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

4 次查看（过去 30 天）
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.

请先登录，再回答此问题。

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.
Professional Interests
  • Technical Services and Consulting
  • Embedded Systems | Firmware Developement | Simulations
  • Electrical and Electronics Engineering
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.

请先登录，再进行评论。

类别

在 Help Center 和 File Exchange 中查找有关 Spectral Measurements 的更多信息

标签

产品


版本

R2021b

Community Treasure Hunt

Find the treasures in MATLAB Central and discover how the community can help you!

Start Hunting!

Translated by