A review of Noise Suppression Technology for Real-Time Speech Enhancement

International Research Journal of Engineering and Technology (IRJET) e-ISSN: 2395-0056
Volume: 09 Issue: 07 | July 2022 www.irjet.net p-ISSN: 2395-0072
© 2022, IRJET | Impact Factor value: 7.529 | ISO 9001:2008 Certified Journal | Page 335
A review of Noise Suppression Technology for Real-Time
Speech Enhancement
Keshav Patta1, Hitesh Tiwari2, Mohit Kumar Tiwari3, Prof. Vaishali Gatty4
1,2,3 PG Student, Department of M.C.A. VESIT, Mumbai, Maharashtra, India.
4Professor Vaishali Gatty, Department of M.C.A, VESIT, Mumbai, Maharashtra, India.
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Abstract - Despite noise suppression being a mature space
in the signal process, it remains hugely captivated by the fine
calibration of reckoner algorithms and parameters. In this
paper, we demonstrate a Real-Time learning approach to
noise suppression. We tend to focus powerfully on keeping the
quality as low as potential, while still achieving high-quality
increased speech. A huge number ofcommunicationprograms
or systems haveintroduced next-generation noise-cancelingAI
as an alternative to reduce background sounds/noise in their
online meetings or work and madethisprocesswellorganized.
Yet, noise suppression is far more prominent than active noise
canceling systems out there in existing systems and providesa
better result. Currently leadingtechcompaniestochoose noise
suppression for communication applications to offer better
result, even though these system does notyield100%accuracy
it is still far superior than the other conventional systems.
Key Words: Noise Detection, Noise Suppression, Deep
Learning, Voice Detector, Microphone,
1. INTRODUCTION
For decades we hear that AI is the future and with its help,
Noise suppression has gained much interest in the field.
Despite important enhancements in quality, the high-level
structure has remained principally equivalent. The noise
spectrum estimation technique is the backbone of noise
suppression AI, it is derived by a voice activity detector
(VAD). It has three component which needs correct
estimators and is tough to tune. for instance,thecrudeinitial
noise estimators and also spectral estimators thatsupported
spectral subtraction are replaced by a lot of correct noise
estimators and spectral amplitude estimators. Despite the
enhancements, these estimators have remained tough to
style and have needed important manual calibration effort.
that's why recent advances in deep learning techniques are
appealing for noise suppression.
Active noise-canceling technology has been the focus of the
world for many years because of its easy implementation
and the technology sector see it as a compelling option, but
since it has drawback on hazardous level (security of user)
noise suppression technology has been seen as an only
solution for that problem. Noise suppression in devices is
achieved with twin microphonesorquad(Omni-directional)
microphones, that square measure accustomedpull inaudio
signals (noise, speech, alarms, etc.) from the encompassing
setting. These audio signals square measure sent to a digital
signal processor (DSP) with algorithms to assist separate
and suppressing background signals. Also, with advanced
algorithms, this could embody uninflected and enhancing
speech thus you'll hear and be detected clearly inspiteofthe
noise around you. Noise suppression AI also achieved
prominent results through deep learning which can detect
human voice between different noises given as an input,
these results show the advancement in Artificial Intelligence
in today’s world.
2. Voice Activity Detector (VAD)
Voice activity detection (VAD) can be defined as a technique
during which the presence or absence of a human voice is
detected. The detectionisaccustomedtotriggeringamethod.
VAD has been applied in speech-controlled applications and
devices like smartphones, which can be operated by voice
commands.MostcommonVADalgorithmsarebasedonusing
amplitude and are very efficient namely Short-Time average
Energy (STE) and Zero-crossing Rate (ZCR) but these
techniques do not work in loud or noisy environments.Using
spectral energy for this process is the best approach since it
works with high accuracyinhighspeechnoisyenvironments,
some of the spectral energy techniques are long-term
spectral envelope, Mel-frequency cepstral coefficients
(MFCCs), linear predictive coding coefficients (LPCC).
The detection process has been carried out by the technique
chosen by the engineer, in the spectralenergybasetechnique
when input comes with speech + noise, it sets the idle
frequency range in which speech can be recognized and set
the condition to remove everything else when spectral
energy is less than zero. In the next part, sum up all the parts
and set a threshold for energy to recognize active and non-
active parts in the detection, and create a length filter for
removing non-active segments, and in the last part, it creates
a buffer on both sides (e.g. 0.5 seconds) to complete the
process.

3. Noise Estimation Algorithm
The need for noise Estimation is to calculate the noise in the
speech and subtract the estimated value to get the desired
speech. With this basic idea noise estimation algorithm has
been created, the working of the algorithm is very simple in
the initial part it divides the speech signal into small parts
and starts at the beginning. The first part of the signal starts
being estimated for noise and compared with the next part
and so on when the noise signal gets estimatedina fewparts
it starts subtracting that signal from the rest of the speech
signal to get the desired output.
There are three types of noise estimation algorithms, first is
a histogram-based algorithm in which the speech signal is
divided into bins, and each bin contains a speech signal with
noise. The spectral energy in each bin has been calculatedas
the density of the noise and once it’s been calculated the
algorithm subtracts it from the speech signal and a clean
signal can be achieved. The second algorithm is the Minimal
tracking Algorithm in this algorithm we track the minimum
band frequency of the noise in the speech spectrum and
calculate the spectral energy similar to the previous
algorithm, this algorithm has two types 1 - Minimum
statistics (MS) Noise Estimation and 2 - Continuous Spectral
Minimum. Time Recursive Algorithm is the third algorithm
that uses the knowledge that speech signal has a non-
constant effect on the spectrum signal in some parts, those
parts then been calculated for signal-noise ratio (SNR) and
each part will give differentSNRstocalculatewithwhich will
give a high value and a low value. The noise considers being
absent at high and low values by which we can get the
average spectrum signal to subtract from the speech signal,
Time recursive algorithm is further derived in minima
controlled recursive averaging (MCRA) algorithm.
4. Conventional use of Noise Suppression
Technology
Current generation devices like computers, tablets,
smartphones, and many more are the prime example of the
Conventional use of Noise Suppression Technology.
Conventionally devise takes thenoiseasinputand processes
it to produce clear sound for the end-user.
For past decades devices have been improving vigorously
and competing with each other to improve more and more,
old devices like mobile phone have bad reception and user
have to suffer from background noises a lot. Later
smartphones introduce Active Noise-canceling technology
which can cancel the background noises and provide better
reception still the results were not promising since all the
devices have to rely on the hardware of that device.
Smartphones are equipped with multiple microphones to
process the incoming sound at different positions (e.g., the
first mic at bottom of the device, the second at top of the
device) to capture user sound at a different angle, after
processing input data (sound) from all the mics background
noise can be detected by matching the data and removing it
in the process and in the end, the output will be a clean
audible sound.
All this process is done by a combination ofsimplehardware
and software but it has some major flaws, imagine if the
device is only receiving data of background sound or the
device has been put far apart then it won’t get the desired
input to the process. This type of flaw compels the tech
industry to improve its hardware and software even more
but doing so also increases the cost of devicesandalsohasto
upgrade them with the latest technology.
Conventional smartphone flaws in using the mic for ANCisa
huge problem, especially in the condition where the user is
doing exercise away from the device or walking in the
market, but to some extent, these problems are solved by
new devices like a smartwatch, Bluetooth earphone, etc. but
still, the accuracy is not up to the mark and in some extreme
condition noise suppression simple fail to achieve its desire
output.
5. Background Noise Separation using Deep
Learning
Using deep learning to separatenoisefromthesoundismost
effective and revolutionized way. A study in 2015 says that
using a regression method we can create a model which
learns to produce a signal-noise ratio (SNR) for every single
spectrum and the produced signal will only give a human
voice leaving extra noise. But still, it was not perfect since it
was the initial model.
With time many more models have been introduced but the
conventional approach is a three-part process that consists
of Data gathering, training the data, and Deduction. In data
gathering, users gather a huge dataset of loud noises, with
different variations of human-animal background noise, etc.
training the data is the process in which we train the model
according to the dataset and achieve the highest accuracy
possible. Lastly, come deduction where final trainedweights
are created to detect human voice with high accuracy and
mask out all the noises.

By working on different models and creating a distinctive
model we have reached a time where we can get exceptional
results with the current-generationmodel andalsowithhigh
accuracy of 99.0%. with such a model, next-generation
devices can suppress sound using a single microphone
design.
Fig- Prosess of Deep Learning Algorithm.
6. Single Microphone Design over Multi
Microphone Design
Multi-microphone styles havea coupleofvital shortcomings.
1 – This design is more centric on smartphones whichhavea
conventional design to be used in close proximitytotheuser
(near the mouth)
2 – This conventional design put the tech industry at a
disadvantage by creating an expensive design with costly
components and software.
3 – The sound input can only be processed at the device end
so the software cannot be too precise or the result cannot be
too accurate to achieve cheap cost
If all the process can be done on the software side of a device
with the help of a single microphone in a smartphone it will
put the hardware cost as cheaper side and also will be cost-
effective. Currently extracting a person’svoicefroma speech
signal is a very difficult task even for an algorithm, no
perfectly accurate model is available yet.
The conventional model of digital signal tries to learn
continuously from the human voice by processing it bit by
bit, these models work well in some cases but fail to achieve
high accuracy in real-world trials.
Nonstationary sound can be extremely tough to recognize
when compared to a person’s speech since the signal is very
unpredictable and very thin (e.g. whistle or short ring )
whereas, Stationary sound has the same regressive
characteristic as a person’s voice with the difference in the
signal, conventional algorithms can be highly successful
while working on such sound. To achieve high accuracy
regardless of the type of sound we need to improve the
conventional algorithm models and it’s not too far when
using deep learning can achieve it.
7. Leading names in Noise SuppressionTechnology
Microsoft Teams
Noise Suppression Technology has come a long way since it
has been introduced especially after the effects of covid-19.
Microsoft has introduced a superior suppressiontechnology
for its meeting application (Teams) using artificial
intelligence to recognize and separate unwanted noise from
calls.
Achieving it was not a simple task Microsoft have to collect
over a thousand hours of sound data and categorized it into
human voice or just a noise. Microsoft has collected sound
data over dozens of languages and hundreds of other noises
to achieve high accuracy. They trained their artificial
intelligence model with this data to recognize the person’s
voice communication over the call.
Nvidia
A new way of the noise suppression model is being
developed by Nvidia using DL. Nvidia has created RTX-voice
which is very identical to its competitor Krisp.ai, which also
used a virtual AI base system to suppress noises. RTX-voice
currently runs on the latest windows only (10 - 11) and
needs a high-end GPU to process.
This extraordinary growth in this industry is a result of the
high need for clear communication over distance meetings
on online calls.
Krisp.ai
When main competitors were creating a single platform-
centric AI-based model, an AI-based company created a
platform-independent AI solution to work with many
connective applications used for meeting out in the market.
Krisp.ai give its solution to a single user or an organization
on call platform from each end (outgoing-incoming).
The most fascinating feature of krisp.ai is that it not only
blocks the outgoing extra sound from the surroundings but
also blocks the incoming surroundingsoundandonlygivesa
filtered clean voice even when the opposite user has not
krisp.ai. it has trained to suppress unwanted sounds
immediately on the call while it’s passing
8. Moving to the Cloud
Considering all the model is completelysoftware-centric,the
question arises whether they can be transferred over a
cloud, it can be simply answered as ‘yes’. N number of
devices support the cloud and hence the model can work on

them, it is also very convenient to use on both incoming and
outgoing calls over a connectivity application. Today it is
possible to migrate over the cloud but it was still a dream
decade ago due to the heavy hardware requirements of the
suppression system. An original equipment manufacturer
must meet the standard set by mobile companies to achieve
high-quality communication, sadly even today multi-
microphone design is the only solution we get. Still, we can
get a high level of noise suppression thanks to DL in the
cloud which can support a singular microphone design
9. CONCLUSIONS
In this paper we have tried to demonstrate how the noise
suppression technology has been replacing Conventional
Active Noise Canceling Technology and get the summarized
knowledge of voice activity detection which help to detect
voice in the transmission we also learned about different
noise estimation algorithm which is important to estimate
the noise present in a signal. Learning about the leading
company and technology was the important part of this
study and we learn how conventional methodarebeingused
but still yield mediocre result. Next generation need to be
shifted over single microphone design from multi
microphone design if industry need to be innovative as well
as cost effective, it can be achieved with the help of cloud
technology and we are sure noise suppression technology
will yield high accuracy then any current available
technologies.
A superior noise suppression technology can help user
experience grow like never before, it can help improve
customer service (prevent backgroundnoisedisturbance on
important calls), it can be used in restaurant’s for new gen
ordering system over voice commands (such technology is
being used in today’s date with some flows), personal
smartphone assistant is common nowadays just imagine it
with more higher accuracy to command such system with
any hassle of repeating our self over and over again. The
more dynamic these model gets the more possibility can be
open with such system, deep learning has a potential to
create flawless suppression system.
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