SbSOF
Overview
Subband SOF detector
Discussion
This module implements a general purpose second order filter that operates on subband data and that is capable of realizing many different filter types. The function can operate on real or complex data. For complex data, the function filters the real and imaginary components separately.
The module performs internal smoothing allowing the filters to be updated without introducing clicks. In all cases, the module implements an underlying second order filter. First order filters are realized by setting some of the second order coefficients to zero.
The behavior of the cascade filters are controlled by the filterType arrary. The variable, filterType is an integer in the range from 0 to 14 inclusive.
After specifying filterType, adjust the filter parameters by setting the fields, gain, freq, and Q. Some filter parameters are not applicable to all filter types. The variable, freq controls the center frequency of the filter, gain determines the boost or cut in dB, and Q determines how sharp the filter is. Low Q values lead to broad filters and high Q values lead to narrow filters.
The following table discusses the various filter types and which variables are active in each case.
filterType=0, Simple pass through with unity gain.
filterType=1, Linear gain. [gain].
filterType=2, 1st order Butterworth low pass filter. [freq].
filterType=3, 2nd order Butterworth low pass filter. [freq].
filterType=4, 1st order Butterworth high pass filter. [freq].
filterType=5, 2nd order Butterworth high pass filter. [freq].
filterType=6, 1st order allpass filter. [freq].
filterType=7, 2nd order allpass filter. [freq, Q].
filterType=8, 2nd order low shelf. It allows you to vary the gain of the low frequencies. The high frequencies have a gain of 1.0. [freq, gain].
filterType=9, 2nd order low shelf with Q. It allows you to vary the gain of the low frequencies. The high frequencies have a gain of 1.0. [freq, gain, Q].
filterType=10, 2nd order high shelf. It allows you to vary the gain of the high frequencies. The low frequencies have a gain of 1.0. [freq, gain].
filterType=11, 2nd order high shelf with Q. It allows you to vary the gain of the high frequencies. The low frequencies have a gain of 1.0. [freq, gain, Q].
filterType=12, 2nd order peaking filter. It has unity gain except around the specified frequency. By varying gain, you can get a peak or a notch in the frequency band. [freq, gain, Q].
filterType=13, 2nd order notch filter. It has unity gain except around the specified frequency. At the specified frequency, the filter has a true notch with -inf dB gain. [freq, Q].
filterType=14, 2nd order bandpass filter. It has unity gain at the specified frequency and falls off in both directions. The bandwidth of the filter is determined by Q. [freq, Q].
filterType=15, 1st order Bessel low pass filter. [freq].
filterType=16, 1st order Bessel high pass filter. [freq].
filterType=17, 1st order asymmetrical low shelf. [freq, gain].
filterType=18, 1st order asymmetrical high shelf. [freq, gain].
filterType=19, 1st order symmetrical low shelf. [freq, gain].
filterType=20, 1st order symmetrical high shelf. [freq, gain].
filterType=21, 2nd order Butterworth low pass filter with variable Q. [freq, Q].
filterType=22, 2nd order Butterworth high pass filter with variable Q. [freq, Q].
Type Definition
typedef struct _ModuleSbSOF
{
ModuleInstanceDescriptor instance; // Common Audio Weaver module instance structure
INT32 filterType; // Selects the type of filter that is implemented by the module: Bypass=0, Gain=1, Butter1stLPF=2, Butter2ndLPF=3, Butter1stHPF=4, Butter2ndHPF=5, Allpass1st=6, Allpass2nd=7, Shelf2ndLow=8, Shelf2ndLowQ=9, Shelf2ndHigh=10, Shelf2ndHighQ=11, PeakEQ=12, Notch=13, Bandpass=14, Bessel1stLPF=15, Bessel1stHPF=16, AsymShelf1stLow=17, AsymShelf1stHigh=18, SymShelf1stLow=19, SymShelf1stHigh=20, VariableQLPF=21, VariableQHPF=22.
FLOAT32 freq; // Cutoff frequency of the filter, in Hz.
FLOAT32 gain; // Amount of boost or cut to apply, in dB if applicable.
FLOAT32 Q; // Specifies the Q of the filter, if applicable.
FLOAT32 smoothingTime; // Time constant of the smoothing process.
INT32 updateActive; // Specifies whether the filter coefficients are updating (=1) or fixed (=0).
FLOAT32 b0; // Desired first numerator coefficient.
FLOAT32 b1; // Desired second numerator coefficient.
FLOAT32 b2; // Desired third numerator coefficient.
FLOAT32 a1; // Desired second denominator coefficient.
FLOAT32 a2; // Desired third denominator coefficient.
FLOAT32 current_b0; // Instantaneous first numerator coefficient.
FLOAT32 current_b1; // Instantaneous second numerator coefficient.
FLOAT32 current_b2; // Instantaneous third numerator coefficient.
FLOAT32 current_a1; // Instantaneous second denominator coefficient.
FLOAT32 current_a2; // Instantaneous third denominator coefficient.
FLOAT32 smoothingCoeff; // Smoothing coefficient. This is computed based on the smoothingTime, sample rate, and block size of the module.
FLOAT32* state; // State variables. 2 per channel.
} ModuleSbSOFClass;
Variables
Properties
Name | Type | Usage | isHidden | Default value | Range | Units |
filterType | int | parameter | 0 | 0 | 0:22 | |
freq | float | parameter | 0 | 250 | 1:20000 | Hz |
gain | float | parameter | 0 | 6 | -24:24 | dB |
Q | float | parameter | 0 | 1 | -20:20 | |
smoothingTime | float | parameter | 0 | 10 | 0:1000 | msec |
updateActive | int | parameter | 1 | 1 | 0:1 | |
b0 | float | derived | 1 | 1 | Unrestricted | |
b1 | float | derived | 1 | 0 | Unrestricted | |
b2 | float | derived | 1 | 0 | Unrestricted | |
a1 | float | derived | 1 | 0 | Unrestricted | |
a2 | float | derived | 1 | 0 | Unrestricted | |
current_b0 | float | state | 1 | 1 | Unrestricted | |
current_b1 | float | state | 1 | 0 | Unrestricted | |
current_b2 | float | state | 1 | 0 | Unrestricted | |
current_a1 | float | state | 1 | 0 | Unrestricted | |
current_a2 | float | state | 1 | 0 | Unrestricted | |
smoothingCoeff | float | derived | 1 | 0.06449 | Unrestricted | |
state | float* | state | 1 | [64 x 1] | Unrestricted |
Pins
Input Pins
Name: in
Description: audio input
Data type: float
Channel range: Unrestricted
Block size range: Unrestricted
Sample rate range: Unrestricted
Complex support: Real and Complex
Output Pins
Name: out
Description: audio output
Data type: float
MATLAB Usage
File Name: sb_sof_module.m
M=sb_sof_module(NAME)
Subband second order filter in which each subband is separately
filtered. The same coefficients are applied to each subband and
each subband has its own set of state variables. In the case of
complex data, the module filters the real and imaginary components
separately. Arguments:
NAME - name of the module.
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