Wave & MP3 Formats

The term "format" indicates the coding pattern used to store data in a file.

The term "audio file" indicates a file that contains data used to represent sounds, music, and songs.

Audio files come in a variety of formats. The two most popular audio formats are "Wave" file format (synonymous usage: Wave file and Wave format), and MPEG file format (abbreviated in this documentation as MP3). Both formats are digitized representations of "analog" sounds. A simple difference between them is that a Wave file represents the sounds (waveforms) as they occurred in real-time, whereas an MP3 file is essentially a compressed version of a Wave file. Consequently an MP3 file is much, much shorter than a Wave file.

The MP3 compression is performed according to a specification lay down by the Moving Picture Experts Group (MPEG, pronounced 'em-peg'). This compression is performed with the help of a "codec", so called because it helps to compress and decompress an audio file.

The MP3 format is a highly compressed type of audio that takes up far less disk space than Wave files. It is also a "lossy" compression algorithm, meaning that (in theory) your MP3 files will not sound quite so good as your wave files. Practically speaking, you probably won't notice the difference unless you choose a really low bitrate for the MP3 conversion!

Parameters for Wave-format files:

Sampling methodology:

This describes how the analog signal was sampled to derive the digitized value. This is almost always PCM.

Sampling rate:

The number of samples evaluated per second. Practical unit is kilo hertz (kHz). Common rates are 12 kHz, 24kHz, and 44.1kHz.

Sampling resolution:

The number of "bits" used to express the numerical value of the sample. The unit is bits per sample (bpS). Common resolutions are 8bpS and 16bpS or simply 8-bit and 16-bit.

Number of channels:

The number of parallel data streams which compose a single performance. Common numbers are 1 (monophonic) and 2 (stereophonic).

Parameters for MP3-format files:

Codec:

The compression-decompression algorithm (methodology) used to record and playback the MP3 audio file.

Bitrate:

This is the compression rate. It is the number of bits, which will be used to express 1 second of sound. The practical unit is kilobits per second (kbps). Common bitrates are 128kbps and 256kbps, but severe compression can go right down to 20kbps.

Sampling rate:

The sampling rate is the number of sound samples generated per second. The practical unit is kilohertz (kHz). Typical rates are: 48kHz, 44.1kHz, and 32kHz.

As a simple comparison of an audio file before and after compression:

Sampling resolution: 16bpS

Sampling frequency: 44.1kHz

Size for 1 second of audio:

16bpS X 44100S = 705600 bits

With MP3 compression:

@128kbps:=> 128000b; comp. ratio = 5.5:1

@ 20kbps:=> 20000b; comp. ratio = 35.25:1

This is based on simple arithmetic; the actual compression ratio can go much higher than that depending on the content of the song! In practical terms the typical compression ratio is around twice the arithmetically-derived ratio !

[NOTE: Some Wave-format files are indeed compressed, but these (generally) use proprietary compression algorithms and hence are not popular.]