code tidy up
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david
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113
src/server.ts
113
src/server.ts
@ -1,56 +1,26 @@
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// @ts-ignore
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import { nonstandard, RTCPeerConnection, MediaStreamTrack, MediaStream, RTCSessionDescription } from '@roamhq/wrtc';
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import { spawn, ChildProcessWithoutNullStreams } from 'child_process';
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import { Readable } from 'stream';
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import * as ws from 'ws'
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import * as fs from 'fs';
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import { i420ToRgba, RTCVideoSource } from '@roamhq/wrtc/types/nonstandard';
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// import { RTCVideoSource } from '@roamhq/wrtc/types/nonstandard';
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// type RTCVideoSource = nonstandard.RTCVideoSource;
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// const RTCVideoSource = wrtc.nonstandard.RTCVideoSource;
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// const {mediaDevices} = wrtc
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import { MediaStream, MediaStreamTrack, nonstandard, RTCPeerConnection } from '@roamhq/wrtc';
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import { ChildProcessWithoutNullStreams, spawn } from 'child_process';
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import * as ws from 'ws';
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// Constants
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// const VIDEO_DEVICE = '/dev/video0'; // Video source device
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const VIDEO_DEVICE = '/dev/dvb/adapter0/dvr0'; // Video source device
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const WIDTH = 640; // Video width
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const HEIGHT = 480; // Video height
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const FRAME_SIZE = WIDTH * HEIGHT * 1.5; // YUV420p frame size (460800 bytes)
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class VideoStream extends Readable {
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private device: fs.ReadStream;
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constructor(devicePath: string) {
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super();
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this.device = fs.createReadStream(devicePath);
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}
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_read(size: number): void {
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const chunk = this.device.read(size);
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if (chunk === null) {
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this.push(null); // Signal end of stream
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} else {
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this.push(chunk);
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}
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}
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}
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// Function to start FFmpeg and capture raw video
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function startFFmpeg(): ChildProcessWithoutNullStreams {
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const startFFmpeg = (): ChildProcessWithoutNullStreams => {
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const p = spawn('ffmpeg', [
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'-loglevel', 'debug',
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'-i', VIDEO_DEVICE, // Input device
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'-i', VIDEO_DEVICE,
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// Video
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'-map', '0:v:0',
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'-vf', `scale=${WIDTH}:${HEIGHT}`, // Scale video resolution
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'-vcodec', 'rawvideo', // Output raw video codec
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'-pix_fmt', 'yuv420p', // Pixel format for WebRTC
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'-f', 'rawvideo', // Output format
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'pipe:3', // Pipe to stdout
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'-vf', `scale=${WIDTH}:${HEIGHT}`,
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'-vcodec', 'rawvideo',
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'-pix_fmt', 'yuv420p',
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'-f', 'rawvideo',
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'pipe:3',
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// Audio
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'-map', '0:a:0',
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@ -62,12 +32,10 @@ function startFFmpeg(): ChildProcessWithoutNullStreams {
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], {
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stdio: ['ignore', 'pipe', 'pipe', 'pipe', 'pipe']
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// detached: true
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});
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process.on('SIGINT', () => {
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console.log('🔻 Server shutting down... KILLING');
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let b = p.kill('SIGINT');
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// let b = process.kill(p.pid)
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console.log('🔻 Server shutting down...');
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p.kill('SIGINT');
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process.exit(0);
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});
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@ -78,49 +46,33 @@ function startFFmpeg(): ChildProcessWithoutNullStreams {
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});
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process.on('exit', () => {
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p.kill('SIGHUP'); //this one
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let b = p.kill('SIGTERM');
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console.log("b ", b)
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p.kill('SIGTERM');
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});
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return p;
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}
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// const videoSource =
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let frameBuffer = Buffer.alloc(0);
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const ffmpegProcess = startFFmpeg();
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const videoSource = new nonstandard.RTCVideoSource();
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const audioSource = new nonstandard.RTCAudioSource();
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// Function to create a WebRTC PeerConnection
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async function createPeerConnection(): Promise<RTCPeerConnection> {
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const createPeerConnection = async (): Promise<RTCPeerConnection> => {
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const peerConnection = new RTCPeerConnection({ iceServers: [] });
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const videoStream = ffmpegProcess.stdio[3]; // pipe:3
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const audioStream = ffmpegProcess.stdio[4]; // pipe:4
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// Start FFmpeg and pipe video frames to the source
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videoStream.on('data', (chunk: Buffer) => {
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// Push video frames to the RTCVideoSource
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frameBuffer = Buffer.concat([frameBuffer, chunk]);
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while (frameBuffer.length >= FRAME_SIZE) {
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const frameData = frameBuffer.slice(0, FRAME_SIZE);
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frameBuffer = frameBuffer.slice(FRAME_SIZE); // Keep remaining data
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frameBuffer = frameBuffer.slice(FRAME_SIZE);
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const frame: nonstandard.RTCVideoFrame = {
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width: WIDTH,
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height: HEIGHT,
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data: new Uint8Array(frameData),
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}
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videoSource.onFrame(frame);
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}
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});
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@ -129,8 +81,6 @@ async function createPeerConnection(): Promise<RTCPeerConnection> {
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// console.error('FFmpeg Error:', data.toString());
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});
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videoStream.on('exit', (code) => {
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console.log(`FFmpeg exited with code ${code}`);
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});
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@ -143,16 +93,13 @@ async function createPeerConnection(): Promise<RTCPeerConnection> {
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audioBuffer = Buffer.concat([audioBuffer, chunk]);
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while (audioBuffer.length >= AUDIO_FRAME_SIZE) {
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const frameData = audioBuffer.slice(0, AUDIO_FRAME_SIZE);
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// const sampleBuffer = Buffer.from(frameData.buffer.; // makes an isolated buffer
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audioBuffer = audioBuffer.slice(AUDIO_FRAME_SIZE);
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const samples = new Int16Array(480);
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for (let i = 0; i < 480; i++) {
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samples[i] = frameData.readInt16LE(i * 2);
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}
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audioSource.onData({
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samples: samples,
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samples,
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sampleRate: 48000,
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bitsPerSample: 16,
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channelCount: 1,
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@ -165,26 +112,18 @@ async function createPeerConnection(): Promise<RTCPeerConnection> {
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// console.error('FFmpeg Error:', data.toString());
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});
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audioStream.on('exit', (code) => {
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console.log(`FFmpeg exited with code ${code}`);
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});
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// Add the track to the PeerConnection
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const track: MediaStreamTrack = videoSource.createTrack();
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const track1 = audioSource.createTrack();
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console.log('vdei src ', videoSource.isScreencast)
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const videoTrack: MediaStreamTrack = videoSource.createTrack();
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const audioTrack: MediaStreamTrack = audioSource.createTrack();
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const stream = new MediaStream()
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stream.addTrack(track)
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stream.addTrack(track1);
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console.log('enabled ', track.enabled, track.id, track.kind, track.label, track.readyState);
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// track.
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console.log('get', stream.getVideoTracks()[0].id)
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peerConnection.addTrack(track, stream);
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peerConnection.addTrack(track1, stream);
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stream.addTrack(videoTrack)
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stream.addTrack(audioTrack);
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peerConnection.addTrack(videoTrack, stream);
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peerConnection.addTrack(audioTrack, stream);
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return peerConnection;
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}
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@ -195,12 +134,9 @@ const wss = new ws.WebSocketServer({ port: 8080 });
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wss.on('connection', async (ws: ws.WebSocket) => {
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const peerConnection: RTCPeerConnection = await createPeerConnection();
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// const source = new RTCVideoSource();
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console.log('Client connected');
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ws.on('message', async (message: Buffer) => {
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const { type, data } = JSON.parse(message.toString());
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console.log("message type", type)
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if (type == 'offer') {
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await peerConnection.setRemoteDescription(data);
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@ -210,14 +146,12 @@ wss.on('connection', async (ws: ws.WebSocket) => {
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}
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if (type === 'ice-candidate') {
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console.log('type ice')
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await peerConnection.addIceCandidate(data);
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}
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});
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peerConnection.oniceconnectionstatechange = () => {
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console.log('ICE connection state:', peerConnection.iceConnectionState);
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if (peerConnection.iceConnectionState === 'failed') {
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console.error('ICE connection failed');
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}
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@ -226,7 +160,6 @@ wss.on('connection', async (ws: ws.WebSocket) => {
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// Send ICE candidates to the client
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peerConnection.onicecandidate = ({ candidate }) => {
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console.log("onicecandidate")
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if (candidate) {
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ws.send(JSON.stringify({ type: 'ice-candidate', data: candidate }));
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}
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@ -28,11 +28,9 @@
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pc.ontrack = (event) => {
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console.log("Received track event", event.streams);
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video.srcObject = event.streams[0];
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// video.muted = false;
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};
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pc.onicecandidate = ({ candidate }) => {
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// console.log("pc.onicecandidate")
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if (candidate) {
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ws.send(JSON.stringify({ type: 'ice-candidate', data: candidate })); // Use 'candidate' instead of 'ice-candidate'
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}
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@ -46,15 +44,11 @@
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pc.addTransceiver('audio', { direction: 'recvonly' })
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const offer = await pc.createOffer();
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await pc.setLocalDescription(offer);
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// ws.send(JSON.stringify(offer));
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console.log("on open ")
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// ws.send(JSON.stringify(offer));
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ws.send(JSON.stringify({ type: 'offer', data: offer }));
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}
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ws.onmessage = async (message) => {
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const msg = JSON.parse(message.data);
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console.log("onmessage type:", msg.type, msg)
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if (msg.type === 'answer') {
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await pc.setRemoteDescription(msg.data);
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