Audio compression

Introduction
There is a lot of confusion surrounding the terms audio compression, audio
encoding, and audio decoding. This section will give you an overview
what audio coding (another one of these terms…) is all about.

The purpose of audio compression

Up to the advent of audio compression, high-quality digital audio data took
  a lot of hard disk space to store. Let us go through a short example.
You want to, say, sample your favorite 1-minute song and store it on your
  harddisk. Because you want CD quality, you sample at 44.1 kHz, stereo,
  with 16 bits per sample.
44100 Hz means that you have 44100 values per second coming in from your sound
  card (or input file). Multiply that by two because you have two channels. Multiply
  by another factor of two because you have two bytes per value (that’s what 16
  bit means). The song will take up 44100 samples/s · 2 channels
  · 2 bytes/sample · 60 s/min ~ 10 MBytes of storage space
  on your harddisk.
If you wanted to download that over the internet, given an average 56k modem
  connected at 44k (which is a typical case), it would take you (at least) 10000000
  bytes · 8 bits/byte / (44000 bits/s) · / (60 s/min) ~ 30 minutes
Just to download one minute of music!
Digital audio coding, which - in this context - is synonymously called digital
  audio compression as well, is the art of minimizing storage space (or channel
  bandwidth) requirements for audio data. Modern perceptual audio coding techniques
  (like MPEG Layer III) exploit the properties of the human ear (the perception
  of sound) to achieve a size reduction by a factor of 11 with little or no perceptible
  loss of quality.
Therefore, such schemes are the key technology for high quality low bit-rate
  applications, like soundtracks for CD-ROM games, solid-state sound memories,
  Internet audio, digital audio broadcasting systems, and the like.

The two parts of audio compression

Audio compression really consists of two parts. The first part, called encoding,
  transforms the digital audio data that resides, say, in a WAVE file, into a
  highly compressed form called bitstream. To play the bitstream on your
  soundcard, you need the second part, called decoding. Decoding takes
  the bitstream and re-expands it to a WAVE file.
The program that effects the first part is called an audio encoder.
  LAME is such an encoder . The program that does the second part is called
  an audio decoder. One well-known MPEG Layer III decoder is Xmms,
  another mpg123. Both can be found on ww.mp3-tech.org .
Compression ratios, bitrate and quality

It has not been explicitly mentioned up to now: What you end up with after
  encoding and decoding is not the same sound file anymore: All superflous information
  has been squeezed out, so to say. It is not the same file, but it will
  sound the same - more or less, depending on how much compression had
  been performed on it.
Generally speaking, the lower the compression ratio achieved, the better the
  sound quality will be in the end - and vice versa. Table 1.1
  gives you an overview about quality achievable.
Because compression ratio is a somewhat unwieldy measure, experts use the
  term bitrate when speaking of the strength of compression. Bitrate denotes
  the average number of bits that one second of audio data will take up in your
  compressed bitstream. Usually the units used will be kbps, which is kbits/s,
  or 1000 bits/s. To calculate the number of bytes
  per second of audio data, simply divide the number of bits per second by eight.

   
 
  Table 1.1: Bitrate versus sound quality
 
    Bitrate
    Bandwidth
    Quality comparable to or better than
 
 
    16 kbps
    4.5 kHz
    shortwave radio
 
 
    32 kbps
    7.5 kHz
    AM radio
 
 
    96 kbps
    11 kHz
    FM radio
 
 
    128 kbps
    16 kHz
    near CD
 
 
    160-180 kbps

      (variable bitrate)
    20 kHz
    perceptual transparency
 
 
    256 kbps
    22 kHz
    studio