Sound is either analog or digital. Analog has an infinite possibility of values whereas digital has to be a single value at a time depending on how accurately your measuring it. We speak in analog but computers can only read digital signals so we must first convert our analog voices into digital signals if we want to store/play them throgh a computer. We use a transducer to do this which could be a microphone.
We use a microphone to pick up the analog sound then a ADC converts it to digital signals using the lower part of the possible frequencires it could be. The computer then uses a DAC to play it back through a speaker.
The two factors that influence the quality of the sound are the sampling rate which is how often per second you take a sample of the sound and the sound resalution which is the number of bits availiable to encode each sample you take.
We must sample the frequency of the sound at least twice the frequency of the highest part of the sample; so if the frequency is 200hz - 2khz then we should sample it at 4khz per second and store it using 16, 32,64 bits for each sample.
We use mpeg files to store sound and they discard any sound/frequency that we can't here so the file size is about 10% the size of the original WAV file type.
Digital sound is easier to mix because it can only be certain values and this can help music companies and djs.
To synthesis sound using MIDI we store the instructions to make the sound not the sound itself, this is like storing drawing instructions for vector graphics.
Again, an excellent summary, to the point, in your own words and accurate - well done
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