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• ffmpeg not working with filenames that have whitespace

  1er avril 2017, par cmw

  I’m using FFMPEG to measure the duration of videos stored in an Amazon S3 Bucket.

  I’ve read the FFMPEG docs, and they explicitly state that all whitespace and special characters need to be escaped, in order for FFMPEG to handle them properly :

  See docs 2.1 and 2.1.1 : https://ffmpeg.org/ffmpeg-utils.html

  However, when dealing with files whose filenames contain whitespace, ffmpeg fails to render a result.

  I’ve tried the following, with no success

  ffmpeg -i "http://s3.mybucketname.com/videos/my\ video\ file.mov" 2>&1 | grep Duration | awk '{print $2}' | tr -d
  
  ffmpeg -i "http://s3.mybucketname.com/videos/my video file.mov" 2>&1 | grep Duration | awk '{print $2}' | tr -d
  
  ffmpeg -i "http://s3.mybucketname.com/videos/my'\' video'\' file.mov" 2>&1 | grep Duration | awk '{print $2}' | tr -d
  
  ffmpeg -i "http://s3.mybucketname.com/videos/my\ video\ file.mov" 2>&1 | grep Duration | awk '{print $2}' | tr -d

  However, if I strip out the whitespace in the filename – all is well, and the duration of the video is returned.

  Any help is appreciated !

• rtmp audio out of sync, http works fine

  21 janvier 2014, par marca

  We have encoded and distributed videos for some years now, using FFMPEG to produce h.264/mp4 files that have been working great for us. We have been using HTML mode and fall-backed to flash for browsers that does not support it natively using flowplayer.

  We use cloudfront to serve our files from a s3 bucket and have been using http progressive streaming.

  Recently we started distribute the files in flashmode over rtmp instead, using a cloudfront streaming distribution pointing to the same amazon s3 bucket.

  All good for some weeks, until yesterday when we notice a couple of files with audio sync issues in rtmp mode.
  The same file have no sync problems in flash with direct url to file.

  What can be the case ?

  Not working when streamed via RTMP, but file work with http streaming/progressive.
  You see the sync issue 15 sec's into the video.
  rtmp ://s2xe2avk54qztf.cloudfront.net:1935/cfx/st/mp4:95fvOY255bdPspO3z6tEvGi3Em7/default.mp4
  http://media.shootitlive.com/95fvOY255bdPspO3z6tEvGi3Em7/default.mp4

  Another file that have no sync issue at all.
  rtmp ://s2xe2avk54qztf.cloudfront.net:1935/cfx/st/mp4:P4EuH2TZxfV6BvpupP6dxrrs7gD/default.mp4
  http://media.shootitlive.com/P4EuH2TZxfV6BvpupP6dxrrs7gD/default.mp4

  Both files have the same format for video and audio and have been encoded the exact same way with ffmpeg. It's not player related as we see the audio sync issue on several players and when playing stream in VLC.

• WebRTC books – a brief review

  1er janvier 2014, par silvia

  I just finished reading Rob Manson’s awesome book “Getting Started with WebRTC” and I can highly recommend it for any Web developer who is interested in WebRTC.

  Rob explains very clearly how to create your first video, audio or data peer-connection using WebRTC in current Google Chrome or Firefox (I think it also now applies to Opera, though that wasn’t the case when his book was published). He makes available example code, so you can replicate it in your own Web application easily, including the setup of a signalling server. He also points out that you need a ICE (STUN/TURN) server to punch through firewalls and gives recommendations for what software is available, but stops short of explaining how to set them up.

  Rob’s focus is very much on the features required in a typical Web application :

  • video calls
  • audio calls
  • text chats
  • file sharing

  In fact, he provides the most in-depth demo of how to set up a good file sharing interface I have come across.

  Rob then also extends his introduction to WebRTC to two key application areas : education and team communication. His recommendations are spot on and required reading for anyone developing applications in these spaces.

  —

  Before Rob’s book, I have also read Alan Johnson and Dan Burnett’s “WebRTC” book on APIs and RTCWEB protocols of the HTML5 Real-Time Web.

  Alan and Dan’s book was written more than a year ago and explains that state of standardisation at that time. It’s probably a little out-dated now, but it still gives you good foundations on why some decisions were made the way they are and what are contentious issues (some of which still remain). If you really want to understand what happens behind the scenes when you call certain functions in the WebRTC APIs of browsers, then this is for you.

  Alan and Dan’s book explains in more details than Rob’s book how IP addresses of communication partners are found, how firewall holepunching works, how sessions get negotiated, and how the standards process works. It’s probably less useful to a Web developer who just wants to implement video call functionality into their Web application, though if something goes wrong you may find yourself digging into the details of SDP, SRTP, DTLS, and other cryptic abbreviations of protocols that all need to work together to get a WebRTC call working.

  —

  Overall, both books are worthwhile and cover different aspects of WebRTC that you will stumble across if you are directly dealing with WebRTC code.