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• filesize is not growing as expected

  29 juillet 2019, par Oracy Martos

  I am trying to record a stream here on my machine to study ffmpeg library,
  but with(out) success.

  I have a file watcher to clean up bugged streams, that cleanup each 3 minutes files that have been not changed less then 3 minutes.

  The real problem is, if I use the command below :

  /usr/bin/ffmpeg -i http://sysrad.net:10090/ -y test.mp3

  this command doesn’t have any kind of codec or audio transformation, so my target file (test.mp3) become 256k quickly, but, if I use this command below :

  /usr/bin/ffmpeg -i http://sysrad.net:10090/ -y -b:a 8k -ac 1 -ar 11025 test.mp3

  My target file (test.mp3) keep 0k until the record has 256k, I am not sure if this is an Unix problem or ffmpeg problem.

  Other information, if I run in loop :

  while true; do wc -l teste.mp3; sleep 0.5; done;

  test.mp3 file keeps 0 rows, until has 256k size...

  I have no idea how to workaround that, to get the real time file size for each 1k that ffmpeg get from stream with those codecs, does you guys have any idea how can I handle that ?

  Thanks !!!!

• vvcdec : add CTU parser

  5 décembre 2023, par Nuo Mi

  vvcdec : add CTU parser
  Co-authored-by : Xu Mu <toxumu@outlook.com>
  
  Co-authored-by : Frank Plowman <post@frankplowman.com>
  
  Co-authored-by : Shaun Loo <shaunloo10@gmail.com>
  
  Co-authored-by : Wu Jianhua <toqsxw@outlook.com>
  

  • [DH] libavcodec/vvc/vvc_ctu.c
  • [DH] libavcodec/vvc/vvc_ctu.h

• WebRTC books – a brief review

  30 décembre 2013, 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.