Introduction
Sounds that reach our ears are signals in the form of air pressure. This air pressure is a continuous signal, i.e. it is defined for all time instances and can have any value on a reasonable domain. These kind of signals cannot be stored directly on a computer, which uses numbers to perform calculations. The signal first needs to be approximated in two ways in order to allow for computations on a computer and to limit the required memory.
Module overview
The concepts covered in this module are:
- Mathematical description sampling process [⯈] - The conversion of a sample frequency can give rise to several undesired artifacts. The correct filtering of the signal is therefore of crucial importance.
- Basic building blocks of multirate signal processing [⯈] - In order to perform the sample conversion, several common filter architectures can be utilized, which will be discussed in this section.
Summary
A continuous-time signal with frequencies not higher than can be reconstructed exactly from its samples , if samples are taken at a rate , that is larger than .
If the sampling frequency is lower than two times the highest frequency of the analog signal , an overlap of spectral components may occur and cannot be recovered from its samples .
This overlap in frequency domain is called aliasing.
C/D conversion
Note: In practice analog LPF needed before input to prevent frequencies .
D/C conversion
Interpolation equations ideal LPF
Practice D/C conversion: Ideal LPF Zero Order Hold filter:
Relation analog vs discrete-time filter
Bandlimited and LPF in D/C
Sample Rate Conversion
SRD
SRI