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• X264—Why does "—pass 1 —stats" make the file bigger ?

  4 mai 2016, par Codecodeup

  I am using --pass 1 --stats <stats file="file" location="location"></stats> to generate the stats file for H.264 encoding with x264. The command is like this :

  <x264 binary="binary"> <input file="file" /> -o <output file="file"> --preset veryslow --crf 27 
  
    --tune ssim  --scenecut 0 --pass 1 --stats <stats file="file" location="location">
  
  </stats></output></x264>

  When I ran this command with and without --pass 1 --stats <stats file="file" location="location"></stats>, the difference between the sizes of the final output files are very different. The one with --pass 1 --stats <stats file="file" location="location"></stats> is much bigger. What is the reason ? Thanks.

• seeking mp4 with audiostream inaccurate in mobile safari

  23 février 2015, par dubbelj

  I have an issue with seeking an MP4 videofile on iOS (ipad). We use currentTime to skip through the video. This is working very well on most browsers, and also for h264 files without an audiostream on iOS. When I add an audiostream to the mp4 file however, the seeking becomes very inaccurate. This leads to all sorts of issues as the application bases a lot around timing.

  So we definitely concluded that a file that is encoded (using ffmpeg) with the exact same encoding settings but without audio is working well.

  These are the ffmpeg parameters I’m using when encoding with audio (added breaks for readability) :

  ffmpeg
  
  -i source.mov 
  
  -force_key_frames 00:00:04.840,00:00:04.880,00:00:04.920,00:00:04.960,00:00:05.000,00:00:05.040,00:00:05.080,00:00:14.760,00:00:14.800,00:00:14.840,00:00:14.880,00:00:14.920,00:00:14.960,00:00:15.000,00:00:24.760,00:00:24.800,00:00:24.840,00:00:24.880,00:00:24.920,00:00:24.960,00:00:25.000,00:00:31.720,00:00:31.760,00:00:31.800,00:00:31.840,00:00:31.880,00:00:31.920,00:00:31.960
  
  -c:a libvo_aacenc
  
  -b:a 128k 
  
  -ar 44100 
  
  -c:v libx264 
  
  -preset slow 
  
  -crf 21 
  
  -g 25 
  
  -r 25 
  
  -maxrate 2000k 
  
  -bufsize 2000k 
  
  -pix_fmt yuv420p 
  
  -movflags +faststart 
  
  -tune zerolatency 
  
  -vstats 
  
  encoded.mp4

  Things I’ve tried :

  • all available aac and mp3 codecs
  • different audio and video bitrates
  • different audio frequencies
  • with without zerolatency faststart flags
  • maximum amount of keyframes in file
  • settings keyframes only at (and around) the area where I need to skip to

  Any ideas anyone ?

• Can I stream WebM video to a video element through WebSocket ?

  5 février 2015, par Ravenstine

  I’d like to be able to capture my desktop using FFmpeg and send the video through a WebSocket to the client where the video stream can be played in an HTML5 video tag. It seems that the current way to accomplish this is by sending JPG data and placing it on a canvas element. I would rather stream actual video and audio data before resorting to that.

  Below is my server code. Note that I am running Ruby 2.2.0 on Linux(Debian Stable).

  require 'bundler'
  
  Bundler.require
  
  
  
  EM.run do
  
    EM::WebSocket.run host: "0.0.0.0", port: 9393 do |ws|
  
  
  
      cmd = "ffmpeg -y -f x11grab -s 1600x900 -r 15 -i :0.0 -tune fastdecode -b:v 150k -threads 4 -f webm -"
  
      @handler = EM.popen3(cmd, stdout: Proc.new { |data| 
  
        puts(data)
  
        ws.send(Base64.encode64(data))
  
      }, stderr: Proc.new{|err|})
  
  
  
      @handler.callback do
  
        puts "hey"
  
      end
  
  
  
      @handler.errback do |err_code|
  
        puts err_code
  
      end
  
  
  
      ws.onopen do |handshake|
  
        puts "WebSocket connection open"
  
      end
  
  
  
      ws.onclose do
  
        puts "Connection closed"
  
        # @handler.kill('TERM', true)
  
      end
  
  
  
    end
  
  
  
    class SimpleView < Sinatra::Base
  
      set :public_folder, './'
  
      configure do
  
        set :threaded, false
  
      end
  
      get '/' do
  
        send_file
  
      end
  
    end
  
  
  
    EM.run do
  
      Rack::Server.start({
  
        app:    Rack::Builder.app{map('/'){ run SimpleView.new }},
  
        server: 'thin',
  
        Host:   '0.0.0.0',
  
        Port:   '8181'
  
      })
  
    end
  
  
  
  end

  Here is the client code(JavaScript) :

  var stream = new WebSocket('ws://localhost:9393')
  
  var videoElement = document.querySelector("#desktop")
  
  var videoSource = document.querySelector("source")
  
  window.MediaSource = window.MediaSource || window.WebKitMediaSource;
  
  var mediaSource = new MediaSource()
  
  videoElement.src = window.URL.createObjectURL(mediaSource)
  
  
  
  stream.onopen = function(){
  
    console.log('connection open')
  
  }
  
  
  
  stream.onclose = function(){
  
    console.log('connection closed')
  
  }
  
  
  
  mediaSource.addEventListener('sourceopen', function(e){
  
    var sourceBuffer = mediaSource.addSourceBuffer('video/webm; codecs="vp8,vorbis"')
  
  
  
    stream.onmessage = function(e){
  
      var byteCharacters = atob(e.data)
  
  
  
      var byteNumbers = new Array(byteCharacters.length)
  
      for (var i = 0; i < byteCharacters.length; i++) {
  
        byteNumbers[i] = byteCharacters.charCodeAt(i)
  
      }
  
  
  
      var byteArray = new Uint8Array(byteNumbers)
  
  
  
      sourceBuffer.appendStream(byteArray)
  
  
  
    }
  
  
  
  }, false)

  What’s happening is I’m capturing the stdout of FFmpeg, converting each chunk of data to base64, sending it through the websocket, then the client tries to decode the base64 chunk into something a source buffer can understand and play the mediasource in the video element.

  Though it doesn’t surprise me that this doesn’t work, I would still like it to and I’m hoping maybe there’s just one little thing that I’m missing. I get nothing except black inside the video element.

  NOTE : I am using the original FFmpeg, and NOT Avconv. I compiled it with all codecs enabled.

  The full source is available at https://github.com/Ravenstine/simpleview.