MATLAB: Using DSB-SC for a wav file.

dsbsc

Hello everyone..!!

I have a assignment to do for my lab which is about to remove the noise from a wav file. I did try the type that the teacher gave us but it doesnt seem to work. can anyone help me ?

clear all;
close all;
[data,fs] = audioread ('mixed.wav');
fc=13; %% hz
t=1:2; %% time
u=3*pi/2; %% phase
m = data * cos(2*pi*fc*t+u);
sound (m,fs);

Best Answer

If you intend to create a double sideband -suppressed carrier signal, you need to use element-wise multiplication:

m = data .* cos(2*pi*fc*t+u);

Of course the two vectors must have the same sizes, or the operation will throw an error to that effect.

See the documentation section on Array vs. Matrix Operations (link) for details.

Related Solutions

MATLAB: Changing frequency of .wav

One way to do what I believe you want to do is to heterodyne your initial signal with the difference between it and the the new frequency. This is called ‘amplitude modulation’, and specifically here ‘double sideband suppressed carrier’ modulation. I then filter out the lower sideband to produce a ‘single-sideband suppressed carrier’ signal at the desired (upper sideband) frequency.

Don’t worry about the details. I’m simply explaining what my code does and how it works. My code requires the Signal Processing Toolbox to design and implement the filter.

The modulation technique multiplies two sinusoids of different frequencies together to produce sinusiods with the sum and difference of those frequencies. The mathmatical background is given in the Wikipedia entry on Product-to-sum and sum-to-product identities.

The code:

Fs = 4.41E4;                                % Sampling Frequency (Hz)
tmax = 5;                                   % Time Duration Of Signal (sec)
t = linspace(0, tmax, tmax*Fs);             % Time Vector
f = 220;                                    % Original Frequency
s = sin(2*pi*f*t + cos(15*pi*t)*pi/2);      % Original Signal
sound(s, Fs)                                % Listen To Original Signal
pause(tmax*2)                               % Wait For It To Finish
Fc = 261.6;                                 % Desired Output Frequency   
carrier = sin(2*pi*(Fc-f)*t);               % Generate Carrier 
sm = s .* carrier;                          % Modulate (Produces Upper & Lower Sidebands
Fn = Fs/2;                                  % Design High-Pass Filter To Eliminate Lower Sideband
Wp = Fc/Fn;
Ws = Wp*0.8;
[n,Wn] = buttord(Wp, Ws, 1, 10);
[b,a] = butter(n,Wn,'high');
[sos,g] = tf2sos(b,a);
smf = 2*filtfilt(sos,g,sm);                 % Output Is Suppressed-Carrier Upper Sideband Signal
sound(smf, Fs)                              % Listen To New Signal
Fn1 = Fs/2;                                 % Compute & Plot Fourier Series Of Both
Ft1 = fft(s)/length(s);
Fv1 = linspace(0, 1, fix(length(Ft1)/2)+1)*Fn1;
Ix1 = 1:length(Fv1);
Fn2 = Fs/2;
Ft2 = fft(smf)/length(smf);
Fv2 = linspace(0, 1, fix(length(Ft2)/2)+1)*Fn2;
Ix2 = 1:length(Fv2);
figure(1)
plot(Fv1, abs(Ft1(Ix1)))
grid
hold on
plot(Fv2, abs(Ft2(Ix2)))
hold off
axis([0  500   ylim])

MATLAB: Spectrum of fm modulaton

Hello AbdAlla, The generation of a real fm signal is not a linear process, but there is no problem with finding the resulting spectrum by fft since the fft can find the frequency components of any signal. The fft works out best with modulation by a tone or the sum of a small number of tones, AND when the carrier and all the tones have an exact number of cycles in the time array.

(revised) Here is a small example:

F = 1e6;                 % sampling frequency
f0 = 20000;              % carrier
f1 = 200;                % modulating tone
beta = 1;                % modulation index  
% signal, spectrum
N = 1e6;
t = (0:N-1)*(1/F);       % delt = 1us
y = cos(2*pi*f0*t+beta*cos(2*pi*f1*t));
z = fftshift(fft(y))/N;
f = (-N/2:N/2-1)*(F/N);
% predicted sidebands for the positive frequencies only
n = 4;
sidamp = (1/2)*besselj(0:n,beta);
sidamp = [fliplr(sidamp(2:end)), sidamp];
sidf = (f0-n*f1):f1:(f0+n*f1);
figure(1)
plot (f,abs(z))
xlim([-2*f0 2*f0])
figure(2)
plot(f,abs(z),sidf,sidamp,'o')
xlim([f0-f1*10 f0+f1*10])            % sidebands at the pos frequencies

Fig 1 shows positive and negative frequencies. Most of the time people double the abs(amplitudes) and show positive frequencies only, in which case the sideband calculation does not have the factor of 1/2.

Best Answer

Related Solutions

MATLAB: Changing frequency of .wav

MATLAB: Spectrum of fm modulaton

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