树莓派建立webRTC视频传输

最新推荐文章于 2024-09-03 20:21:00 发布

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最新推荐文章于 2024-09-03 20:21:00 发布

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文章标签： webrtc 音视频

本文链接：https://blog.csdn.net/weixin_46084533/article/details/140850439

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要将树莓派上的摄像头流通过 WebRTC 传输到 HTML 页面中，你需要调整你的WebRTC信令服务器和树莓派客户端代码。以下是你需要做的步骤：

1. 修改树莓派客户端

在树莓派的客户端代码中，你需要将摄像头流推送到 WebRTC 的 RTCPeerConnection 中。以下是更新后的代码示例：

const io = require('socket.io-client');
const { exec } = require('child_process');

const socket = io('http://<你的公网服务器IP>:3000');

// WebRTC 连接
let peerConnection = new RTCPeerConnection({
    iceServers: [{ urls: 'stun:stun.l.google.com:19302' }]
});

// 处理 ICE 候选
socket.on('candidate', (candidate) => {
    peerConnection.addIceCandidate(new RTCIceCandidate(candidate));
});

// 开始摄像头流
function startCameraStream() {
    const ffmpeg = exec(`raspivid -o - -t 0 -vf -hf -n | ffmpeg -re -i pipe:0 -c:v copy -f mpegts udp://localhost:1234`);

    ffmpeg.stdout.on('data', (data) => {
        // 将视频流添加到 peerConnection
        let stream = new MediaStream();
        let track = new MediaStreamTrack(data);
        stream.addTrack(track);
        peerConnection.addTrack(track);
    });
}

// 启动摄像头流
startCameraStream();

// 处理 offer
socket.on('offer', async (offer) => {
    await peerConnection.setRemoteDescription(new RTCSessionDescription(offer));
    const answer = await peerConnection.createAnswer();
    await peerConnection.setLocalDescription(answer);
    socket.emit('answer', answer);
});

// 处理 answer
socket.on('answer', (answer) => {
    peerConnection.setRemoteDescription(new RTCSessionDescription(answer));
});

2. 修改 HTML 页面

在 HTML 页面中，你需要确保 WebRTC 连接正常，并可以接收到视频流。下面是更新后的 HTML 示例：

<!DOCTYPE html>
<html lang="en">
<head>
    <meta charset="UTF-8">
    <meta name="viewport" content="width=device-width, initial-scale=1.0">
    <title>WebRTC Camera Stream</title>
</head>
<body>
    <h1>WebRTC Camera Stream</h1>
    <video id="video" autoplay playsinline style="width: 100%; max-width: 800px;"></video>
    <script src="/socket.io/socket.io.js"></script>
    <script>
        const video = document.getElementById('video');
        const socket = io();

        const peerConnection = new RTCPeerConnection({
            iceServers: [{ urls: 'stun:stun.l.google.com:19302' }]
        });

        peerConnection.onicecandidate = (event) => {
            if (event.candidate) {
                socket.emit('candidate', event.candidate);
            }
        };

        peerConnection.ontrack = (event) => {
            video.srcObject = event.streams[0];
        };

        async function startStream() {
            const offer = await peerConnection.createOffer();
            await peerConnection.setLocalDescription(offer);
            socket.emit('offer', offer);
        }

        socket.on('offer', async (offer) => {
            await peerConnection.setRemoteDescription(new RTCSessionDescription(offer));
            const answer = await peerConnection.createAnswer();
            await peerConnection.setLocalDescription(answer);
            socket.emit('answer', answer);
        });

        socket.on('answer', (answer) => {
            peerConnection.setRemoteDescription(new RTCSessionDescription(answer));
        });

        socket.on('candidate', (candidate) => {
            peerConnection.addIceCandidate(new RTCIceCandidate(candidate));
        });

        // 启动视频流
        startStream();
    </script>
</body>
</html>