Digital sound describes sound recording and reproduction systems which work by using a digital representation of the audio waveform. The adaptation of digital sound technology occurred during the 1980’s. Before this time audio signals were created through the use of analogue oscillators. These signals were processed with analogue filters and effects. This approach was replaced with completely digital audio hardware systems, which the signal was generated as digital information and was only converted to analogue electric signal in the last stage by a digital to analogue converter. If not created digitally then files would be converted from analogue to digital to be stored, mixed with other digital signals, manipulated by a digital signal processor, and transferred across interconnects as numerical information. When received digital to analogue converters will accurately reconstruct an analogue signal from the digital data stream of numbers. (en.wikipedia.org)
     Why use digital audio instead of analogue? There are a number of different advantages of digital over analogue sound. These include storage on an inexpensive media, nondestructive editing, transmission via phone line or internet, multiple copying without quality loss even across different formats, and unlike analogue which is prone to degradation when carried from medium to medium, digital data if carried without numerical error, by theory will not change. All digital recorders, digital mixers, and digital processors must use either built in converters or outboard dedicated converters to input analogue audio and play it back. (apogeedigital.com)
     Digital sound quality is determined by three main factors, these factors are sampling rate, bit rate, and compression. Sampling rate is the number of times each second the computer stores information about the sound wave, the higher the sampling rate the better the sound quality that will be heard. Bit rate refers to a computer binary number system of ones and zeros which combine together to create long strings called bits, the higher the bit length the better the quality. Compression refers to a computer using different schemes to eliminate repetitive information when storing data in turn making files smaller and easier to download. Any form of compression will have an affect on the sound quality of the reproduced sound so users must balance the desire for high quality and the need to minimize the size of the recorded file. (camil.music.uiuc.edu)
    There are many different kinds of digital sound files that can be used. These include au, aif, mp3, ra, and wav files. AU files are the most common sound format on the web. This file type was created by Sun Microsystems. Most web browsers include capability to play AU files directly so it is a good choice for internet work that will be received by a large net audience.  This type of file can be larger than other types of audio files so it is best used for short sound clips for effective download times.  The format specifies arbitrary sampling rates and multi-channel sounds. Normal sound formats use linear encoding but AU file format uses u-law and a-law which is logarithmic. Logarithmic means that the spacing between different sound levels grows larger as the values increase which provides a larger dynamic range than normal 8-bit samples, more equivalent to 12-bit samples, but the format suffers from more noise than linear encodings.
    AIF is another fairly common sound format found on the Web. This standard Mac Digital audio requires the same programs as .au to play. Because the format does not support any kind of compression, it tends to produce large files. The AIF format consists of a series of so-called data chunks that store information about the recorded sound: sample size, sampling rate, stereo/mono etc... The data chunks are followed by the actual sound data. The sound data is stored as samples, either single samples in a mono sound file or pairs of samples in a stereo file. Both are lossy algorithms, but provide reasonable quality with great space savings. Lossy compression means the compressed sound will not sound exactly like the original.  AIF files can be played by many audio players such as Quicktime and can also be easily converted to other formats such as CD audio files and mp3.
    MP3 is the most popular file format on the Web for distributing CD-quality music. A 1 Mb file is equal to about one minute of music. This type of file requires an MP3 player. MPEG audio is a standard for high-quality audio and video files that has gained widespread use. The MP3 format uses compression to minimize file sizes, while retaining good audio quality. On average, files are reduced to about 10% of their original uncompressed size, a three or four minute stereo clip that would take up about 30 to 40 megabytes on an audio CD will only be about three or four megabytes as an MP3. MP3 uses a lossy compression scheme based on perceptual encodings, which can achieve high rates of compression without a noticeable decrease in quality. These techniques contribute to the near CD audio quality that has made the MP3 format extremely popular.
    RA is a proprietary streaming audio format. RealAudio allows you to play sound files in real-time and requires the use of a RealPlayer. It was one of the first audio and video format specifically designed for Internet use and has gained widespread use. The files are played using a free “plug-in” that is available from their Web site and you can obtain a free encoder to convert files from WAV to Real Audio format. Real Audio files use compression schemes to make the file small for Internet use, some loss of quality is apparent, but you can control what type of compression you wish to use for the file, depending on the target audience and recorded contents. Most web servers include Real Audio Servers which means that audio can be streamed. In other words you don't have to download a complete file in order to listen to it and you can get 'instant play'.
    WAV is the native sound format for Windows. Because the WAV format is used in the Windows operating systems for recording and playback of recorded sound an advantage to using WAV is that the file type is already located on your user’s machine if it is a Windows computer. A disadvantage is that many people on the Internet will be unable to play the file if they are using another operating system and the fact that WAV files are typically larger, taking longer for the end user to download.  The Microsoft WAV uses Adaptive Delta Pulse Code Modulation which returns a high quality signal with very little processing power required for fast decoding. WAV files may be compressed or uncompressed, but even when compressed are still comparatively large. (w3schools.com)
    As you can easily see digital sound is not only much better for quality sound of files but it has many advantages over analogue. Today all that we use is digital files for obvious reasons, however it is surprising to see how many people do even know what digital audio is or how it compares to analogue. I guess for the majority of people as long as it works they don’t really care how it works.