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• shell script : ffmpeg can't either read arguments or filenames, if spaces in path- or filename

  9 juillet 2014, par Nicolas

  I’m currently working on a bash shell script on OS X to get a custom tailored Pashua GUI/automatization package, using ffmpeg in the back, to transcode videos, label them, zip them, upload them, organize them and so on. Everything is working like a charm so far.

  My problem now is that I can’t make ffmpeg take files or paths that have spaces in them IF I use variables in variables. But that (I think) is a must in order to get ffmpeg to transcode multiple files at once depending on the formats and checkboxes the user ticks on.

  I made a little example snippet so you can see my dilemma.

  No spaces in path or filename / works great :

  #!/bin/bash
  
  
  
  infile='/Users/nicolas/Desktop/MASTER_[DE]_Test_140626.mov'
  
  
  
  gen_outfile () {
  
      outfile="${1/MASTER/TRANSCODE}"
  
      ffmpeg_cl="-s $2 -c:v libx264 -preset slow -crf 18 -pix_fmt yuv420p -c:a aac -b:a 240k -strict -2 $outfile"
  
  }
  
  
  
  gen_outfile "$infile" 1920x1080
  
  
  
  ./ffmpeg_stat -i "$infile" $ffmpeg_cl

  The output is as expected and the file is called "TRANSCODE_[DE]_Test_140626.mov"

  Filename variable with spaces - directly fed to ffmpeg :

  This works as well but isn’t really usable for me because I need to have the option of variable numbers of output files and arguments generated dynamically.

  #!/bin/bash
  
  
  
  infile_full='/Users/nicolas/Google Drive/encode-o-mat/MASTER_[DE] Test  140626.mov'
  
  
  
  gen_outfile () {
  
      outfile="${1/MASTER/TRANSCODE}"
  
      ffmpeg_cl="-s $2 -c:v libx264 -preset slow -crf 18 -pix_fmt yuv420p -c:a aac -b:a 240k -strict -2"
  
  }
  
  
  
  gen_outfile "$infile_full" 1920x1080
  
  
  
  ./ffmpeg_stat -i "$infile_full" $ffmpeg_cl "$outfile"

  The output is as expected and the file is called "TRANSCODE_[DE] Test 140626.mov".
  But this way is just not an option.

  This here does not work ! Filename is split up and incomplete :

  #!/bin/bash
  
  
  
  infile_full='/Users/nicolas/Google Drive/encode-o-mat/MASTER_[DE] Test  140626.mov'
  
  
  
  gen_outfile () {
  
      outfile="${1/MASTER/TRANSCODE}"
  
      ffmpeg_cl="-s $2 -c:v libx264 -preset slow -crf 18 -pix_fmt yuv420p -c:a aac -b:a 240k -strict -2 $outfile"
  
  }
  
  
  
  gen_outfile "$infile_full" 1920x1080
  
  
  
  ./ffmpeg_stat -i "$infile_full" $ffmpeg_cl

  Ffmpeg gives this error : "Unable to find a suitable output format for ’/Users/nicolas/Google’".
  Of course.

  This doesn’t work as well ! Quoting and such :

  ./ffmpeg_stat -i "$infile_full" "$ffmpeg_cl"

  If I put the $ffmpeg_cl variable in various quotes ffmpeg doesn’t understand the arguments anymore ! This is one example output :

  "Unrecognized option ’s 1920x1080 -c:v libx264 -preset slow -crf 18 -pix_fmt yuv420p -c:a aac -b:a 240k -strict -2 /Users/nicolas/Google Drive/encode-o-mat/TRANSCODE_[DE] Test 140626.mov’.
  Error splitting the argument list : Option not found
  Conversion failed !"

  It’s really important that the script can handle path- and filenames with spaces because I can’t change the environment in which that tool is supposed to run. All ideas and solutions are more than welcome !

• 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.

• 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.