Design Nyquist filter
Filtering / Filter Designs
dspfdesign
This block brings the filter design capabilities of the filterbuilder
function to the Simulink® environment.
See Nyquist Filter Design — Main Pane for more information about the parameters of this block. The Data Types and Code Generation panes are not available for blocks in the DSP System Toolbox™ Filter Designs library.
Parameters of this block that do not change filter order or structure are tunable.
This button opens the Filter Visualization Tool (fvtool
) from the
Signal Processing Toolbox™ product. You can use the tool to display:
Magnitude response, phase response, and group delay in the frequency domain.
Impulse response and step response in the time domain.
Pole-zero information.
The tool also helps you evaluate filter performance by providing information about filter order, stability, and phase linearity. For more information on FVTool, see the Signal Processing Toolbox documentation.
In this group, you specify your filter format, such as the impulse response and the filter order.
Specifies the location of the center of the transition region between the passband and the stopband. The center of the transition region, Fc, is calculated using the value for Band:
Fc = Fs/(2·Band).
The default value, 2
, corresponds to a halfband
filter.
Select either FIR
or
IIR
from the drop-down list.
FIR
is the default. When you choose
an impulse response, the design methods and structures you can use
to implement your filter change accordingly. These options are both
available only when Band is
2
. For values of Band
greater than 2
, only FIR designs are
supported.
Note
The design methods and structures for FIR filters are not the same as the methods and structures for IIR filters.
Select either Minimum
(the default) or
Specify
from the drop-down list.
Selecting Specify
enables the
Order option (see the following sections)
so you can enter the filter order.
Select Single-rate
,
Decimator
,
Interpolator
, or
Sample-rate converter
. Your choice
determines the type of filter as well as the design methods and
structures that are available to implement your filter. By default,
the block specifies a single-rate filter.
Selecting Decimator
or
Interpolator
activates
the Decimation Factor or the
Interpolation Factor options
respectively.
Selecting Sample-rate
converter
activates both
factors.
Enter the filter order. This option is enabled only if
Specify
was selected for
Filter order mode.
Enter the decimation factor. This option is enabled only if the
Filter type is set to
Decimator
or Sample-rate
converter
. The default value is 2.
Enter the interpolation factor. This option is enabled only if the
Filter type is set to
Interpolator
or
Sample-rate converter
. The default
value is 2.
The parameters in this group allow you to specify your filter response curve.
Select the filter features that the block uses to define the frequency response characteristics.
Use this parameter to specify whether your frequency settings are
normalized or in absolute frequency. Select Normalized
(0–1)
to enter frequencies in normalized form. This
behavior is the default. To enter frequencies in absolute values, select
one of the frequency units from the drop-down
list—Hz
,
kHz
, MHz
, or
GHz
. Selecting one of the unit options
enables the Input sample rate parameter.
Fs, specified in the units you selected for Frequency units, defines the sampling frequency at the filter input. When you provide an input sampling frequency, all frequencies in the specifications are in the selected units as well. This parameter is available when you select one of the frequency options from the Frequency units list.
Specify the width of the transition between the end of the passband and the edge of the stopband. Specify the value in normalized frequency units or the absolute units you select in Frequency units.
Parameters in this group specify the filter response in the passbands and stopbands.
Specify the units for any parameter you provide in magnitude specifications. From the drop-down list, select one of the following options:
Linear
— Specify the
magnitude in linear units.
dB
— Specify the
magnitude in decibels (default)
Squared
— Specify the
magnitude in squared units.
Enter the filter attenuation in the stopband in the units you choose for Magnitude units, either linear or decibels.
The parameters in this group allow you to specify the design method and structure of your filter.
Lists the design methods available for the frequency and magnitude
specifications you entered. When you change the specifications for a
filter, such as changing the impulse response, the methods available
to design filters changes as well. The default IIR design method is
Butterworth
, and the default FIR
method is Kaiser window
.
Selecting this parameter directs the design to scale the filter coefficients to reduce the chances that the inputs or calculations in the filter overflow and exceed the representable range of the filter. Clearing this option removes the scaling. This parameter applies only to IIR filters.
The options for each design are specific for each design method. This section does not present all of the available options for all designs and design methods. There are many more that you encounter as you select different design methods and filter specifications. The following options represent some of the most common ones available.
Density factor controls the density of the frequency grid over which the design method optimization evaluates your filter response function. The number of equally spaced points in the grid is the value you enter for Density factor times (filter order + 1).
Increasing the value creates a filter that more closely approximates an ideal equiripple filter but increases the time required to design the filter. The default value of 20 represents a reasonable trade between the accurate approximation to the ideal filter and the time to design the filter.
When you select this parameter, the design method
determines and design the minimum order filter to meet
your specifications. Some filters do not provide this
parameter. Select Any
,
Even
, or
Odd
from the drop-down
list to direct the design to be any minimum order, or
minimum even order, or minimum odd order.
Stopband shape lets you specify how the stopband changes with increasing frequency. Choose one of the following options:
Flat
—
Specifies that the stopband is flat. The
attenuation does not change as the frequency
increases.
Linear
—
Specifies that the stopband attenuation changes
linearly as the frequency increases. Change the
slope of the stopband by setting
Stopband decay.
When you set Stopband shape, Stopband decay specifies the amount of decay applied to the stopband. the following conditions apply to Stopband decay based on the value of Stopband Shape:
When you set Stopband
shape to Flat
,
Stopband decay has no affect
on the stopband.
When you set Stopband
shape to Linear
,
enter the slope of the stopband in units of
dB/rad/s. The block applies that slope to the
stopband.
When you set Stopband
shape to 1/f
, enter
a value for the exponent n in
the relation (1/f)n to
define the stopband decay. The block applies the
(1/f)n relation to the
stopband to result in an exponentially decreasing
stopband attenuation.
For the filter specifications and design method you select, this parameter lists the filter structures available to implement your filter.
Select this check box to implement the filter as a subsystem of basic Simulink blocks. Clear the check box to implement the filter as a high-level subsystem. By default, this check box is cleared.
The high-level implementation provides better compatibility across various filter structures, especially filters that would contain algebraic loops when constructed using basic elements. On the other hand, using basic elements enables the following optimization parameters:
Optimize for zero gains — Terminate chains that contain Gain blocks with a gain of zero.
Optimize for unit gains — Remove Gain blocks that scale by a factor of one.
Optimize for delay chains — Substitute delay chains made up of n unit delays with a single delay by n.
Optimize for negative gains — Use subtraction in Sum blocks instead of negative gains in Gain blocks.
Specify how the block should process the input. The available options may vary depending on he settings of the Filter Structure and Use basic elements for filter customization parameters. You can set this parameter to one of the following options:
Columns as channels (frame based)
—
When you select this option, the block treats each column of the input
as a separate channel.
Elements as channels (sample based)
—
When you select this option, the block treats each element of the
input as a separate channel.
When the Filter type parameter specifies a multirate filter, select the rate processing rule for the block from following options:
Enforce single-rate processing
— When you select this option, the block maintains the
sample rate of the input.
Allow multirate processing
—
When you select this option, the block adjusts the rate at the
output to accommodate an increased or reduced number of samples.
To select this option, you must set the Input
processing parameter to Elements as
channels (sample based)
.
Select this check box to enable the specification of coefficients using MATLAB® variables. The available coefficient names differ depending on the filter structure. Using symbolic names allows tuning of filter coefficients in generated code. By default, this check box is cleared.
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Output |
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