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• FFmpeg : How to generate a .mp4 with h.265 Codec ?

  5 mai 2014, par skull3r7

  I want to encode a video file to H.265. The last version of ffmpeg should be support H.265 (Source).

  However, I could not find any information about the exact command.

  I do not know, which library I should indicate after -vcodec.

  EDIT : I use the lastest Zeranoe FFmpeg Build (git-c78a416 (2013-10-26)).

• How can I encode and segment audio files without having gaps (or audio pops) between segments when I reconstruct it ?

  16 mai 2013, par fenduru

  I'm working on a web application that requires streaming and synchronization of multiple audio files. For this, I am using the Web Audio API over HTML5 audio tags because of the importance of timing audio.

  Currently, I'm using FFMPEG's segmentation feature to encode and segment the audio files into smaller chunks. The reason I am segmenting them is so I can start streaming from the middle of the file instead of starting from the beginning (otherwise I would've just split the files using UNIX split, as shown here. The problem is that when I string the audio segments back together, I get an audio pop between segments.

  If I encode the segments using a PCM encoding (pcm_s24le) in a .wav file, the playback is seamless, which leads me to believe that the encoder is padding either the beginning or the end of the file. Since I will be dealing with many different audio files, using .wav would require far too much bandwidth.

  I'm looking to one of the following solutions to the problem :

  • How can I segment encoded audio files seamlessly,
  • How can I force an encoder to NOT pad audio frames using ffmpeg (or another utility), or
  • What is a better way to stream audio (starting at an arbitrary track time) without using an audio tag ?

  System Information

  • Custom node.js server
  • Upon upload of an audio file, node.js pipes the data into ffmpeg's encoder
  • Need to use HTML5 Web Audio API supported encoding
  • Server sends audio chunks 1 at a time through a WebSockets socket

  Thanks in advance. I've tried to be as clear as possible but if you need clarification I'd be more than willing to provide it.

• Video Conferencing in HTML5 : WebRTC via Socket.io

  
  5 février 2013, 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.