Freerip. Free, easy and converts to and between Ogg, MP3 and Wav.

Welcome to the world of higher quality audio ABF, it comes with better equipment you start hearing things like that, or more noticeably noticing what is missing from the recordings in an A/B test. Even I still have problems with 320 sometimes, but for example in a side by side test that I later confirmed with an reverse polarity test, of a classical piano recording I could notice some of the attack of the keys missing and other very minor details. It is tough for me at 320 to hear the difference though. Go below 320 and heck yes I can hear the difference and despise it.

As such for high quality audio ripping, on Mac and Linux you can use CDParanoia for the rip, and the encode it in FLAC. On Windows EAC is the best there is. On Linux the abcde bash scripts does a good job of ripping(Via CDParanoia), encoding(If you dont like FLAC it can also do OGG etc) tag from FreeCDDB etc. On Mac there is a GUI called Max I have been leaning towards looking into. It is also a FE to CDParanoia, and like abcde will encode in a variety of formats including FLAC and take care of the tagging for you. I havent tried it out yet, I have been meaning to, but at a glance at the technologies behind it it looks pretty good.

Seablade

Guess you guys have never heard of LAME. Here's a link for your edification:

http://wiki.hydrogenaudio.org/index.php?title=LAME

LAME 'Ain't A MP3 Encoder', or so its name goes. It is, perhaps, the best available MP3 encoder, but MP3 is not 'high quality'...

Thank you to Olyteddy, but just for 'edification's' purpose I will feel free to explain WHY MP3 is not a very high quality codec.

For the record I have used, and still use on rare occasions LAME, when I compress MP3s it is what I choose to use.

However on to the topic..

MP3 is a lossy perceptual compression format. This means several things, one that 'lossy' part up there should be self explanatory, it does NOT preserve the exact audio. What it does has to do with the perceptual part, where it useses alogrithms to determine audio frequencies it thinks you dont hear or wont notice. This is far from an exact science, in fact can be off at times, however some examples might be one frequency that might mask another due to it being a harmonic of the other, and as such the lesser frequency needs to be a certain level for you to hear it. As I mentioned though, this is not an exact science and the second frequency combined with a third one might create interference of a harmonic nature to create a fourth one. As you can probably guess this gets very complex very fast.

Now the other way that MP3s use to lessen audio file size is they chop off above 16kHz. No perceptual compression, they just delete it. Now on your average car stereo where you are competing with your engine noise and have cruddy paper cone speakers and a poor quality head unit, this may not make a difference. Put it into a somewhat decent listening environment though and it certainly does. Now how much of that you can hear depends on several things, one your listening environment that I have already mentioned, as well as your own personal hearing, considering how many people cause hearing loss by listening to things on earphones to loud, which some studies have claimed is worse than going to concerts due to repeated exposure this might make a surprisingly small difference to some, to others though it makes a large difference. And the last thing that makes a difference I already alluded to as well, the quality of your componets. Listening on a pair of decent headphones instead of 5 buck off the shelf headphones makes a huge amount of difference. As well as the difference between decent speakers and paper coned ones you can buy in the bargain bin at a Radio Shack.

Now compare all this to an uncomressed .wav file at 16/44.1 which is what you get on a CD. 44.1 was chosen because of Nyquist which states to reproduce a frequency requires only 2 samples of that frequency. Now what it ignores is the same harmonic interference I mentioned before which works with higher frequencies as well as lower to create new frequencies, so even it is not perfect, in fact it could be argued digital is incapable of being perfect, thus leading to many analog vs digital debates, which is another topic entirely. But at any rate with CDs you get a VERY close representation of the signal, close enough that probably 95-99 percent of people cant hear the difference, and at this time the benefits outweigh the drawbacks. But at any rate the signal ranges the full 20Hz-20KHz range of human hearing in the reproduction of audio signal, already beating out MP3s. Not to mention also it has those signals that MP3 removes(Assuming it hasnt been compressed to MP3 or another perceptual codec before and then converted to uncompressed). This allows for a much better representation of the signal.

Finally we come to what I use myself, which is a lossless compression codec. There are a few out there, FLAC is a popular one as is now Apple Lossless. I personally prefer FLAC as it is open but that is another topic. At any rate lossless compression in theory has everything that the uncompressed file will have, in reality there is a very slight difference determined by the decompression of the file usually, and how good a quality decompression algorithm is used, so players to save CPU or because of bugs will not be as good a quality it has been found, but this is very rarely an issue.

File sizes also tend to vary, for instance on a 1 gig flash card I can fit approx 7 hours of 320 kbps stereo MP3s. I know this because I have been using this to record longer operas that I dont have a CF card large enough to store in uncompressed .wavs yet.

Obviously a CD stores 60-70 minutes of Uncompressed audio.

And finally FLAC on average compresses things to about 1/2 the file size of the uncompressed file, though this can vary both ways depending on the file.

128 MP3 tends to be about 1/10th the size for the record. ITunes I believe uses 192 AAC files which is also a lossy perceptual compression.

At any rate this VERY brief and simplified lesson on quality and compression algorithms has been brought to you by Carmen, which I am currently working on.

Seablade

By the way, the above is why 'MP3' and 'High Quality' are Oxymorons, just in case it isnt obvious

And if you need links to your 'edification', I am certain Google can provide quite a few if you google Lossless compression, Lossy compression, MP3, FLAC, AAC, and a variety of other topics.

Of course you can post back here and depending on how bored I am I may post some up as well, as it is I have typed a bit much anyways

Seablade

PS All the programs I mentioned before I believe can be configured to Use liblame as well as libogg libflac etc depending on what compression you choose to use for whatever reason.

Yeah, I'm not bored enough to post back here either. Maybe tomorrow....