Lecture Three Flashcards
What is the function of a filter”?
To remove unwanted frequency components from a signal and hence improve the signal to noise ratio
What are the types of filters?
Hardware (analogue) filters (analogue signal) Digital Filters (digital signal)
What is the best domain representation of filtering?
Frequency Domain
Signal and noise can be well discriminated in this domain
What is the amplitude response of the ideal filter?
The amplitude response of the ideal filter is a rectangular function (bandpass) to seperate signal from noise.
An ideal filter will produce no phase shift.
How can the filter be mathematically represented?
The frequency domain representation of the filtered signal is the product of the fourier transform of the input signal X(f) and the filter response H(f)
i.e multiple desired frequency by one and noise by zero
How is a filter on the frequency domain described?
Pass band (signal desired) Stopband (noise removal)
What are the other types of filters and their characteristics?
Low pass filter (stopband high Hz)
High pass filter (passband high Hz)
Band Pass filter (keep a band of Hz)
Band Stop (notch) Filter (remove a notch of noise)
What are the characteristics of the ideal filter?
- Gain is frequency independent in the pass band and 0 in the stop band (i.e multiple by one and zero)
- Produce no phase shift, this may not always be possibly but a linear phase shift in the passband represents a simple delay. i.e the same delay at all frequencies. (radians in shift)
- Doesnt alter quality of signal
What happens in reality with digital filters?
The filter is unlikely to remove all frequencies outside of the passband. Instead the response is likely to roll off at the cut off frequency.
How do we generate the time domain properties of a filter?
Use fourier transform properties
Describe the relation of the time and frequency domain using notation;
We stated that
Hz domain;
Y(f) = X(f) x H(f) Hz response
i.e multiple the Hz domain to remove noise
Time Domain;
y(t) = x (t) . h(t) Impulse response
i.e convolution with the impulse response to remove noise
y(t) = filtered signal x(t) = unfiltered signal h(t) = filter . = process of convolution
The impulse response h(t) for a filter H(f) specified in the Hz domain is simply the inverse fourier transformation
What is convolution?
The filter data set is formed by convolving the the raw data with the filter weights.
These filter weights are sampled representations of h(t) the impulse response of the filter.
i.e three points moving average
y(i) = 0.25x(i-1) + 0.5x(i) + 0.25x(i+1)
i.e it moves along the line of data multiplying it by x amount of numbers either side to average it. Removes noise between samples
DONT NEED TO REPRODUCE EQUATION
Redescribe convolution in terms fo y values
Each value of Y is evaluated as the weighted average of the corresponding input data according to filter weights h(k)
h(k) = 1 for low pass and 0 for high pass
Whats another name for a filter that involves convolution?
It is because NON-RECURSIVE filters are implemented using finite set of filter weights that they are also called finite impulse response (FIR) filters
How do we analyse the performance of filters in the time domain?
We need to obtain the Hz response we fourier transform its impulse response