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Converting Media Files

JTLanguage only supports a limited number of media file formats. These are (using a standard format identification name called a "Mime" type to identify them):

  • audio/mpeg3
  • audio/ogg
  • video/mp4
  • video/ogg
  • image/jpeg
  • image/png
  • image/gif
  • text/plaintext
  • application/pdf

If you can't produce media in one of these formats, you will need to run a converter tool to convert your media files to one of the above formats.

There are many converter tools generally available. I use a free one called "ffmpeg", which is kind of the end-all for converters, although it can be intimitating to use. Do a web search for "ffmpeg download windows", subsituting the O/S you use.

Or, if you have sufficient trust (it's generally recommended you don't click on executable file links on websites), here's a link to a copy of the executable for Windows that I use on the web server when it converts files: ffmpeg.exe (Right-click on the link and select "Save target as".)

In some cases, you could just run ffmpeg like this:

    ffmpeg -i (input file) (output file)

But unfortunately, sometimes various configurations (specifically the bit rate and sample rate) in the media don't translate well. It might still work in JTLanguage, but it might choke when the audio visualization is done in the audio recorder or mapper in JTLanguage. Therefore, the safe way is to explicitly tell ffmpeg these parameters.

First, to find out the parameters, first run ffmpeg with just the input file, and no output. For example, if my file is called "myfile.aac", do this:

    ffmpeg -i myfile.aac

ffmpeg will spit out some messages like this:

ffmpeg version N-64304-g7f5d75f-Sherpya Copyright (c) 2000-2014 the FFmpeg developers
  built on Jun 29 2014 18:17:37 with gcc 4.9 (GCC)
[aac @ 06795e40] Estimating duration from bitrate, this may be inaccurate
Input #0, aac, from 'myfile.aac':
  Duration: 00:00:47.36, bitrate: 119 kb/s
    Stream #0:0: Audio: aac, 48000 Hz, mono, fltp, 120 kb/s
At least one output file must be specified

Then, taking note of the bitrate (119000) and sample rate (48000) you can do the real conversion like this:

    ffmpeg -i myfile.aac -b:a 126000 -ar 48000 myfile.mp3

Note that you might only need to get the parameters once, assuming other files created the same way will have the same parameters.

Before converting lots of files, you probably want to convert and upload one file, and test it in the player and mapper, to make sure it works.

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