Question

我的目标是拥有一个带有视频流的Web应用程序。经过一番研究，我找到了WebRTC并能够运行第一个原型。我查看过的代码可以在与this blog匹配的this Repo中找到。

如果我理解正确，那么我在这里使用对等通信。因此，在一个选项卡中，将要求用户授予访问默认摄像机的权限。此视频流将在我的实际网站中显示。到目前为止，它工作得很好，而且没有时间延迟，这是很好的（低延迟是关键因素）。

这是服务器：

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}`));

这是发布者：

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);
}

下一步，我想使用gstreamer代替摄像机。但是，由于我找不到很多信息，我仍然不清楚为此需要更改/添加什么。

-> 我是否需要像janus这样的服务器？还是可以使用 gstreamer流就像网络摄像头一样作为navigator.mediadevice？

如果您需要更多详细信息，我们将很高兴添加缺少的信息。现在，WebRTC对我来说似乎是一个困难的概念，所以请问这是一个非常基本的问题。

一个可能有用的信息：对于该网站，我正在使用vue.js / javascript以及通过websocket通信的flask服务器。对于使用WebRTC进行的测试，我还拥有上面存储库（还有节点服务器）中的所有内容。

提前谢谢！

通过WebRTC将视频流传输到网站（网络摄像头/ Gestreamer）

0 个答案: