Can you help us debug and fix this code entirely?

%on line sound importing or recourding 
recObj = audiorecorder; 
recordblocking(recObj, 15); 
play(recObj); 
y = getaudiodata(recObj); 
plot(y); 
play(recObj); 
y = getaudiodata(recObj); 
plot(y); 
%code of AEC 
M=4001; 
fs=8000; 
[B,A]=cheby2(4,20,[0.1, 0.7]); 
Hd=dfilt.df2t([zeros(1,6) B]); 
hFVT=fvtool(Hd); 
set(hFVT, 'color' ,[1 1 1]) 
v=340; 
H = filter(Hd,log(0.99*rand(1,M)+0.01).* ... 
sign (randn(1,M)).*exp(-0.002*(1:M))); 
H = H/norm(H)*4; % Room Impulse Response 
plot(0:1/fs:0.5,H); 
xlabel('Time [sec]' ); 
ylabel('Amplitude' ); 
title( 'Room Impulse Response' ); 
set(gcf, 'color' , [1 1 1]) 
figure(1) 
hold on  
load nearspeech 
n = 1: length (v); 
t=n/fs; 
plot(t,v); 
axis([0 33.5 -1 1]); 
xlabel('Time [sec]' ); 
ylabel('Amplitude' ); 
25 
title('Near-end speech signal' ); 
set(gcf, 'color' , [1 1 1]) 
figure(2) 
hold on  
load farspeech 
x =x( 1: length (x)); 
dhat = filter(H,1,x); 
plot(t,dhat); 
axis([0 33.5 -1 1]);xlabel('Time [sec]' ); 
ylabel('Amplitude' ); 
title('Far-End speech Signal' ); 
set(gcf, 'color' , [1 1 1]) 
figure(3) 
hold on  
d=dhat + v+0.001*randn(length(v),1); 
plot(t,d); 
axis([0 33.5 -1 1]); 
xlabel('Time [sec]' ); 
ylabel('Amplitude' ); 
title('Microphon Signal' ); 
set(gcf, 'color' , [1 1 1]) 
figure(4) 
hold on  
mu=0.025; 
W0 = zeros(1,2048); 
del = 0.01; 
lam = 0.98; 
x = x(1:length(W0)*floor(length(x)/length(W0))); 
d = d(1:length(W0)*floor(length(d)/length(W0))); 
% Construct the Frequency-Domain Adaptive Filter 
fdafilt = dsp.FrequencyDomainAdaptiveFilter('Length',32,'StepSize',mu); 
[y,e] = fdafilt(x,d); 
n = 1:length(e);  
t = n/fs; 
pos = get(gcf,'Position'); 
set(gcf,'Position',[pos(1), pos(2)-100,pos(3),(pos(4)+111)]) 
subplot(3,1,1); 
26 
plot(t,v(n),'g'); 
([0 33.5 -1 1]); 
xlabel('Time [sec]'); 
ylabel('Amplitude'); 
title('Near-End Speech Signal of MR.ABERA'); 
subplot(3,1,2); 
plot(t,d(n),'b'); 
axis([0 33.5 -1 1]); 
ylabel('Amplitude'); 
title('Microphone Signal Mr. Amex + Mr.Abera'); 
subplot(3,1,3); 
plot(t,v(n),'r'); 
axis([0 33.5 -1 1]); 
ylabel('Amplitude');axis 
title('Output of Acoustic Echo Canceller');  
set(gcf, 'color' , [1 1 1]) 
#%Normalized LMS method 
FrameSize = 102; NIter = 14; 
lmsfilt2 = dsp.LMSFilter('Length',11,'Method','Normalized LMS', ... 
'StepSize',0.005); 
firfilt2 = dsp.FIRFilter('Numerator', fir1(10,[.05, .075])); 
sinewave = dsp.SineWave('Frequency',0.001, ... 
'SampleRate',1,'SamplesPerFrame',FrameSize); 
TS = dsp.TimeScope('TimeSpan',FrameSize*NIter,'TimeUnits','Seconds',... 
'YLimits',[-3 3],'BufferLength',2*FrameSize*NIter, ... 
'ShowLegend',true,'ChannelNames', ... 
{'echo signal', 'Filtered signal'}); 
%% 
% Pass the echo input signal into the LMS filter and view the filtered  
% output in the time scope. 
for k = 1:NIter 
x = randn(FrameSize,1);       
% Input signal 
d = firfilt2(x) + sinewave(); % echo + Signal 
[y,e,w] = lmsfilt2(x,d); 
TS([d,e]);          
% echo = channel 1; Filtered = channel 2 
end 
27 
#% convergence performance of regular NLMS 
x = 0.1*randn(500,1); 
[b,err,res] = fircband(12,[0 0.4 0.5 1], [1 1 0 0], [1 0.2],...  
{'w' 'c'}); 
d = filter(b,1,x); 
lms_normalized = dsp.LMSFilter(13,'StepSize',mu,... 
'Method','Normalized LMS','WeightsOutputPort',true); 
[~,e1,~] = lms_normalized(x,d); 
plot([e1]); 
title('NLMS Conversion Performance'); 
legend('NLMS Derived Filter Weights'); 
#% convergence performance of regular LMS 
x = 0.1*randn(500,1); 
[b,err,res] = fircband(12,[0 0.4 0.5 1], [1 1 0 0], [1 0.2],...  
{'w' 'c'}); 
d = filter(b,1,x); 
lms_normalized = dsp.LMSFilter(13,'StepSize',mu,... 
'Method','LMS','WeightsOutputPort',true); 
[~,e2,~] = lms_normalized(x,d); 
plot([e2]); 
title('LMS Conversion Performance'); 
legend('LMS Derived Filter Weights'); 
#% comparing the LMS and NLMS convergence performance 
x = 0.1*randn(500,1); 
[b,err,res] = fircband(12,[0 0.4 0.5 1], [1 1 0 0], [1 0.2],...  
{'w' 'c'}); 
d = filter(b,1,x); 
lms = dsp.LMSFilter(13,'StepSize',mu,'Method',... 
'Normalized LMS','WeightsOutputPort',true); 
lms_normalized = dsp.LMSFilter(13,'StepSize',mu,... 
28 
'Method','Normalized LMS','WeightsOutputPort',true); 
lms_nonnormalized = dsp.LMSFilter(13,'StepSize',mu,... 
'Method','LMS','WeightsOutputPort',true); 
[~,e1,~] = lms_normalized(x,d); 
[~,e2,~] = lms_nonnormalized(x,d); 
plot([e1,e2]); 
title('Comparing the LMS and NLMS Conversion Performance'); 
legend('NLMS Derived Filter Weights', ... 
'LMS Derived Filter Weights','Location', 'NorthEast'); 

main issue we keep facing is

Error using load

Unable to find file or directory 'nearspeech'.

Error in OG (line 30)

load nearspeech

Answer 1

[Walter Roberson]

You can get nearspeech.mat and farspeech.mat as part of the example

openExample('audio/AcousticEchoCancellationAecExample')

[Barnabas]

we rly dont understand MATLAB can you debug and send me the corrected script good sir?

[Barnabas]

its for our undergrad thesis and we just cannot figure it out

[Walter Roberson]

I do not have necessary toolboxes.

[Barnabas]

i can provide. @Bernabas is my handle on telegram. if youre willing and got spare time.

[Walter Roberson]

I have no reason to doubt the correctness of the code from https://www.scribd.com/document/563550610/project-paper