Documentation |
Design pulse shaping filter
This block brings the filter design capabilities of the filterbuilder function to the Simulink^{®} environment.
See Pulse-shaping Filter Design Dialog Box—Main Pane for more information about the parameters of this block. The Data Types and Code 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 the shape and length of the filter.
Select the shape of the impulse response from the following options:
Raised Cosine
Square Root Raised Cosine
Gaussian
This specification is only available for raised cosine and square root raised cosine filters. For these filters, select one of the following options:
Minimum— This option will result in the minimum-length filter satisfying the user-specified Frequency specifications.
Specify order—This option allows the user to construct a raised cosine or square root cosine filter of a specified order by entering an even number in the Order input box. The length of the impulse response will be Order+1 .
Specify symbols—This option enables the user to specify the length of the impulse response in an alternative manner. If Specify symbols is chosen, the Order input box changes to the Number of symbols input box.
Specify the oversampling factor. Increasing the oversampling factor guards against aliasing and improves the FIR filter approximation to the ideal frequency response. If Order is specified in Number of symbols, the filter length will be Number of symbols*Samples per symbol+1. The product Number of symbols*Samples per symbol must be an even number.
If a Gaussian filter is specified, the filter length must be specified in Number of symbols and Samples per symbol. The product Number of symbols*Samples per symbol must be an even number. The filter length will be Number of symbols*Samples per symbol+1.
In this group, you specify the frequency response of the filter. For raised cosine and square root raised cosine filters, the frequency specifications include:
The rolloff factor takes values in the range [0,1]. The smaller the rolloff factor, the steeper the transition in the stopband.
The frequency units are normalized by default. If you specify units other than normalized, the block assumes that you wish to specify an input sampling frequency, and enables this input box. The choice of frequency units are: Normalized (0 to 1), Hz, kHz, MHz, GHz
For a Gaussian pulse shape, the available frequency specifications are:
This option allows the user to specify the width of the Gaussian filter. Note that this is independent of the length of the filter. The bandwidth-time product (BT) must be a positive real number. Smaller values of the bandwidth-time product result in larger pulse widths in time and steeper stopband transitions in the frequency response.
The frequency units are normalized by default. If you specify units other than normalized, the block assumes that you wish to specify an input sampling frequency, and enables this input box. The choice of frequency units are: Normalized (0 to 1), Hz, kHz, MHz, GHz
If the Order mode is specified as minimum, the magnitude units may be selected from:
dB — Specify the magnitude in decibels (default).
Linear — Specify the magnitude in linear units.
The only design method available for FIR pulse-shaping filters is the window method.
For the filter specifications and design method you select, this parameter lists the filter structures available to implement your filter. FIR filters use direct-form structure.
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.