Burning Ogg Vorbis tracks to audio CD

Spikey, I agree that Vorbis is the better format, but the simple thing is that MP3 is ubiquitous and considered "good enough" by most. And while you assert that it is not a filesize/quality issue, in the majority of cases that's exactly the question: How many bits does this codec take to represent audio with audible transparency? MP3's approximation of human psychoacoustics is good enough to produce humanly-transparent audio, and so is Vorbis' model. (Although, there are some uncommon samples that will make either codec artifact at just about any bitrate.)

Additionally, while there may be many Vorbis-compatible players, people who have invested hundreds of dollars in portable or car-based MP3 players will not be so eager to make yet another pricey purchase just so they can listen to Vorbis files. Even some Vorbis-compatible hardware players don't have full compability... I know for sure that certain iRiver players can't play Ogg Vorbis files that contain frames larger than a certain value (I can't remember right now... I think it may be any frame > 320Kbps, or maybe even 160Kbps). You, Spikey, might care to use Vorbis, but honestly you cannot expect everyone else to be so eager about it, just as I can't expect everyone to care enough to use my audio ripping guide. Some people just want their music on their hard drive fast, quality be damned.

Also, I find it hard to believe that, knowing all of this you do about Vorbis, you did not realize that all codecs are (and indeed must be) decoded to get at the audio data inside. Saying Vorbis -> WAV degrades quality is like saying a ZIPed .DOC -> .DOC degrades the quality of the Word document. And it would also imply that simply playing an MP3 degrades the quality of the audio going to your speakers (since decoding converts the audio to raw PCM, the kind of data that WAV files contain, and the only kind of data your soundcard can understand)... which makes no sense.

Also, I'd like to clarify the explanation someone made, that while correct in result, is totally incorrect in process. Re-encoding loses quality based on the simple principle that continuously approximating (encoding an MP3) based on each previous approximation (the existing MP3) will result in an increasing deviation from the original (CD/WAV). Just like playing "telephone", or faxing faxes. The explanation that one encode removes inaudible data, thus other encodes must find other data to remove (with the implication that it will be audible) is completely false. I suppose a better way to describe lossy compression, to avoid such confusion, is that it represents data in an approximate fashion such that it will likely never be exactly equal to the original. There is really no active "removal", it's more of a... selective, approximate preservation.

Jam it back in, in the dark.

It sounds like you still don't quite understand what decoding is. Decoding should never be lossy, because its simply a "translation" from that codec's arbitrarily-structured data compression to the standard raw data format that your hardware or other software can process. If you tried to pass actual MP3 data to your soundcard, or MPEG data directly to your video card, you'd only get gibberish out of it. Compressed files are only useful for one thing: being stored on your hard disk (or other permanent storage). To get any actual use from the file, the data must first be decompressed, usually into the RAM. (In the case of playing compressed media, it is decompressed in a "stream", continuously decoding what's about to play and discarding from memory what has already played.) There is no output of a compressed file without decoding, so saying that decompressing introduces loss to the output is pretty comparable to declaring 1 is not equal to 1. "... which makes no sense."

The only process that introduces loss is the encoding phase... when the source audio is approximated into that lossy file. And, of course, lossless compression formats will encode with no loss, exactly preserving the source. (Also of note, WAV is neither lossless or lossy per se, because it is not a compressed format at all. It contains PCM data that can be sent directly to the soundcard.)

You say that Vorbis is far superior at recreating the original WAV than MP3... I'll give you the benefit of the doubt, and assume that you really meant Vorbis is superior at recreating the sound of the original WAV as heard by the human ear, because anything else in the original WAV is worthless as far as we're concerned. However, this does not do anything to counter my point that MP3s at a higher bitrate will achieve the same identical-sounding effect (aka "transparency"), so it's still just a bitrate efficiency issue.

But what it mostly comes down to -- and don't ignore this, it's the most important piece -- is many people like to be able to share their music. If you just have your music in Ogg Vorbis, then you will find that far fewer people will be receptive to your sharing... even though its a free offer! The majority thinks MP3 is just fine and would consider getting their computers to play Vorbis as much more of a hassle than the additional disk space required by MP3s of similar quality. And that's if they even care about the difference in audio quality...

As a post-note, I must say I have been amazed at how good music sounds at low quality settings with the latest aoTuV version, compared to the horrendous MP3 quality I remember at those sub-64Kbps bitrates. And I don't doubt that Vorbis reaches audible transparency "earlier" than MP3 in most cases... but the compatibility thing (having both portable and car-stereo MP3-CD players) and the sharing thing make MP3 my personal lossy format of choice. Although, I say lossy choice because I encode all of my albums to the lossless FLAC format, usually only transcoding to lossy when the audio is to be used beyond my computer.

There's nowhere I can't reach.