Code covered by the BSD License
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ACdsgn(Fs)
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ArgStruct=parseArgs(args,ArgS...
Helper function for parsing varargin.
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[Fsn, p, q, errors]=get_p_q2(...
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[SP, f, bin_size, num_average...
% pressure_spectra: Calculates an accurate estimate of the pressure spectra
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[SP, f, num_averages_out]=spe...
% spectra_estimate: Is a rough estimate of the pressure spectra
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[SP2, mean_array2]=sub_mean(S...
% sub_mean: Removes the running average from a time record given a sampling rate and high pass cutoff frequency.
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[SPa]=test_pressure_spectra(d...
% test_spectra_estimate: runs demos for the pressure spectra program.
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[bin_size, num_averages_out, ...
% number_of_averages: Calculates the number of points not overlapped from the array size, bin size, and number of averages
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[bz, az]=bessel_digital(Fs, F...
% bessel_digital: creates a digital low pass bessel filter of order n
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[cfa, SP, f, f_cal]=mic_calib...
% mic_calib: Uses a flat top window to calibrate using A-weighted or Linear weighting
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[ftwcf]=window_correction_fac...
% window_correction_factor: Computes the factor for calibrating a Fourier Transform given specific processing parameters
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[m2]=geospace(a, b, n, flag)
% geospace: caculates a geometric sequence or psuedogeometric sequence from a to b with n elements
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[prms]=rms_val(p, dim)
% rms_val: Calculates the rms value along a specific dimension
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[varargout]=convert_double(va...
% This program converts the inputs into double precision arrays. Then
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[w]=flat_top(N, type)
% Flat top windows are used for calibration, because the wide main lobe
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[y, x, a]=match_height_and_sl...
% match_height_and_slopes2: creates a quartic with specifed height and slope at the end points.
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[y2, num_settle_pts, settling...
% filter_settling_data: Creates data to append to a time record for settling a filter
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[y2]=remove_filter_settling_d...
% remove_filter_settling_data: removes data added to time records to settle the filter
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[yAC, errors]=ACweight_time_f...
% ACweight_time_filter: Applies an A or C weighting time filter to a sound recording
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[y]=moving(x,m,fun)
MOVING will compute moving averages of order n (best taken as odd)
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[y_out, b, a]=bessel_antialia...
% bessel_antialias: applies an antialiasing digital Bessel filter
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[y_out, t_out, b, a]=bessel_d...
% bessel_down_sample: applies an antialiasing digital Bessel filter
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[y_out, x_out, y_in]=resample...
% resample_interp3: resamples using interp1 with additional options
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h=subaxis(varargin)
SUBAXIS Create axes in tiled positions. (just like subplot)
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loc=LMSloc(X)
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View all files
from
Calibrated Spectral Analysis
by Edward Zechmann
Simple Fourier Spectral Analysis of sound pressure time record.
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| [bz, az]=bessel_digital(Fs, Fcutoff, n)
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function [bz, az]=bessel_digital(Fs, Fcutoff, n)
% % bessel_digital: creates a digital low pass bessel filter of order n
% %
% % Syntax:
% %
% % [bz, az]=bessel_digital(Fs, Fcutoff, n);
% %
% %
% % *********************************************************************
% %
% % Description
% %
% % Applies an antialiasing digital Bessel filter. Assumes that Fs_cutoff
% % will be the Nyquist Frequency for downsampling. 5th order Bessel
% % filter is default.
% %
% %
% % *********************************************************************
% %
% % Input Variables
% %
% % Fs=50000; % (Hz) sampling rate in Hz.
% % % default is 50000 Hz.
% %
% % Fs_cutoff=10000; % (Hz) Low frequency cutoff for application of
% % % antialising filter.
% % % default is Fs_cutoff=10000; %(Hz)
% %
% % n=3; % is the order of the digital Bessel filter.
% % % Default is 3 for a 3rd order Bessel filter.
% % % default is n=3;
% %
% %
% %
% % **********************************************************************
% %
% % Output Variables
% %
% % bz is an array of feedforward filter coefficients.
% %
% % az is an array of feedbackfilter coefficients.
% %
% %
% % *********************************************************************
% %
% % Subprograms
% %
% % This program requires the Matlab Signal Processing Toolbox
% %
% %
% % *********************************************************************
% %
% % bessel_digital is written by Edward Zechmann
% %
% % date 8 July 2010
% %
% % modified 13 July 2010 Update Comments
% %
% % modified 5 August 2010 Update Comments
% %
% %
% %
% % *********************************************************************
% %
% % Please feel free to modify this code.
% %
% % See also: resample, downsample, upsample, upfirdn
% %
if (nargin < 1 || isempty(Fs)) || ~isnumeric(Fs)
Fs=50000;
end
if (nargin < 2 || isempty(Fcutoff)) || ~isnumeric(Fcutoff)
Fcutoff=1000;
end
if (nargin < 3 || isempty(n)) || ~isnumeric(n)
n=3;
end
% Define an analog Bessel filter
% on a unit samling rate
Wo=1;
[b, a]=besself(n, Wo);
% Apply the impulse invariance transformation to transform the analog
% filter into a digital filter.
[bz, az] = impinvar(b, a, Fs/Fcutoff);
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