Command-line Based OGG to MP3 (batch) Converter for Ubuntu Linux – ogg2mp3

MP3 is an old lossy proprietary audio codec (compression format) and when compared to the open source answer to it (OGG Vorbis), it has many drawbacks.

For instance, OGG can hold more than 2 audio channels where MP3 can only hold 2, it can give better quality outputs in the same bitrate, it also compresses better than MP3 at both lower and higher bitrates and the list goes on ;-). But MP3 is still one of the widely used formats and is supported by most audio players etc.

There are a few that supports OGG Vorbis too but that’s nowhere near the level of MP3 support. So in that sense, even if it sucks 😉 if you have a large collection of OGG Vorbis files, then to play it in your portable audio player for instance, you will have to convert them into MP3 and for that you can try ‘ogg2mp3.

As easy as 123 😉 …

The only issue that I have with it is that it is really sensitive. Meaning that if one of your OGG files has a bit of a improper coding in its meta-data header, it won’t encode it. Other than that, it’s an extremely useful tool.


Main features …

*. It’s actually a tool that uses the ‘lame’ (default MP3 encoder) and most of its commands are the commands of ‘lame’.

*. Supports average and constant bitrate modes.

*. Channel output support include: stereo, mono, dual-mono etc.

*. Other than entering a bitrate, you can change the quality by using a number which enables a preset (0 to 9, lover the better. Default level is 5).

*. Individual and batch file converting.

If interested, you can install “ogg2mp3” in Ubuntu 12.04 Precise Pangolin, 11.10 Oneiric Ocelot, 11.04 Natty Narwhal, 10.10 and 10.04 by first getting the “.deb” file from this ogg2mp3 download page.

Once the download completes, just double click on it and Ubuntu Software Center should install it for you.

A simple example … (singe file mode)

Let’s say that I have a Vorbis audio called “1.ogg”. Then to convert it to MP3 format using an average audio bitrate of say “96 kbit/s” I’d use the below command.

ogg2mp3 1.ogg -a 96

If I wanted to use a variable bitrate, then I’d use the below command instead.

ogg2mp3 1.ogg -v 6

The number represents the quality (read the manual of ‘lame’).

Batch conversion …

First put all the files that you want convert into a single folder (let’s call it “temp”). Then simply give “ogg2mp3” the folder path with the appropriate audio parameters (bitrate, channels etc) and it will automatically convert one by one.

In this instance I’ll use the below command.

ogg2mp3 /home/gayan/temp -a 96


For more information please read its manual (including the actual converting tool called “lame” by using the below commands.

man ogg2mp3

man lame

Other than its a too sensitive nature where it won’t encode a file if its meta-tag is not set according to general standards as mentioned above, ‘ogg2mp3a is a very useful tool.

An RHCE, 'Linux' user with 14+ years of experience. Extreme lover of Linux and FOSS. He is passionate to test every Linux distribution & compare with the previous release to write in-depth articles to help the FOSS community.

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