I am getting the following error
‘MyTopLevelModel/SubsystemA/Model’ has multiple sample times. Only constant (inf) or inherited (-1) sample times are allowed in function call subsystem ‘MyTopLevelModel/SubsystemA’.
More info of my setup
SubsystemA has a function call which has a trigger port connected to a port set as "output function call with sample time of 0.001. All my other blocks have sample time -1 (inheritet).
If I go to the model explorer the compiled sample time are not [-1 0] some of the have set the [0.001 0]
Any idea what could be causing this error ?I am getting the following error
‘MyTopLevelModel/SubsystemA/Model’ has multiple sample times. Only constant (inf) or inherited (-1) sample times are allowed in function call subsystem ‘MyTopLevelModel/SubsystemA’.
More info of my setup
SubsystemA has a function call which has a trigger port connected to a port set as "output function call with sample time of 0.001. All my other blocks have sample time -1 (inheritet).
If I go to the model explorer the compiled sample time are not [-1 0] some of the have set the [0.001 0]
Any idea what could be causing this error ? I am getting the following error
‘MyTopLevelModel/SubsystemA/Model’ has multiple sample times. Only constant (inf) or inherited (-1) sample times are allowed in function call subsystem ‘MyTopLevelModel/SubsystemA’.
More info of my setup
SubsystemA has a function call which has a trigger port connected to a port set as "output function call with sample time of 0.001. All my other blocks have sample time -1 (inheritet).
If I go to the model explorer the compiled sample time are not [-1 0] some of the have set the [0.001 0]
Any idea what could be causing this error ? error, time MATLAB Answers — New Questions

​

Hi,
Trying to manually add contour labels over a filled contour plot over a narrow range of values.
Matlab returns the error:
Error using hgconvertunits
The reference object is invalid.
Any ideas how I can manually add labels to a tiledlayout?
I also assume there is no workaround for avoiding the contours from striking through the labels when added manually?
Thank you
a = 0.98;
b = 1.02;
Z = round(((b-a).*rand(99,99) + a),2);
XY = [-0.375:.125:.375];
no_levels = 8;
npoints = [99 99]; % define improved resolution
Xfine = linspace(XY(1),XY(end),npoints(1)); % y coords
Yfine = linspace(XY(1),XY(end),npoints(2)); % z coords
[Xfine Yfine] = meshgrid(Xfine,Yfine); % create mesh

tiledlayout(1,2)

for i = 1:2
nexttile(1)

[C,H] = contourf(Xfine,Yfine,Z,no_levels,’-‘);
% clabel(C,H,’interpreter’,’latex’,’FontSize’,14);
% H.LevelList = round(H.LevelList,1); % rounds levels to 1st decimal place
clabel(C,H,"manual",’FontSize’, 22)

hold on
ylim([-0.5 0.5]);
y = ylim;
yticks([y(1):.25:y(2)])
xticks([-.5:.25:.5])
xlim([-.55 .55])
axis square
endHi,
Trying to manually add contour labels over a filled contour plot over a narrow range of values.
Matlab returns the error:
Error using hgconvertunits
The reference object is invalid.
Any ideas how I can manually add labels to a tiledlayout?
I also assume there is no workaround for avoiding the contours from striking through the labels when added manually?
Thank you
a = 0.98;
b = 1.02;
Z = round(((b-a).*rand(99,99) + a),2);
XY = [-0.375:.125:.375];
no_levels = 8;
npoints = [99 99]; % define improved resolution
Xfine = linspace(XY(1),XY(end),npoints(1)); % y coords
Yfine = linspace(XY(1),XY(end),npoints(2)); % z coords
[Xfine Yfine] = meshgrid(Xfine,Yfine); % create mesh

tiledlayout(1,2)

for i = 1:2
nexttile(1)

[C,H] = contourf(Xfine,Yfine,Z,no_levels,’-‘);
% clabel(C,H,’interpreter’,’latex’,’FontSize’,14);
% H.LevelList = round(H.LevelList,1); % rounds levels to 1st decimal place
clabel(C,H,"manual",’FontSize’, 22)

hold on
ylim([-0.5 0.5]);
y = ylim;
yticks([y(1):.25:y(2)])
xticks([-.5:.25:.5])
xlim([-.55 .55])
axis square
end Hi,
Trying to manually add contour labels over a filled contour plot over a narrow range of values.
Matlab returns the error:
Error using hgconvertunits
The reference object is invalid.
Any ideas how I can manually add labels to a tiledlayout?
I also assume there is no workaround for avoiding the contours from striking through the labels when added manually?
Thank you
a = 0.98;
b = 1.02;
Z = round(((b-a).*rand(99,99) + a),2);
XY = [-0.375:.125:.375];
no_levels = 8;
npoints = [99 99]; % define improved resolution
Xfine = linspace(XY(1),XY(end),npoints(1)); % y coords
Yfine = linspace(XY(1),XY(end),npoints(2)); % z coords
[Xfine Yfine] = meshgrid(Xfine,Yfine); % create mesh

tiledlayout(1,2)

for i = 1:2
nexttile(1)

[C,H] = contourf(Xfine,Yfine,Z,no_levels,’-‘);
% clabel(C,H,’interpreter’,’latex’,’FontSize’,14);
% H.LevelList = round(H.LevelList,1); % rounds levels to 1st decimal place
clabel(C,H,"manual",’FontSize’, 22)

hold on
ylim([-0.5 0.5]);
y = ylim;
yticks([y(1):.25:y(2)])
xticks([-.5:.25:.5])
xlim([-.55 .55])
axis square
end contour labels, tiledlayout MATLAB Answers — New Questions

​

Hi all!
My question may be simple but I don’t get to perform Bonferroni correction on ttest2 comparison of two random independent samples with equal column number but different row number.
I want to perform ttest2 comparison between each column of my two samples, therefore getting 10 p-values (see example below). But i get the following error "Index in position 1 is invalid. Array indices must be positive integers or logical values".
Once I overcome this error, I would like to perform Bonferroni correction on ttest2 comparison.
So many thanks!!! I’m using Matlab R2022b
% Note that r1 and r2 differ in row number, not column number
r1 = randi(125,200,10); % Random sample 1
r2 = randi(155,100,10); % Random sample 2

% Perform ttest2 comparison, getting 10-pvalues by comparing the 10 columns of r1 and r2
[h,p,ci,stats] = ttest2(r1(i,:),r2(i,:)); % ttest2 comparison of two random, independent samples
results = multcompare(stats,"CriticalValueType","bonferroni") % Bonferroni correction on ttest2Hi all!
My question may be simple but I don’t get to perform Bonferroni correction on ttest2 comparison of two random independent samples with equal column number but different row number.
I want to perform ttest2 comparison between each column of my two samples, therefore getting 10 p-values (see example below). But i get the following error "Index in position 1 is invalid. Array indices must be positive integers or logical values".
Once I overcome this error, I would like to perform Bonferroni correction on ttest2 comparison.
So many thanks!!! I’m using Matlab R2022b
% Note that r1 and r2 differ in row number, not column number
r1 = randi(125,200,10); % Random sample 1
r2 = randi(155,100,10); % Random sample 2

% Perform ttest2 comparison, getting 10-pvalues by comparing the 10 columns of r1 and r2
[h,p,ci,stats] = ttest2(r1(i,:),r2(i,:)); % ttest2 comparison of two random, independent samples
results = multcompare(stats,"CriticalValueType","bonferroni") % Bonferroni correction on ttest2 Hi all!
My question may be simple but I don’t get to perform Bonferroni correction on ttest2 comparison of two random independent samples with equal column number but different row number.
I want to perform ttest2 comparison between each column of my two samples, therefore getting 10 p-values (see example below). But i get the following error "Index in position 1 is invalid. Array indices must be positive integers or logical values".
Once I overcome this error, I would like to perform Bonferroni correction on ttest2 comparison.
So many thanks!!! I’m using Matlab R2022b
% Note that r1 and r2 differ in row number, not column number
r1 = randi(125,200,10); % Random sample 1
r2 = randi(155,100,10); % Random sample 2

% Perform ttest2 comparison, getting 10-pvalues by comparing the 10 columns of r1 and r2
[h,p,ci,stats] = ttest2(r1(i,:),r2(i,:)); % ttest2 comparison of two random, independent samples
results = multcompare(stats,"CriticalValueType","bonferroni") % Bonferroni correction on ttest2 multcompare, bonferroni, ttest2, two independent samples MATLAB Answers — New Questions

​

I am unable to input the dirichlet condition into this non homogenous ode. Please point out my mistake.

%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
%%%% -u" + u = – 5x + 4 0 < x <= 1 %%%%
%%%% u(0) = 0, u(1) = 2 %%%%
%%%% Exact solution u = (exp(x)*(3*exp(1) + 4))/(exp(2) – 1) – 5*x – (exp(-x)*(3*exp(1) + 4*exp(2)))/(exp(2) – 1) + 4 %%%%
%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
clear all;
close all;
format long;
N = 20;
x0 = 0;
xN = 1;
h = 1/N;
for j = 1:N+1
X(j) = x0 + (j-1)*h;
end
f = @(x)(- 5*x +4);
A = zeros(N+1,N+1);
F = zeros(N+1,1);
%%%% Local stiffness matrix %%%%
a = [1/h -1/h ; -1/h 1/h] ;
for i=1:N
phi1 = @(x)(X(i+1)-x)/h; %%%% linear basis function %%%%
phi2 = @(x)(x-X(i))/h; %%%% linear basis function %%%%
f1 = @(x)f(x)*phi1(x); %%%% integrand for load vector %%%%
f2 = @(x)f(x)*phi2(x); %%%% integrand for load vector %%%%
v(1,i) = gauss(f1,X(i),X(i+1),1); %%%% element wise values of
v(2,i) = gauss(f2,X(i),X(i+1),1); %%%% load vector
end
%%%% Assembling %%%%
for i=1:N
A([i i+1],[i i+1]) = A([i i+1],[i i+1]) + a;
F([i i+1],1) = F([i i+1],1) + v([1 2],i);
end
%%%% Dirichlet Boundary condition %%%%
% F(N+1,1) = F(N+1,1)+2;
fullnodes = [1:N+1];
%%%%% Dirichlet boundary condition %%%%%
freenodes=setdiff(fullnodes,[1]);
Uh = zeros(N+1,1);
Uh(N+1,1) = 2;
%%%% Approximate solution %%%%
Uh(freenodes)=A(freenodes,freenodes)F(freenodes,1);
%%%% Exact solution %%%%
U = zeros(N+1,1);
for i =1:N+1
U(i) = (exp(X(i))*(3*exp(1) + 4))/(exp(2) – 1) – 5*X(i) – (exp(-X(i))*(3*exp(1) + 4*exp(2)))/(exp(2) – 1) + 4;
end
error = max(abs(U-Uh));
[U Uh]
plot(X,U,X,Uh,’o’)I am unable to input the dirichlet condition into this non homogenous ode. Please point out my mistake.

%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
%%%% -u" + u = – 5x + 4 0 < x <= 1 %%%%
%%%% u(0) = 0, u(1) = 2 %%%%
%%%% Exact solution u = (exp(x)*(3*exp(1) + 4))/(exp(2) – 1) – 5*x – (exp(-x)*(3*exp(1) + 4*exp(2)))/(exp(2) – 1) + 4 %%%%
%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
clear all;
close all;
format long;
N = 20;
x0 = 0;
xN = 1;
h = 1/N;
for j = 1:N+1
X(j) = x0 + (j-1)*h;
end
f = @(x)(- 5*x +4);
A = zeros(N+1,N+1);
F = zeros(N+1,1);
%%%% Local stiffness matrix %%%%
a = [1/h -1/h ; -1/h 1/h] ;
for i=1:N
phi1 = @(x)(X(i+1)-x)/h; %%%% linear basis function %%%%
phi2 = @(x)(x-X(i))/h; %%%% linear basis function %%%%
f1 = @(x)f(x)*phi1(x); %%%% integrand for load vector %%%%
f2 = @(x)f(x)*phi2(x); %%%% integrand for load vector %%%%
v(1,i) = gauss(f1,X(i),X(i+1),1); %%%% element wise values of
v(2,i) = gauss(f2,X(i),X(i+1),1); %%%% load vector
end
%%%% Assembling %%%%
for i=1:N
A([i i+1],[i i+1]) = A([i i+1],[i i+1]) + a;
F([i i+1],1) = F([i i+1],1) + v([1 2],i);
end
%%%% Dirichlet Boundary condition %%%%
% F(N+1,1) = F(N+1,1)+2;
fullnodes = [1:N+1];
%%%%% Dirichlet boundary condition %%%%%
freenodes=setdiff(fullnodes,[1]);
Uh = zeros(N+1,1);
Uh(N+1,1) = 2;
%%%% Approximate solution %%%%
Uh(freenodes)=A(freenodes,freenodes)F(freenodes,1);
%%%% Exact solution %%%%
U = zeros(N+1,1);
for i =1:N+1
U(i) = (exp(X(i))*(3*exp(1) + 4))/(exp(2) – 1) – 5*X(i) – (exp(-X(i))*(3*exp(1) + 4*exp(2)))/(exp(2) – 1) + 4;
end
error = max(abs(U-Uh));
[U Uh]
plot(X,U,X,Uh,’o’) I am unable to input the dirichlet condition into this non homogenous ode. Please point out my mistake.

%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
%%%% -u" + u = – 5x + 4 0 < x <= 1 %%%%
%%%% u(0) = 0, u(1) = 2 %%%%
%%%% Exact solution u = (exp(x)*(3*exp(1) + 4))/(exp(2) – 1) – 5*x – (exp(-x)*(3*exp(1) + 4*exp(2)))/(exp(2) – 1) + 4 %%%%
%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
clear all;
close all;
format long;
N = 20;
x0 = 0;
xN = 1;
h = 1/N;
for j = 1:N+1
X(j) = x0 + (j-1)*h;
end
f = @(x)(- 5*x +4);
A = zeros(N+1,N+1);
F = zeros(N+1,1);
%%%% Local stiffness matrix %%%%
a = [1/h -1/h ; -1/h 1/h] ;
for i=1:N
phi1 = @(x)(X(i+1)-x)/h; %%%% linear basis function %%%%
phi2 = @(x)(x-X(i))/h; %%%% linear basis function %%%%
f1 = @(x)f(x)*phi1(x); %%%% integrand for load vector %%%%
f2 = @(x)f(x)*phi2(x); %%%% integrand for load vector %%%%
v(1,i) = gauss(f1,X(i),X(i+1),1); %%%% element wise values of
v(2,i) = gauss(f2,X(i),X(i+1),1); %%%% load vector
end
%%%% Assembling %%%%
for i=1:N
A([i i+1],[i i+1]) = A([i i+1],[i i+1]) + a;
F([i i+1],1) = F([i i+1],1) + v([1 2],i);
end
%%%% Dirichlet Boundary condition %%%%
% F(N+1,1) = F(N+1,1)+2;
fullnodes = [1:N+1];
%%%%% Dirichlet boundary condition %%%%%
freenodes=setdiff(fullnodes,[1]);
Uh = zeros(N+1,1);
Uh(N+1,1) = 2;
%%%% Approximate solution %%%%
Uh(freenodes)=A(freenodes,freenodes)F(freenodes,1);
%%%% Exact solution %%%%
U = zeros(N+1,1);
for i =1:N+1
U(i) = (exp(X(i))*(3*exp(1) + 4))/(exp(2) – 1) – 5*X(i) – (exp(-X(i))*(3*exp(1) + 4*exp(2)))/(exp(2) – 1) + 4;
end
error = max(abs(U-Uh));
[U Uh]
plot(X,U,X,Uh,’o’) finite element method MATLAB Answers — New Questions

​

I have image of anulus, I need to find center and radius of inner and outer arcI have image of anulus, I need to find center and radius of inner and outer arc I have image of anulus, I need to find center and radius of inner and outer arc image processing, circle fitting, binary image processing MATLAB Answers — New Questions

​

I managed to program a plugin that does a bit of reverb and a low pass filter. The plugin itself just shows me the spectrum of a signal during the time it is played but I need the entire spectrogram of that signal. Is there an option to plot the entire spectrogram with the audioTestBench function?

classdef FDNreverb2 < audioPlugin
properties
preDelayT = 0; % pre delay [ms]
reverbTime = 1.0; % reverb time [s]
roomSize = 5;
mix = 70; % mix of wet and dry signal [Percent], 100 % -> only wet
order = enumFDNorder.order8; % order of FDN, should be power of 2
in_coeff = 1/2; % coeff for in and output lines — just for now one value for all
out_coeff = 0.7;
end

properties (Access = private)
N = 8; % order of FDN
a = 1.1; % multiplying factor for delays
cSound = 343.2; % speed of sound

primeDelays = [2 3 5 7 11 13 17 19 23 29 31 37 41 43 47 53 59 61 …
67 71 73 79 83 89 97 101 103 107 109 113 127 131];

% following values have to be set manually for Interface
maxPreDelay = 0.2; % max pre delay [s] (200ms)
maxRoomSize = 20.0; % max room size [m]

% following variables initialized in reset method

fs; % sample rate
A; % feedback matrix
b; % input gain coefficients
c; % output gain coefficients
f; % feedback lines after matrix
v; % signal before delay – v_i(n) = b_i * x(n) + f_i(n)
s; % output lines
d; % signal to be sent to matrix A
buffDelayLines; % buffer delay lines, initialized in reset method
m; % delay in samples of each delay line
g; % gain coefficients of delay lines
maxDelay; % maximum delay in delay lines
buffInput; % input buffer for pre delay
pBuffDelayLines; % pointer for delay lines buffer
pBuffInput; % pointer for input buffer
preDelayS; % preDelay in samples
alpha;
v_prev2;
v_prev1;
v_filt_prev1;
v_filt_prev2;

end

properties(Constant)
PluginInterface = audioPluginInterface( …
audioPluginParameter(‘preDelayT’, …
‘DisplayName’, ‘Pre Delay [ms]’, …
‘Mapping’, {‘int’, 0, 200}, …
‘Style’, ‘rotary’, …
‘Layout’, [1,1]), …
audioPluginParameter(‘roomSize’, …
‘DisplayName’, ‘Room Size [m]’, …
‘Mapping’, {‘lin’, 1.0, 20.0}, …
‘Style’, ‘rotary’, …
‘Layout’, [1,2]), …
audioPluginParameter(‘reverbTime’, …
‘DisplayName’, ‘Reverb Time [s]’, …
‘Mapping’, {‘lin’, 0.1, 5.0}, …
‘Style’, ‘rotary’, …
‘Layout’, [1,3]), …
audioPluginParameter(‘order’, …
‘DisplayName’, ‘Order of FDN’, …
‘Mapping’, {‘enum’, ‘8’, ’16’, ’32’}, …
‘Layout’, [3,2]), …
audioPluginParameter(‘in_coeff’, …
‘DisplayName’, ‘Input Gain’, …
‘Mapping’, {‘lin’, 0.0, 1.0}, …
‘Style’, ‘rotary’, …
‘Layout’, [3,3]), …
audioPluginParameter(‘out_coeff’, …
‘DisplayName’, ‘Output Gain’, …
‘Mapping’, {‘lin’, 0.0, 1.0}, …
‘Style’, ‘rotary’, …
‘Layout’, [3,4]), …
audioPluginParameter(‘mix’, …
‘DisplayName’, ‘Mix’, …
‘Mapping’, {‘int’, 0, 100}, …
‘Style’, ‘rotary’, …
‘Layout’, [3,1]), …
audioPluginGridLayout( …
‘RowHeight’, [100 100 100 100 100 100], …
‘ColumnWidth’,[100, 100 100 100 100 100], …
‘RowSpacing’, 10, …
‘ColumnSpacing’, 10, …
‘Padding’, [10 10 10 10]) …
);
end

methods

function plugin = FDNreverb
init(plugin)
end

function out = process(plugin, in)
out = zeros(size(in));
%Butterworth coefficients
[bx, ax] = butter(2, 3000/(plugin.fs/2)); % 2. Ordnung, Grenzfrequenz 1000 Hz
b0 = bx(1);
b1 = bx(2);
b2 = bx(3);
a1 = ax(2);
a2 = ax(3);
% rawVn(1) = b0*plugin.v(1);
% rawVn(2) = b0*plugin.v(2) + b1*plugin.v(1) – a1* rawVn(1);

% Define the cutoff frequency and calculate alpha
cutoffFreq = 1000; % Example cutoff frequency in Hz
plugin.alpha = (2 * pi * cutoffFreq) / (plugin.fs + 2 * pi * cutoffFreq);

% Initialize the previous output for the filter
prevY = zeros(plugin.N, 1);

for i = 1:size(in,1)
% Summieren der L/R – Kan�le
inL = in(i,1);
inR = in(i,2);
inSum = (inL + inR)/2;
plugin.buffInput(plugin.pBuffInput + 1) = inSum;

% loop over delay lines
for n=1:plugin.N
% d_n = gain * delayed v_n
for k=1:plugin.N
plugin.d(k) = plugin.g(k) * plugin.buffDelayLines(k, mod(plugin.pBuffDelayLines + plugin.m(k), plugin.maxDelay +1) + 1);
end
% f_n = A(n,:) * d’
plugin.f(n) = plugin.A(n,:) * plugin.d(:);
% v_n with pre delay
rawVn = plugin.b(n) * plugin.buffInput(mod(plugin.pBuffInput + plugin.preDelayS, (plugin.maxPreDelay * plugin.fs + 1)) + 1) + plugin.f(n);

% Filter anwenden
plugin.v(n) = b0 * rawVn + b1 * plugin.v_prev1(n) + b2 * plugin.v_prev2(n) …
– a1 * plugin.v_filt_prev1(n) – a2 * plugin.v_filt_prev2(n);

% Update der vorherigen Filterwerte
plugin.v_prev2(n) = plugin.v_prev1(n);
plugin.v_prev1(n) = rawVn;
plugin.v_filt_prev2(n) = plugin.v_filt_prev1(n);
plugin.v_filt_prev1(n) = plugin.v(n);

plugin.buffDelayLines(n, plugin.pBuffDelayLines + 1) = plugin.v(n);

% output lines
plugin.s(n) = plugin.c(n) * plugin.d(n);

out(i,:) = out(i,:) + real(plugin.s(n));

end

% Assign to output
out(i,1) = plugin.mix/100 * out(i,1) + (1.0 – plugin.mix/100) * in(i,1);
out(i,2) = plugin.mix/100 * out(i,2) + (1.0 – plugin.mix/100) * in(i,2);

calculatePointer(plugin);
end
end

function calculatePointer(plugin)
if (plugin.pBuffDelayLines==0)
plugin.pBuffDelayLines = plugin.maxDelay;
else
plugin.pBuffDelayLines = plugin.pBuffDelayLines – 1;
end

if (plugin.pBuffInput==0)
plugin.pBuffInput = plugin.maxPreDelay * plugin.fs;
else
plugin.pBuffInput = plugin.pBuffInput – 1;
end
end

function set.in_coeff(plugin, val)
plugin.in_coeff = val;
plugin.b = ones(plugin.N,1) * val;
end

function set.out_coeff(plugin, val)
plugin.out_coeff = val;
plugin.c = ones(plugin.N,1) * val;
end

function set.order(plugin, val)
plugin.order = val;
switch (plugin.order)
case enumFDNorder.order8
plugin.N = 8;
case enumFDNorder.order16
plugin.N = 16;
case enumFDNorder.order32
plugin.N = 32;
end
reset(plugin)
end

function set.reverbTime(plugin, val)
plugin.reverbTime = val;
calculateGainCoeffs(plugin, plugin.reverbTime)
calculateDelays(plugin)
%sprintf(‘Set Reverb Time: %f’, plugin.reverbTime)
%disp([‘Set Reverb Time: ‘, num2str(plugin.reverbTime), ‘ s.’]);
end

function set.preDelayT(plugin, val)
plugin.preDelayT = val;
calculatePreDelayS(plugin, plugin.preDelayT)
%disp([‘Set Predelay: ‘, int2str(plugin.preDelayT), ‘ ms.’]);
end

function set.mix(plugin, val)
plugin.mix = val;
%disp([‘Set Mix value: ‘, int2str(plugin.mix), ‘ %.’]);
end

function calculateMatrix(plugin)
plugin.A = generateFDNmatrix(plugin.N);
end

function calculateDelays(plugin)
% calculate sample delays dependent on room size (val) and
% sample rate (fs)
% m_1 = trace of sound / cSound * fs

M = ceil(0.15 * plugin.reverbTime * plugin.fs);
%disp([‘M = ‘, int2str(M)]);
plugin.m = zeros(plugin.N,1);
interval = M/(plugin.N*4);
%test = 0;
for i=1:plugin.N
tmp = interval/2 + (i-1) * interval;
e = floor(0.5 + log(tmp)/log(plugin.primeDelays(i)));
plugin.m(i) = plugin.primeDelays(i)^(e);
end
end

function calculateGainCoeffs(plugin, val)
for i=1:plugin.N
plugin.g(i) = 10^((-3) * (plugin.m(i)/plugin.fs) / val);
end
end

function calculatePreDelayS(plugin, val)
plugin.preDelayS = val * plugin.fs / 1000;
end

function init(plugin)
plugin.fs = getSampleRate(plugin);

% initialize buffDelayLines & pointer
%plugin.maxDelay = floor(plugin.maxRoomSize/plugin.cSound * plugin.fs * plugin.a^(plugin.N)); %probably higher than actual max delay as actual delays get rounded down
%plugin.maxDelay = ceil(0.15 * 5 * plugin.fs); % maximum possible delay for max reverb time = 5s
plugin.maxDelay = 131^2;
plugin.buffDelayLines = zeros(plugin.N, plugin.maxDelay + 1);
plugin.pBuffDelayLines = plugin.maxDelay;

%initialize filter elements
plugin.v_prev1 = zeros(plugin.N,1);
plugin.v_prev2 = zeros(plugin.N,1);
plugin.v_filt_prev2 = zeros(plugin.N,1);
plugin.v_filt_prev1 = zeros(plugin.N,1);

% initialize
plugin.A = zeros(plugin.N);
plugin.b = ones(plugin.N,1) * plugin.in_coeff; % input gain coefficients
plugin.c = ones(plugin.N,1) * plugin.out_coeff; % output gain coefficients
plugin.f = zeros(plugin.N,1); % feedback lines after matrix
plugin.v = zeros(plugin.N,1); % signal before delay – v_i(n) = b_i * x(n) + f_i(n)
plugin.s = zeros(plugin.N,1); % output lines
plugin.d = zeros(plugin.N,1); % signal to be sent fsto matrix A
plugin.m = zeros(plugin.N,1); % delay in samples of each delay line
plugin.g = zeros(plugin.N,1); % gain coefficients of delay lines
plugin.buffInput = zeros((plugin.maxPreDelay * plugin.fs) + 1, 1); % input buffer for pre delay, 0.2 = max pre delay of 200ms
plugin.preDelayS = 0; % pre delay in samples
plugin.pBuffInput = plugin.maxPreDelay * plugin.fs; % pointer for input buffer

% calculate sample delays of delay lines
calculateDelays(plugin)

% calculate gain coeffs of delay lines
calculateGainCoeffs(plugin, plugin.reverbTime)

% calculate FDN matrix based on set order
calculateMatrix(plugin)
end

function reset(plugin)
init(plugin)
end

function generateFDNmatrix(order)
% generates FDN matrix with evenly distributed eigenvalues at
% the unit circle
eig_nr = order/2;
FDNmatrix = zeros(order, order);
M = orth((rand(order,order)));
BC_n = zeros(order, order, eig_nr);
DtD_n = zeros(order, order, eig_nr);
k1 = 1;
for k=2:2:eig_nr*2
x = M(:,k-1);
y = M(:,k);
x = x / sqrt(2);
y = y / sqrt(2);
B = x * x.’;
C = y * y.’;
D = x * y.’;
BC_n(1:order,1:order,k1) = B + C;
DtD_n(1:order,1:order,k1) = D.’ – D;
w = pi * k / (eig_nr * 2 + 2);
temp = 2 * BC_n(:,:,k1) * cos(w) + 2 * DtD_n(:,:,k1) * sin(w);
FDNmatrix = FDNmatrix + temp;
k1 = k1 + 1;
end
end
end
endI managed to program a plugin that does a bit of reverb and a low pass filter. The plugin itself just shows me the spectrum of a signal during the time it is played but I need the entire spectrogram of that signal. Is there an option to plot the entire spectrogram with the audioTestBench function?

classdef FDNreverb2 < audioPlugin
properties
preDelayT = 0; % pre delay [ms]
reverbTime = 1.0; % reverb time [s]
roomSize = 5;
mix = 70; % mix of wet and dry signal [Percent], 100 % -> only wet
order = enumFDNorder.order8; % order of FDN, should be power of 2
in_coeff = 1/2; % coeff for in and output lines — just for now one value for all
out_coeff = 0.7;
end

properties (Access = private)
N = 8; % order of FDN
a = 1.1; % multiplying factor for delays
cSound = 343.2; % speed of sound

primeDelays = [2 3 5 7 11 13 17 19 23 29 31 37 41 43 47 53 59 61 …
67 71 73 79 83 89 97 101 103 107 109 113 127 131];

% following values have to be set manually for Interface
maxPreDelay = 0.2; % max pre delay [s] (200ms)
maxRoomSize = 20.0; % max room size [m]

% following variables initialized in reset method

fs; % sample rate
A; % feedback matrix
b; % input gain coefficients
c; % output gain coefficients
f; % feedback lines after matrix
v; % signal before delay – v_i(n) = b_i * x(n) + f_i(n)
s; % output lines
d; % signal to be sent to matrix A
buffDelayLines; % buffer delay lines, initialized in reset method
m; % delay in samples of each delay line
g; % gain coefficients of delay lines
maxDelay; % maximum delay in delay lines
buffInput; % input buffer for pre delay
pBuffDelayLines; % pointer for delay lines buffer
pBuffInput; % pointer for input buffer
preDelayS; % preDelay in samples
alpha;
v_prev2;
v_prev1;
v_filt_prev1;
v_filt_prev2;

end

properties(Constant)
PluginInterface = audioPluginInterface( …
audioPluginParameter(‘preDelayT’, …
‘DisplayName’, ‘Pre Delay [ms]’, …
‘Mapping’, {‘int’, 0, 200}, …
‘Style’, ‘rotary’, …
‘Layout’, [1,1]), …
audioPluginParameter(‘roomSize’, …
‘DisplayName’, ‘Room Size [m]’, …
‘Mapping’, {‘lin’, 1.0, 20.0}, …
‘Style’, ‘rotary’, …
‘Layout’, [1,2]), …
audioPluginParameter(‘reverbTime’, …
‘DisplayName’, ‘Reverb Time [s]’, …
‘Mapping’, {‘lin’, 0.1, 5.0}, …
‘Style’, ‘rotary’, …
‘Layout’, [1,3]), …
audioPluginParameter(‘order’, …
‘DisplayName’, ‘Order of FDN’, …
‘Mapping’, {‘enum’, ‘8’, ’16’, ’32’}, …
‘Layout’, [3,2]), …
audioPluginParameter(‘in_coeff’, …
‘DisplayName’, ‘Input Gain’, …
‘Mapping’, {‘lin’, 0.0, 1.0}, …
‘Style’, ‘rotary’, …
‘Layout’, [3,3]), …
audioPluginParameter(‘out_coeff’, …
‘DisplayName’, ‘Output Gain’, …
‘Mapping’, {‘lin’, 0.0, 1.0}, …
‘Style’, ‘rotary’, …
‘Layout’, [3,4]), …
audioPluginParameter(‘mix’, …
‘DisplayName’, ‘Mix’, …
‘Mapping’, {‘int’, 0, 100}, …
‘Style’, ‘rotary’, …
‘Layout’, [3,1]), …
audioPluginGridLayout( …
‘RowHeight’, [100 100 100 100 100 100], …
‘ColumnWidth’,[100, 100 100 100 100 100], …
‘RowSpacing’, 10, …
‘ColumnSpacing’, 10, …
‘Padding’, [10 10 10 10]) …
);
end

methods

function plugin = FDNreverb
init(plugin)
end

function out = process(plugin, in)
out = zeros(size(in));
%Butterworth coefficients
[bx, ax] = butter(2, 3000/(plugin.fs/2)); % 2. Ordnung, Grenzfrequenz 1000 Hz
b0 = bx(1);
b1 = bx(2);
b2 = bx(3);
a1 = ax(2);
a2 = ax(3);
% rawVn(1) = b0*plugin.v(1);
% rawVn(2) = b0*plugin.v(2) + b1*plugin.v(1) – a1* rawVn(1);

% Define the cutoff frequency and calculate alpha
cutoffFreq = 1000; % Example cutoff frequency in Hz
plugin.alpha = (2 * pi * cutoffFreq) / (plugin.fs + 2 * pi * cutoffFreq);

% Initialize the previous output for the filter
prevY = zeros(plugin.N, 1);

for i = 1:size(in,1)
% Summieren der L/R – Kan�le
inL = in(i,1);
inR = in(i,2);
inSum = (inL + inR)/2;
plugin.buffInput(plugin.pBuffInput + 1) = inSum;

% loop over delay lines
for n=1:plugin.N
% d_n = gain * delayed v_n
for k=1:plugin.N
plugin.d(k) = plugin.g(k) * plugin.buffDelayLines(k, mod(plugin.pBuffDelayLines + plugin.m(k), plugin.maxDelay +1) + 1);
end
% f_n = A(n,:) * d’
plugin.f(n) = plugin.A(n,:) * plugin.d(:);
% v_n with pre delay
rawVn = plugin.b(n) * plugin.buffInput(mod(plugin.pBuffInput + plugin.preDelayS, (plugin.maxPreDelay * plugin.fs + 1)) + 1) + plugin.f(n);

% Filter anwenden
plugin.v(n) = b0 * rawVn + b1 * plugin.v_prev1(n) + b2 * plugin.v_prev2(n) …
– a1 * plugin.v_filt_prev1(n) – a2 * plugin.v_filt_prev2(n);

% Update der vorherigen Filterwerte
plugin.v_prev2(n) = plugin.v_prev1(n);
plugin.v_prev1(n) = rawVn;
plugin.v_filt_prev2(n) = plugin.v_filt_prev1(n);
plugin.v_filt_prev1(n) = plugin.v(n);

plugin.buffDelayLines(n, plugin.pBuffDelayLines + 1) = plugin.v(n);

% output lines
plugin.s(n) = plugin.c(n) * plugin.d(n);

out(i,:) = out(i,:) + real(plugin.s(n));

end

% Assign to output
out(i,1) = plugin.mix/100 * out(i,1) + (1.0 – plugin.mix/100) * in(i,1);
out(i,2) = plugin.mix/100 * out(i,2) + (1.0 – plugin.mix/100) * in(i,2);

calculatePointer(plugin);
end
end

function calculatePointer(plugin)
if (plugin.pBuffDelayLines==0)
plugin.pBuffDelayLines = plugin.maxDelay;
else
plugin.pBuffDelayLines = plugin.pBuffDelayLines – 1;
end

if (plugin.pBuffInput==0)
plugin.pBuffInput = plugin.maxPreDelay * plugin.fs;
else
plugin.pBuffInput = plugin.pBuffInput – 1;
end
end

function set.in_coeff(plugin, val)
plugin.in_coeff = val;
plugin.b = ones(plugin.N,1) * val;
end

function set.out_coeff(plugin, val)
plugin.out_coeff = val;
plugin.c = ones(plugin.N,1) * val;
end

function set.order(plugin, val)
plugin.order = val;
switch (plugin.order)
case enumFDNorder.order8
plugin.N = 8;
case enumFDNorder.order16
plugin.N = 16;
case enumFDNorder.order32
plugin.N = 32;
end
reset(plugin)
end

function set.reverbTime(plugin, val)
plugin.reverbTime = val;
calculateGainCoeffs(plugin, plugin.reverbTime)
calculateDelays(plugin)
%sprintf(‘Set Reverb Time: %f’, plugin.reverbTime)
%disp([‘Set Reverb Time: ‘, num2str(plugin.reverbTime), ‘ s.’]);
end

function set.preDelayT(plugin, val)
plugin.preDelayT = val;
calculatePreDelayS(plugin, plugin.preDelayT)
%disp([‘Set Predelay: ‘, int2str(plugin.preDelayT), ‘ ms.’]);
end

function set.mix(plugin, val)
plugin.mix = val;
%disp([‘Set Mix value: ‘, int2str(plugin.mix), ‘ %.’]);
end

function calculateMatrix(plugin)
plugin.A = generateFDNmatrix(plugin.N);
end

function calculateDelays(plugin)
% calculate sample delays dependent on room size (val) and
% sample rate (fs)
% m_1 = trace of sound / cSound * fs

M = ceil(0.15 * plugin.reverbTime * plugin.fs);
%disp([‘M = ‘, int2str(M)]);
plugin.m = zeros(plugin.N,1);
interval = M/(plugin.N*4);
%test = 0;
for i=1:plugin.N
tmp = interval/2 + (i-1) * interval;
e = floor(0.5 + log(tmp)/log(plugin.primeDelays(i)));
plugin.m(i) = plugin.primeDelays(i)^(e);
end
end

function calculateGainCoeffs(plugin, val)
for i=1:plugin.N
plugin.g(i) = 10^((-3) * (plugin.m(i)/plugin.fs) / val);
end
end

function calculatePreDelayS(plugin, val)
plugin.preDelayS = val * plugin.fs / 1000;
end

function init(plugin)
plugin.fs = getSampleRate(plugin);

% initialize buffDelayLines & pointer
%plugin.maxDelay = floor(plugin.maxRoomSize/plugin.cSound * plugin.fs * plugin.a^(plugin.N)); %probably higher than actual max delay as actual delays get rounded down
%plugin.maxDelay = ceil(0.15 * 5 * plugin.fs); % maximum possible delay for max reverb time = 5s
plugin.maxDelay = 131^2;
plugin.buffDelayLines = zeros(plugin.N, plugin.maxDelay + 1);
plugin.pBuffDelayLines = plugin.maxDelay;

%initialize filter elements
plugin.v_prev1 = zeros(plugin.N,1);
plugin.v_prev2 = zeros(plugin.N,1);
plugin.v_filt_prev2 = zeros(plugin.N,1);
plugin.v_filt_prev1 = zeros(plugin.N,1);

% initialize
plugin.A = zeros(plugin.N);
plugin.b = ones(plugin.N,1) * plugin.in_coeff; % input gain coefficients
plugin.c = ones(plugin.N,1) * plugin.out_coeff; % output gain coefficients
plugin.f = zeros(plugin.N,1); % feedback lines after matrix
plugin.v = zeros(plugin.N,1); % signal before delay – v_i(n) = b_i * x(n) + f_i(n)
plugin.s = zeros(plugin.N,1); % output lines
plugin.d = zeros(plugin.N,1); % signal to be sent fsto matrix A
plugin.m = zeros(plugin.N,1); % delay in samples of each delay line
plugin.g = zeros(plugin.N,1); % gain coefficients of delay lines
plugin.buffInput = zeros((plugin.maxPreDelay * plugin.fs) + 1, 1); % input buffer for pre delay, 0.2 = max pre delay of 200ms
plugin.preDelayS = 0; % pre delay in samples
plugin.pBuffInput = plugin.maxPreDelay * plugin.fs; % pointer for input buffer

% calculate sample delays of delay lines
calculateDelays(plugin)

% calculate gain coeffs of delay lines
calculateGainCoeffs(plugin, plugin.reverbTime)

% calculate FDN matrix based on set order
calculateMatrix(plugin)
end

function reset(plugin)
init(plugin)
end

function generateFDNmatrix(order)
% generates FDN matrix with evenly distributed eigenvalues at
% the unit circle
eig_nr = order/2;
FDNmatrix = zeros(order, order);
M = orth((rand(order,order)));
BC_n = zeros(order, order, eig_nr);
DtD_n = zeros(order, order, eig_nr);
k1 = 1;
for k=2:2:eig_nr*2
x = M(:,k-1);
y = M(:,k);
x = x / sqrt(2);
y = y / sqrt(2);
B = x * x.’;
C = y * y.’;
D = x * y.’;
BC_n(1:order,1:order,k1) = B + C;
DtD_n(1:order,1:order,k1) = D.’ – D;
w = pi * k / (eig_nr * 2 + 2);
temp = 2 * BC_n(:,:,k1) * cos(w) + 2 * DtD_n(:,:,k1) * sin(w);
FDNmatrix = FDNmatrix + temp;
k1 = k1 + 1;
end
end
end
end I managed to program a plugin that does a bit of reverb and a low pass filter. The plugin itself just shows me the spectrum of a signal during the time it is played but I need the entire spectrogram of that signal. Is there an option to plot the entire spectrogram with the audioTestBench function?

classdef FDNreverb2 < audioPlugin
properties
preDelayT = 0; % pre delay [ms]
reverbTime = 1.0; % reverb time [s]
roomSize = 5;
mix = 70; % mix of wet and dry signal [Percent], 100 % -> only wet
order = enumFDNorder.order8; % order of FDN, should be power of 2
in_coeff = 1/2; % coeff for in and output lines — just for now one value for all
out_coeff = 0.7;
end

properties (Access = private)
N = 8; % order of FDN
a = 1.1; % multiplying factor for delays
cSound = 343.2; % speed of sound

primeDelays = [2 3 5 7 11 13 17 19 23 29 31 37 41 43 47 53 59 61 …
67 71 73 79 83 89 97 101 103 107 109 113 127 131];

% following values have to be set manually for Interface
maxPreDelay = 0.2; % max pre delay [s] (200ms)
maxRoomSize = 20.0; % max room size [m]

% following variables initialized in reset method

fs; % sample rate
A; % feedback matrix
b; % input gain coefficients
c; % output gain coefficients
f; % feedback lines after matrix
v; % signal before delay – v_i(n) = b_i * x(n) + f_i(n)
s; % output lines
d; % signal to be sent to matrix A
buffDelayLines; % buffer delay lines, initialized in reset method
m; % delay in samples of each delay line
g; % gain coefficients of delay lines
maxDelay; % maximum delay in delay lines
buffInput; % input buffer for pre delay
pBuffDelayLines; % pointer for delay lines buffer
pBuffInput; % pointer for input buffer
preDelayS; % preDelay in samples
alpha;
v_prev2;
v_prev1;
v_filt_prev1;
v_filt_prev2;

end

properties(Constant)
PluginInterface = audioPluginInterface( …
audioPluginParameter(‘preDelayT’, …
‘DisplayName’, ‘Pre Delay [ms]’, …
‘Mapping’, {‘int’, 0, 200}, …
‘Style’, ‘rotary’, …
‘Layout’, [1,1]), …
audioPluginParameter(‘roomSize’, …
‘DisplayName’, ‘Room Size [m]’, …
‘Mapping’, {‘lin’, 1.0, 20.0}, …
‘Style’, ‘rotary’, …
‘Layout’, [1,2]), …
audioPluginParameter(‘reverbTime’, …
‘DisplayName’, ‘Reverb Time [s]’, …
‘Mapping’, {‘lin’, 0.1, 5.0}, …
‘Style’, ‘rotary’, …
‘Layout’, [1,3]), …
audioPluginParameter(‘order’, …
‘DisplayName’, ‘Order of FDN’, …
‘Mapping’, {‘enum’, ‘8’, ’16’, ’32’}, …
‘Layout’, [3,2]), …
audioPluginParameter(‘in_coeff’, …
‘DisplayName’, ‘Input Gain’, …
‘Mapping’, {‘lin’, 0.0, 1.0}, …
‘Style’, ‘rotary’, …
‘Layout’, [3,3]), …
audioPluginParameter(‘out_coeff’, …
‘DisplayName’, ‘Output Gain’, …
‘Mapping’, {‘lin’, 0.0, 1.0}, …
‘Style’, ‘rotary’, …
‘Layout’, [3,4]), …
audioPluginParameter(‘mix’, …
‘DisplayName’, ‘Mix’, …
‘Mapping’, {‘int’, 0, 100}, …
‘Style’, ‘rotary’, …
‘Layout’, [3,1]), …
audioPluginGridLayout( …
‘RowHeight’, [100 100 100 100 100 100], …
‘ColumnWidth’,[100, 100 100 100 100 100], …
‘RowSpacing’, 10, …
‘ColumnSpacing’, 10, …
‘Padding’, [10 10 10 10]) …
);
end

methods

function plugin = FDNreverb
init(plugin)
end

function out = process(plugin, in)
out = zeros(size(in));
%Butterworth coefficients
[bx, ax] = butter(2, 3000/(plugin.fs/2)); % 2. Ordnung, Grenzfrequenz 1000 Hz
b0 = bx(1);
b1 = bx(2);
b2 = bx(3);
a1 = ax(2);
a2 = ax(3);
% rawVn(1) = b0*plugin.v(1);
% rawVn(2) = b0*plugin.v(2) + b1*plugin.v(1) – a1* rawVn(1);

% Define the cutoff frequency and calculate alpha
cutoffFreq = 1000; % Example cutoff frequency in Hz
plugin.alpha = (2 * pi * cutoffFreq) / (plugin.fs + 2 * pi * cutoffFreq);

% Initialize the previous output for the filter
prevY = zeros(plugin.N, 1);

for i = 1:size(in,1)
% Summieren der L/R – Kan�le
inL = in(i,1);
inR = in(i,2);
inSum = (inL + inR)/2;
plugin.buffInput(plugin.pBuffInput + 1) = inSum;

% loop over delay lines
for n=1:plugin.N
% d_n = gain * delayed v_n
for k=1:plugin.N
plugin.d(k) = plugin.g(k) * plugin.buffDelayLines(k, mod(plugin.pBuffDelayLines + plugin.m(k), plugin.maxDelay +1) + 1);
end
% f_n = A(n,:) * d’
plugin.f(n) = plugin.A(n,:) * plugin.d(:);
% v_n with pre delay
rawVn = plugin.b(n) * plugin.buffInput(mod(plugin.pBuffInput + plugin.preDelayS, (plugin.maxPreDelay * plugin.fs + 1)) + 1) + plugin.f(n);

% Filter anwenden
plugin.v(n) = b0 * rawVn + b1 * plugin.v_prev1(n) + b2 * plugin.v_prev2(n) …
– a1 * plugin.v_filt_prev1(n) – a2 * plugin.v_filt_prev2(n);

% Update der vorherigen Filterwerte
plugin.v_prev2(n) = plugin.v_prev1(n);
plugin.v_prev1(n) = rawVn;
plugin.v_filt_prev2(n) = plugin.v_filt_prev1(n);
plugin.v_filt_prev1(n) = plugin.v(n);

plugin.buffDelayLines(n, plugin.pBuffDelayLines + 1) = plugin.v(n);

% output lines
plugin.s(n) = plugin.c(n) * plugin.d(n);

out(i,:) = out(i,:) + real(plugin.s(n));

end

% Assign to output
out(i,1) = plugin.mix/100 * out(i,1) + (1.0 – plugin.mix/100) * in(i,1);
out(i,2) = plugin.mix/100 * out(i,2) + (1.0 – plugin.mix/100) * in(i,2);

calculatePointer(plugin);
end
end

function calculatePointer(plugin)
if (plugin.pBuffDelayLines==0)
plugin.pBuffDelayLines = plugin.maxDelay;
else
plugin.pBuffDelayLines = plugin.pBuffDelayLines – 1;
end

if (plugin.pBuffInput==0)
plugin.pBuffInput = plugin.maxPreDelay * plugin.fs;
else
plugin.pBuffInput = plugin.pBuffInput – 1;
end
end

function set.in_coeff(plugin, val)
plugin.in_coeff = val;
plugin.b = ones(plugin.N,1) * val;
end

function set.out_coeff(plugin, val)
plugin.out_coeff = val;
plugin.c = ones(plugin.N,1) * val;
end

function set.order(plugin, val)
plugin.order = val;
switch (plugin.order)
case enumFDNorder.order8
plugin.N = 8;
case enumFDNorder.order16
plugin.N = 16;
case enumFDNorder.order32
plugin.N = 32;
end
reset(plugin)
end

function set.reverbTime(plugin, val)
plugin.reverbTime = val;
calculateGainCoeffs(plugin, plugin.reverbTime)
calculateDelays(plugin)
%sprintf(‘Set Reverb Time: %f’, plugin.reverbTime)
%disp([‘Set Reverb Time: ‘, num2str(plugin.reverbTime), ‘ s.’]);
end

function set.preDelayT(plugin, val)
plugin.preDelayT = val;
calculatePreDelayS(plugin, plugin.preDelayT)
%disp([‘Set Predelay: ‘, int2str(plugin.preDelayT), ‘ ms.’]);
end

function set.mix(plugin, val)
plugin.mix = val;
%disp([‘Set Mix value: ‘, int2str(plugin.mix), ‘ %.’]);
end

function calculateMatrix(plugin)
plugin.A = generateFDNmatrix(plugin.N);
end

function calculateDelays(plugin)
% calculate sample delays dependent on room size (val) and
% sample rate (fs)
% m_1 = trace of sound / cSound * fs

M = ceil(0.15 * plugin.reverbTime * plugin.fs);
%disp([‘M = ‘, int2str(M)]);
plugin.m = zeros(plugin.N,1);
interval = M/(plugin.N*4);
%test = 0;
for i=1:plugin.N
tmp = interval/2 + (i-1) * interval;
e = floor(0.5 + log(tmp)/log(plugin.primeDelays(i)));
plugin.m(i) = plugin.primeDelays(i)^(e);
end
end

function calculateGainCoeffs(plugin, val)
for i=1:plugin.N
plugin.g(i) = 10^((-3) * (plugin.m(i)/plugin.fs) / val);
end
end

function calculatePreDelayS(plugin, val)
plugin.preDelayS = val * plugin.fs / 1000;
end

function init(plugin)
plugin.fs = getSampleRate(plugin);

% initialize buffDelayLines & pointer
%plugin.maxDelay = floor(plugin.maxRoomSize/plugin.cSound * plugin.fs * plugin.a^(plugin.N)); %probably higher than actual max delay as actual delays get rounded down
%plugin.maxDelay = ceil(0.15 * 5 * plugin.fs); % maximum possible delay for max reverb time = 5s
plugin.maxDelay = 131^2;
plugin.buffDelayLines = zeros(plugin.N, plugin.maxDelay + 1);
plugin.pBuffDelayLines = plugin.maxDelay;

%initialize filter elements
plugin.v_prev1 = zeros(plugin.N,1);
plugin.v_prev2 = zeros(plugin.N,1);
plugin.v_filt_prev2 = zeros(plugin.N,1);
plugin.v_filt_prev1 = zeros(plugin.N,1);

% initialize
plugin.A = zeros(plugin.N);
plugin.b = ones(plugin.N,1) * plugin.in_coeff; % input gain coefficients
plugin.c = ones(plugin.N,1) * plugin.out_coeff; % output gain coefficients
plugin.f = zeros(plugin.N,1); % feedback lines after matrix
plugin.v = zeros(plugin.N,1); % signal before delay – v_i(n) = b_i * x(n) + f_i(n)
plugin.s = zeros(plugin.N,1); % output lines
plugin.d = zeros(plugin.N,1); % signal to be sent fsto matrix A
plugin.m = zeros(plugin.N,1); % delay in samples of each delay line
plugin.g = zeros(plugin.N,1); % gain coefficients of delay lines
plugin.buffInput = zeros((plugin.maxPreDelay * plugin.fs) + 1, 1); % input buffer for pre delay, 0.2 = max pre delay of 200ms
plugin.preDelayS = 0; % pre delay in samples
plugin.pBuffInput = plugin.maxPreDelay * plugin.fs; % pointer for input buffer

% calculate sample delays of delay lines
calculateDelays(plugin)

% calculate gain coeffs of delay lines
calculateGainCoeffs(plugin, plugin.reverbTime)

% calculate FDN matrix based on set order
calculateMatrix(plugin)
end

function reset(plugin)
init(plugin)
end

function generateFDNmatrix(order)
% generates FDN matrix with evenly distributed eigenvalues at
% the unit circle
eig_nr = order/2;
FDNmatrix = zeros(order, order);
M = orth((rand(order,order)));
BC_n = zeros(order, order, eig_nr);
DtD_n = zeros(order, order, eig_nr);
k1 = 1;
for k=2:2:eig_nr*2
x = M(:,k-1);
y = M(:,k);
x = x / sqrt(2);
y = y / sqrt(2);
B = x * x.’;
C = y * y.’;
D = x * y.’;
BC_n(1:order,1:order,k1) = B + C;
DtD_n(1:order,1:order,k1) = D.’ – D;
w = pi * k / (eig_nr * 2 + 2);
temp = 2 * BC_n(:,:,k1) * cos(w) + 2 * DtD_n(:,:,k1) * sin(w);
FDNmatrix = FDNmatrix + temp;
k1 = k1 + 1;
end
end
end
end spectrogram, reverb, filter, audio toolbox MATLAB Answers — New Questions

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Need to send a sample packet from can0 itself as loopback test or send from CAN0 to CAN1 in NVIDIA Jetson AGX Orin.Need to send a sample packet from can0 itself as loopback test or send from CAN0 to CAN1 in NVIDIA Jetson AGX Orin. Need to send a sample packet from can0 itself as loopback test or send from CAN0 to CAN1 in NVIDIA Jetson AGX Orin. can communication, nvidia jetson MATLAB Answers — New Questions

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I don’t know what I am supposed to do with the second part of question 3 and also don’t know what to do with #4. This is my first time ever taking a class about coding so I’m super lost.<</matlabcentral/answers/uploaded_files/1765134/Screenshot%202024-09-02%20at%206.21.33%E2%80%AFPM.png>>

I don’t know what I am supposed to do with the second part of question 3 and also don’t know what to do with #4. This is my first time ever taking a class about coding so I’m super lost. <</matlabcentral/answers/uploaded_files/1765134/Screenshot%202024-09-02%20at%206.21.33%E2%80%AFPM.png>>

I don’t know what I am supposed to do with the second part of question 3 and also don’t know what to do with #4. This is my first time ever taking a class about coding so I’m super lost. vector, vectors, variable MATLAB Answers — New Questions

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Hi all I’ve 1-hour EEG data with a sampling frequency 291hz.I’ve installed EEGlab v14.1.1 version and tried to load my data files of ‘.EEG file’,’text’ and ‘excel’formats, but none of them are loading to EEGlab.It’s showing the following error. Please help me to slove this issue since I’m new to this EEGlab softwareHi all I’ve 1-hour EEG data with a sampling frequency 291hz.I’ve installed EEGlab v14.1.1 version and tried to load my data files of ‘.EEG file’,’text’ and ‘excel’formats, but none of them are loading to EEGlab.It’s showing the following error. Please help me to slove this issue since I’m new to this EEGlab software Hi all I’ve 1-hour EEG data with a sampling frequency 291hz.I’ve installed EEGlab v14.1.1 version and tried to load my data files of ‘.EEG file’,’text’ and ‘excel’formats, but none of them are loading to EEGlab.It’s showing the following error. Please help me to slove this issue since I’m new to this EEGlab software eeg, eeglab, signal processing MATLAB Answers — New Questions

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I observed that ‘filtfilt’ suffers from an undesired behavior when I provide IIR bandpass filters having steep transition bands. Specifically, the output signal exhibits excessive transient response; nevertheless, this behaviour does not emerge if I use a ‘home-made’ version of zerophase filtering based on ‘filter’. I guess that it is not a numerical issue involving the filter coefficients or structure, nor the ‘filter’ function.

% design bandpass filter having transition bandwidth of 200 Hz (Fs = 8000)
bp = designfilt(‘bandpassiir’, ‘StopbandFrequency1’, 50, ‘PassbandFrequency1’, 250,…
‘PassbandFrequency2’, 3600, ‘StopbandFrequency2’, 3700, ‘StopbandAttenuation1’, 30,…
‘PassbandRipple’, 0.1, ‘StopbandAttenuation2’, 30, ‘SampleRate’, 8000, ‘DesignMethod’, ‘cheby2’);
% check stability
assert(isstable(bp),’Unstable filter’);
% apply filtfilt to a random (white) long input signal; output signal shows an undesirable transient
x = randn(2^20,1);
y = filtfilt(bp,x);
% apply ‘home-made’ filtfilt to the same input; output signal shows a more accptable transient
y2 = flipud(filter(bp,flipud(filter(bp,x))));
% compare effects
figure; semilogy(abs(y-y2));

As a rule of thumb, the effect grows as the transition bands get narrower, while it tends to vanish as they get broader.

Where is the problem? Have I missed some recommendations or hints in the function’s help?I observed that ‘filtfilt’ suffers from an undesired behavior when I provide IIR bandpass filters having steep transition bands. Specifically, the output signal exhibits excessive transient response; nevertheless, this behaviour does not emerge if I use a ‘home-made’ version of zerophase filtering based on ‘filter’. I guess that it is not a numerical issue involving the filter coefficients or structure, nor the ‘filter’ function.

% design bandpass filter having transition bandwidth of 200 Hz (Fs = 8000)
bp = designfilt(‘bandpassiir’, ‘StopbandFrequency1’, 50, ‘PassbandFrequency1’, 250,…
‘PassbandFrequency2’, 3600, ‘StopbandFrequency2’, 3700, ‘StopbandAttenuation1’, 30,…
‘PassbandRipple’, 0.1, ‘StopbandAttenuation2’, 30, ‘SampleRate’, 8000, ‘DesignMethod’, ‘cheby2’);
% check stability
assert(isstable(bp),’Unstable filter’);
% apply filtfilt to a random (white) long input signal; output signal shows an undesirable transient
x = randn(2^20,1);
y = filtfilt(bp,x);
% apply ‘home-made’ filtfilt to the same input; output signal shows a more accptable transient
y2 = flipud(filter(bp,flipud(filter(bp,x))));
% compare effects
figure; semilogy(abs(y-y2));

As a rule of thumb, the effect grows as the transition bands get narrower, while it tends to vanish as they get broader.

Where is the problem? Have I missed some recommendations or hints in the function’s help? I observed that ‘filtfilt’ suffers from an undesired behavior when I provide IIR bandpass filters having steep transition bands. Specifically, the output signal exhibits excessive transient response; nevertheless, this behaviour does not emerge if I use a ‘home-made’ version of zerophase filtering based on ‘filter’. I guess that it is not a numerical issue involving the filter coefficients or structure, nor the ‘filter’ function.

% design bandpass filter having transition bandwidth of 200 Hz (Fs = 8000)
bp = designfilt(‘bandpassiir’, ‘StopbandFrequency1’, 50, ‘PassbandFrequency1’, 250,…
‘PassbandFrequency2’, 3600, ‘StopbandFrequency2’, 3700, ‘StopbandAttenuation1’, 30,…
‘PassbandRipple’, 0.1, ‘StopbandAttenuation2’, 30, ‘SampleRate’, 8000, ‘DesignMethod’, ‘cheby2’);
% check stability
assert(isstable(bp),’Unstable filter’);
% apply filtfilt to a random (white) long input signal; output signal shows an undesirable transient
x = randn(2^20,1);
y = filtfilt(bp,x);
% apply ‘home-made’ filtfilt to the same input; output signal shows a more accptable transient
y2 = flipud(filter(bp,flipud(filter(bp,x))));
% compare effects
figure; semilogy(abs(y-y2));

As a rule of thumb, the effect grows as the transition bands get narrower, while it tends to vanish as they get broader.

Where is the problem? Have I missed some recommendations or hints in the function’s help? filtfilt, numerical stability, transient effects MATLAB Answers — New Questions

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