An Audio Classification Approach using Feature
extraction neural network classification Approach.
Krishna Kumar, Kapil Chaturvedi
Department of CSE, SIRTS, BHOPAL
Abstract: Audio signal and audio data is an
important information data. It make useful to
understand behaviour and understanding of
existing or past scenario. There are multiple
features in sound signal which help in such analysis.
There are many approach such as wavelet analysis,
chroma, centroid based approach and MFCC is the
main approach for such feature extraction. In this
paper there is a sound classification using the
proper feature extraction then followed by proper
classification using fast efficient neural network
based approach is proposed. This algorithm process
and classify the sound input dataset and help in
high classification rate classification. The execution
of this system is performed using Matlab 2018b and
result execution performance shows the efficiency of
algorithm over traditional classification solution.
Keywords: Audio classification, neural network,
MFCC, frequency , audio signals.
I INTRODUCTION
Audio signals are important part in any kind of
communication media today, where the speech
recognition help understanding and visualizing of
various scenario. There are many classification
approaches which work towards the analysis of audio
classification. The approaches are mainly work with
various parameters involved in the speech signals.
Audio information generated from different resources
plays role while working with vocal human voice or
with instrumental voice analysis. Thus previous
approaches worked with limited properties of audio
signal or any of speech input.
This paper work with the audio classification with
various content available in it. The recognition using
the approaches such as classification , regression and
other approaches work with identification of such
analysis is discussed.
The objective of the information mining procedure is to
find learning from enormous databases and change into
a human reasonable configuration. It is the procedure
by which we consequently dole out an individual thing
to one of various classifications or classes, in view of
its qualities.
For our situation:
(1) The things are audio sign (for example audios,
tracks, extracts);
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(2) Their attributes are the highlights we separate from
them (MFCC, chroma, centroid);
(3) The classes (for example speakers, instruments,
telephones, audio condition) fit the issue definition
The intricacy lies in finding a suitable connection
among highlights and classes.
Figure 1: Audio Signals Frequency.
Machine learning and its other classification
approaches worked with different signal analysis
policy. Signal frequency work with audio signal
analysis with its classification efficiency. Audio
information help working with its pitch analysis and
several other auto processing values.
AUDIO CLASSIFICATION
Quick increment in the measure of audio information
requests for an electronic strategy which permits
effective and robotized content-based grouping and
recovery of audio database [1], [2], [3], [4]. Be that as
it may, a crude audio sign information is a featureless
gathering of bytes with most simple fields connected,
for example, name, group, testing rate. This does not
promptly permit content-based grouping and recovery.
While audio substance might be portrayed by utilizing
catchphrases and content, such data has so far been
made physically.Ongoing discoveries from audio-
related physiology, psychoacoustics, and discourse
recognition, recommend that the audio-related
framework re-encodes acoustic range as far as ghostly
and transient balances. The perceptual job of
exceptionally low recurrence balances takes after quiet
message bearing waves regulating higher recurrence
transporters. The presence of adjustments in discourse
has been confirm. Dynamic data given by the balance
range incorporates quick and slower time-changing
amounts, for example, pitch, phonetic and syllabic rates
of discourse, beat of music, and so forth [8]. In
addition, discourse clarity relies upon the uprightness
of the moderate spectro-worldly vitality tweaks.
• Recognition accuracy: Audio recognition and its
analysis tool work with the fundamental of working
procedure. Ccuracy recognition from the voice sample
plays an important role of understanding more on
accuracy parameter.
• Recognition speed –An absolute recognition speed is
always an factor which plays an important role while
dealing with real-time analysis of audio.
• Pre-processing: Part conversation and frame
conversation of voice is performed with the pre-
processing module.
• HMM Training: Working with pattern understanding
over the voice sample is performed using the HMM
module which is duly performed while collecting the
sample unit.
• HMM Recognition: A pattern matching and analysis
process work with the HMM recognition while it deals
the recognition of signal based on HMM training.
ii) Artificial Neural Network Classier(ANN) Neural
network approach with its network layer , distribution
and working with different analysis platform work with
the ANN architecture [4]
• Speech breakup, noise cancellation and processing of
actual data work with pre-processing unit..
• Two kinds of acoustic features are extracted, from the
speech signal. They are Mel Frequency Cep- strum
Coefficients (MFCC) and Linear Predictive Coding
coefficients (LPCC).
II LITERATURE REVIEW
In this paper [11] proposed a content ward speaker
confirmation framework which uses distinctive sort of
data for settling on choice in regards to the personality
of guaranteed speaker. The benchmark framework
utilized is (DTW) strategy for coordinating. Discovery
for the area of the end point is critical for the
presentation of DTW based format coordinating. A
technique dependent on vowel beginning point (VOP)
is proposed finding the end purpose of an expression.
The proposed strategy for speaker check utilized the
suprasegmental and source highlights, other than the
phantom highlights. The suprasegmental highlights, for
example, pitch and length are removed from the
distorting way data of the DTW calculation. Highlights
of the excitation source, removed utilizing neural
system models are likewise utilized for content ward
speaker confirmation framework.
In this paper [12] built up a system which is effective at
segregating discourse from music on communicate FM
radio. The computational straightforwardness of the
methodology could fit wide application including the
capacity to consequently change channels when ads
show up. The calculation gives the capacity to
vigorously recognize the two classes discourse and
music and runs effectively continuously. The severe
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utilization of time-area 14 highlights evades the
requirement for a FFT processor to figure phantom
highlights. Given that this calculation has just been
created in an exploratory manner, it is normal that
improved exhibition past that detailed is conceivable.
In this paper [13] provided details regarding
development of a constant PC framework fit for
recognizing discourse signals from music flag over a
wide scope of advanced audio information. They have
inspected 13 highlights proposed to quantify
thoughtfully particular properties of discourse and
additionally music flag, and joined them in a few
multidimensional grouping structures. They have given
broad information on framework execution and the
cross-approved preparing/test arrangement used to
assess the framework. For the datasets right now being
used, the best classifier arranges with 5.8% blunder on
a casing by-outline premise, and 1.4% mistake when
coordinating long (2.4 second) fragments of audio.
In this paper [14] introduced a technique for quickly
deciding the qualities of audio examples, utilizing a
directed tree-based vector quantizer prepared to
amplify common data (MMI). Such a measure has
demonstrated effective for talker recognizable proof,
and the expansion from discourse to general audio, for
example, music, is clear. A classifier that recognizes
discourse from music and non-vocal audios is
exhibited, just as test results appearing flawless
grouping exactness might be accomplished on a little
corpus utilizing significantly under two seconds for
every test audio record. The procedures displayed here
might be reached out to different applications and
areas, for example, audio recovery by-comparability,
melodic type order, and programmed division of
persistent audio.
In this paper [15] analyzed the separation accomplished
by a few unique highlights utilizing regular planning
and test sets and a comparable classifier. The database
gathered for these tests consolidates talk from thirteen
vernaculars and music from wherever all through the
world. For every circumstance the flows in the
component space were shown by a Support vector
machine (SVM). Tests were finished on four sorts of
feature, sufficiency, cepstra, pitch and zero
convergences. For every circumstance the subordinate
of the component was furthermore used and found to
improve execution. The best execution came about due
to using the cepstra and delta cepstra which gave a
comparable slip-up rate (EER) of 1.2%. This was
eagerly trailed by institutionalized adequacy and delta
plentifulness. This in any case used an essentially less
capricious model. The pitch and delta pitch gave an
EER of 4% which was better than the zero convergence
which conveyed an EER of 6%.
In this paper [16] displayed a various leveled
framework for audio order and recovery dependent on
audio substance examination. The framework
comprises of three phases. The main stage is known as
the coarse-level audio grouping and division where
audio accounts are characterized and divided into
discourse, music, a few kinds of natural audios, and
quietness, in light of morphological and measurable
investigation of worldly bends of brief time highlights
of audio sign. In the subsequent stage, ecological
audios are additionally grouped into definite classes,
for example, adulation, downpour, feathered creatures
audio, and so forth. This fine-level order depends on
time-recurrence investigation of audio flag and
utilization of shrouded Markov model (HMM) for
grouping.
III PROBLEM DEFINITION
• Non-uniform availability of voice sample and
its process issue plays the major limitation
where a proper pre-processing technique is
required.
• Working with the proper grammer level check
with automated system, working with updated
dictionary help working with analysis is major
limitation due to limited library.
• In practice, efficient classification selection
,working with proper procedure is very much
lack and needful in approach.
Figure 2: Flow chart of proposed algorithm.
In the figure 2 above, the flow steps of complete
approach is discussed, which discuss about the various
steps involve in the process.
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• The problem of pattern recognition, which
traditionally followed the framework of Bayes
and required estimation of distributions for the
data, was transformed into an optimization
problem involving minimization of the
empirical recognition error.
• The second problem is the choice of k,
choosing k large generally results in a linear
classifier whereas small k results in nonlinear
ones. This influences the generalization
capability of the k NN classifier. The optimal
k can be found by using for instance the leave
out one method on the training set. A
disadvantage of this method is its large
computing power requirement, since for
classifying an object its distance to all the
objects in the learning set has to be calculated.
• Both pitch and loudness appeared to be an
issue for spectral flatness due to the
perception of crescendos and decrescendos
resulting from the source separation of noise
and pitch (perceived fading or increase in one
or the other).
Proposed Architecture
There are components which are described below with
their functionality.
Dataset document extracted from the web resource and
stored over the local disk. Initializing MATLAB library
and components for the data processing.
• Input file selection which is needed to match
with existing image.
 • Pre-processing the image with grayscale
conversation and further using the binarization
over it.
• Measuring features using Feature extraction
Histogram of oriented gradients (HOG)
• Filtering of color and intensity data using the
filter kernels. Further block normalization.
• Performing CNN using the number of layer
selection and number of kernel selection.
Classification of data is performed using the
traditional CNN approach.
• Performing the Hybrid proposed approach
which take advantage of SENet and CNN
layer concept of better classification. In this
scenario the quick process occurrence is
performed using suppression of less used
information. It tries to add weights to each and
every feature map in the layer.
• Thus the SE approach along with CNN is
going to work as proposed solution for quick
process with high accuracy.
• Confusion matrix computation and returning
the result parameter comparison analysis.
IV SIMULATION PLATFORM
In order to perform proper simulation over the
technique, Matlab as simulation tool is used here with
audio library. Thus working with network layer NN
library tool and audio analysis tool helped to process
the implementation and result analysis part.
MATLAB deals with:
1. Basic stream control
programming dialect
2. How to compose contents (primary
capacities) with matlab
3. How to compose capacities with
matlab
4. How to utilize the debugger
5. How to utilize the graphical interface
6. Examples of helpful contents and
capacities for audio signal handling
In the wake of finding out about matlab we will have
the capacity to utilize matlab as an apparatus to assist
us with our maths, gadgets, flag and audio signal
preparing, measurements, neural systems, control and
mechanization.
MATLAB FUNCTIONS:
imread: Read audio signals from illustrations records.
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Linguistic structure:
A = imread(filename,fmt)
[X,map] = imread(filename,fmt)
[...] = imread(filename)
[...] = imread(...,idx) (TIFF as it were)
[...] = imread(...,ref) (HDF as it were)
RESULT ANALYSIS:
This section discuss about the comparison parameters
and their analysis along with traditional algorithm. The
algorithm Proposed CNN is compared with spectro-
temportal features, deep learning and SVM approach of
audio classification. Hence the solution discuss the
efficiency of proposed approach over existing
solutions.
Computation Parameters:
In order to perform study and efficiency of proposed
algorithm. The comparison is performed using
accuracy and Error rate analysis. The confusion matrix
computation and further analysis of proposed approach
is performed. Following are the parameters and its
discussion formulae which is used for parameter
computation.
Accuracy:
Accuracy (ACC) is calculated as the number of all
correct predictions divided by the total number of the
dataset. The best accuracy is 1.0, whereas the worst is
0.0.
and
Figure 3: Accuracy
computation formulae.
The figure 3 above shows the equation formulae which
is used for computing the accuracy in the execution.
Accuracy is calculated as the total number of two
correct predictions (TP + TN) divided by the total
number of a dataset (P + N).
Error Rate:
Error rate (ERR) is calculated as the number of all
incorrect predictions divided by the total number of the
dataset. The best error rate is 0.0, whereas the worst is
1.0.
 Figure 5: Graphical analysis of comparative study
between proposed and traditional solution

In the figure 5 above, the comparison analysis between

the proposed CNN and traditional approaches is
performed. The execution result shows the efficiency
of proposed system.

Figure 4: Error
computation formulae
The figure 4 above shows the equation formulae which Comparison Analysis
is used for computing the Error rate in the execution. between the Proposed and
RESULT COMPARISON:
Following table and graph shows the statically and Previous published work

Accuracy and Error rate %

graphical analysis of approach.
95
90
Table 1: Comparison among obtained results. 85
80
Method Features Accuracy Error Rate 75
70
(EER) Accuracy Error Rate
Proposed CNN 89.2 77.8
Proposed CNN Audio 89.2 77.8
SVM 86.7 78.5

Spectro- Audio 87.4 89.5

Figure 6: Comparison Analysis between previous and
temporal
proposed work
feature ()

In the figure 6 above, the comparison analysis between

the proposed CNN and Previously paper SVM
Deep Learning Audio 85.2 78.9
algorithm is performed. The execution result shows the
efficiency of proposed system.
SVM Audio 86.7 78.5
V CONCLUSION

Sound is the signal form which is always around us and

The table 1 above shows the comparison analysis many of the latest technology, applications are working
between the algorithm and proposed CNN approach. on the sound system. Today many advantages for
The comparison made shows the efficiency of especially abled people also involve sound technology.
proposed algorithm in terms of accuracy and error rate This approach works with signals and such
efficient along with other algorithms. classification according to requirement, using them is
required. Signal analysis given an important
understanding of different motions. Working with
different chords and its visualization provides the
Comparison Analysis actual classification which can provide a proper study
between the Proposed about the input signals. Finding a proper technique
using which a better classification can be perform is
and other algorithms always a challenging issue working with dynamic
operation. This research discuss about the sound
Accuracy and Error rate %

95 classification using its different middle features and

90
85 their analysis. The recent researcher discussed their
80
75 algorithm such as support vector machine, feature
70
Accuracy Error Rate based solution and other rule based solution for the
Proposed CNN 89.2 77.8 analysis of audio signals. The presented approach
which is SENet based CNN is given which the parallel
Spectro-temporal is computing oriented fast approach for more accuracy
87.4 89.5
feature and less error rate. The approach is implemented using
Deep Learning 85.2 78.9 the MATLAB tool and further the parameter such as
SVM 86.7 78.5 Accuracy and Error rate is computed using
classification confusion matrix. Result obtained shows
the high value of accuracy and less value of error rate

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from the given proposed CNN solution. Thus the
approach is efficient and can be used in further for
better classification.
VI FUTURE WORK
The scope of the research work in the area of content
based audio classification is not limited by any means.
There is a provision of expansion of the research both
vertically (i.e., increase in the number of classes) and
laterally (i.e., adding more unique features to each
class). This will require more feature sets and more
rigorous analysis. An optimum feature set for each
audio class can be created by including certain unique
features which may improve the performance of the
classifier.
Following are the future work possibility with sound
classification:
1. Working with mobile devices and
implementing them with the voice input from
real time devices can be done.
2. It can be used with especially abled people
where the application can be very useful to
understand the person behavior.
3. This approach can be used in robotics for auto
analysis and performing the different task
according to the user requirements.
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