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• Encoding Webm video with VP9 codec with ffmpeg

  29 juin 2017, par Michau

  I am trying to encode a Webm video for embedding in HTML5 tag. According to information from Wikipedia and other places, Chrome and Firefox should support Webm with both VP9 and VP9 codecs.

  When I encode the vide with the VP8 codec, the video plays fine in both browsers. Here is the ffmpeg command I use :

  ffmpeg -i source.mp4 -f webm -vcodec libvpx -acodec libvorbis -b:v 10M -crf 50 -y result.webm

  However, I wanted to use VP9 instead of VP8 to reduce the file size. So I changed the command to use the VP9 codec :

  ffmpeg -i source.mp4 -f webm -vcodec libvpx-vp9 -acodec libvorbis -b:v 10M -crf 50 -y result.webm

  The output video file is produced, but it doesn’t play both in Chrome and FireFox. The only thing I changed was the "vcodec" option.

  So my question is, how can I encode Webm video with VP9 codec using ffmpeg so that it is playable in Web browsers ?

• How can I improve the up-time of my coffee pot live stream ?

  26 avril 2017, par tww0003

  Some Background on the Project :

  Like most software developers I depend on coffee to keep me running, and so do my coworkers. I had an old iPhone sitting around, so I decided to pay homage to the first webcam and live stream my office coffee pot.

  The stream has become popular within my company, so I want to make sure it will stay online with as little effort possible on my part. As of right now, it will occasionally go down, and I have to manually get it up and running again.

  My Setup :

  I have nginx set up on a digital ocean server (my nginx.conf is shown below), and downloaded an rtmp streaming app for my iPhone.

  The phone is set to stream to example.com/live/stream and then I use an ffmpeg command to take that stream, strip the audio (the live stream is public and I don’t want coworkers to feel like they have to be careful about what they say), and then make it accessible at rtmp://example.com/live/coffee and example.com/hls/coffee.m3u8.

  Since I’m not too familiar with ffmpeg, I had to google around and find the appropriate command to strip the coffee stream of the audio and I found this :
  

  ffmpeg -i rtmp://localhost/live/stream -vcodec libx264 -vprofile baseline -acodec aac -strict -2 -f flv -an rtmp://localhost/live/coffee

  Essentially all I know about this command is that the input stream comes from, localhost/live/stream, it strips the audio with -an, and then it outputs to rtmp://localhost/live/coffee.

  I would assume that ffmpeg -i rtmp://localhost/live/stream -an rtmp://localhost/live/coffee would have the same effect, but the page I found the command on was dealing with ffmpeg, and nginx, so I figured the extra parameters were useful.

  What I’ve noticed with this command is that it will error out, taking the live stream down. I wrote a small bash script to rerun the command when it stops, but I don’t think this is the best solution.

  Here is the bash script :

  while true;
  
  do
  
          ffmpeg -i rtmp://localhost/live/stream -vcodec libx264 -vprofile baseline -acodec aac -strict -2 -f flv -an rtmp://localhost/live/coffee
  
          echo 'Something went wrong. Retrying...'
  
          sleep 1
  
  done

  I’m curious about 2 things :

  1. What is the best way to strip audio from an rtmp stream ?
  2. What is the proper configuration for nginx to ensure that my rtmp stream will stay up for as long as possible ?

  Since I have close to 0 experience with nginx, ffmpeg, and rtmp streaming any help, or tips would be appreciated.

  Here is my nginx.conf file :

  worker_processes  1;
  
  
  
  events {
  
      worker_connections  1024;
  
  }
  
  
  
  
  
  http {
  
      include       mime.types;
  
      default_type  application/octet-stream;
  
  
  
      sendfile        on;
  
  
  
      keepalive_timeout  65;
  
  
  
      server {
  
          listen       80;
  
          server_name  localhost;
  
  
  
          location / {
  
              root   html;
  
              index  index.html index.htm;
  
          }
  
  
  
          error_page   500 502 503 504  /50x.html;
  
          location = /50x.html {
  
              root   html;
  
          }
  
  
  
          location /stat {
  
                  rtmp_stat all;
  
                  rtmp_stat_stylesheet stat.xsl;
  
                  allow 127.0.0.1;
  
          }
  
          location /stat.xsl {
  
                  root html;
  
          }
  
          location /hls {
  
                  root /tmp;
  
                  add_header Cache-Control no-cache;
  
          }
  
          location /dash {
  
                  root /tmp;
  
                  add_header Cache-Control no-cache;
  
                  add_header Access-Control-Allow-Origin *;
  
          }
  
      }
  
  }
  
  
  
  rtmp {
  
  
  
      server {
  
  
  
          listen 1935;
  
          chunk_size 4000;
  
  
  
          application live {
  
              live on;
  
  
  
              hls on;
  
              hls_path /tmp/hls;
  
  
  
              dash on;
  
              dash_path /tmp/dash;
  
          }
  
      }
  
  }

  edit :
  I’m also running into this same issue : https://trac.ffmpeg.org/ticket/4401

• GOP size for realtime video stream

  7 janvier 2017, par MadMass

  I’m working on a kind of rich remote desktop system, with a video stream of the desktop encoded using avcodec/x264. I have to set manually the GOP size for the stream, and so far I was using a size of fps/2.
  But I’ve just read the following on Wikipedia :

  This structure [Group Of Picture@ suggests a problem because the fourth frame (a P-frame) is needed in order to predict the second and the third (B-frames). So we need to transmit the P-frame before the B-frames and it will delay the transmission (it will be necessary to keep the P-frame).

  It means I’m creating a lot of latency since the client needs to receive at least half of the GOP to output the first frame following the I frame. What is the best strategy for the GOP size if I want the smallest latency possible ? A gop of 1 picture ?