python与html的websocket音频通讯2_web向python端传实时音频-CSDN博客

本文链接：https://blog.csdn.net/qq_15011887/article/details/126925504

这一遍，写网页端传声音到python端走websocket。

不二话，上代码打样。
python音频服务端

#复制粘贴即可，细节自己百度，主要是跑通程序
import base64
import asyncio
import websockets
import cv2
import math
import np
import pyaudio
import wave
import time
import threading
import matplotlib.pyplot as plt

chunk = 4096# Record in chunks of 1024 samples
sample_format = pyaudio.paInt16 # 16 bits per sample
channels = 2 #两声道
fs = 44100  # Record at 44100 samples per second
p = pyaudio.PyAudio()  # 创建音频对象

#image = cv2.imread("1.png")#image :是返回提取到的图片的值
#麦克风
micstream = p.open(format=sample_format,#音频输入流创建
                   channels=channels,
                   rate=fs,
                   frames_per_buffer=chunk,
                   input=True)
#喇叭
stream2 = p.open(format=sample_format,#音频输出流创建
                channels=channels,
                rate=fs,
                frames_per_buffer=chunk,
                output=True)


async def echo(websocket, path):
    print(websockets.serve)
    async for message in websocket:
        stream2.write(message)#播放 
        #m=micstream.read(chunk)#录音
        y=[]
        for i in range(1):#这里和上一篇数字不一样请注意。15次录音会阻塞线程。需要开新线程处理。范例用1次可忽略阻塞影响。
            m=micstream.read(chunk)
            y.append(m)
            
        if len(message )>1:await websocket.send(y)#发送录音

        
        
async def main():
    # start a websocket server
    async with websockets.serve(echo, "192.168.1.102", 8765):
        await asyncio.Future()  # run forever

asyncio.run(main())

html网页端，上代码
点击设备授权就开始跑代码

<!DOCTYPE html>
<html lang="en">
<head>
    <meta charset="UTF-8">
    <meta http-equiv="X-UA-Compatible" content="IE=edge, chrome=1">
    <meta name="viewport" content="width=device-width, initial-scale=1.0, minimum-scale=1.0, maximum-scale=1.0,user-scalable=no">
    <meta name="format-detection" content="telephone=no">
    <title>音量</title>
    <style>
	
        
		
    </style>
</head>
<body>
<div id="status">预备开始</div>
<audio id="video" src=""></audio>
<a onclick='aa()'>开始</a> <a onclick='bb()'>结束</a>
<script type="text/javascript">
if (navigator.mediaDevices && navigator.mediaDevices.getUserMedia) {
            // 获取用户的 media 信息
             navigator.mediaDevices.getUserMedia({
	video: false,                                   
    audio: {
        sampleRate: 44100, // 采样率
        channelCount: 2,   // 声道
        volume: 1.0,        // 音量
		sampleBits:16
		}
    }).then((stream) => {
	var audioContext= new AudioContext();
	const socket = new WebSocket('ws://192.168.1.102:8765');
                // 将麦克风的声音输入这个对象
                mediaStreamSource = audioContext.createMediaStreamSource(stream);
                // 创建一个音频分析对象，采样的缓冲区大小为4096，输入和输出都是单声道
                scriptProcessor = audioContext.createScriptProcessor(4096,2,2);
                // 将该分析对象与麦克风音频进行连接
                mediaStreamSource.connect(scriptProcessor);
                // 此举无甚效果，仅仅是因为解决 Chrome 自身的 bug
                scriptProcessor.connect(audioContext.destination);
				//scriptProcessor.start();
                // 开始处理音频
				
				var c=[]
				var i2=0;
                scriptProcessor.onaudioprocess = function(e) {
                    // 获得缓冲区的输入音频，转换为包含了PCM通道数据的32位浮点数组
                    let buffer = e.inputBuffer.getChannelData(0);
                    let buffer1 = e.inputBuffer.getChannelData(1);
					var v = new Int16Array(buffer.length*2)//Int32Array(buffer.length*2);
					for (var i=0;i<buffer.length;i++){
					v[i2]=Math.trunc(buffer[i]*30000);
					i2++;
					v[i2]=Math.trunc(buffer1[i]*30000);
					i2++;
                    //v[i+1]=Math.trunc(buffer1[Math.floor(i/2)]*30000);
					}
					i2=0

					//c.push(v)
				//b= new Int16Array(v)
					console.log(c.length,v)
					socket.send(v)
					//if (c.length>10){socket.send(c);c=[]}


                };           
            }).catch((error) => {

            });
        } else {

        }

</script>
</body>
</html>