I am about to resume ripping my CD collection to my iMac now that I've gotten a larger drive to work with. Of late, I've been reading much about XLD and it's merits, so I think I'm going to try that. As for formats, I've read that Apple Lossless, and wav files actually alter the sound in some cases in relation to the originals. At least to some peoples ears.
I want transparency. I am considering AIFF. My existing iTunes library seems to be a mixture of AAC, MP3 & wav.
Would someone please explain the basic differences between AAC, AIFF, & MP3 as it relates to compression, files size, transparency, sonic quality, etc.
I want to maintain quality, as well as maintain compatibility with an iPod. Thanks.
I ripped my vast collection of CDs to ALAC (Apple Lossless) and can discern no difference in sound, it is after all lossless and should be identical to the CD. They are both digital versions of the same music, just in different formats.
I prefer lossless no matter whether it is ALAC, FLAC or APE because I love Classical, Jazz and other music where detail is important.
ALAC lets iTunes manage my music and convert it to the best format for any portable devices on which I may listen to it.
Unless you are listening to your music on Hi-Fi equipment, with top speakers in an otherwise quiet setting, the whole issue of fidelity is frankly a load of old bollocks.
We live in very noisy environments, with most people listening through earpieces out in public. The fact you can hear anything beyond the background muzak you'd get in a lift is a miracle.
To test just how well your audio experience is, play something with normal voice eg an audiobook and see how much you can make out in the surroundings and with the equipment you have. In most cases you can't get most of it. So really fine nuances in pitch and tone are moot.
AIFF and
WAV are exactly the same as the original CD.
ALAC is exactly the same but compressed to about half the size, it is uncompressed on the fly, which makes no difference to you the listener, but saves hard drive space.
AAC uses a slightly better compression and frequency algorithm that edges out MP3 at lower sampling rates, which is why it is used in DAB, x264 video and telecommunications, to get the smallest file size. You can choose the sampling rate, the lower the number the smaller the file but also the worse the sound. You get metallic artifacts in the sound at too low levels (under 64 kbs), particularly with music. Voice requires a smaller frequency range and can be encoded at a much smaller sampling rate (32kbs really is the minimum for me).
MP3 is not as good as AAC but if it has variable sampling rate gets close. AAC's main trick is actually variable sampling rates. MP3s ubiquity is its main advantage.
OGG-Vorbis is another good and open source format, but nowhere as widespread and supported.
Every format depends on good software to create it, as a great deal depends on the trade-offs made to compress lossy sound.
Except for lossless files, everything else damages and reduces the quality of the sound. Recompressing or changing the format of music from one lossy format to another is particular bad for the sound. Think of it like layering a screened pattern over another screened pattern, the artifacts just get very noticeable.
It is very easy to test all of this for yourself. RIP some CD with which you are extremely familiar to all different formats and bitrates, labelling them as you go. Then play them back on the same hardware and see what your impressions are.