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I want to add PSNR, MSError and SNR to my ready code.

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Mert Sari
Mert Sari 2024-1-16
回答： Mert Sari 2024-1-17
Hello everyone, I'm doing research on an assignment that does LPC compression to a Speech file.
Here you see the code;
clear all;
clc;
%TAKING INPUT WAVEFILE,
a1 = 'C:\Users\user\Desktop\WAV\a1.wav';
[y, Fs] =audioread(a1);
% x=wavrecord(,);
%LENGTH (IN SEC) OF INPUT WAVEFILE,
t=length(y)./Fs;
sprintf('Processing the wavefile "%s"', a1)
sprintf('The wavefile is %3.2f seconds long', t)
%THE ALGORITHM STARTS HERE,
M=10; %prediction order
[aCoeff, pitch_plot, voiced, gain] = f_ENCODER(y, Fs, M); %pitch_plot is pitch periods
synth_speech = f_DECODER (aCoeff, pitch_plot, voiced, gain);
Additionally, I want this code to calculate PSNR, MSError and SNR.
dis=numel(y)-numel(A2);
A2=[A2,zeros(1,dis)];
PSNR = psnr(A2,y)
MSError=mse(A2,y)
SNR=snr(A2,y)
I found such a code on the internet. But I don't know what I should write instead of "A2" or for the others.

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采纳的回答

Mert Sari
Mert Sari 2024-1-17
I solved the problem, it works fine in this set of code;
s0=snr(y);
disp(" Original audio SNR in dB " + s0)
s1=snr(synth_speech);
disp(" LPC audio SNR in dB " + s1)

更多回答（1 个）

Hassaan
Hassaan 2024-1-16
编辑：Hassaan 2024-1-16
clear all;
clc;
% TAKING INPUT WAVEFILE
a1 = 'C:\Users\user\Desktop\WAV\a1.wav';
[y, Fs] = audioread(a1);
% LENGTH (IN SEC) OF INPUT WAVEFILE
t = length(y) / Fs;
disp(['Processing the wavefile "', a1, '"'])
disp(['The wavefile is ', sprintf('%3.2f', t), ' seconds long'])
% THE ALGORITHM STARTS HERE
M = 10; % Prediction order
[aCoeff, pitch_plot, voiced, gain] = f_ENCODER(y, Fs, M); % Encoder function
synth_speech = f_DECODER(aCoeff, pitch_plot, voiced, gain); % Decoder function
% Ensure the original and reconstructed signals are of the same length
dis = numel(y) - numel(synth_speech);
% Get the size of the synth_speech array
[synth_rows, synth_cols] = size(synth_speech);
% Ensure the zero-padding array has the same number of columns
zero_padding = zeros(dis, synth_cols);
% Pad the shorter signal
synth_speech_padded = [synth_speech; zero_padding];
% Calculate PSNR, MSE, and SNR
PSNR = psnr(synth_speech_padded, y);
MSError = immse(synth_speech_padded, y); % Mean Squared Error
SNR = snr(synth_speech_padded, y);
% Display the results
disp(['PSNR: ', num2str(PSNR), ' dB']);
disp(['Mean Squared Error: ', num2str(MSError)]);
disp(['SNR: ', num2str(SNR), ' dB']);
-----------------------------------------------------------------------------------------------------------------------------------------------------
If you find the solution helpful and it resolves your issue, it would be greatly appreciated if you could accept the answer. Also, leaving an upvote and a comment are also wonderful ways to provide feedback.
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Feel free to contact me.
  4 个评论
显示 2更早的评论
Image Analyst
Image Analyst 2024-1-16
@Mert Sari please attach 'C:\Users\user\Desktop\WAV\a1.wav' so we can try it ourselves.
Mert Sari
Mert Sari 2024-1-16
After the changes, I tried again and this time the error I experienced was;
Error using psnr>checkImages (line 97)
A and REF must have the same size.
Error in psnr (line 69)
checkImages(A,ref);
Error in MAIN (line 23)
PSNR = psnr(synth_speech_padded, y);
Here f_ENCODER code;
%ENCODER PORTION
%here, fs = sampling frequency
% aCoeff = LP coefficients
% pitch = pitch periods
% v = voiced or unvoiced decision bit
% g = gain of frames
function [aCoeff, pitch_plot, voiced, gain] = f_ENCODER(x, fs, M);
if (nargin<3), M = 10; end %prediction order=10;
%INITIALIZATION;
b=1; %index no. of starting data point of current frame
fsize = 30e-3; %frame size
frame_length = round(fs .* fsize); %=number data points in each framesize
%of "x"
N= frame_length - 1; %N+1 = frame length = number of data points in
%each framesize
%VOICED/UNVOICED and PITCH; [independent of frame segmentation]
[voiced, pitch_plot] = f_VOICED (x, fs, fsize);
%FRAME SEGMENTATION for aCoeff and GAIN;
for b=1 : frame_length : (length(x) - frame_length),
y1=x(b:b+N); %"b+N" denotes the end point of current frame.
%"y" denotes an array of the data points of the current
%frame
y = filter([1 -.9378], 1, y1); %pre-emphasis filtering
%aCoeff [LEVINSON-DURBIN METHOD];
[a, tcount_of_aCoeff, e] = func_lev_durb (y, M); %e=error signal from lev-durb proc
aCoeff(b: (b + tcount_of_aCoeff - 1)) = a; %aCoeff is array of "a" for whole "x"
%GAIN;
pitch_plot_b = pitch_plot(b); %pitch period
voiced_b = voiced(b);
gain(b) = f_GAIN (e, voiced_b, pitch_plot_b);
end
I uploaded a1.wav.

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