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)
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
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