Sound

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Converting analogue to digital sound

The following image demonstrates how sound files are created when they are inputted through devices such as a microphone:

600px-Analogue Digital Conversion.png

Representing Sound

To store sound a digitizer is needed to convert the analogue sound into digital sound. An analogue to digital converter carries out these conversions. 16 bit ADC is enough for CD quality sound.

Sampling Rate

The sampling rate or frequency is the number of samples taken per second. It is measured in hertz (Hz)?. The higher the sampling rate the more accurate the representation of the sound.

Sampling Resolution

The sampling resolution is the number of bits assigned to each sample. The number of bits assigned allows for a wider range of sounds to be displayed. For example, if only one bit is used it can only be either 1 or 0, giving an option of only two different sounds, where as if 8 bits are used there are a possible 255 different sounds that can be recorded and then replayed.

Nyquist's Theorem

In 1928 Harry Nyquist found that in order to sample any sound you must use a sampling rate at double the frequency of the original sound. The result will be the closest possible to the original sound.

Audio Compression

Audio compression removes sounds and frequencies which we can't hear.

Synthesising Sound

Sound can be synthesised with MIDI (Musical Instrument Digital Interface), which records information about each note - such as duration, pitch, tempo, instrument and volume - and recreates that note when played. When using MIDI it is hard to replecate the proper soud as it would have to be played through note by note and would be a synthetic sound, in some cases it's good like in pop music but in other cases usually not. A MIDI link can hold up to 16 channels of information which can be routed to a seperate device for each channel.