Variable Bandwidth FIR Filter
Design tunable bandwidth FIR filter
Libraries:
DSP System Toolbox /
Filtering /
Filter Designs
Description
The Variable Bandwidth FIR Filter block filters each channel of the input signal over time using the specified FIR filter specifications. This block offers tunable filter design parameters, which enable you to tune the filter characteristics while the simulation is running.
The block designs the FIR filter according to the filter parameters specified in the block dialog box. The output port properties, such as datatype, complexity, and dimension, are identical to the input port properties.
This block also supports SIMD code generation. For details, see Code Generation.
Ports
Input
x — Data input
vector  matrix
Specify the data input as a vector or a matrix. The block treats each column of the input signal as a separate channel. If the input is a twodimensional signal, the first dimension represents the channel length (or frame size) and the second dimension represents the number of channels. If the input is a onedimensional signal, then the block interprets it as having a single channel.
The block accepts variablesize input signals, that is, you can change the size of each input channel during simulation but you cannot change the number of channels.
This port is unnamed until you select one of these parameters:
Specify cutoff frequency from input port
Specify center frequency from input port
Specify bandwidth from input port
Data Types: single
 double
Complex Number Support: Yes
Fcut — Filter cutoff frequency
positive scalar
Specify the cutoff frequency of the FIR filter as a real positive scalar in Hz or in normalized frequency units (since R2023a).
Dependencies
To enable this port, select the Specify cutoff frequency from input port parameter.
Data Types: single
 double
Fc — Filter center frequency
positive scalar
Specify the center frequency of the FIR filter as a real positive scalar in Hz or in normalized frequency units (since R2023a).
Dependencies
To enable this port, select the Specify center frequency from input port parameter.
Data Types: single
 double
BW — Filter bandwidth
positive scalar
Specify the bandwidth of the FIR filter as a real positive scalar in Hz or in normalized frequency units (since R2023a).
Dependencies
To enable this port, select the Specify bandwidth from input port parameter.
Data Types: single
 double
Output
Port_1 — Filtered output
vector  matrix
Filtered output, returned as a vector or a matrix. The size, data type, and complexity of the output signal matches that of the input signal.
Data Types: single
 double
Complex Number Support: Yes
Parameters
FIR filter order — FIR filter order
30
(default)  positive integer
Specify the order of the FIR filter as a positive integer scalar.
Filter type — Filter type
Lowpass
(default)  Highpass
 Bandpass
 Bandstop
Specify the type of FIR filter. You can set this parameter to:
Lowpass
Highpass
Bandpass
Bandstop
Specify cutoff frequency from input port — Specify cutoff frequency from input port
off
(default)  on
When you select this check box, specify the cutoff frequency through the Fcut port. When you clear this check box, specify the cutoff frequency in the block dialog box through the Filter cutoff frequency parameter.
Dependency
To enable this parameter, set Filter type to
Lowpass
or
Highpass
.
Filter cutoff frequency — Filter cutoff frequency
512 (default)  positive scalar
Specify the cutoff frequency of the FIR filter as a real positive scalar in Hz or in normalized frequency units (since R2023a).
If you set the Sample rate mode parameter to:
Specify on dialog
orInherit from input port
–– The value of the filter cutoff frequency is in Hz and must be less than half the value of the input sample rate.Use normalized frequency (0 to 1)
–– The value of the filter cutoff frequency is in normalized frequency units. The value must be a positive scalar less than1.0
.
(since R2023a)
Tunable: Yes
Dependencies
To enable this parameter:
Set Filter type to
Lowpass
orHighpass
.Clear the Specify cutoff frequency from input port parameter.
Specify center frequency from input port — Specify center frequency from input port
off
(default)  on
When you select this check box, specify the center frequency through the Fc port. When you clear this check box, specify the center frequency in the block dialog box through the Filter center frequency parameter.
Dependencies
To enable this parameter, set Filter type to
Bandpass
or
Bandstop
.
Filter center frequency — Filter center frequency
44100/4 (default)  positive scalar
Specify the center frequency of the FIR filter as a real positive scalar in Hz or in normalized frequency units (since R2023a).
If you set the Sample rate mode parameter to:
Specify on dialog
orInherit from input port
–– The value of the filter center frequency is in Hz and must be less than half the value of the input sample rate.Use normalized frequency (0 to 1)
–– The value of the filter center frequency is in normalized frequency units. The value must be a positive scalar less than1.0
.
(since R2023a)
Tunable: Yes
Dependencies
To enable this parameter:
Set Filter type to
Bandpass
orBandstop
.Clear the Specify center frequency from input port parameter.
Specify bandwidth from input port — Specify bandwidth from input port
off
(default)  on
When you select this check box, specify the filter bandwidth through the BW port. When you clear this check box, specify the filter bandwidth in the block dialog box through the Filter bandwidth parameter.
Dependency
To enable this parameter, set Filter type to
Bandpass
or
Bandstop
.
Filter bandwidth — Filter bandwidth
7680 (default)  positive scalar
Specify the bandwidth of the FIR filter as a real positive scalar in Hz or in normalized frequency units (since R2023a).
If you set the Sample rate mode parameter to:
Specify on dialog
orInherit from input port
–– The value of the filter bandwidth is in Hz and must be less than half the value of the input sample rate.Use normalized frequency (0 to 1)
–– The value of the filter bandwidth is in normalized frequency units. The value must be a positive scalar less than1.0
.
(since R2023a)
Tunable: Yes
Dependencies
To enable this parameter:
Set Filter type to
Bandpass
orBandstop
.Clear the Specify bandwidth from input port parameter.
Window function — Window function
Hann
(default)  Hamming
 Chebyshev
 Kaiser
Specify the window function used to design the FIR filter. You can set this parameter to:
Hann
Hamming
Chebyshev
Kaiser
Chebyshev window sidelobe attenuation (dB) — Chebyshev window sidelobe attenuation
60 (default)  positive scalar
Specify the sidelobe attenuation of the Chebyshev window as a real positive scalar.
Dependencies
To enable this parameter, set Window function to
Chebyshev
.
Kaiser window parameter — Kaiser window parameter
0.5 (default)  real scalar
Specify the Kaiser window parameter as a real scalar.
Dependencies
To enable this parameter, set Window function to
Kaiser
.
Sample rate mode — Mode to specify the input sample rate
Specify on dialog
(default)  Inherit from input port
 Use normalized frequency (0 to 1)
Since R2023a
Specify the input sample rate using one of these options:
Specify on dialog
–– Specify the input sample rate in the block dialog box using the Input sample rate (Hz) parameter.Inherit from input port
–– The block inherits the sample rate from the input signal as N / Ts, where N is the frame size of the input signal and Ts is the sample time of the input signal.Use normalized frequency (0 to 1)
–– Specify the filter cutoff frequency, center frequency, and the filter bandwidth in normalized frequency units (0 to 1).
Input sample rate (Hz) — Input sample rate
44100 (default)  positive scalar
Specify the sample rate of the input signal as a positive scalar in Hz.
Dependencies
To enable this parameter, set
the Sample rate mode parameter to
Specify on dialog
. (since R2023a)
View Filter Response — View Filter Response
button
Open the dynamic filter visualizer and display the magnitude response of the variable bandwidth FIR filter. The response is based on the parameters you select in the block dialog box. To update the magnitude response while the dynamic filter visualizer is running, modify the parameters in the dialog box and click Apply.
You can configure the plot settings and the signal measurements from the interface of the visualizer.
On the Plot tab, the Configuration section allows you to modify the plot settings.
On the Measurements tab, you can measure the signal statistics, place data cursors, and display the peak values of the selected signal.
For more details on the dynamic filter visualizer interface and its tools,
see dsp.DynamicFilterVisualizer
.
Simulate using — Simulate using
Code generation
(default)  Interpreted execution
Specify the type of simulation to run. You can set this parameter to:
Code generation
(default)Simulate model using generated C code. The first time you run a simulation, Simulink^{®} generates C code for the block. The C code is reused for subsequent simulations, as long as the model does not change. This option requires additional startup time but provides faster simulation speed than
Interpreted execution
.Interpreted execution
Simulate model using the MATLAB^{®} interpreter. This option shortens startup time but has slower simulation speed than
Code generation
.
Block Characteristics
Data Types 

Multidimensional Signals 

VariableSize Signals 

Algorithms
FIR Transformations
All transformations assume a lowpass filter of length 2N+1.
Consider an ideal lowpass brickwall filter with normalized cutoff frequency ω_{c1}. By taking the inverse discrete Fourier transform of the ideal frequency response, and clipping the resulting sequence to length 2N+1, the impulse response is:
$$\begin{array}{l}forn=0\\ {h}_{LP}(n)=\frac{{\omega}_{c}}{\pi}.w(0)\\ for\text{1}\le \text{n}\le \text{N}\\ {h}_{LP}(n)=\frac{\mathrm{sin}({\omega}_{c}n)}{\pi n}.w(n)\end{array}$$
where w(n) is the window vector. Tune the lowpass filter coefficients to a new cutoff frequency ω_{c2} as follows:
$$\begin{array}{l}for\text{}n=0\\ {h}_{LP}(n)=\frac{{\omega}_{c2}}{\pi}.\omega (0)\\ for\text{}1\text{}\le \text{}\leftn\right\text{}\le \text{}N\\ {h}_{LP}(n)=\frac{\mathrm{sin}({\omega}_{c2}n)}{\pi n}.\omega (n)\end{array}$$
You do not need to recompute the window every time you tune the cutoff frequency.
Assuming a lowpass filter with normalized 6dB cutoff frequency ω_{c}, you can obtain a highpass filter with the same cutoff frequency by taking the complementary of the lowpass frequency response: H_{HP}(e^{jω}) = 1 − H_{LP}(e^{jω})
Taking the inverse discrete Fourier transform of the above response, you have the following highpass filter coefficients:
$$\begin{array}{l}for\text{}n=0\\ {h}_{hp}(n)=1{h}_{LP}(n)\\ for\text{}1\le \text{}\leftn\right\text{}\le \text{}N\\ {h}_{hp}(n)={h}_{LP}(n)\end{array}$$
Obtain a bandpass filter centered at frequency ω_{0} by shifting the lowpass response:
H_{BP}(e^{jω}) = H_{LP}(e^{j(ω–ω0})) + H_{LP}(e^{j(ω–ω0)})
The bandwidth of the resulting bandpass filter is 2ω_{c}, as measured between the two cutoff frequencies of the bandpass filter. The equivalent bandpass filter coefficients are then:
$$\begin{array}{l}{h}_{BP}(n)=({e}^{j{\omega}_{0}n}+{e}^{j{\omega}_{0}n}){h}_{LP}(n)\\ \text{whichcanbewrittenas:}\\ {h}_{BP}(n)=2\mathrm{cos}({\omega}_{0}n){h}_{LP}(n)\end{array}$$
You can transform a lowpass filter to a bandstop filter by combining the highpass and bandpass transformations. First make the filter bandpass by shifting the lowpass response, and then invert it to get a bandstop response centered at ω_{0}.
H_{BS}(e^{jω}) = 1 – (H_{LP}(e^{j(ω–ω0)}) + H_{LP}(e^{j(ω+ω0)}))
This yields the following coefficients:
$$\begin{array}{l}for\text{}n=0\\ {h}_{BS}(n)=12\mathrm{cos}({\omega}_{0}n){h}_{LP}(n)\\ for\text{}1\le \text{}\leftn\right\text{}\le \text{}N\\ {h}_{BS}(n)=2\mathrm{cos}({\omega}_{0}n){h}_{LP}(n)\end{array}$$
You can combine these transformations to transform a lowpass filter to a lowpass, highpass, bandpass, or bandstop filter with arbitrary cutoffs.
For example, to transform a lowpass filter with cutoff ω_{c1} to a highpass with cutoff ω_{c2}, you first apply the lowpasstolowpass transformation to get a lowpass filter with cutoff ω_{c2}, and then apply the lowpasstohighpass transformation to get the highpass with cutoff ω_{c2}.
To get a bandpass filter with center frequency ω_{0} and bandwidth β, we first apply the lowpasstolowpass transformation to go from a lowpass with cutoff ω_{c} to a lowpass with cutoff β/2, and then apply the lowpasstobandpass transformation to get the desired bandpass filter. You can use the same approach for a bandstop filter.
References
[1] Jarske, P., Y. Neuvo, and S. K. Mitra. "A Simple Approach to the Design of Linear Phase FIR Digital Filters with Variable Characteristics." Signal Processing 14, no. 4 *(1988): 313326.
Extended Capabilities
C/C++ Code Generation
Generate C and C++ code using Simulink® Coder™.
The Variable Bandwidth FIR Filter block supports SIMD code generation using
Intel^{®} AVX2 code replacement library when the input signal has a data type of
single
or double
.
The SIMD technology significantly improves the performance of the generated code. For more information, see SIMD Code Generation. To generate SIMD code from this block, see Use Intel AVX2 Code Replacement Library to Generate SIMD Code from Simulink Blocks.
Version History
Introduced in R2015aR2023a: Support for normalized frequencies
When you set the Sample rate mode parameter to
Use normalized frequency (0 to 1)
, you can specify
the filter cutoff frequency, center frequency, and filter bandwidth in normalized
frequency units (0 to 1).
See Also
Objects
Blocks
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