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For non-tech people, what this means, basically, is that the file went through a compression state and then was uncompressed with no change.

Example:

A WAV file could look like this: ABCDEFGHIJK - This is called "lossless" - all of the digital information is there, which is then translated back into audio information

To elaborate on this... The WAV file is the raw, uncompressed, full digitized data of the music itself. This is the actual music itself in the digitized form as it would pretty much be on a standard Red Book audio CD itself. Of course, there could be arguments made that some quality gets lost in the actual digitizing process of the analog waveform itself, but that is going way beyond the scope of this. Believe me, there are plenty of folks that argue back and forth about that aspect, which I am not about to get into right now.

The MP3 would look like this : ACDGK - This is "lossy" - you'll see "B, E, and F" are gone. When you listen to this, the computer knows B,E and F are missing, and ignores them completely. Normally, you would not notice these missing gaps.

The 'B', 'E', and 'F' are completely thrown away, thus cannot be restored completely. Most playback/decoding algorithms will just simply ignore that, since it is not really there anymore. However, some of the better, higher-end playback equipment/software can make a "best guess" at what those missing pieces actually were and attempt to fill it back in. This is called "interpolation". I think my Elan Via!DJ music server does this, but cannot find anything in the documentation that actually says so. When done correctly, it would be really hard-pressed for most folks to tell the difference between that and the original, especially with high bit-rate MP3 files, since there is more information available in those files with which to work with in making a really good guess. The parts that are typically thrown away in MP3 files are the really high and really low frequencies which most play-back equipment would not be able to play anyways. That is why it is typically the high frequencies (i.e, the cymbols, and such), or the low frequencies (the bass) that suffers and where one can typically hear the compression artifacts.


The FLAC file would look like this: ACDFHJK - Bigger than the MP3 file, but still with missing pieces. HOWEVER, when it's run through the program that plays it BACK, it uses an algorythm to fill in what's missing. This is "decompression" and allows it to be "lossless".

And the reason why these files are typically bigger is because there is additional data embedded with the actual compressed data that basically describes the missing pieces, thus basically 'telling' the playback/decoding algorithm how to put those missing pieces back to get the original uncompressed data back, unchanged.
 
I always thought that a FLAC was just a compressed (lossless) WAV file. I've been known to use Winzip on a WAV file and it was about the same size as a FLAC of the same song. Using the new compression algorithm for Winzip (the new ".zipx" format) and it's even less than FLAC.