Audio Compression can be one of two categories, lossless or lossy. Lossless compression algorithms return audio data that is identical to the original audio information. Lossless compression sounds wonderful, however it does not yield an extremely high compression ratio. Typically, lossless routines give a ratio no higher than 2:1 for realtime playback for audio with a full dynamic range.
Lossy Audio Compression can yield a variety of different rates of compression based on the ability of your hardware and software to decode and playback the music audio data in real-time. There are various compression schemes to achieve different results; for example, many codecs remove portions of the audio signal that human hearing is less sensitive to. To some peoples ears the resulting audio has distinguishing artifacts or characteristics that change the listening experience. The final "quality" level is largely a matter of personal perception.
Each different encoder/decoder (CODEC) has different strengths and weaknesses. Typically, ENCODING takes a long period of time and large amounts of processing power. DECODING must be real time or faster -- able to decode and play simultaneously -- to be useful for an "on demand" digital audio playback system.
MPEG
The Moving Pictures Experts Group (MPEG) put a considerable amount of research into ways of compressing video and audio to make movies fit onto compact discs.
A wide variety of software and hardware encoders and decoders currently exist.
For a very technical and detailed review of MPEG, you can read the online FAQ about MPEG.
Strangely (notice the "P" for pictures) the compression levels for video are well over 20:1, and MPEG audio compression loses quality quickly at compression levels higher than a 6:1 compression ratio.
MPEG Audio compression continues to degrade until it becomes absolutely dismal at 24:1 compression.
An extension to MPEG-1 and MPEG-2 audio called 'Layer 3' (commonly known by its file extension, MP3) has been developed allowing greater compression levels and more BIT DEPTH options.
Software routines for MPEG audio decoding are fast to decode with relatively low quality loss, but generally speaking it takes a LONG time to encode.
MPEG-4 has been ratified as a standard, and new codecs will become available in the not-too-distant future.
The compression ratio for MPEG compression can be adjusted across a wide range of quality levels, however sound quality drops as the compression is increased.
Since MPEG audio uses a perceptual model, the exact perceived amount of lost quality varies for each individual listener.
Trying to maintain the highest quality (without extremely noticeable signal loss) yields the following compression ratios:
Layer 1 = 1:4 (384 kbps for stereo)
Layer 2 = 1:6 or 1:8 (256 kbps or 192 kbps for a stereo signal)
Layer 3 = 1:10 or 1:12 (128 kbps or 112 kbps for a stereo signal)
MPEG I Layer 3 by Fraunhofer
This format was created by the Fraunhofer IIS, and is marketed by Opticom.
Online info is located at http://www.iis.fhg.de/amm/techinf/layer3/index.html
Their product, MP3 Producer, is the reference MP3 encoder.
The "free" MP3 ACM codec that Microsoft provides does not have the full range of compression options for encoding.
The Producer "professional" version does have the complete range of bit rate options, along with the abiltiy to save raw MP3 files or MP3 files in WAV format.
Many MP3 advocates would like you to believe that MP3 is "true CD Quality audio."
Unfortunately, that claim isn't quite true.
A more factual claim would be "near CD quality at bit rates of 128 kbps or higher."
The quality issues with MP3 are typically not because of a limitation of the compression format, but a standard practice of compressing the audio to a bit rate lower than 128 kbps.
MPEG I Layer 3 VBR by Xing
The Xing MPEG encoder has a feature called VBR (Variable Bit Rate) encoding.
Acknowledging the limitations of extreme audio compression, they have created an encoder that will adapt to use less compression for complex audio signals.
This format has the advantage of encoding very quickly, as well as using less compression when appropriate.
The drawback to this format is the irregular time/size issues created when using random access playback.
TwinVQ
The Transform-domain Weighted Interleave Vector Quantization format is more commonly known by its file extension, VQF.
The plus side of this format is higher compression than MP3 while providing better subjective audio quality.
(e.g. VQF @ 12:1 compression is closer to CD quality than MP3 @ 10:1)
The hardware requirement of VQF is higher, including much longer encoding times and more CPU cycles required for playback.
