtx.createBuffer(1, 4096, 48000)
        // 也是由于只创建了一个音轨，可以直接取到0
        const nowBuffering = myArrayBuffer.getChannelData(0)
        // 通过循环，将接收过来的数据赋值给简单音频对象
        for (let i = 0; i < 4096; i++) {
          nowBuffering[i] = buffer[i]
        }
        // 使用AudioBufferSourceNode播放音频
        const source = this.audioCtx.createBufferSource()
        source.buffer = myArrayBuffer
        const gainNode = this.audioCtx.createGain()
        source.connect(gainNode)
        gainNode.connect(this.audioCtx.destination)
        var muteValue = 1
        if (!this.play) { // 是否静音
          muteValue = 0
        }
        gainNode.gain.setValueAtTime(muteValue, this.audioCtx.currentTime)
        source.start()
      }
      this.ws.onerror = (e) => {
        console.log('发生错误', e)
      }
      this.ws.onclose = () => {
        console.log('socket closed')
      }
	
    },
    // 开始对讲
    startCall() {

      this.play = true
      this.audioCtx = new AudioContext()
      this.initWs1()
	
      // 该变量存储当前MediaStreamAudioSourceNode的引用
      // 可以通过它关闭麦克风停止音频传输

      // 创建一个ScriptProcessorNode 用于接收当前麦克风的音频
      this.scriptNode = this.audioCtx.createScriptProcessor(4096, 1, 1)
      navigator.mediaDevices
        .getUserMedia({ audio: true, video: false })
        .then((stream) => {
          this.mediaStack = stream
          this.source = this.audioCtx.createMediaStreamSource(stream)

          this.source.connect(this.scriptNode)
          this.scriptNode.connect(this.audioCtx.destination)
        })
        .catch(function (err) {
          /* 处理error */
          console.log('err', err)
        })
      // 当麦克风有声音输入时，会调用此事件
      // 实际上麦克风始终处于打开状态时，即使不说话，此事件也在一直调用
      this.scriptNode.onaudioprocess = (audioProcessingEvent) => {
        const inputBuffer = audioProcessingEvent.inputBuffer
		// console.log("inputBuffer",inputBuffer);
        // 由于只创建了一个音轨，这里只取第一个频道的数据
        const inputData = inputBuffer.getChannelData(0)
		console.log("调用")
        // 通过socket传输数据，实际上传输的是Float32Array
        if (this.ws.readyState === 1) {
			
			
			// console.log("发送的数据",inputData);
          this.ws.send(inputData)
        }
      }
    },
    // 关闭麦克风
    stopCall() {
      this.play = false
      this.mediaStack.getTracks()[0].stop()
      this.scriptNode.disconnect()
    }
  }
}
</script>

package com.example.software.config;
import org.springframework.context.annotation.Bean;
import org.springframework.context.annotation.Configuration;
import org.springframework.web.socket.config.annotation.EnableWebSocket;
import org.springframework.web.socket.server.standard.ServerEndpointExporter;

/**
 * @Description: websocket配置
 */
@Configuration
@EnableWebSocket
public class WebSocketConfig {
    @Bean
    public ServerEndpointExporter serverEndpointExporter() {
        return new ServerEndpointExporter();
    }
}

package com.example.software.service.webscoket;

import cn.hutool.core.map.MapUtil;
import cn.hutool.core.util.StrUtil;
import cn.hutool.json.JSONUtil;
import lombok.extern.slf4j.Slf4j;
import org.springframework.beans.factory.annotation.Autowired;
import org.springframework.stereotype.Controller;

import javax.websocket.*;
import javax.websocket.server.PathParam;
import javax.websocket.server.ServerEndpoint;
import java.io.BufferedInputStream;
import java.io.IOException;
import java.io.InputStream;
import java.nio.ByteBuffer;
import java.util.ArrayList;
import java.util.HashMap;
import java.util.List;
import java.util.Map;
import java.util.concurrent.ConcurrentHashMap;

/**
 * @Author:wf
 * @Date 2023/5/14 13:55
 * 消息收发
 **/
@Controller
@ServerEndpoint(value = "/video/{senderID}/{recipientId}")
@Slf4j
public class WebSocketServer {


    /** 当前在线连接数。应该把它设计成线程安全的 */
    private static  int onlineCount = 0;
    /** 存放每个客户端对应的MyWebSocket对象。实现服务端与单一客户端通信的话，其中Key可以为用户标识 */
    private static ConcurrentHashMap<String, Session> webSocketSet = new ConcurrentHashMap<String, Session>();

    /** 与某个客户端的连接会话，需要通过它来给客户端发送数据 */
    private Session WebSocketsession;
    /** 当前发消息的人员编号 */
    private String senderID = "";


    /**
     * 连接建立成功调用的方法
     * @param param 发送者ID，是由谁发送的
     * @param WebSocketsession 可选的参数。session为与某个客户端的连接会话，需要通过它来给客户端发送数据
     */
    @OnOpen
    public void onOpen(@PathParam(value = "senderID") String param, @PathParam(value = "recipientId") String recipientId,Session WebSocketsession) {
        System.out.println("人员-------**-------编号:"+param+"：加入聊天");
        System.out.println("盆友是:"+recipientId+"");

        //接收到发送消息的人员编号
        senderID = param;
        System.out.println("senderID："+senderID);
        //设置消息大小最大为10M，这种方式也可以达到效果，或者使用下面的    @OnMessage(maxMessageSize=5242880)
        //The default buffer size for text messages is 8192 bytes.消息超过8192b,自动断开连接

//        WebSocketsession.setMaxTextMessageBufferSize(10*1024*1024);
//        WebSocketsession.setMaxBinaryMessageBufferSize(10*1024*1024);


        //加入map中，绑定当前用户和socket
        webSocketSet.put(param, WebSocketsession);
        //在线数加1
        addOnlineCount();
    }


    /**
     * 连接关闭调用的方法
     */
    @OnClose
    public void onClose() {
        if (StrUtil.isNotBlank(senderID)) {
            //从set中删除
            webSocketSet.remove(senderID);
            //在线数减1
            subOnlineCount();
        }
    }


    /**
     * 收到客户端消息后调用的方法
     *
     *设置最大接收消息大小
     */
    @OnMessage(maxMessageSize=5242880)
    public void onMessage(@PathParam(value = "senderID") String senderID ,@PathParam(value = "recipientId") String recipientId,InputStream inputStream) {
        System.out.println(senderID+":发送给"+recipientId+"的消息-->"+inputStream);

        try {
            byte[] buff = new byte[inputStream.available()];
            inputStream.read(buff, 0, inputStream.available());
            Session session = webSocketSet.get("2");
            synchronized (session) {
                //给2号发送
                session.getBasicRemote().sendBinary(ByteBuffer.wrap(buff));
            }
        } catch (Exception e) {
            e.printStackTrace();
        }
    }




    /**
     * 发生错误时调用
     *
     * @param session
     * @param error
     */
    @OnError
    public void onError(Session session, Throwable error) {
        error.printStackTrace();
    }


    /**
     * 为指定用户发送消息
     *
     * @param message 消息内容
     * @throws IOException
     */
    public void sendMessage(String message) throws IOException {
        //加同步锁，解决多线程下发送消息异常关闭
        synchronized (this.WebSocketsession){
            this.WebSocketsession.getBasicRemote().sendText(message);
        }
    }

    /**
     * 获取当前在线人数
     * @return 返回当前在线人数
     */
    public static synchronized int getOnlineCount() {
        return onlineCount;
    }

    /**
     * 增加当前在线人数
     */
    public static synchronized void addOnlineCount() {
        WebSocketServer.onlineCount++;
    }

    /**
     * 减少当前在线人数
     */
    public static synchronized void subOnlineCount() {
        WebSocketServer.onlineCount--;
    }

}

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