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• AppRTC : Google’s WebRTC test app and its parameters

  23 juillet 2014, par silvia

  If you’ve been interested in WebRTC and haven’t lived under a rock, you will know about Google’s open source testing application for WebRTC : AppRTC.

  When you go to the site, a new video conferencing room is automatically created for you and you can share the provided URL with somebody else and thus connect (make sure you’re using Google Chrome, Opera or Mozilla Firefox).

  We’ve been using this application forever to check whether any issues with our own WebRTC applications are due to network connectivity issues, firewall issues, or browser bugs, in which case AppRTC breaks down, too. Otherwise we’re pretty sure to have to dig deeper into our own code.

  Now, AppRTC creates a pretty poor quality video conference, because the browsers use a 640×480 resolution by default. However, there are many query parameters that can be added to the AppRTC URL through which the connection can be manipulated.

  Here are my favourite parameters :
  

  • hd=true : turns on high definition, ie. minWidth=1280,minHeight=720
  • stereo=true : turns on stereo audio
  • debug=loopback : connect to yourself (great to check your own firewalls)
  • tt=60 : by default, the channel is closed after 30min – this gives you 60 (max 1440)

  For example, here’s how a stereo, HD loopback test would look like : https://apprtc.appspot.com/?r=82313387&hd=true&stereo=true&debug=loopback .

  This is not the limit of the available parameter, though. Here are some others that you may find interesting for some more in-depth geekery :

  • ss=[stunserver] : in case you want to test a different STUN server to the default Google ones
  • ts=[turnserver] : in case you want to test a different TURN server to the default Google ones
  • tp=[password] : password for the TURN server
  • audio=true&video=false : audio-only call
  • audio=false : video-only call
  • audio=googEchoCancellation=false,googAutoGainControl=true : disable echo cancellation and enable gain control
  • audio=googNoiseReduction=true : enable noise reduction (more Google-specific parameters)
  • asc=ISAC/16000 : preferred audio send codec is ISAC at 16kHz (use on Android)
  • arc=opus/48000 : preferred audio receive codec is opus at 48kHz
  • dtls=false : disable datagram transport layer security
  • dscp=true : enable DSCP
  • ipv6=true : enable IPv6

  AppRTC’s source code is available here. And here is the file with the parameters (in case you want to check if they have changed).

  Have fun playing with the main and always up-to-date WebRTC application : AppRTC.

  UPDATE 12 May 2014

  AppRTC now also supports the following bitrate controls :

  • arbr=[bitrate] : set audio receive bitrate
  • asbr=[bitrate] : set audio send bitrate
  • vsbr=[bitrate] : set video receive bitrate
  • vrbr=[bitrate] : set video send bitrate

  Example usage : https://apprtc.appspot.com/?r=&asbr=128&vsbr=4096&hd=true

• h264 : increase MAX_SLICES to 32

  5 décembre 2014, par Vittorio Giovara

  h264 : increase MAX_SLICES to 32
  H264 streams from CounterPath Bria and some Cisco phones need an
  
  increased pool for correct decoding.
  Bug-Id : 645
  
  Sample-Id : bria-h264.mka
  

  • [DBH] libavcodec/h264.h

• Reading a stream of images from rtsp using ffmpeg

  15 février 2015, par Andrew Simpson

  I had been using a wrapper around VLC to stream images from my ip camera to my WinForm Desktop App using the RTSP protocol.

  This works well.

  However, I wanted to see if could achieve the same thing using just FFMPEG.

  I have done so but the results are poor.

  The FPS is slow.

  This is my code :

      private Process process = new Process();
  
      private FileStream baseStream = null;
  
      string file = @"C:\bin\ffmpeg.exe";
  
      string arguments = @" -i rtsp://admin:admin@192.168.0.8:554/video_1 -an -f image2 -s 360x240 -vframes 1 -";
  
  
  
      private void button1_Click(object sender, EventArgs e)
  
      {
  
          try
  
          {
  
              process.StartInfo.FileName = file;
  
              process.StartInfo.Arguments = arguments;
  
              process.StartInfo.CreateNoWindow = true;
  
              process.StartInfo.RedirectStandardError = true;
  
              process.StartInfo.RedirectStandardOutput = true;
  
              process.StartInfo.UseShellExecute = false;
  
  
  
              Task.Run(() => { Feeder(); });
  
          }
  
          catch (Exception ex)
  
          {
  
              //log error here
  
          }
  
      }
  
  
  
      private void Feeder()
  
      {
  
          try
  
          {
  
              process.Start(); 
  
              baseStream = process.StandardOutput.BaseStream as FileStream;
  
              int lastRead = 0;
  
              using (MemoryStream ms = new MemoryStream())
  
              {
  
                  byte[] buffer = new byte[500];
  
                  do
  
                  {
  
                      lastRead = baseStream.Read(buffer, 0, buffer.Length);
  
                      ms.Write(buffer, 0, lastRead);
  
                  } while (lastRead > 0);
  
  
  
                  pictureBox1.Invoke(new Action(() => pictureBox1.Image = Image.FromStream(ms)));
  
                  ms.Close();
  
              }
  
          }
  
          catch (Exception ex)
  
          {
  
              //log error here
  
          }
  
          Task.Run(() => { Feeder(); });
  
      }

  Now I had considered streaming a constant flow of images rather than just taking 1 at a time.
  I can easily identify markers for the JPEG header.

  But, I would have to use

  baseStream.ReadByte

  instead which without trying it would be slow as I would have look at every single byte ?

  Can anyone suggest what I should do ?

  Thanks