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• It’s Piwik Analytics Survey time : Your Opinion Matters !

  10 juillet 2014, par Piwik Core Team — Community

  It’s been almost two years since our last survey. We were so happy to get the participation we did from the community last time and hope that you will once again share your thoughts !

  

  Fill in the survey

  Looking back on the previous survey, several features were implemented as a direct result :

  • New app design with Morpheus theme
  • Piwik Marketplace for Plugins
  • Visitor Profile
  • Performance and Reliability Improvements
  • Theming
  • Migrated from SVN to Github !
  • And much more…

  Now it is time to ask for your opinion again. Sure it’s only 20 short questions, but your time you take and energy you place in this endeavour will shape Piwik development and features for the near future. We really look forward to your ideas, your opinions and your suggestions. To get started click the box (or link) below :

  

  Fill in the survey !

  By the way, if you haven’t already make sure you sign up for our newsletter to make sure you get the latest Piwik news and information to your inbox ! You can subscribe here.

• rtmppkt : Handle extended timestamp field even for one-byte header

  5 mars 2014, par Martin Panter

  rtmppkt : Handle extended timestamp field even for one-byte header
  Related fix in "rtmpdump" :
  
  https://repo.or.cz/w/rtmpdump.git/commitdiff/79459a2
  Adobe’s RTMP specification (21 Dec 2012), section 5.3.1.3 ("Extended
  
  Timestamp"), says "this field is present in Type 3 chunks". Type 3 chunks are
  
  those with the one-byte header size.
  This resolves intermittent hangs and segfaults caused by the read function,
  
  and also includes an untested fix for the write function.
  The read function was tested with ABC (Australia) News 24 streams, however
  
  they are probably restricted to only Australian internet addresses. Some of
  
  the packets at the start of these streams seem to contain junk timestamp
  
  fields, often requiring the extended field. Test command :
  avplay rtmp ://cp81899.live.edgefcs.net/live/news24-med@28772
  Signed-off-by : Martin Storsjö <martin@martin.st>
  

  • [DH] libavformat/rtmppkt.c
  • [DH] libavformat/rtmppkt.h

• Video Conferencing in HTML5 : WebRTC via Socket.io

  
  1er janvier 2014, par silvia

  Six months ago I experimented with Web sockets for WebRTC and the early implementations of PeerConnection in Chrome. Last week I gave a presentation about WebRTC at Linux.conf.au, so it was time to update that codebase.

  I decided to use socket.io for the signalling following the idea of Luc, which made the server code even smaller and reduced it to a mere reflector :

   var app = require(’http’).createServer().listen(1337) ;
   var io = require(’socket.io’).listen(app) ;
   io.sockets.on(’connection’, function(socket) 
  
           socket.on(’message’, function(message) 
  
           socket.broadcast.emit(’message’, message) ;
  
       ) ;
  
   ) ;
  

  Then I turned to the client code. I was surprised to see the massive changes that PeerConnection has gone through. Check out my slide deck to see the different components that are now necessary to create a PeerConnection.

  I was particularly surprised to see the SDP object now fully exposed to JavaScript and thus the ability to manipulate it directly rather than through some API. This allows Web developers to manipulate the type of session that they are asking the browsers to set up. I can imaging e.g. if they have support for a video codec in JavaScript that the browser does not provide built-in, they can add that codec to the set of choices to be offered to the peer. While it is flexible, I am concerned if this might create more problems than it solves. I guess we’ll have to wait and see.

  I was also surprised by the need to use ICE, even though in my experiment I got away with an empty list of ICE servers – the ICE messages just got exchanged through the socket.io server. I am not sure whether this is a bug, but I was very happy about it because it meant I could run the whole demo on a completely separate network from the Internet.

  The most exciting news since my talk is that Mozilla and Google have managed to get a PeerConnection working between Firefox and Chrome – this is the first cross-browser video conference call without a plugin ! The code differences are minor.

  Since the specification of the WebRTC API and of the MediaStream API are now official Working Drafts at the W3C, I expect other browsers will follow. I am also looking forward to the possibilities of :

  • multi-peer video conferencing like the efforts around webrtc.io,
  • the media stream recording API,
  • and the peer-to-peer data API.

  The best places to learn about the latest possibilities of WebRTC are webrtc.org and the W3C WebRTC WG. code.google.com has open source code that continues to be updated to the latest released and interoperable features in browsers.

  The video of my talk is in the process of being published. There is a MP4 version on the Linux Australia mirror server, but I expect it will be published properly soon. I will update the blog post when that happens.