ffmpeg - 将 Webrtc 轨道流转换为 Video 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，在服务器端我正在获取该轨道并将轨道流分配给视频源。到目前为止一切都很好，但问题是现在我包括 AI(人工智能)，现在我想将我的跟踪流转换为 URL，可能是 UDP/RTSP/RTP 等。所以 AI 将使用该 URL 进行对象检测。我不知道我们如何将轨道流转换为 URL。
虽然有几个像 https://ffmpeg.org/ 这样的包和 RTP 到 Webrtc 等，我正在使用 Nodejs、Socket.io 和 Webrtc，您可以在下面查看我的客户端和服务器端代码以获取和发布流，我正在关注 github 代码 https://github.com/Basscord/webrtc-video-broadcast .
现在我主要关心的是跟踪作为视频标签的 URL，是否可能或请建议，任何帮助将不胜感激。
服务器.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}`));

广播.js
这是发射流(轨道)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);
}

远程服务器.js
此代码正在跟踪并分配给视频标签

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-to-webrtc做你想要的。
不幸的是，您需要运行某种服务器来实现这一点，它不能全部在浏览器中。如果您不想使用 WebRTC，也可以通过其他协议(protocol)(通过 MediaRecorder 捕获)上传。