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Digital to Analogue Convertor - Analogue to Digital Convertor

Digital - Has two states ON or OFF - Like a light.
Analogue - Has limitless states between a minimum (usually 0) and a maximum.

Why is everything Digital?

Digital signals are much easier to transport around and are highly resistant to loss due to poor transmission links. Analogue signals vary from nothing to maximum and very quiet signals (close to the "nothing" end) tend to get swamped by interference and noise as they travel through the transmission medium. Hence the trend is to convert very delicate analogue signals to the almost "bomb proof" digital version as soon as possible.


To get from one method of transmission to the other requires a convertor - either Analogue to Digital (A-D) or the reverse, Digital to Analogue (D-A). No process is perfect and the quality of conversion has a direct bearing on the accuracy (and hence musical quality) of the resultant signal.

An Analogy - Suppose you were given the task of building a curved wall for your garden - around a tree trunk for example and you were given large blocks (building breeze blocks) to do it with. What would the completed structure look like?
It would look circular from a distance, but close up it would be a collection of straight bits arranged in a circle and would look very rough!
Now if we used normal house bricks instead - what would the difference be?
This time the wall is much closer to being circular - yes it is still a collection af straight bits arranged in a circle but now the circle looks much better.
If we were to use even smaller bricks - the circle would start to look perfect. BUT - it will never be absolutely circular when we use bricks with straight edges.

Similarly with convertors - the more bits we use then the smaller is the size of brick. So an 8 bit convertor is like using large breeze blocks whilst a 24 bit convertor can be likened to house bricks. No doubt in time the bit rates will rise higher to give an even better approximation. An additional complication is sampling frequency. It is all well and good sampling the signal using lots of bits, that means we can measure the amplitude very precisely, BUT, only at a single instant in time. At the next instant, the signal might be louder or quieter so we need to measure it again, and again. Obviously the more samples we take, the more accurate is our digital representation of the original signal but at the expense of a higher sampling rate which implies much more data to process.


It is possible to apply filters to the convertor output (D-A) to get a better approximation. Using our analogy above this is like plastering over the bricks with mortar to smooth out the irregularities. Unfortunately, no matter how good the builder is, there will always be rough bits on the wall - it will never be a perfect circle. Similarly our convertor may have a number of filter circuits built in to smooth out the rough bits but it cannot achieve perfection. Of course the higher the bit rate, the less there is to fill in and smooth over so obviously we need to focus on bit rate as an initial guide to sound quality.

Choosing a Convertor

Most equipment these days have convertors built in so the issue of choosing one never arises. However, some systems will allow the insertion of a stand alone unit and so what should we look for? Like any other piece of equipment, the basic specifications tell us a lot about technical performance but our ears tell us most and again it is impractical to personally test every convertor in your own listening space. So the best we can do is to compare specifications.

With convertors, it is the high frequencies that present most problems. The convertor is like a person taking a series of photographs - one after the other. Each "picture" happens in an instant of time (very fast) and records exactly what is seen. Likewise the convertor (A-D) takes a "photograph" of the audio, then goes away and works on it. First, the convertor looks at the sample and compares it with a table it has of values. It finds the nearest value and then looks up on another table what the numeric equivalent is in digital speak. So the original sample is converted into a digital "word" and sent out as a data stream. The convertor then takes another "photograph" and repeats the procedure to send another digital word out and so on. The analogue input signal (our music) is thus converted to a continuous stream of digital information which consists of a very large number of ON or OFF pulses which represent our music.

At the very highest frequencies of music (the overtones) our convertor even though working very fast progressively struggles to take very good "photographs" - the action is hapenning just too quickly and compromise is the order of the day. So high frequencies tend to get altered (blurred is a good word) by the process and start to sound wrong. The quality of the convertor is therefore assesed by how musical it sounds and especially at the higher frequencies.

In general then, the higher the conversion bits ie 24Bit and the higher the bit rate ie 96K, the better will be the audio quality. Note though that at these rates, a huge amount of digital information is needed and systems need to cope with very high clock rates. It just so happens that the higher the clock rate, the more prone digital equipment is to radiate interference (just like a radio station) and the more likely analogue and especially low level (small signals) analogue is to pick it up in the form of interference. Analogue and digital mix very badly and the design to seperate out one from the other very specialised.


Distortion - Is the key parameter and needs to extremely low as distortion due to digital abberations is awful.
Noise - Another key parameter as it relates to the smallest "photograph" allowed - ie the very first value on the table after value 0 (no signal at all). Go for 10dB quieter than normal - more if possible.
Dynamic Range - the difference between the quietest and loudest outputs allowed. The higher the figure the better.
Others - All sorts of digital measurements here - jitter etc. This relates to how badly the digital signal can be before the end product - audio output - collapses. The better the error correction and lower the jitter then better will be the output.

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