1 Introduction

The process of converting analog signal to digital signal involves two steps: sampling the input analog signal and quantization of the sampled signal to a digital value. Sampling converts the continuous-time signal to discrete-time samples spaced by the sampling frequency whereas quantization digitizes the sampled signal to a discrete amplitude value. Theoretically, the Sampling Theorem dictates that the sampling frequency (FS) can be chosen at least twice as high as the maximum input signal frequency content (FB). However, any out-of-band signal, interference, or noise from N×FS ± FB (N = 1, 2, 3, and so forth) is folded back into the desired band, overlapping it with the input signal. Figure 1-1 depicts this folding of the out-of-band frequencies greater than FS into the in-band frequency (FB). To limit the interference of these out-of-band frequencies with the in-band frequencies, a low-pass filter is used, which greatly reduces the out-of-band frequencies before the input of the analog-to-digital converter (ADC). By selecting large order low pass filters, the aliasing components are greatly attenuated, but higher order filters come at higher cost.

Figure 1-1 Folding Back of Noise and Interferer Due to Sampling