将Webrtc跟踪流转换为视频标签中的URL(RTSP/UDP/RTP/Http)

Question

我是WebRTC的新手，我已经完成了客户端/服务器的连接，从客户端我选择WebCam，并使用Track将流发布到服务器，在服务器端我获得该轨迹并将轨迹流分配给视频源。到目前为止一切都很好，但问题是现在我包括了AI(人工智能(，现在我想将我的曲目流转换为URL，可能是UDP/RTSP/RTP等。所以AI将使用该URL进行对象检测。我不知道如何将音轨流转换为URL。尽管有几个包https://ffmpeg.org/和RTP到Webrtc等，我使用Nodejs、Socket.io和Webrtc，下面你可以检查我的客户端和服务器端代码，以获取和发布流，我遵循github代码https://github.com/Basscord/webrtc-video-broadcast.现在我主要关心的是将track作为视频标签的URL，这是可能的还是不可能的，或者请提出建议，任何帮助都将不胜感激。

Server.js

这是nodejs服务器代码

const express = require("express");
const app = express();
let broadcaster;
const port = 4000;
const http = require("http");
const server = http.createServer(app);
const io = require("socket.io")(server);
app.use(express.static(__dirname + "/public"));
io.sockets.on("error", e => console.log(e));
io.sockets.on("connection", socket => {
socket.on("broadcaster", () => {
broadcaster = socket.id;
socket.broadcast.emit("broadcaster");
});
socket.on("watcher", () => {
socket.to(broadcaster).emit("watcher", socket.id);
});
socket.on("offer", (id, message) => {
socket.to(id).emit("offer", socket.id, message);
});
socket.on("answer", (id, message) => {
socket.to(id).emit("answer", socket.id, message);
});
socket.on("candidate", (id, message) => {
socket.to(id).emit("candidate", socket.id, message);
});
socket.on("disconnect", () => {
socket.to(broadcaster).emit("disconnectPeer", socket.id);
});
});
server.listen(port, () => console.log(`Server is running on port ${port}`));

Broadcast.js这是发射流(track(的代码

const peerConnections = {};
const config = {
iceServers: [
{
urls: ["stun:stun.l.google.com:19302"]
}
]
};
const socket = io.connect(window.location.origin);
socket.on("answer", (id, description) => {
peerConnections[id].setRemoteDescription(description);
});
socket.on("watcher", id => {
const peerConnection = new RTCPeerConnection(config);
peerConnections[id] = peerConnection;
let stream = videoElement.srcObject;
stream.getTracks().forEach(track => peerConnection.addTrack(track, stream));
peerConnection.onicecandidate = event => {
if (event.candidate) {
socket.emit("candidate", id, event.candidate);
}
};
peerConnection
.createOffer()
.then(sdp => peerConnection.setLocalDescription(sdp))
.then(() => {
socket.emit("offer", id, peerConnection.localDescription);
});
});
socket.on("candidate", (id, candidate) => {
peerConnections[id].addIceCandidate(new RTCIceCandidate(candidate));
});
socket.on("disconnectPeer", id => {
peerConnections[id].close();
delete peerConnections[id];
});
window.onunload = window.onbeforeunload = () => {
socket.close();
};
// Get camera and microphone
const videoElement = document.querySelector("video");
const audioSelect = document.querySelector("select#audioSource");
const videoSelect = document.querySelector("select#videoSource");
audioSelect.onchange = getStream;
videoSelect.onchange = getStream;
getStream()
.then(getDevices)
.then(gotDevices);
function getDevices() {
return navigator.mediaDevices.enumerateDevices();
}
function gotDevices(deviceInfos) {
window.deviceInfos = deviceInfos;
for (const deviceInfo of deviceInfos) {
const option = document.createElement("option");
option.value = deviceInfo.deviceId;
if (deviceInfo.kind === "audioinput") {
option.text = deviceInfo.label || `Microphone ${audioSelect.length + 1}`;
audioSelect.appendChild(option);
} else if (deviceInfo.kind === "videoinput") {
option.text = deviceInfo.label || `Camera ${videoSelect.length + 1}`;
videoSelect.appendChild(option);
}
}
}
function getStream() {
if (window.stream) {
window.stream.getTracks().forEach(track => {
track.stop();
});
}
const audioSource = audioSelect.value;
const videoSource = videoSelect.value;
const constraints = {
audio: { deviceId: audioSource ? { exact: audioSource } : undefined },
video: { deviceId: videoSource ? { exact: videoSource } : undefined }
};
return navigator.mediaDevices
.getUserMedia(constraints)
.then(gotStream)
.catch(handleError);
}
function gotStream(stream) {
window.stream = stream;
audioSelect.selectedIndex = [...audioSelect.options].findIndex(
option => option.text === stream.getAudioTracks()[0].label
);
videoSelect.selectedIndex = [...videoSelect.options].findIndex(
option => option.text === stream.getVideoTracks()[0].label
);
videoElement.srcObject = stream;
socket.emit("broadcaster");
}
function handleError(error) {
console.error("Error: ", error);
}

RemoteServer.js此代码正在跟踪并分配给视频标记

let peerConnection;
const config = {
iceServers: [
{
urls: ["stun:stun.l.google.com:19302"]
}
]
};
const socket = io.connect(window.location.origin);
const video = document.querySelector("video");
socket.on("offer", (id, description) => {
peerConnection = new RTCPeerConnection(config);
peerConnection
.setRemoteDescription(description)
.then(() => peerConnection.createAnswer())
.then(sdp => peerConnection.setLocalDescription(sdp))
.then(() => {
socket.emit("answer", id, peerConnection.localDescription);
});
peerConnection.ontrack = event => {
video.srcObject = event.streams[0];
};
peerConnection.onicecandidate = event => {
if (event.candidate) {
socket.emit("candidate", id, event.candidate);
}
};
});
socket.on("candidate", (id, candidate) => {
peerConnection
.addIceCandidate(new RTCIceCandidate(candidate))
.catch(e => console.error(e));
});
socket.on("connect", () => {
socket.emit("watcher");
});
socket.on("broadcaster", () => {
socket.emit("watcher");
});
socket.on("disconnectPeer", () => {
peerConnection.close();
});
window.onunload = window.onbeforeunload = () => {
socket.close();
};

Answer 1

rtp-webrtc正是您想要的。

不幸的是，你需要运行某种服务器才能实现这一点，它不可能全部在浏览器中。如果你不想使用WebRTC，你也可以通过其他协议上传(通过MediaRecorder捕获(。