Some advice on audio formats and converting tracks

Question

_*Some thoughts about audio formats and conversions. These are all supported by iTunes unless stated otherwise.*_
General principles : there is no point in converting an already compressed track into a higher bit rate - it may seem as if this would give an improved quality, but once information has been lost through compression, there is no recovering it. The sound MAY be improved subjectively through the use of an appropriate EQ setting.
This is not a definitive list : other formats, for example the one used by Real Player, and the ATRAC minidisc codec used by Sony, are not listed : many of the following can be used in iTunes, or are reasonably well-known.
To use one of the available formats, they should be set in the Importing tag in *iTunes Preferences* (Advanced prior to iTunes 8, now in General). Once set, the format is used for importing CDs, but can also be used to convert tracks already in iTunes (using the Advanced menu : +Convert Selection to xxx+ ). It is not used when dropping files onto iTunes, or using Add To Library.
1. _Lossless formats_
Certain high-quality audio formats are "Lossless". This means that if the track is re-converted into the same format (after editing, for example), it will not degrade no matter how many times that is done.
.AIFF - a format (common in Macs) representing full CD quality. Bit-rates are over 1400 kbps; +1 minute of music is approx 10MB.+
.WAV - more or less the same as .AIFF but a Windows format.
*Apple Lossless* - a 'once-only compression', applied to .AIFF or .WAV tracks to retain quality but reduce file sizes. A little (totally inaudible) information is lost, but also a more efficient 'codec' (compression algorithm) is involved. +1 minute = approx 5MB.+
.FLAC - similar to Apple Lossless, but less proprietary (a more open standard). Not supported by iTunes.
2. _'Lossy' formats_
These formats are so-called because each time the conversion is done, the track is re-compressed, and more information is lost even if the file size remains the same. Various quality settings can be chosen, from around 320 kbps (high) down to below 128 kbps, though 128 is reckoned to be the lowest acceptable bit rate to listen to music in stereo.
In general, the higher the input quality, the higher the output quality when 'stepping down'. Therefore an .AIFF track converted directly to 128kbps 'lossy' will sound better than a 256kbps track re-compressed to the same 128kbps.
.MP3 - a standard audio compression format that has been around for many years. Common everywhere, and supported by virtually all contemporary music players. The codec has been improved so that modern MP3s sound markedly better than earlier versions. +1 minute @ 128 kbps = approx 1MB.+
.AAC - a superior codec to MP3 (though the quality gap has narrowed), which has been chosen by Apple as the iTunes format of choice. Nowadays, a track at 128 kbps .AAC will sound around the same quality as the same track at 160 kbps MP3. +1 minute @ 128 kbps = approx 1MB.+
.WMA - Windows' own proprietary lossy codec. It seems to have a bright and sparkly sound compared to MP3, but lacks a decent 'bottom end', i.e. the bass tones are somewhat lacking. Not supported by iTunes. (Can be played using VLC).
.OGG - the open standard Ogg Vorbis format, often held to be superior to MP3 but little-used and therefore not supported in many platforms and players, including iTunes.
When considering which format to use, the most important things to remember are
1. how will these tracks be played back? (on a hi-fi? an iPod? expensive earphones?)
2. the quality will depend finally on the judgement of the listener (some ears are better at telling small differences in quality than others)
3. how much space is available to store (and expand) the music library?
As for quality, a good way to decide is to "audition" different formats. Using one CD track known to you, try importing it at several different formats and bit rates (using iTunes Preferences). Then just play them, see which sounds best to you. In this way I came to decide for myself that AAC @ 256kbps was the best for me +(1 minute = approx 2MB).+

Answer 1

The Windows version of iTunes supports importing of WMA files, converting them to one of the other formats. Mac iTunes offers now support of WMA files at all.
Good point - and if I knew more about iTunes on Windows I would amend that and post a different version of the article in their forum.
I accept the point about .WAV files being used also for compressed formats but didn't want to make it too complicated; the most common use for .WAV I believe, is full quality audio.
On the subject of Lossless compressed audio (FLAC and AL), Wikipedia has this :
_*Difficulties in lossless compression of audio data*_
+It is difficult to maintain all the data in an audio stream and achieve substantial compression.+ +First, the vast majority of sound recordings are highly complex, recorded from the real world. As one of the key methods of compression is to find patterns and repetition, more chaotic data such as audio doesn't compress well. In a similar manner, photographs compress less efficiently with lossless methods than simpler computer-generated images do. But interestingly, even computer generated sounds can contain very complicated waveforms that present a challenge to many compression algorithms. This is due to the nature of audio waveforms, which are generally difficult to simplify without a (necessarily lossy) conversion to frequency information, as performed by the human ear.+
+The second reason is that values of audio samples change very quickly, so generic data compression algorithms don't work well for audio, and strings of consecutive bytes don't generally appear very often. However, convolution with the filter [-1 1] (that is, taking the first difference) tends to slightly whiten (decorrelate, make flat) the spectrum, thereby allowing traditional lossless compression at the encoder to do its job; integration at the decoder restores the original signal.+
+Codecs such as FLAC, Shorten and TTA use linear prediction to estimate the spectrum of the signal. At the encoder, the estimator's inverse is used to whiten the signal by removing spectral peaks while the estimator is used to reconstruct the original signal at the decoder.+
This seems to imply that even Lossless codecs make changes, but it IS a very complex subject. If I could still edit the main article I would remove that bit about AL losing information.