Digital and Analog Filters - MATLAB & Simulink
MATLAB Filter Design Wizard for AD9361 [Analog …
A new digital resonant filter structure is also proposed for subtractive synthesis. It is a modified version of the nonlinear digital Moog ladder filter introduced previously by Huovilainen (2004). The new structure reduces the computational cost of the nonlinear digital Moog filter by using a single nonlinearity instead of five nonlinear functions inside filter sections. The new digital Moog filter structure also decouples fairly well the cutoff and the resonance parameters and offers several response types by selecting a weighted sum of different output points.
MATLAB Filter Design Wizard for AD9361
All of the a's are zero for a FIR filter. The main advantage of IIR filters is that they can produce a steeper slope for a given number of coefficients. The main advantage of FIR filters is that the group delay is constant. This provides the capability of obtaining both a steep cutoff and perfect phase response. This is impossible to achieve with an analog filter.
It covers the synthesis of analog filters free ..
Design and implementation of digital subtractive synthesis are more demanding than is generally understood, because imitating analog electronics with digital processing is not as easy as it may seem. One problem is aliasing caused by sampling of analog waveforms that have rapid changes. The spectra of such waveforms contain infinitely high frequencies, and the signals are thus not band-limited. Another difficulty is that analog filters do not obey simple linear theory. With high signal levels they generate distortion. This does not naturally occur in digital processing, but it must be designed and implemented on purpose (; ).
Filter Design - Circuit Sage - Analog/RF Design …
It is important to realize that a digital filter is a completely different animal than an analog filter. The order of an analog filter directly determines the slope (dB per octave) of the filter attenuation beyond cutoff. The number of taps of a digital filter are related to the slope, but the slopes are very dependent on the cutoff frequency. For a given sample rate, a high-pass at 300 Hz requires twice as many taps as a high-pass at 600 Hz, to achieve the same slope. And for a 300 Hz high-pass it takes almost 40 taps to provide the same slope as a 4th order analog. For a given number of taps, high-pass and low-pass filters have different slopes. It is easy to to achieve a perfect phase response with a digital filter, and impossible to achieve with any analog filter of order greater than one. It is possible to emulate a analog filter response with a digital filter if you want, but why?