The document analyzes a method for comparing two audio files to detect human errors using fast Fourier transforms (FFT). It describes using FFT to convert audio files from the time domain to the frequency domain. It then calculates the mean squared error (MSE) between the normalized spectral densities of the two files. A low MSE would indicate the files are identical, while a higher MSE shows a difference. The document provides the steps and flowchart used, and includes examples of Matlab and Labview code implementing the comparison method on identical and non-identical audio files.
These slides deal with the basic problem of channel equalization and exposes the issue related to it and shows how it can be balanced by the usage of effective and robust algorithms.
fast-Fourier-transform-presentation and Fourier transform for wave
in
signal possessing for
physics and
geophysics
spectra analysis
periodic and non periodic wave
data sampling
The Nyquist frequency
These slides deal with the basic problem of channel equalization and exposes the issue related to it and shows how it can be balanced by the usage of effective and robust algorithms.
fast-Fourier-transform-presentation and Fourier transform for wave
in
signal possessing for
physics and
geophysics
spectra analysis
periodic and non periodic wave
data sampling
The Nyquist frequency
Presentation of Mr.Vibin Chander, CEO, Shabari Software Solutions, CBE. Delivered during the Faculty development Program on NS2 organized by Department of Computer Science, Rathinam College of Arts and Science (Autonomous), Eachanari, Coimbatore - 641021.
Deep Learning: Recurrent Neural Network (Chapter 10) Larry Guo
This Material is an in_depth study report of Recurrent Neural Network (RNN)
Material mainly from Deep Learning Book Bible, http://www.deeplearningbook.org/
Topics: Briefing, Theory Proof, Variation, Gated RNNN Intuition. Real World Application
Application (CNN+RNN on SVHN)
Also a video (In Chinese)
https://www.youtube.com/watch?v=p6xzPqRd46w
Deep Learning - Overview of my work IIMohamed Loey
Deep Learning Machine Learning MNIST CIFAR 10 Residual Network AlexNet VGGNet GoogleNet Nvidia Deep learning (DL) is a hierarchical structure network which through simulates the human brain’s structure to extract the internal and external input data’s features
Presentation of Mr.Vibin Chander, CEO, Shabari Software Solutions, CBE. Delivered during the Faculty development Program on NS2 organized by Department of Computer Science, Rathinam College of Arts and Science (Autonomous), Eachanari, Coimbatore - 641021.
Deep Learning: Recurrent Neural Network (Chapter 10) Larry Guo
This Material is an in_depth study report of Recurrent Neural Network (RNN)
Material mainly from Deep Learning Book Bible, http://www.deeplearningbook.org/
Topics: Briefing, Theory Proof, Variation, Gated RNNN Intuition. Real World Application
Application (CNN+RNN on SVHN)
Also a video (In Chinese)
https://www.youtube.com/watch?v=p6xzPqRd46w
Deep Learning - Overview of my work IIMohamed Loey
Deep Learning Machine Learning MNIST CIFAR 10 Residual Network AlexNet VGGNet GoogleNet Nvidia Deep learning (DL) is a hierarchical structure network which through simulates the human brain’s structure to extract the internal and external input data’s features
Question I Part (a) 1.SOURCE CODE LISTING for numb.docxcatheryncouper
Question I
Part (a)
1.SOURCE CODE LISTING for numbers 0 - 9:
In this program we are going to load the .wav files into MATLAB code and we can be able to play the required key tones from 0 to 9 by
executing the code and listen through the system output devices.
In order to execute the program, the .wav files must be present in the current directory in order to load or play the dial tones.
The time domain plot for the required key tone (including two frequency components) will be observed where the axis will be time on x axis &
Amplitude on Y axis.
Later the frequency Domain plot (indicating the dual frequencies) will be observed using fourier transform
of the time domain of the key tones and with the sampling rate of 8000 Hz .
In this plot the Dual tone Multiple Frequencies is observed in the plot (DTMF Concept) as each keytone is comprised of two frequency
components, one low frequency & one high frequency components. i.e. the Dual frequencies can be viewed separately in the plot for each
number from 0 - 9.
%% Program
% To enter the number manually and listen the tone.
function program_aa
fprintf('Type 1 = Auto Play \n');
fprintf('Type 2 = Manual Play\n');
Type = input('\n enter the type of Input : Type = ');
%% Automatic Input Type
if Type == 1
% loading the .wav files
dial0= 'dial0.wav';
dial1= 'dial1.wav';
dial2= 'dial2.wav';
dial3= 'dial3.wav';
dial4= 'dial4.wav';
dial5= 'dial5.wav';
dial6= 'dial6.wav';
dial7= 'dial7.wav';
dial8= 'dial8.wav';
dial9= 'dial9.wav';
% reading the loaded .wav files
keypad_0 = wavread(dial0);
keypad_1 = wavread(dial1);
keypad_2 = wavread(dial2);
keypad_3 = wavread(dial3);
keypad_4 = wavread(dial4);
keypad_5 = wavread(dial5);
keypad_6 = wavread(dial6);
keypad_7 = wavread(dial7);
keypad_8 = wavread(dial8);
keypad_9 = wavread(dial9);
% playing the .wav files using output sources
wavplay(keypad_0)
wavplay(keypad_1)
wavplay(keypad_2)
wavplay(keypad_3)
wavplay(keypad_4)
wavplay(keypad_5)
wavplay(keypad_6)
wavplay(keypad_7)
wavplay(keypad_8)
wavplay(keypad_9)
end
%% Manual Input Type
if Type == 2
N = input('enter the number of keypad : N = ');
fprintf('\n The Number Entered = %g\n',N);
Fs = 8000; % Sampling Rate is 8 kHz
%% To load and Play for number 0
if N == 0
dial0= 'dial0.wav';
keypad_0 = wavread(dial0);
wavplay(keypad_0,Fs)
% Fourier Transformation
FFT = 2^nextpow2(length(keypad_0)); % Next power of 2 from length of y
Y = fft(keypad_0,FFT)/length(keypad_0);
f = 8000/2*linspace(0,1,FFT/2);
y = 2*abs(Y(1:FFT/2));
fprintf('\n The Low frequency component in Hz = %g\n',941);
fprintf('\n The High frequency component in Hz = %g\n',1336);
fprintf('\n The Fourier transform Max value = %g\n',norm(y));
% Plot Fourier Transform magnitude spectrum.(Frequency Domain)
subplot (2,1,1)
plot(f,y) ;
...
Ensemble Empirical Mode Decomposition: An adaptive method for noise reductionIOSR Journals
Abstract:Empirical mode decomposition (EMD), a data analysis technique, is used to denoise non-stationary and non-linear processes. The method does not require any pre & post processing of signal and use of any specified basis functions. But EMD suffers from a problem called mode mixing. So to overcome this problem a new method known as Ensemble Empirical mode decomposition (EEMD) has been introduced. The presented paper gives the detail of EEMD and its application in various fields. EEMD is a time–space analysis method, in which the added white noise is averaged out with sufficient number of trials; and the averaging process results in only the component of the signal (original data). EEMD is a truly noise-assisted data analysis (NADA) method and represents a substantial improvement over the original EMD. Keywords –Data analysis, Empirical mode decomposition, intrinsic mode function, mode mixing, NADA,
Ensemble Empirical Mode Decomposition: An adaptive method for noise reduction IOSR Journals
Empirical mode decomposition (EMD), a data analysis technique, is used to denoise non-stationary
and non-linear processes. The method does not require any pre & post processing of signal and use of any
specified basis functions. But EMD suffers from a problem called mode mixing. So to overcome this problem a
new method known as Ensemble Empirical mode decomposition (EEMD) has been introduced. The presented
paper gives the detail of EEMD and its application in various fields. EEMD is a time–space analysis method, in
which the added white noise is averaged out with sufficient number of trials; and the averaging process results
in only the component of the signal (original data). EEMD is a truly noise-assisted data analysis (NADA)
method and represents a substantial improvement over the original EMD.
Data Compression using Multiple Transformation Techniques for Audio Applicati...iosrjce
IOSR Journal of Computer Engineering (IOSR-JCE) is a double blind peer reviewed International Journal that provides rapid publication (within a month) of articles in all areas of computer engineering and its applications. The journal welcomes publications of high quality papers on theoretical developments and practical applications in computer technology. Original research papers, state-of-the-art reviews, and high quality technical notes are invited for publications.
Real-time DSP Implementation of Audio Crosstalk Cancellation using Mixed Unif...CSCJournals
For high fidelity sound reproduction, it is necessary to use long filter coefficients in audio crosstalk cancellation. To implement these long filters on real-time DSP processors, conventional overlap save technique suffers from more computational power as well as processing delay. To overcome these technical problems, mixed uniform partitioned convolution technique is proposed. This method is derived by combining uniform partitioned convolution with mixed filtering technique. With the proposed method, it is possible to perform audio crosstalk cancellation even at the order of ten thousand filter taps with less computations and short processing delay. The proposed technique was implemented on 32-bit floating point DSP processor and design was provided with efficient memory management to achieve optimization in computational complexity. The computational comparison of this method with conventional methods shows that the proposed technique is very efficient for long filters
Study of the release of heat by targeted gold nanoparticles exposed to RF fields. Study the physical properties of gold nanoparticles influence their RF thermal delivery which will aid in the further development of nanoscale materials for the treatment of cancer and various biomedical applications
Welcome to WIPAC Monthly the magazine brought to you by the LinkedIn Group Water Industry Process Automation & Control.
In this month's edition, along with this month's industry news to celebrate the 13 years since the group was created we have articles including
A case study of the used of Advanced Process Control at the Wastewater Treatment works at Lleida in Spain
A look back on an article on smart wastewater networks in order to see how the industry has measured up in the interim around the adoption of Digital Transformation in the Water Industry.
Explore the innovative world of trenchless pipe repair with our comprehensive guide, "The Benefits and Techniques of Trenchless Pipe Repair." This document delves into the modern methods of repairing underground pipes without the need for extensive excavation, highlighting the numerous advantages and the latest techniques used in the industry.
Learn about the cost savings, reduced environmental impact, and minimal disruption associated with trenchless technology. Discover detailed explanations of popular techniques such as pipe bursting, cured-in-place pipe (CIPP) lining, and directional drilling. Understand how these methods can be applied to various types of infrastructure, from residential plumbing to large-scale municipal systems.
Ideal for homeowners, contractors, engineers, and anyone interested in modern plumbing solutions, this guide provides valuable insights into why trenchless pipe repair is becoming the preferred choice for pipe rehabilitation. Stay informed about the latest advancements and best practices in the field.
NO1 Uk best vashikaran specialist in delhi vashikaran baba near me online vas...Amil Baba Dawood bangali
Contact with Dawood Bhai Just call on +92322-6382012 and we'll help you. We'll solve all your problems within 12 to 24 hours and with 101% guarantee and with astrology systematic. If you want to take any personal or professional advice then also you can call us on +92322-6382012 , ONLINE LOVE PROBLEM & Other all types of Daily Life Problem's.Then CALL or WHATSAPP us on +92322-6382012 and Get all these problems solutions here by Amil Baba DAWOOD BANGALI
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Industrial Training at Shahjalal Fertilizer Company Limited (SFCL)MdTanvirMahtab2
This presentation is about the working procedure of Shahjalal Fertilizer Company Limited (SFCL). A Govt. owned Company of Bangladesh Chemical Industries Corporation under Ministry of Industries.
Final project report on grocery store management system..pdfKamal Acharya
In today’s fast-changing business environment, it’s extremely important to be able to respond to client needs in the most effective and timely manner. If your customers wish to see your business online and have instant access to your products or services.
Online Grocery Store is an e-commerce website, which retails various grocery products. This project allows viewing various products available enables registered users to purchase desired products instantly using Paytm, UPI payment processor (Instant Pay) and also can place order by using Cash on Delivery (Pay Later) option. This project provides an easy access to Administrators and Managers to view orders placed using Pay Later and Instant Pay options.
In order to develop an e-commerce website, a number of Technologies must be studied and understood. These include multi-tiered architecture, server and client-side scripting techniques, implementation technologies, programming language (such as PHP, HTML, CSS, JavaScript) and MySQL relational databases. This is a project with the objective to develop a basic website where a consumer is provided with a shopping cart website and also to know about the technologies used to develop such a website.
This document will discuss each of the underlying technologies to create and implement an e- commerce website.
Overview of the fundamental roles in Hydropower generation and the components involved in wider Electrical Engineering.
This paper presents the design and construction of hydroelectric dams from the hydrologist’s survey of the valley before construction, all aspects and involved disciplines, fluid dynamics, structural engineering, generation and mains frequency regulation to the very transmission of power through the network in the United Kingdom.
Author: Robbie Edward Sayers
Collaborators and co editors: Charlie Sims and Connor Healey.
(C) 2024 Robbie E. Sayers
1. Results Review of Detecting of Human Errors Algorithm for audio files
Author: Minh Anh Nguyen
Email: minhanhnguyen@q.com
Purpose
The purpose of this analysis is to determine if the method of comparing two audio files can be
used for detecting of human errors for channel-not-loaded-correctly issue. There are many
algorithms which can be used to compare two audio files. These algorithms are: Discrete Fourier
Transform (DFT), Short Time Fourier Transform (STFT), Wavelet, and Fast Fourier Transform
(FFT). In this project, FFT algorithm is used to compare two audio files.
Why use FFT
The fast Fourier transform (FFT) is an algorithm for converting a time-domain signal into a
frequency-domain representation of the relative amplitude of different frequency regions in the
signal. The FFT is an implementation of the Fourier transform. The Fourier transform is one of
the most useful mathematical tools for many fields of science and engineering. The Fourier
transform displays the frequency components within a time series of data. The result of the FFT
contains the frequency data and the complex transformed result. The FFT works perfectly when
analyzing exactly one cycle of a tone.
How FFT works
The FFT takes a chunk of time called a frame (number of samples) and considers that chunk to
be a single period of a repeating waveform. Most sounds are locally stationary, which means that
the sound does look like a regularly repeating function in a short period of time.
The FFT is the one-dimensional Fourier transform. If assuming a signal is saved as an array in
the variable X, then preforming fft(X) will return the Fourier transform of X as a vector of the
same size as X. However, the values returned by fft(X) are frequencies.
Applications of the FFT
There are many applications of FFT. The FFT algorithm tends to be better suited to analyzing
digital audio recordings than for filtering or synthesizing sounds. The FFT is used to determine
the frequency of a signal, to try to recognize different kinds of sound, etc.
Problem with FFT
The FFT function automatically places some restrictions on the time series to generate a
meaningful, accurate frequency response. The FFT function uses (n/2) log2 (n), it requires that
the length of the time series or total number of data points precisely equal to a 2n. Therefore, FFT
can only calculate with a fixed length waveform such as 512 points, or 1024 points, or 2048
points, etc. Another problem with using the FFT for processing sounds is that the digital
recordings must be broken up into chunks of n samples, where n always has to be an integer
2. power of 2. When the audio is broken up into chunks like this and processed with the FFT, the
filtered result will have discontinuities which cause a clicking sound in the output at each chunk
boundary. For example, if the recording has a sampling rate of 44,100 Hz, and the blocks have a
size n = 1024, then there will be an audible click every 1024 /(44,100 Hz) = 0.0232 seconds,
which is extremely annoying to say the least. The input signal must be repeats periodically and
that the periodic length is equal to the length of the actual input; otherwise leakage will occur
and cause both the amplitude and position of a frequency measurement to be inaccurate.
Techniquesfor comparing two audio files
Step 1: Load audio files
– Read in two audio files into the workspace.
Step2: Truncate both signals so that their durations are equivalent.
Step 3: Perform FFT
– Compute normalized energy spectral density (ESD) from DFT's two signals
Step 4: Compute mean-square-error (MSE)
– Compute mean-square-error (MSE) between normalized ESD's of two signals
– Two perfectly identical signals will obviously have MSE of zero.
Step 5: Display error message on the monitor/computer screen, if MSE value is not zero.
Flow chart for comparing two audiofiles
3. Figure1. Flow chart for comparing two audio files
Audio test in Simulation mode
The following steps are the procedure to perform audio test in simulation mode
1. Make sure audio files are loaded into the workspace without any issues.
2. Plot and listen to audio files, make sure a correct file is loaded.
3. Verify that the FFT value is correct.
4. Positive test: load two audio files which have 2 different signals and verify that
the test results show the difference.
5. Negative test: load two audio files which have 2 similar signals and verify that the
test result show the same thing or match.
Software
Matlab R2015b
Labview 2014
4. Summaryof Results
The following figure illustrates the difference between the displays of the FFT of the repeats
periodically input signals and the results of the techniques and flow chart for comparing two
audio files above.
Figure2. Matlab results for comparing two audio files which are identical
Figure3. Matlab results for comparing two audio files which are not identical
5. Figure4. Matlab results for comparing two audio files which are not identical
Figure5. Labview results for comparing two audio files which are identical
6. Figure6. Labview results for comparing two audio files which are not identical
These results showed that the method of comparing two audio files can be used for detecting of
human errors for channel-not-loaded-correctly issue. FFT of a time domain signal takes the
samples and calculate a new set of numbers representing the frequencies, amplitudes, and phases
of the sine waves that make up the sound.
Matlab Code for comparing two audio files.
% MatLab Code:
% Author: Minh Anh Nguyen (minhanhnguyen@q.com)
% Expects a .wav soundfile as input.
% usage: Soundfunction('myfile.wav')
% This function will read in two wav files, perform fft, and compare these
files. it
% will display on the screen whether or not the file are identifcal.
% this function is also perfom autocorrelation
% This beginning part just defines the function to be used in MatLab: takes a
% "wav" sound file as an input, and spits out a graph.
function [Soundfunction] = Soundfunction( file1, file2 );
clf % Clears the graphic screen.
close all;% Close all figures (except those of imtool.)
clc;% Clear the command window.
fontSize = 20;
7. fontSize1 = 14;
% Reads in the sound files, into a big array called y1 and y2.
%y = wavread( file );
[y1, fs1]= audioread( file1 );
[y2, fs2]= audioread( file2 );
% Normalize y1; that is, scale all values to its maximum. Note how simple it
is
% to do this in MatLab.
yn1 = y1/max(abs(y1));
yn2 = y1/max(abs(y1));
sound1 = y1;
sound2 = y2;
%% Do not modify here
%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
%%%%%% time of sound
N = length(sound1); % number of points to analyze
ls = size (sound1); % find the length of the data per second
nbits = 2^(length(sound1));
t1 = (0:1:length(sound1)-1)/fs1; %% time1
fx = fs1*(0:N/2-1)/N; %Prepare freq data for plot
T1 = 1/fs1 % period between each sample
N2 = length(sound2);
ls2 = size (sound2); % find the length of the data per second
t2 = (0:1:length(sound2)-1)/fs2;
fx2 = fs2*(0:N2/2-1)/N2; %Prepare freq data for plot
T2 = 1/fs2 % period between each sample
%%%%% cut signal for comparing
% cut signal to same length
voice1 = y1(fs1*1 : fs1*9);
voice2 = y2(fs2*1 : fs2*9);
%% find new length
N2x = length(voice2);
N1x = length(voice1);
% find new time
t1x = (0:1:length(voice1)-1)/fs1;
t2x = (0:1:length(voice2)-1)/fs2;
%% find new frequency
f2x = fs2*(0:N2x/2-1)/N2x;
f1x = fs1*(0:N1x/2-1)/N1x;
%% fft of cut signal
8. NFFT1x = 2 ^ nextpow2(N1x);
Y1x = fft(voice1, NFFT1x)/ N1x;
f1xx = (fs1/ 2 * linspace(0, 1, NFFT1x / 2+1))'; % Vector containing
frequencies in Hz
STFFT1x = ( 2 * abs(Y1x(1: NFFT1x / 2+1))); % Vector containing corresponding
amplitudes
NFFT2x = 2 ^ nextpow2(N2x);
Y2x = fft(voice2, NFFT2x) / N2x;
f2xx = (fs2 / 2 * linspace(0, 1, NFFT2x / 2+1))'; % Vector containing
frequencies in Hz
STFFT2x = ( 2 * abs(Y2x(1: NFFT2x / 2+1))); % Vector containing corresponding
amplitudes
%% plot for the cut signal
%% plot for the cut signal
figure; subplot (3,2,1); plot(t1x, voice1);
str1=sprintf('Plot Sound1 with sampling rate = %d Hz and number sample = %d',
fs1, N1x);
title(str1);
xlabel('time (sec)'); ylabel('relative signal strength'); grid on;
subplot (3,2,2); plot(t2x, voice2);
str2=sprintf('Plot Sound2 with sampling rate = %d Hz and number sample = %d',
fs2, N2x);
title(str2); xlabel('time (sec)','Fontsize', fontSize); ylabel('relative
signal strength','Fontsize', fontSize); grid on;
%% fft of cut signal
subplot (3,2,3); plot(f1xx, STFFT1x); title ('Single-sided Power spectrum for
Sound1 with same length','Fontsize', fontSize1);
ylabel ('Magnitude [dB]','Fontsize', fontSize); xlabel ('frequency
[Hz]','Fontsize', fontSize); grid on;
subplot (3,2,4); plot(f2xx, STFFT2x); title ('Single-sided Power spectrum for
Sound2 with same length','Fontsize', fontSize1);
ylabel ('Magnitude [dB]','Fontsize', fontSize); xlabel ('frequency
[Hz]','Fontsize', fontSize); grid on;
%% corlation
[C1, lag1] = xcorr(abs((fft(voice1))),abs((fft(voice2)) ));
figure, plot(lag1/fs1,C1);
ylabel('Amplitude'); grid on
title('Cross-correlation between Sound1 and sound2 files')
%% calculate mean
% Calculate the MSE
D= voice1 - voice2;
MSE=mean(D.^2);
%msgbox(strcat('MSE value is= ',mat2str(),' MSE'));
msgbox(strcat('MSE value is= ',mat2str(MSE)));
if MSE ==0;
9. msgbox('no error. Signal are the same');
disp('Signal are the same');
else
disp('Signal are not the same');
msgbox('Test error. Please check your set up');
end
Labview Code for comparing two audio files.