Recherche avancée

Sur d’autres sites (8209)

• Receiving UDP streams on my web server

  25 novembre 2013, par user3032143

  I have a Winform C# Desktop application.

  This is my overall aim :

  I have a constant stream of images from which I acquire using a VLC wrapper receiving a RTSP stream from my IP camera.

  I am doing image processing on these separate jpegs and at the same time I am wanting to upload these jpegs to my web server so my User can view these streaming jpegs live a video.

  Now, I have accomplished this so far by uploading each jpeg to my server using a [Web method]. But, I am trying to push my knowledge to make it more efficient.

  Now, i know if I use a video encoder - like OGG (used because it is Open Source) I can use ffmpeg called from my client code using the Process class so to convert images to that video format.

  Doing this saves a lot of memory when comparing that 1 ogg file to the separate bytes added up in total from all the individual jpegs.

  These are the arguments I pass to ffmpeg to achieve that :

  -f image2 -r 10 -i {location of jpegs}+"\img%05d.jpg -crf 23  -y -r 10 -f outputfile.ogg

  Now, I could take this a step further and not output to a physical file but instead to the base stream of the Process class. I would use these arguments to accomplish that :

  -f image2 -r 10 -i {location of jpegs}+"\img%05d.jpg -crf 23  -y -r 10 -f ogg -

  and in my code I would get the memory stream like so :

  mStandardOutput = serverBuild.StandardOutput.BaseStream;
  
  mStandardOutput.BeginRead(mReadBuffer, 0, mReadBuffer.Length, StandardOutputReadCallback, null);
  
  serverBuild.WaitForExit();
  
  data = mStandardOutputMs.ToArray();
  
  mStandardOutput.Close();

  Now ultimately, I would like to replace :

  -i {location of jpegs}+"\img%05d.jpg

  with a constant flow of jpegs in a memory stream like so :

  ffmpeg -f mjpeg -i - -r 10 -c:v libtheora -q:v 7 -f ogg -

  .. by over-writing the stdin...

  But I have not done this yet because I want to 1st try getting the ogg to be received within my web server.

  From there I would extract the jpegs to be accessible somehow via my web application written in asp.net 4.0.

  But first thing is first I want to just see if I can receive the UDP stream from my client.
  So, I create a test C# application to open and listen write from the client stream..
  this is my code :

  Socket socket = new Socket(AddressFamily.InterNetwork, SocketType.Dgram, ProtocolType.Udp);
  
  public Form1()
  
  {
  
     InitializeComponent();
  
  
  
     socket.SetSocketOption(SocketOptionLevel.Socket, SocketOptionName.ReuseAddress, true);
  
     string ip = "My Server IP Address";
  
     int port = 3000;
  
     socket.Bind(new IPEndPoint(IPAddress.Parse(ip), port));
  
  }
  
  
  
  private void button1_Click(object sender, EventArgs e)
  
  {
  
     try
  
     {
  
        var buffer = new byte[1024];
  
        while (true)
  
        {
  
           Application.DoEvents();
  
           int numBytesReceived = socket.Receive(buffer);
  
           if (numBytesReceived > 0)
  
           {
  
              File.WriteAllBytes("c:\\udp\\test.ogg", buffer);
  
           }
  
        }
  
     }
  
     catch (Exception _ex)
  
     {
  
        MessageBox.Show(_ex.ToString());
  
     }
  
  }

  But, I get this error :

  A message sent on a datagram socket was larger than the internal message buffer or some other network limit, or the buffer used to receive a datagram into was smaller than the datagram itself.

  What am I doing wrong please ?

  Thanks

• avcodec/vp9dsp : add DC only versions for idct/idct.

  22 novembre 2013, par Clément Bœsch

  avcodec/vp9dsp : add DC only versions for idct/idct.
  before :
  
    ./ffmpeg -v 0 -nostats -i /samples/vp9/etv.webm -f null -  16.29s user 0.02s system 99% cpu 16.323 total
  
    ./ffmpeg -v 0 -nostats -i /samples/vp9/etv.webm -f null -  16.32s user 0.01s system 99% cpu 16.351 total
  
    ./ffmpeg -v 0 -nostats -i /samples/vp9/etv.webm -f null -  16.27s user 0.05s system 99% cpu 16.335 total
  after :
  
    ./ffmpeg -v 0 -nostats -i /samples/vp9/etv.webm -f null -  15.22s user 0.03s system 99% cpu 15.257 total
  
    ./ffmpeg -v 0 -nostats -i /samples/vp9/etv.webm -f null -  15.20s user 0.02s system 99% cpu 15.237 total
  
    ./ffmpeg -v 0 -nostats -i /samples/vp9/etv.webm -f null -  15.19s user 0.02s system 99% cpu 15.227 total
  

  • [DH] libavcodec/vp9dsp.c

• FFmpeg : How to estimate number of samples in audio stream ?

  15 août 2017, par Marius

  I’m currently writing a small application that’s making use of the FFmpeg library in order to decode audio files (especially avformat and swresample) in C++.

  Now I need the total number of samples in an audio stream. I know that the exact number can only be found out by actually decoding all the frames, I just need an estimation. What is the preferred method here ? How can I find out the duration of a file ?