ODISHA UNIVERSITY OF TECHNOLOGY AND RESEARCH , BHUBANESWAR

DIGITAL SIGNAL
PROCESSING MINI PROJECT
ELECTRONICS AND INSTRUMENTATION ENGINEERING
6TH SEMESTER, GROUP A2

SUBMITTED BY:-

NACHIKET MOHANTY(1901106508)

PRACHETAS PADHI(1901106510)

SABYASACHI SAHOO(1901106517)

SANATANA ROUT(1901106521)

SUBMITTED TO:-

Mr.MADHAB CHANDRA TRIPATHY

Mrs.PRANGYA PRADHAN
 ACKNOWLEDGEMENTS

I am pleased to acknowledge Mr. Madhab Chandra Tripathy and for his invaluable
guidance during the course of this seminar. I extend my sincere thanks to Mrs. Prangya
Pradhan, who continuously helped me throughout the process and without his
guidance, the completion would have been an uphill task. They were a persevering
boost which helped me tackle with the pressure and their contribution in stimulating
suggestions and encouragement, helped me to coordinate my seminar especially in
writing of this report. Moreover, the constructive criticism guided me in best of
direction to flaunt my unsurpassable skills.
 ABSTRACT

Traditional analog hearing aids are similar to a simple radio. They can be tuned and
adjusted for volume, bass and treble. But hearing loss is not just a technical loss of
volume. Rather, hearing deficiency can increase sensitivity and reduce tolerance to
certain sounds while diminishing sensitivity to others. For instance, digital technology
can tell the difference between speech and background noise, allowing one in while
filtering out the other. Approximately 10% of the world's population suffers from some
type of hearing loss, yet only a small percentage of this statistic use a hearing aid. The
stigma associated with wearing a hearing aid, customer dissatisfaction with hearing aid
performance, and the cost associated with a high performance solution are all causes of
low market penetration. Through the use of digital signal processing, digital hearing aid
now offers what the analog hearing aid cannot offer. It proposes the possibility of
performing signal-to noise enhancement, flexible gain-processing, digital feedback
reduction, etc. In this paper, the simulation of simple digital hearing aid was developed
using MATLAB programming language. The implementation of this configurable digital
hearing aid (DHA) system includes the noise reduction filter, frequency shaper function,
and amplitude compression function. This digital hearing aid system is design to adapt
for mild and moderate hearing loss patient since different gain can be set to map
different levels of hearing loss.
 INDEX

Acknowledgement . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . ii
Abstract . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
iii
Project Idea . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
5
Software Used . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
5
Procedure . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5
Inscription . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5
MatLab Code. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 6-8

Observation And Diagram . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .9-11

Description . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1
2
Conclusion . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
12
PROJECT IDEA:-
Hearing aids System for Impaired People.

SOFTWARE USED:-
MATLAB

PROCEDURE:-
● First of all we have taken an input speech sound.

● To make the simulation real we have added some noise to the input speech signal.

● A noise reduction filter that is wavelet filter is used to eliminate to eliminate the noise.

● An frequency shaper is designed to correct for the loss of hearing at certain

frequencies. It applies high gain for higher frequencies and vice versa.

●An Amplitude compression function is used to do the task of controlling the overall
gain of a speech amplification system. Amplitude compression will ensure that the
amplified signal will not exceed saturation power.

● After the amplitude compression the output is observed in the spectrogram.

● The above steps are done by code using MatLab.

INSCRIPTION:-
FOR INPUTS:-

● A Speech sound

FOR OUTPUTS:-

● Noise added sound

● Denoise Sound

● Frequencies Shaped Sound

● Amplitude Shaped Sound

●Spectrogram Of Original Signal

● Spectrogram Of Output Signal

MATLAB CODE:
clc
clear all
close all
[INPUT]
% y = getaudiodata(recObj);
% input= 'Counting-16-44p1-mono-15secs.wav';
disp('input sound')
% input = 'audio.wav';
% [in,fs] = audioread(input);
% [y,fs] = audioread(input);
load handel.mat
fs=Fs;
y = y(:, 1);
%info = audioinfo(input);
sound(y);
pause(10);
figure,plot(y);
title('input');
xlabel('samples');
ylabel('amplitude');

[NOISE ADDITION]
y = awgn(y,40);
noi = y;
figure,plot(y);
xlabel('samples');
ylabel('amplitude');
title('awgn');
disp('playing added noise...');
sound(y);
pause(10)
 [DENOISE]
%'Fp,Fst,Ap,Ast' (passband frequency, stopband frequency,
passband ripple, stopband attenuation)
hlpf =
fdesign.lowpass('Fp,Fst,Ap,Ast',3.0e3,3.5e3,0.5,50,fs);
D = design(hlpf);
freqz(D);
x = filter(D,y);
disp('playing denoised sound');
figure,plot(x);
title('denoise');
sound(x,fs);
xlabel('samples');
ylabel('amplitude');
pause(10)

[FREQUENCY SHARPER USING PASS BAND]

y = y*100;T = 1/fs;
len = length(x);
p = log2(len);
p = ceil(p);
N = 2^p;
f1 =
fdesign.bandpass('Fst1,Fp1,Fp2,Fst2,Ast1,Ap,Ast2',2000,3000,
4000,5000,60,2,60,2*fs);
hd = design(f1,'equiripple');
y = filter(hd,x);
freqz(hd)
%%
disp('playing frequency shaped...');
sound(y,fs);
pause(10);

[AMPLITUDE SHARPER]
disp('amplitude shaper')
out1=fft(y);
phse=angle(out1);
mag=abs(out1)/N;
[magsig,~]=size(mag);
threshold=1000;
out=zeros(magsig,1);
for i=1:magsig/2
if(mag(i)>threshold)
mag(i)=threshold;mag(magsig-i)=threshold;
end
out(i)=mag(i)*exp(j*phse(i));
out(magsig-i)=out(i);
end
%%
outfinal=real(ifft(out))*10000;
disp('playing amplitude shaped...');
sound(outfinal,fs);
pause(10);

[OUTPUT]
load handel.mat

figure;
subplot(2,1,1);
specgram(noi);
title('Spectrogram of Original Signal');

subplot(2,1,2);
specgram(outfinal);
title('Spectrogram of Adjusted Signal');
OBSERVATION AND DIAGRAM:-
DESCRIPTION:-
● When the input section of the code is evaluated the input speech sound would play.

● For making the simulation real WE have added AWGN(Additive White Gussain Noise ).

●When the noise addition section is evaluated noise added sound would play .

● After that for denoisation we have used a wavelet filter. When the section of
denoisation was run the output of denoise sound was shown and the sound would play.

●Frequency Sharper and Amplitude Compression is done to obtained the desire output.

●All the outputs are obtained in the figure window.

CONCLUSION:-
 The newer digital aid is more capable of fine-tuning the sound without distorting
the quality. In this digital hearing aids system implementation using MATLAB,
sound processing is digitalized. Thus, it is possible to refine the sound signal , for
instance by reducing nooise and improving speech signals.

 This digitalization makes its possible to precisely analyze and filter the signals.
The signals can be processed in one or more frequency channels. At the end , the
digital signal is again converted to its analog form. The benefits of using digital
aids can improve quality of life by improving sound quality.