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• Where to (long time) host Spring Boot Application with Data Base Backup and Linux Root Access [closed]

  22 mai 2024, par Lord Helmchen

  I developed a small application for my father. It uses Spring Boot, MySQL and FFMPEG, which I currently installed on Linux.

  


  I want to host it, deploy it automatically, have a back up and root access for FFMPEG installation.

  


  It runs smoothly locally on Windows / Linux, now I want to host it somewhere.

  


  What I would like to have :

  


  

  • Ease of deployment : I got experience in adminstration of linux root servers, but I look for something easy to integrate and maybe automatically deploy it from Github or Gitlab

  


  • Backup : I want to backup the database ideally to another service provider in case something goes wrong.

  


  • Linux : One Part of it, amongs others is to convert different audio formats using ffmpeg.
So, (I think) I need linux root access as well.

  


  • Time Horzion : I would like to make sure it still runs in ten+ years, so it should be a reliable provider where I only update the application from time to time if needed.

  


  • Money : As it is only for personal use at this moment, I don't want to invest a fortune.

  


  


  What provider and deployment pipeline would you recommend to me ?

  


• Loading and unloading C jni library based on when it's needed

  30 septembre 2014, par Alin

  I finally managed to compile ffmpeg for android and I’ve been able to use it in my app.

  Here is the scenario of my app :

  • I show the user a gridview with thumbnails of images and videos
  • the user can click on a cell and it is taken to image/video details where he can see the full image or play the video
  • the user can apply an image over an video and this is when ffmpeg is used

  So basically, the user might never actually use the watermarking option or he can do it very rare because the amount of available videos is way smaller than images.

  I am loading the ffmpeg library, first time it is needed by running :

  static {
  
       System.loadLibrary("ffmpeglib");
  
  }

  Now here are my questions :

  1. loading the library like this, uses app’s memory and resources ?
  2. can I unload the library, or better said, is it needed to unload it ? I have not found any java code like System.unloadLibrary to take care of unloading
  3. Since the library might be used rarely, wouldn’t a load => do encoding => unload be a better approach ? Or maybe having it loaded would allow easy reuse since no loading is necessary.
  4. If I use an IntentService to load the library and make the encoding, when the service completes the job, does the library gets unloaded ?

• Live audio using ffmpeg, javascript and nodejs

  8 novembre 2017, par klaus

  I am new to this thing. Please don’t hang me for the poor grammar. I am trying to create a proof of concept application which I will later extend. It does the following : We have a html page which asks for permission to use the microphone. We capture the microphone input and send it via websocket to a node js app.

  JS (Client) :

  var bufferSize = 4096;
  
  var socket = new WebSocket(URL);
  
  var myPCMProcessingNode = context.createScriptProcessor(bufferSize, 1, 1);
  
  myPCMProcessingNode.onaudioprocess = function(e) {
  
    var input = e.inputBuffer.getChannelData(0);
  
    socket.send(convertFloat32ToInt16(input));
  
  }
  
  
  
  function convertFloat32ToInt16(buffer) {
  
    l = buffer.length;
  
    buf = new Int16Array(l);
  
    while (l--) {
  
      buf[l] = Math.min(1, buffer[l])*0x7FFF;
  
    }
  
    return buf.buffer;
  
  }
  
  
  
  navigator.mediaDevices.getUserMedia({audio:true, video:false})
  
                                  .then(function(stream){
  
                                    var microphone = context.createMediaStreamSource(stream);
  
                                    microphone.connect(myPCMProcessingNode);
  
                                    myPCMProcessingNode.connect(context.destination);
  
                                  })
  
                                  .catch(function(e){});

  In the server we take each incoming buffer, run it through ffmpeg, and send what comes out of the std out to another device using the node js ’http’ POST. The device has a speaker. We are basically trying to create a 1 way audio link from the browser to the device.

  Node JS (Server) :

  var WebSocketServer = require('websocket').server;
  
  var http = require('http');
  
  var children = require('child_process');
  
  
  
  wsServer.on('request', function(request) {
  
    var connection = request.accept(null, request.origin);
  
    connection.on('message', function(message) {
  
      if (message.type === 'utf8') { /*NOP*/ }
  
      else if (message.type === 'binary') {
  
        ffm.stdin.write(message.binaryData);
  
      }
  
    });
  
    connection.on('close', function(reasonCode, description) {});
  
    connection.on('error', function(error) {});
  
  });
  
  
  
  var ffm = children.spawn(
  
      './ffmpeg.exe'
  
     ,'-stdin -f s16le -ar 48k -ac 2 -i pipe:0 -acodec pcm_u8 -ar 48000 -f aiff pipe:1'.split(' ')
  
  );
  
  
  
  ffm.on('exit',function(code,signal){});
  
  
  
  ffm.stdout.on('data', (data) => {
  
    req.write(data);
  
  });
  
  
  
  var options = {
  
    host: 'xxx.xxx.xxx.xxx',
  
    port: xxxx,
  
    path: '/path/to/service/on/device',
  
    method: 'POST',
  
    headers: {
  
     'Content-Type': 'application/octet-stream',
  
     'Content-Length': 0,
  
     'Authorization' : 'xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx',
  
     'Transfer-Encoding' : 'chunked',
  
     'Connection': 'keep-alive'
  
    }
  
  };
  
  
  
  var req = http.request(options, function(res) {});

  The device supports only continuous POST and only a couple of formats (ulaw, aiff, wav)

  This solution doesn’t seem to work. In the device speaker we only hear something like white noise.

  Also, I think I may have a problem with the buffer I am sending to the ffmpeg std in -> Tried to dump whatever comes out of the websocket to a .wav file then play it with VLC -> it plays everything in the record very fast -> 10 seconds of recording played in about 1 second.

  I am new to audio processing and have searched for about 3 days now for solutions on how to improve this and found nothing.

  I would ask from the community for 2 things :

  1. Is something wrong with my approach ? What more can I do to make this work ? I will post more details if required.

  2. If what I am doing is reinventing the wheel then I would like to know what other software / 3rd party service (like amazon or whatever) can accomplish the same thing.

  Thank you.