Design halfband filter
This block brings the filter design capabilities of the filterbuilder function to the Simulink® environment.
See Halfband Filter Design Dialog Box — 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.
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.
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 type and order.
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.
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.
Specify the filter response as Lowpass (the default) or Highpass.
Select Single-rate, Decimator, or Interpolator. By default, the block specifies a single-rate filter.
Enter the filter order. This option is enabled only when the Filter order mode is set to Specify.
The parameters in this group allow you to specify your filter response curve. Graphically, the filter specifications for a halfband lowpass filter look similar to those shown in the following figure.
In the figure, the transition region lies between the end of the passband and the start of the stopband. The width is defined explicitly by the value of Transition width.
When Order mode is Specify, set this parameter to Unconstrained or Transition width.
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 Fs 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 Unconstrained (the default), or select Stopband attenuation to constrain the response in the stopband explicitly.
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).
When Magnitude units is Stopband attenuation, 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. For FIR halfband filters, the available design options are equiripple, and Kaiser window. For IIR halfband filters, the available design options are Butterworth, elliptic, and IIR quasi-linear phase.
The following design options are available for FIR halfband filters when the user specifies an equiripple design:
Select the checkbox to specify a minimum-phase design.
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.
1/f — Specifies that the stopband attenuation changes exponentially as the frequency increases, where f is the frequency. Set the power (exponent) for the decay in 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.
Select this check box to scale unit gains between sections in SOS filters. This parameter is available only for SOS filters.
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.
Note: The Inherited (this choice will be removed — see release notes) option will be removed in a future release. See Frame-Based Processing in the DSP System Toolbox Release Notes for more information.
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.