Live RTMP broadcasting

5 files changed, 231 insertions, 6 deletions

diff --git a/README.md b/README.md
index a53c77b..cb11e46 100644
--- a/README.md
+++ b/README.md
@@ -46,6 +46,26 @@ You can also use `liveJoin.js` to live join meeting as a recorder & perform reco
 node liveJoin.js "https://BBB_HOST/bigbluebutton/api/join?meetingId=MEETING_ID...." liveRecord.webm 10 true
 ```
 
+**Live RTMP broadcasting.(Experimental)**
+
+Sometime you may want to broadcast meeting via RTMP. I did some experiment on it & got success but not 100%. To test you can use `ffmpegServer.js` to run websocket server & `liveRTMP.js` to join the meeting. You'll have to edit `rtmpUrl` from `ffmpegServer.js` & port (if need). Then add websocket value as `ffmpegServer` in `liveRTMP.js` file.
+
+1) Frist run websocket server by `node ffmpegServer.js`
+2) Then in another terminal tab
+
+```sh
+node liveJoin.js "https://BBB_HOST/bigbluebutton/api/join?meetingId=MEETING_ID...."
+```
+You can also set duration like this too otherwise it will close after meeting end or kickout:
+
+```sh
+node liveJoin.js "https://BBB_HOST/bigbluebutton/api/join?meetingId=MEETING_ID...." 20
+```
+
+Check the process of websocket server, `ffmpeg` should start sending data to RTMP server.
+
+**Note:**
+If you do nothing in meeting room that time `ffmpeg` may exit with error. Actually I don't have much experience on `ffmpeg` to resolve those problems. Please contribute your experience.
 
 ### How it will work?
 When you will run the command that time `chromium` browser will be open in background & visit the link & perform screen recording. So, if you have set 10 seconds then it will record 10 seconds only. Later it will give you file as webm or mp4.
@@ -53,9 +73,13 @@ When you will run the command that time `chromium` browser will be open in backg
 **Note: It will use extra CPU to process chrome & ffmpeg.** 
 
 
+### Looking for Bigbluebutton shared hosting?
+
+We are offering cheaper [Bigbluebutton shared hosting](https://www.mynaparrot.com/classroom) or Bigbluebutton insallation/configuration service. You can send me email jibon[@]mynaparrot.com
+
 
-Thanks to [@muralikg](https://github.com/muralikg/puppetcam). Everything was copied from there & I did some adjustment. 
+### Thanks to
 
+[puppetcam](https://github.com/muralikg/puppetcam). Most of the parts were copied from there. 
 
-### Looking for Bigbluebutton shared hosting?
-We are offering cheaper [Bigbluebutton shared hosting](https://www.mynaparrot.com/classroom) or Bigbluebutton insallation/configuration service. You can send me email jibon[@]mynaparrot.com
+[Canvas-Streaming-Example](https://github.com/fbsamples/Canvas-Streaming-Example)
\ No newline at end of file
diff --git a/background.js b/background.js
index cb2fb8b..253e81e 100644
--- a/background.js
+++ b/background.js
@@ -2,6 +2,8 @@
 
 let recorder = null;
 let filename = null;
+let ws;
+let liveSteam = false;
 chrome.runtime.onConnect.addListener(port => {
 
   port.onMessage.addListener(msg => {
@@ -10,11 +12,20 @@ chrome.runtime.onConnect.addListener(port => {
       case 'SET_EXPORT_PATH':
         filename = msg.filename
         break
+      case 'FFMPEG_SERVER':
+        let ffmpegServer = msg.ffmpegServer
+        ws = new WebSocket(ffmpegServer);
+        liveSteam = true;
+        break
       case 'REC_STOP':
         recorder.stop()    
         break
       case 'REC_START':
-        recorder.start();
+        if(liveSteam){
+          recorder.start(1000);
+        }else{
+          recorder.start();
+        }
         break
       case 'REC_CLIENT_PLAY':
         if(recorder){
@@ -42,7 +53,7 @@ chrome.runtime.onConnect.addListener(port => {
               }
             }
           }, stream => {
-            var chunks=[];
+            var chunks = [];
             recorder = new MediaRecorder(stream, {
                 videoBitsPerSecond: 2500000,
                 ignoreMutedMedia: true,
@@ -51,10 +62,17 @@ chrome.runtime.onConnect.addListener(port => {
             recorder.ondataavailable = function (event) {
                 if (event.data.size > 0) {
                     chunks.push(event.data);
+                    if(liveSteam){
+                      ws.send(event.data);
+                    }
                 }
             };
 
             recorder.onstop = function () {
+                if(liveSteam){
+                  ws.close();
+                }
+
                 var superBuffer = new Blob(chunks, {
                     type: 'video/webm'
                 });
diff --git a/ffmpegServer.js b/ffmpegServer.js
new file mode 100644
index 0000000..0ab0e6a
--- /dev/null
+++ b/ffmpegServer.js
@@ -0,0 +1,83 @@
+const child_process = require('child_process');
+const WebSocketServer = require('ws').Server;
+const http = require('http');
+
+const server = http.createServer().listen(4000, () => {
+  console.log('Listening...');
+});
+
+const wss = new WebSocketServer({
+	server: server
+});
+
+
+const rtmpUrl = "rtmp://a.rtmp.youtube.com/live2/MyKey";
+
+wss.on('connection', function connection(ws, req) {
+	console.log('connection');
+
+	let auth;
+
+	if ( !(auth = req.url.match(/^\/auth\/(.*)$/)) ) {
+		ws.terminate();
+		return;
+	}
+
+	if(auth[1] !== "mZFZN4yc"){
+		ws.terminate();
+		return;
+	}
+
+	const ffmpeg = child_process.spawn('ffmpeg', [
+
+		// FFmpeg will read input video from STDIN
+    	'-i', '-',
+
+    	// Chromium doesn't support H.264, set the video codec to 'libx264'
+	    // or similar to transcode it to H.264 here on the server.
+	    '-vcodec', 'libx264',
+
+	    //No browser currently supports encoding AAC, so we must transcode the audio to AAC here on the server.
+	    '-acodec', 'aac',
+	    
+	    // FLV is the container format used in conjunction with RTMP
+	    '-f', 'flv',
+
+	    '-max_muxing_queue_size', '99999',
+
+	    //'-vf', 'mpdecimate', '-vsync', 'vfr',
+	    //'-vf', 'mpdecimate,setpts=N/FRAME_RATE/TB',
+	    
+	    // The output RTMP URL.
+	    // For debugging, you could set this to a filename like 'test.flv', and play
+	    // the resulting file with VLC.
+	    rtmpUrl 
+	])
+
+	// If FFmpeg stops for any reason, close the WebSocket connection.
+	ffmpeg.on('close', (code, signal) => {
+		console.log('FFmpeg child process closed, code ' + code + ', signal ' + signal);
+		ws.terminate();
+	});
+
+	ffmpeg.stdin.on('error', (e) => {
+		console.log('FFmpeg STDIN Error', e);
+	});
+
+	ffmpeg.stderr.on('data', (data) => {
+    	console.log('FFmpeg STDERR:', data.toString());
+    });
+
+    // When data comes in from the WebSocket, write it to FFmpeg's STDIN.
+	ws.on('message', (msg) => {
+	    console.log('DATA', msg);
+	    ffmpeg.stdin.write(msg);
+	});
+
+	// If the client disconnects, stop FFmpeg.
+	ws.on('close', (e) => {
+    	ffmpeg.kill('SIGINT');
+    	ws.terminate();
+	});
+
+});
\ No newline at end of file
diff --git a/liveRTMP.js b/liveRTMP.js
new file mode 100644
index 0000000..0fcbfef
--- /dev/null
+++ b/liveRTMP.js
@@ -0,0 +1,99 @@
+const puppeteer = require('puppeteer');
+const Xvfb      = require('xvfb');
+var exec = require('child_process').exec;
+const fs = require('fs');
+const homedir = require('os').homedir();
+const ffmpegServer = "ws://localhost:4000/auth/mZFZN4yc";
+
+var xvfb        = new Xvfb({silent: true});
+var width       = 1280;
+var height      = 720;
+var options     = {
+  headless: false,
+  args: [
+    '--enable-usermedia-screen-capturing',
+    '--allow-http-screen-capture',
+    '--auto-select-desktop-capture-source=bbbrecorder',
+    '--load-extension=' + __dirname,
+    '--disable-extensions-except=' + __dirname,
+    '--disable-infobars',
+    '--no-sandbox',    
+    '--shm-size=1gb',
+    '--disable-dev-shm-usage',
+    `--window-size=${width},${height}`,
+  ],
+}
+
+async function main() {
+    try{
+        xvfb.startSync()
+        var url = process.argv[2],
+            duration = process.argv[3], 
+            exportname = 'liveMeeting.webm'
+
+        if(!url){ url = 'http://tobiasahlin.com/spinkit/' }
+        //if(!duration){ duration = 10 }
+        
+        const browser = await puppeteer.launch(options)
+        const pages = await browser.pages()
+        
+        const page = pages[0]
+
+        page.on('console', msg => {
+            var m = msg.text();
+            console.log('PAGE LOG:', m)
+        });
+
+        await page._client.send('Emulation.clearDeviceMetricsOverride')
+        await page.goto(url, {waitUntil: 'networkidle2'})
+        await page.setBypassCSP(true)
+
+        await page.evaluate((serverAddress) => {
+            console.log("FFMPEG_SERVER");
+            window.postMessage({type: 'FFMPEG_SERVER', ffmpegServer: serverAddress}, '*')
+        }, ffmpegServer)
+
+        await page.waitForSelector('[aria-label="Listen only"]');
+        await page.click('[aria-label="Listen only"]', {waitUntil: 'domcontentloaded'});
+
+        await page.waitForSelector('[id="chat-toggle-button"]');
+        await page.click('[id="chat-toggle-button"]', {waitUntil: 'domcontentloaded'});
+        await page.click('button[aria-label="Users and messages toggle"]', {waitUntil: 'domcontentloaded'});
+        await page.$eval('[class^=navbar]', element => element.style.display = "none");
+
+        await page.$eval('.Toastify', element => element.style.display = "none");
+        await page.waitForSelector('button[aria-label="Leave audio"]');
+        await page.$eval('[class^=actionsbar] > [class^=center]', element => element.style.display = "none");
+        
+        await page.evaluate((x) => {
+            console.log("REC_START");
+            window.postMessage({type: 'REC_START'}, '*')
+        })
+
+        if(duration > 0){
+            await page.waitFor((duration * 1000))
+        }else{
+            await page.waitForSelector('[class^=modal] > [class^=content] > button[description="Logs you out of the meeting"]', {
+                timeout: 0
+            });
+        }
+
+        await page.evaluate(filename=>{
+            window.postMessage({type: 'SET_EXPORT_PATH', filename: filename}, '*')
+            window.postMessage({type: 'REC_STOP'}, '*')
+        }, exportname)
+
+        // Wait for download of webm to complete
+        await page.waitForSelector('html.downloadComplete', {timeout: 0})
+        await page.close()
+        await browser.close()
+        xvfb.stopSync()
+
+        fs.unlinkSync(homedir + "/Downloads/liveMeeting.webm");
+        
+    }catch(err) {
+        console.log(err)
+    }
+}
+
+main()
diff --git a/package.json b/package.json
index fecd5b0..32de7e1 100644
--- a/package.json
+++ b/package.json
@@ -13,6 +13,7 @@
   ],
   "dependencies": {
     "puppeteer": "2.0.0",
-    "xvfb": "^0.2.3"
+    "xvfb": "^0.2.3",
+    "ws": "^7.2.3"
   }
 }