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• What is the best way to capture a screenshot from a udp stream ?

  8 novembre 2017, par yash17

  I’m trying to capture a screenshot from a udp stream using ffmpeg in a Ubuntu 14.04 System.
  Following is the command

  ffmpeg -y -i udp_ip -vframes 1 -q:v 1 test.png

  But the image captured is of very poor resolution and I observed a lag while taking the screenshot.

  Please suggest a best tool or a way to take a screenshot in the fastest way and also of the best image resolution possible.

  edit :
  log files

  ffmpeg version 3.3.2 Copyright (c) 2000-2017 the FFmpeg developers
  
    built with gcc 4.8 (Ubuntu 4.8.4-2ubuntu1~14.04.3)
  
    configuration: --extra-libs=-ldl --prefix=/opt/ffmpeg --mandir=/usr/share/man --enable-avresample --disable-debug --enable-nonfree --enable-gpl --enable-version3 --enable-libopencore-amrnb --enable-libopencore-amrwb --disable-decoder=amrnb --disable-decoder=amrwb --enable-libpulse --enable-libfreetype --enable-gnutls --enable-libx264 --enable-libx265 --enable-libfdk-aac --enable-libvorbis --enable-libtheora --enable-libmp3lame --enable-libopus --enable-libvpx --enable-libspeex --enable-libass --enable-avisynth --enable-libsoxr --enable-libxvid --enable-libvidstab --enable-libwavpack --enable-nvenc
  
    libavutil      55. 58.100 / 55. 58.100
  
    libavcodec     57. 89.100 / 57. 89.100
  
    libavformat    57. 71.100 / 57. 71.100
  
    libavdevice    57.  6.100 / 57.  6.100
  
    libavfilter     6. 82.100 /  6. 82.100
  
    libavresample   3.  5.  0 /  3.  5.  0
  
    libswscale      4.  6.100 /  4.  6.100
  
    libswresample   2.  7.100 /  2.  7.100
  
    libpostproc    54.  5.100 / 54.  5.100
  
  [mpeg2video @ 0x390cc60] Invalid frame dimensions 0x0.
  
      Last message repeated 7 times
  
  Input #0, mpegts, from 'udp://@xxx.xx.xx.xx:xxxx':
  
    Duration: N/A, start: 144.130744, bitrate: 4128 kb/s
  
    Program 1 
  
      Metadata:
  
        service_name    : Program-1   
  
        service_provider: Encoder 
  
      Stream #0:0[0x42]: Video: mpeg2video (Main) ([2][0][0][0] / 0x0002), yuv420p(tv, top first), 720x576 [SAR 16:15 DAR 4:3], 4000 kb/s, 25 fps, 25 tbr, 90k tbn, 50 tbc
  
      Stream #0:1[0x43]: Audio: mp2 ([3][0][0][0] / 0x0003), 48000 Hz, stereo, s16p, 128 kb/s
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (mpeg2video (native) -> png (native))
  
  Press [q] to stop, [?] for help
  
  Output #0, image2, to 'player.png':
  
    Metadata:
  
      encoder         : Lavf57.71.100
  
      Stream #0:0: Video: png, rgb24, 720x576 [SAR 16:15 DAR 4:3], q=2-31, 200 kb/s, 25 fps, 25 tbn, 25 tbc
  
      Metadata:
  
        encoder         : Lavc57.89.100 png
  
  frame=    1 fps=0.0 q=-0.0 Lsize=N/A time=00:00:00.04 bitrate=N/A dup=1 drop=1 speed=0.729x    
  
  video:777kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown

• 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.

• FFmpeg : Multiple x265-params are not recognized

  13 septembre 2022, par zinon

  I'm using ffmpeg in x265 and I want to use multiple x265-params in one encoding.

  



  When I use more than one parameter, ffmpeg does not recognize them.

  



  My script is :

  



  ffmpeg -s:v 1440x1080 -r 25 -i incident_10d_1440x1080_25.yuv -c:v rawvideo \
-pix_fmt yuv420p -c:v libx265 -x265-params "--qp=16:--preset=medium:--psnr" \
out_1440x1080_qp16.mp4


  



  I set quantization parameter value equal to 16.

  



  But my output in terminal contains the following :

  



  x265 [info]: Main profile, Level-4 (Main tier)
x265 [info]: Thread pool created using 4 threads
x265 [info]: Slices                              : 1
x265 [info]: frame threads / pool features       : 2 / wpp(17 rows)
x265 [info]: Coding QT: max CU size, min CU size : 64 / 8
x265 [info]: Residual QT: max TU size, max depth : 32 / 1 inter / 1 intra
x265 [info]: ME / range / subpel / merge         : hex / 57 / 2 / 2
x265 [info]: Keyframe min / max / scenecut / bias: 25 / 250 / 40 / 5.00
x265 [info]: Lookahead / bframes / badapt        : 20 / 4 / 2
x265 [info]: b-pyramid / weightp / weightb       : 1 / 1 / 0
x265 [info]: References / ref-limit  cu / depth  : 3 / on / on
x265 [info]: AQ: mode / str / qg-size / cu-tree  : 1 / 1.0 / 32 / 1
x265 [info]: Rate Control / qCompress            : CRF-28.0 / 0.60


  



  As can be seen I get Rate Control / qCompress : CRF-28.0 / 0.60.

  



  The correct one must be x265 [info]: Rate Control : CQP-16.

  



  When I have only this parameter in x265-params like -x265-params "--qp=16" it's working properly.