Design audio weighting filter
The Audio Weighting Filter block will be removed from DSP System Toolbox™ in a future release. Existing instances of the block continue to run. For new code, use the Audio Weighting Filter block from Audio System Toolbox™ instead.
Filtering / Filter Designs
This block brings the filter design capabilities of the
to the Simulink® environment.
See Audio Weighting Filter Design — 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.
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 format, such as the impulse response and the filter order.
The weighting type defines the frequency response of the filter.
The valid weighting types for this filter are
For definitions of the available weighting types, see the
fdesign.audioweighting reference page.
The filter class describes the frequency-dependent tolerances
specified in the relevant standards , . There are two possible class values:
Class 1 weighting filters have stricter tolerances than class 2 filters.
The filter class value does not affect the design. The class value
is only used to provide a specification mask in
fvtool for the analysis of the filter
design. The default value of this parameter is
The filter class is only applicable for
C weighting filters.
Specify the impulse response type as one of
For A, C , C-message, and ITU-R 468–4 filter,
the only option. For a ITU-T 0.41 weighting filter,
the only option.
Specify the frequency units as Hertz (Hz), kilohertz (kHz),
megahertz (MHz), or gigahertz (GHz). Normalized frequency designs
are not supported for audio weighting filters. The default value of
this parameter is
Specify the input sampling frequency. The units correspond to the setting of the Frequency units parameter.
Valid design methods depend on the weighting type. For type
A and C weighting filters, the only valid design type is
S1.42. This is an IIR design method that follows ANSI
standard S1.42–2001. For a C message filter, the only valid
design method is
Bell 41009, which is an
IIR design method following the Bell System Technical Reference PUB
41009. For a ITU-R 468–4 weighting filter, you can design an
IIR or FIR filter. If you choose an IIR design, the design method
IIR least p-norm. If you choose an FIR
design, the design method choices are
Sampling. For an ITU-T 0.41 weighting filter, the available
FIR design methods are
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.
For the filter specifications and design method you select, this parameter lists the filter structures available to implement your filter. For audio weighting IIR filter designs, you can choose direct form I or II biquad (SOS). You can also choose to implement these structures in transposed form.
For FIR designs, you can choose a direct form, direct-form transposed, direct-form symmetric, or direct-form asymmetric 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.
For more information about sample- and frame-based processing, see Sample- and Frame-Based Concepts.
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.
|Port||Supported Data Types|
 American National Standard Design Response of Weighting Networks for Acoustical Measurements, ANSI S1.42-2001, Acoustical Society of America, New York, NY, 2001.
 Electroacoustics Sound Level Meters Part 1: Specifications, IEC 61672-1, First Edition 2002-05.