This investigation will study the principles and stages behind A/D and D/A conversions, it will also state the different problems encountered in digital audio, and resolve them by deciphering the key issues. As part of the investigation a practical demonstration will be devised in order to understand the relation between bit rate and levels of definition within the processed signal. .
Analogue to digital converters.
The first step in digitalising an analogue waveform is to change an occurrence that happens in continuous time' into one that is chopped into slices of discrete time'.
This principle is used when filming, for example when someone watches a film they see a smooth motion, but in actual fact they only see 24 photos (frames) per second. (See figure 1).
The problem with chopping continuous time into slices of descrete time is that distortion can occur if the slices aren't small enough, this is known as Aliasing. .
The result of aliasing can cause two problems, the first is that frequencies above half the sample rate (Nyquist frequency) may get into the sample and hold circuit, and the second is that a low sample rate will not be capable at capturing all of the frequencies required, thus both causing distortion. (See figures 2 and 3).
In order to solve the first issue an anti-aliasing filter (low-pass filter) is placed just before the sample and hold circuit, this is used to eliminate high frequencies, which is the unwanted energy above the Nyquist, within the audio signal. .
By using Nyquist theorem it is easy to solve the second problem because it states that in order to stop aliasing a sample rate must be at least double the highest frequency that is present in the signal. As our ears can only detect frequencies up to and around 20kHz, a sample rate of 40kHz is required. This is why most sampling devises sample at 44.1kHz. .
Audio is captured by using a device which samples the voltage of the signal at regular time intervals.