It happens that many people have no idea what Digital Signal Processing means, although they hear more and more often these words, today. Considering the name, Digital Signal Processing, people are lead towards thinking that this procedure deals with digital signals. No; Digital Signal Processing is a method of improving the quality of the analog signals, only. Indeed, the naming used is rather improper, because it relates to all types of digital processing, only that the methods, and the techniques used in DSP deal only with signals that are analog in nature. In the digital signals case, we can only compress, encrypt, and translate them to other digital formats; these (different) procedures do not require any DSP techniques. Using the DSP name when referring to digital signals causes confusion.

{mosgoogle}Let’s take each of these one step at a time, and using few practical examples. Suppose that we have an old vinyl record and we want to copy its analog signal on a digital CD, to better protect that recording--CDs are a lot more reliable to hold information unaltered, over time. This means that we need to convert the analog signal to digital format, and the best way of doing it is by using DSP techniques, as follows. First, we need an analog-to-digital hardware module to convert the analog signal into digital format--this is typically a “codec”--then we select a specific scanning frequency, to accomplish this task. Because we work with audio frequencies, a 40 KHz scanning frequency should be sufficient, and please note that the scanning frequency needs to be at least double than the maximum frequency of the original analog signal--the analog audio signals have frequencies within the range of 10 Hz to 16 KHz. After scanning, we have the copy of the analog vinyl record, in digital data format, expressed as a series of digital integer values in binary format.

Unfortunately, our vinyl record is fairly old, and it has a lot of noise on it; this noise is also present on the digital copy, and it needs to be filtered out, before we burn the digital CD. The next step is to take the digital copy--please note this: the digital copy represents the analog signal--and we apply to it a mathematical transformation function: in this way, we change digital data from “time-domain” to the “frequency-domain”. This is done gradually, by chopping digital data into frames of 512, 1024, or 4096 integers in size, and transforming one frame at a time. Once we have the data in frequency-domain, it is easy to filter the noise out, and to select/amplify only the audio frequencies we want; for this we use digital firmware or software filters, which are, in fact, known mathematical algorithms.