This document summarizes and compares several techniques for enhancing the intelligibility of speech signals corrupted by noise. It describes single channel techniques like spectral subtraction, spectral subtraction with oversubtraction, and nonlinear spectral subtraction. It also covers multi-channel techniques such as adaptive noise cancellation and multisensory beamforming. Additionally, it discusses spectral subtraction using adaptive averaging, noise reduction using enhanced Wiener filtering, and other adaptive neuro-fuzzy techniques for speech enhancement. The goal of these techniques is to improve the quality and intelligibility of noisy speech signals.
A REVIEW OF LPC METHODS FOR ENHANCEMENT OF SPEECH SIGNALSijiert bestjournal
This paper presents a review of LPC methods for enhancement of speech signals. The purpose of all method of speech enhancement is to impr ove quality of speech signal by minimizing the background noise . This paper especially comments on Power Spectral Subtraction,Multiband Spectral Subtraction,Non - Linear Spectral Subtraction Method,MMSE Spectral Subtraction Method and Spectral Subtraction based on perceptual properties. As the spectral subtraction produces the Residual noise,Musical noise,Linear Predictive Analysis method is used to enhance the Speech.
Novel Approach of Implementing Psychoacoustic model for MPEG-1 Audioinventy
Research Inventy : International Journal of Engineering and Science is published by the group of young academic and industrial researchers with 12 Issues per year. It is an online as well as print version open access journal that provides rapid publication (monthly) of articles in all areas of the subject such as: civil, mechanical, chemical, electronic and computer engineering as well as production and information technology. The Journal welcomes the submission of manuscripts that meet the general criteria of significance and scientific excellence. Papers will be published by rapid process within 20 days after acceptance and peer review process takes only 7 days. All articles published in Research Inventy will be peer-reviewed.
IJRET : International Journal of Research in Engineering and Technology is an international peer reviewed, online journal published by eSAT Publishing House for the enhancement of research in various disciplines of Engineering and Technology. The aim and scope of the journal is to provide an academic medium and an important reference for the advancement and dissemination of research results that support high-level learning, teaching and research in the fields of Engineering and Technology. We bring together Scientists, Academician, Field Engineers, Scholars and Students of related fields of Engineering and Technology.
A REVIEW OF LPC METHODS FOR ENHANCEMENT OF SPEECH SIGNALSijiert bestjournal
This paper presents a review of LPC methods for enhancement of speech signals. The purpose of all method of speech enhancement is to impr ove quality of speech signal by minimizing the background noise . This paper especially comments on Power Spectral Subtraction,Multiband Spectral Subtraction,Non - Linear Spectral Subtraction Method,MMSE Spectral Subtraction Method and Spectral Subtraction based on perceptual properties. As the spectral subtraction produces the Residual noise,Musical noise,Linear Predictive Analysis method is used to enhance the Speech.
Novel Approach of Implementing Psychoacoustic model for MPEG-1 Audioinventy
Research Inventy : International Journal of Engineering and Science is published by the group of young academic and industrial researchers with 12 Issues per year. It is an online as well as print version open access journal that provides rapid publication (monthly) of articles in all areas of the subject such as: civil, mechanical, chemical, electronic and computer engineering as well as production and information technology. The Journal welcomes the submission of manuscripts that meet the general criteria of significance and scientific excellence. Papers will be published by rapid process within 20 days after acceptance and peer review process takes only 7 days. All articles published in Research Inventy will be peer-reviewed.
IJRET : International Journal of Research in Engineering and Technology is an international peer reviewed, online journal published by eSAT Publishing House for the enhancement of research in various disciplines of Engineering and Technology. The aim and scope of the journal is to provide an academic medium and an important reference for the advancement and dissemination of research results that support high-level learning, teaching and research in the fields of Engineering and Technology. We bring together Scientists, Academician, Field Engineers, Scholars and Students of related fields of Engineering and Technology.
Teager Energy Operation on Wavelet Packet Coefficients for Enhancing Noisy Sp...CSCJournals
In this paper a new thresholding based speech enhancement approach is presented, where the threshold is statistically determined by employing the Teager energy operation on the Wavelet Packet (WP) coefficients of noisy speech. The threshold thus obtained is applied on the WP coefficients of the noisy speech by using a hard thresholding function in order to obtain an enhanced speech. Detailed simulations are carried out in the presence of white, car, pink, and babble noises to evaluate the performance of the proposed method. Standard objective measures, spectrogram representations and subjective listening tests show that the proposed method outperforms the existing state-of-the-art thresholding based speech enhancement approaches for noisy speech from high to low levels of SNR.
In the present-day communications speech signals get contaminated due to
various sorts of noises that degrade the speech quality and adversely impacts
speech recognition performance. To overcome these issues, a novel approach
for speech enhancement using Modified Wiener filtering is developed and
power spectrum computation is applied for degraded signal to obtain the
noise characteristics from a noisy spectrum. In next phase, MMSE technique
is applied where Gaussian distribution of each signal i.e. original and noisy
signal is analyzed. The Gaussian distribution provides spectrum estimation
and spectral coefficient parameters which can be used for probabilistic model
formulation. Moreover, a-priori-SNR computation is also incorporated for
coefficient updation and noise presence estimation which operates similar to
the conventional VAD. However, conventional VAD scheme is based on the
hard threshold which is not capable to derive satisfactory performance and a
soft-decision based threshold is developed for improving the performance of
speech enhancement. An extensive simulation study is carried out using
MATLAB simulation tool on NOIZEUS speech database and a comparative
study is presented where proposed approach is proved better in comparison
with existing technique.
Speech Emotion Recognition is a recent research topic in the Human Computer Interaction (HCI) field. The need has risen for a more natural communication interface between humans and computer, as computers have become an integral part of our lives. A lot of work currently going on to improve the interaction between humans and computers. To achieve this goal, a computer would have to be able to distinguish its present situation and respond differently depending on that observation. Part of this process involves understanding a user‟s emotional state. To make the human computer interaction more natural, the objective is that computer should be able to recognize emotional states in the same as human does. The efficiency of emotion recognition system depends on type of features extracted and classifier used for detection of emotions. The proposed system aims at identification of basic emotional states such as anger, joy, neutral and sadness from human speech. While classifying different emotions, features like MFCC (Mel Frequency Cepstral Coefficient) and Energy is used. In this paper, Standard Emotional Database i.e. English Database is used which gives the satisfactory detection of emotions than recorded samples of emotions. This methodology describes and compares the performances of Learning Vector Quantization Neural Network (LVQ NN), Multiclass Support Vector Machine (SVM) and their combination for emotion recognition.
This paper contains a report on an Audio-Visual Client Recognition System using Matlab software which identifies five clients and can be improved to identify as many clients as possible depending on the number of clients it is trained to identify which was successfully implemented. The implementation was accomplished first by visual recognition system implemented using The Principal Component Analysis, Linear Discriminant Analysis and Nearest Neighbour Classifier. A successful implementation of second part was achieved by audio recognition using Mel-Frequency Cepstrum Coefficient, Linear Discriminant Analysis and Nearest Neighbour Classifier the system was tested using images and sounds that have not been trained to the system to see whether it can detect an intruder which lead us to a very successful result with précised response to intruder.
EFFECT OF MFCC BASED FEATURES FOR SPEECH SIGNAL ALIGNMENTSijnlc
The fundamental techniques used for man-machine communication include Speech synthesis, speech
recognition, and speech transformation. Feature extraction techniques provide a compressed
representation of the speech signals. The HNM analyses and synthesis provides high quality speech with
less number of parameters. Dynamic time warping is well known technique used for aligning two given
multidimensional sequences. It locates an optimal match between the given sequences. The improvement in
the alignment is estimated from the corresponding distances. The objective of this research is to investigate
the effect of dynamic time warping on phrases, words, and phonemes based alignments. The speech signals
in the form of twenty five phrases were recorded. The recorded material was segmented manually and
aligned at sentence, word, and phoneme level. The Mahalanobis distance (MD) was computed between the
aligned frames. The investigation has shown better alignment in case of HNM parametric domain. It has
been seen that effective speech alignment can be carried out even at phrase level.
International Journal of Engineering Research and Applications (IJERA) aims to cover the latest outstanding developments in the field of all Engineering Technologies & science.
International Journal of Engineering Research and Applications (IJERA) is a team of researchers not publication services or private publications running the journals for monetary benefits, we are association of scientists and academia who focus only on supporting authors who want to publish their work. The articles published in our journal can be accessed online, all the articles will be archived for real time access.
Our journal system primarily aims to bring out the research talent and the works done by sciaentists, academia, engineers, practitioners, scholars, post graduate students of engineering and science. This journal aims to cover the scientific research in a broader sense and not publishing a niche area of research facilitating researchers from various verticals to publish their papers. It is also aimed to provide a platform for the researchers to publish in a shorter of time, enabling them to continue further All articles published are freely available to scientific researchers in the Government agencies,educators and the general public. We are taking serious efforts to promote our journal across the globe in various ways, we are sure that our journal will act as a scientific platform for all researchers to publish their works online.
Voice recognition is the process of automatically recognizing who is speaking on the basis of individual information included in speech waves. This technique makes it possible to use the speaker's voice to verify their identity and control access to services such as voice dialing, banking by telephone, telephone shopping, database access services, information services, voice mail, security control for confidential information areas, and remote access to computers.
This document describes how to build a simple, yet complete and representative automatic speaker recognition system. Such a speaker recognition system has potential in many security applications. For example, users have to speak a PIN (Personal Identification Number) in order to gain access to the laboratory door, or users have to speak their credit card number over the telephone line to verify their identity. By checking the voice characteristics of the input utterance, using an automatic speaker recognition system similar to the one that we will describe, the system is able to add an extra level of security.
EFFECT OF DYNAMIC TIME WARPING ON ALIGNMENT OF PHRASES AND PHONEMESkevig
Speech synthesis and recognition are the basic techniques used for man-machine communication. This type
of communication is valuable when our hands and eyes are busy in some other task such as driving a
vehicle, performing surgery, or firing weapons at the enemy. Dynamic time warping (DTW) is mostly used
for aligning two given multidimensional sequences. It finds an optimal match between the given sequences.
The distance between the aligned sequences should be relatively lesser as compared to unaligned
sequences. The improvement in the alignment may be estimated from the corresponding distances. This
technique has applications in speech recognition, speech synthesis, and speaker transformation. The
objective of this research is to investigate the amount of improvement in the alignment corresponding to the
sentence based and phoneme based manually aligned phrases. The speech signals in the form of twenty five
phrases were recorded from each of six speakers (3 males and 3 females). The recorded material was
segmented manually and aligned at sentence and phoneme level. The aligned sentences of different speaker
pairs were analyzed using HNM and the HNM parameters were further aligned at frame level using DTW.
Mahalanobis distances were computed for each pair of sentences. The investigations have shown more than
20 % reduction in the average Mahalanobis distances.
International Journal of Engineering Research and Applications (IJERA) is an open access online peer reviewed international journal that publishes research and review articles in the fields of Computer Science, Neural Networks, Electrical Engineering, Software Engineering, Information Technology, Mechanical Engineering, Chemical Engineering, Plastic Engineering, Food Technology, Textile Engineering, Nano Technology & science, Power Electronics, Electronics & Communication Engineering, Computational mathematics, Image processing, Civil Engineering, Structural Engineering, Environmental Engineering, VLSI Testing & Low Power VLSI Design etc.
Broad phoneme classification using signal based featuresijsc
Speech is the most efficient and popular means of human communication Speech is produced as a sequence
of phonemes. Phoneme recognition is the first step performed by automatic speech recognition system. The
state-of-the-art recognizers use mel-frequency cepstral coefficients (MFCC) features derived through short
time analysis, for which the recognition accuracy is limited. Instead of this, here broad phoneme
classification is achieved using features derived directly from the speech at the signal level itself. Broad
phoneme classes include vowels, nasals, fricatives, stops, approximants and silence. The features identified
useful for broad phoneme classification are voiced/unvoiced decision, zero crossing rate (ZCR), short time
energy, most dominant frequency, energy in most dominant frequency, spectral flatness measure and first
three formants. Features derived from short time frames of training speech are used to train a multilayer
feedforward neural network based classifier with manually marked class label as output and classification
accuracy is then tested. Later this broad phoneme classifier is used for broad syllable structure prediction
which is useful for applications such as automatic speech recognition and automatic language
identification.
A New Speech Enhancement Technique to Reduce Residual Noise Using Perceptual ...IOSR Journals
Abstract- This paper deals with residual musical noise which results from the perceptual speech enhancement
type algorithms and especially using wiener filtering approach. Perceptual speech enhancement techniques
perform better than the non perceptual techniques, most of them still return a trouble residual musical noise.
This is due to that only noise above the noise masking threshold (NMT) is filtered out then noise below the noise
masking threshold (NMT) can become audible if its maskers are filtered. It can affect the performance of
perceptual speech enhancement method that process the audible noise only (Residual noise is still present). In
order to overcome this drawback a new speech enhancement technique is proposed here.The main aim here is
to improve the enhanced speech signal quality provided by perceptual wiener filtering and by controlling the
latter via a second filter regarded as a psychoacoustically motivated weighting factor. The simulation results
gives the information that the performance is improved compared to other perceptual speech enhancement
methods.
HYBRID APPROACH FOR NOISE REMOVAL AND IMAGE ENHANCEMENT OF BRAIN TUMORS IN MA...acijjournal
In medical image processing, Magnetic Resonance Imaging (MRI) is one of significant diagnostic
techniques. It provides high quality of important information about the analysis of human soft tissue when
measured with CT imaging modalities; hence it is suitable for diagnosis at best. However, if it gives quality
of information, image may distorted by noise because of image acquisition device and transmission. The
noises in MR image reduces the quality of image and also damages the segmentation task which can lead
to faulty diagnosis. Noises have to reduce at the same time there is no information loss. This paper propose
a hybrid approach to enhance the brain tumor MRI images using combined features of Anisotropic
Diffusion Filter (ADF) with Modified Decision Based Unsymmetric Trimmed Median Filter (MDBUTMF).
ADF scheme provides a superior performance by removing noise while preserving image details and
enhancing edges. MDBUTMF helps in image denoising as well as preserving edges satisfactorily when the
noise level is high. The performance of this filter is evaluated by carrying out a qualitative comparison of
this method with other filters namely, ADF filter, Modified Decision Algorithm, Median filter, MDBUTMF.
Teager Energy Operation on Wavelet Packet Coefficients for Enhancing Noisy Sp...CSCJournals
In this paper a new thresholding based speech enhancement approach is presented, where the threshold is statistically determined by employing the Teager energy operation on the Wavelet Packet (WP) coefficients of noisy speech. The threshold thus obtained is applied on the WP coefficients of the noisy speech by using a hard thresholding function in order to obtain an enhanced speech. Detailed simulations are carried out in the presence of white, car, pink, and babble noises to evaluate the performance of the proposed method. Standard objective measures, spectrogram representations and subjective listening tests show that the proposed method outperforms the existing state-of-the-art thresholding based speech enhancement approaches for noisy speech from high to low levels of SNR.
In the present-day communications speech signals get contaminated due to
various sorts of noises that degrade the speech quality and adversely impacts
speech recognition performance. To overcome these issues, a novel approach
for speech enhancement using Modified Wiener filtering is developed and
power spectrum computation is applied for degraded signal to obtain the
noise characteristics from a noisy spectrum. In next phase, MMSE technique
is applied where Gaussian distribution of each signal i.e. original and noisy
signal is analyzed. The Gaussian distribution provides spectrum estimation
and spectral coefficient parameters which can be used for probabilistic model
formulation. Moreover, a-priori-SNR computation is also incorporated for
coefficient updation and noise presence estimation which operates similar to
the conventional VAD. However, conventional VAD scheme is based on the
hard threshold which is not capable to derive satisfactory performance and a
soft-decision based threshold is developed for improving the performance of
speech enhancement. An extensive simulation study is carried out using
MATLAB simulation tool on NOIZEUS speech database and a comparative
study is presented where proposed approach is proved better in comparison
with existing technique.
Speech Emotion Recognition is a recent research topic in the Human Computer Interaction (HCI) field. The need has risen for a more natural communication interface between humans and computer, as computers have become an integral part of our lives. A lot of work currently going on to improve the interaction between humans and computers. To achieve this goal, a computer would have to be able to distinguish its present situation and respond differently depending on that observation. Part of this process involves understanding a user‟s emotional state. To make the human computer interaction more natural, the objective is that computer should be able to recognize emotional states in the same as human does. The efficiency of emotion recognition system depends on type of features extracted and classifier used for detection of emotions. The proposed system aims at identification of basic emotional states such as anger, joy, neutral and sadness from human speech. While classifying different emotions, features like MFCC (Mel Frequency Cepstral Coefficient) and Energy is used. In this paper, Standard Emotional Database i.e. English Database is used which gives the satisfactory detection of emotions than recorded samples of emotions. This methodology describes and compares the performances of Learning Vector Quantization Neural Network (LVQ NN), Multiclass Support Vector Machine (SVM) and their combination for emotion recognition.
This paper contains a report on an Audio-Visual Client Recognition System using Matlab software which identifies five clients and can be improved to identify as many clients as possible depending on the number of clients it is trained to identify which was successfully implemented. The implementation was accomplished first by visual recognition system implemented using The Principal Component Analysis, Linear Discriminant Analysis and Nearest Neighbour Classifier. A successful implementation of second part was achieved by audio recognition using Mel-Frequency Cepstrum Coefficient, Linear Discriminant Analysis and Nearest Neighbour Classifier the system was tested using images and sounds that have not been trained to the system to see whether it can detect an intruder which lead us to a very successful result with précised response to intruder.
EFFECT OF MFCC BASED FEATURES FOR SPEECH SIGNAL ALIGNMENTSijnlc
The fundamental techniques used for man-machine communication include Speech synthesis, speech
recognition, and speech transformation. Feature extraction techniques provide a compressed
representation of the speech signals. The HNM analyses and synthesis provides high quality speech with
less number of parameters. Dynamic time warping is well known technique used for aligning two given
multidimensional sequences. It locates an optimal match between the given sequences. The improvement in
the alignment is estimated from the corresponding distances. The objective of this research is to investigate
the effect of dynamic time warping on phrases, words, and phonemes based alignments. The speech signals
in the form of twenty five phrases were recorded. The recorded material was segmented manually and
aligned at sentence, word, and phoneme level. The Mahalanobis distance (MD) was computed between the
aligned frames. The investigation has shown better alignment in case of HNM parametric domain. It has
been seen that effective speech alignment can be carried out even at phrase level.
International Journal of Engineering Research and Applications (IJERA) aims to cover the latest outstanding developments in the field of all Engineering Technologies & science.
International Journal of Engineering Research and Applications (IJERA) is a team of researchers not publication services or private publications running the journals for monetary benefits, we are association of scientists and academia who focus only on supporting authors who want to publish their work. The articles published in our journal can be accessed online, all the articles will be archived for real time access.
Our journal system primarily aims to bring out the research talent and the works done by sciaentists, academia, engineers, practitioners, scholars, post graduate students of engineering and science. This journal aims to cover the scientific research in a broader sense and not publishing a niche area of research facilitating researchers from various verticals to publish their papers. It is also aimed to provide a platform for the researchers to publish in a shorter of time, enabling them to continue further All articles published are freely available to scientific researchers in the Government agencies,educators and the general public. We are taking serious efforts to promote our journal across the globe in various ways, we are sure that our journal will act as a scientific platform for all researchers to publish their works online.
Voice recognition is the process of automatically recognizing who is speaking on the basis of individual information included in speech waves. This technique makes it possible to use the speaker's voice to verify their identity and control access to services such as voice dialing, banking by telephone, telephone shopping, database access services, information services, voice mail, security control for confidential information areas, and remote access to computers.
This document describes how to build a simple, yet complete and representative automatic speaker recognition system. Such a speaker recognition system has potential in many security applications. For example, users have to speak a PIN (Personal Identification Number) in order to gain access to the laboratory door, or users have to speak their credit card number over the telephone line to verify their identity. By checking the voice characteristics of the input utterance, using an automatic speaker recognition system similar to the one that we will describe, the system is able to add an extra level of security.
EFFECT OF DYNAMIC TIME WARPING ON ALIGNMENT OF PHRASES AND PHONEMESkevig
Speech synthesis and recognition are the basic techniques used for man-machine communication. This type
of communication is valuable when our hands and eyes are busy in some other task such as driving a
vehicle, performing surgery, or firing weapons at the enemy. Dynamic time warping (DTW) is mostly used
for aligning two given multidimensional sequences. It finds an optimal match between the given sequences.
The distance between the aligned sequences should be relatively lesser as compared to unaligned
sequences. The improvement in the alignment may be estimated from the corresponding distances. This
technique has applications in speech recognition, speech synthesis, and speaker transformation. The
objective of this research is to investigate the amount of improvement in the alignment corresponding to the
sentence based and phoneme based manually aligned phrases. The speech signals in the form of twenty five
phrases were recorded from each of six speakers (3 males and 3 females). The recorded material was
segmented manually and aligned at sentence and phoneme level. The aligned sentences of different speaker
pairs were analyzed using HNM and the HNM parameters were further aligned at frame level using DTW.
Mahalanobis distances were computed for each pair of sentences. The investigations have shown more than
20 % reduction in the average Mahalanobis distances.
International Journal of Engineering Research and Applications (IJERA) is an open access online peer reviewed international journal that publishes research and review articles in the fields of Computer Science, Neural Networks, Electrical Engineering, Software Engineering, Information Technology, Mechanical Engineering, Chemical Engineering, Plastic Engineering, Food Technology, Textile Engineering, Nano Technology & science, Power Electronics, Electronics & Communication Engineering, Computational mathematics, Image processing, Civil Engineering, Structural Engineering, Environmental Engineering, VLSI Testing & Low Power VLSI Design etc.
Broad phoneme classification using signal based featuresijsc
Speech is the most efficient and popular means of human communication Speech is produced as a sequence
of phonemes. Phoneme recognition is the first step performed by automatic speech recognition system. The
state-of-the-art recognizers use mel-frequency cepstral coefficients (MFCC) features derived through short
time analysis, for which the recognition accuracy is limited. Instead of this, here broad phoneme
classification is achieved using features derived directly from the speech at the signal level itself. Broad
phoneme classes include vowels, nasals, fricatives, stops, approximants and silence. The features identified
useful for broad phoneme classification are voiced/unvoiced decision, zero crossing rate (ZCR), short time
energy, most dominant frequency, energy in most dominant frequency, spectral flatness measure and first
three formants. Features derived from short time frames of training speech are used to train a multilayer
feedforward neural network based classifier with manually marked class label as output and classification
accuracy is then tested. Later this broad phoneme classifier is used for broad syllable structure prediction
which is useful for applications such as automatic speech recognition and automatic language
identification.
A New Speech Enhancement Technique to Reduce Residual Noise Using Perceptual ...IOSR Journals
Abstract- This paper deals with residual musical noise which results from the perceptual speech enhancement
type algorithms and especially using wiener filtering approach. Perceptual speech enhancement techniques
perform better than the non perceptual techniques, most of them still return a trouble residual musical noise.
This is due to that only noise above the noise masking threshold (NMT) is filtered out then noise below the noise
masking threshold (NMT) can become audible if its maskers are filtered. It can affect the performance of
perceptual speech enhancement method that process the audible noise only (Residual noise is still present). In
order to overcome this drawback a new speech enhancement technique is proposed here.The main aim here is
to improve the enhanced speech signal quality provided by perceptual wiener filtering and by controlling the
latter via a second filter regarded as a psychoacoustically motivated weighting factor. The simulation results
gives the information that the performance is improved compared to other perceptual speech enhancement
methods.
HYBRID APPROACH FOR NOISE REMOVAL AND IMAGE ENHANCEMENT OF BRAIN TUMORS IN MA...acijjournal
In medical image processing, Magnetic Resonance Imaging (MRI) is one of significant diagnostic
techniques. It provides high quality of important information about the analysis of human soft tissue when
measured with CT imaging modalities; hence it is suitable for diagnosis at best. However, if it gives quality
of information, image may distorted by noise because of image acquisition device and transmission. The
noises in MR image reduces the quality of image and also damages the segmentation task which can lead
to faulty diagnosis. Noises have to reduce at the same time there is no information loss. This paper propose
a hybrid approach to enhance the brain tumor MRI images using combined features of Anisotropic
Diffusion Filter (ADF) with Modified Decision Based Unsymmetric Trimmed Median Filter (MDBUTMF).
ADF scheme provides a superior performance by removing noise while preserving image details and
enhancing edges. MDBUTMF helps in image denoising as well as preserving edges satisfactorily when the
noise level is high. The performance of this filter is evaluated by carrying out a qualitative comparison of
this method with other filters namely, ADF filter, Modified Decision Algorithm, Median filter, MDBUTMF.
International Journal of Engineering Research and Development (IJERD)IJERD Editor
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Noise reduction in speech processing using improved active noise control (anc...eSAT Journals
Abstract An improved feed forward adaptive Active Noise Control (ANC) scheme is proposed by using Voice Activity Detector (VAD) and wiener filtering method. The ‘speech-plus-noise’ periods and ‘noise-only’ periods are separated using VAD and the unwanted noise is removed by adaptive filtering method. By using Speech Distortion Weighted- Multichannel Wiener Filtering (SDW-MWF) algorithm the noise periods which is present along with the speech signal, is processed and filtered out. The background noise along with the speech samples are removed by the iterative procedure of filtering process. Feed forward based ANC is used to achieve a system with a better noise reduction in speech processing. Adaptive filtering process is carried out and the speech signal without the background noise can be achieved. Key Words: Active Noise Control (ANC), Noise reduction, Adaptive filtering, Feed forward ANC
Noise reduction in speech processing using improved active noise control (anc...eSAT Publishing House
IJRET : International Journal of Research in Engineering and Technology is an international peer reviewed, online journal published by eSAT Publishing House for the enhancement of research in various disciplines of Engineering and Technology. The aim and scope of the journal is to provide an academic medium and an important reference for the advancement and dissemination of research results that support high-level learning, teaching and research in the fields of Engineering and Technology. We bring together Scientists, Academician, Field Engineers, Scholars and Students of related fields of Engineering and Technology
Design and Implementation of Polyphase based Subband Adaptive Structure for N...Pratik Ghotkar
With the tremendous growth in the Digital Signal processing technology, there are many techniques available to remove noise from the speech signals which is used in the speech processing. Widely used LMS algorithm is modified with much advancement but still there are many limitations are introducing. This paper consist of a new approach i.e. subband adaptive processing for noise cancelation in the speech signals. Subband processing employs the multirate signal processing. The polyphase based subband adaptive implementation finds better results in term of MMSE , PSNR and processing time; also the synthesis filter bank is works on the lower data rate which reduces the computational Burdon as compare to the direct implementation of Subband adaptive filter. The normalized least mean squares (NLMS) algorithm is a class of adaptive filter used.
METHOD FOR REDUCING OF NOISE BY IMPROVING SIGNAL-TO-NOISE-RATIO IN WIRELESS LANIJNSA Journal
The signal to noise ratio (SNR) is one of the important measures for reducing the noise.A technique that uses a linear prediction error filter (LPEF) and an adaptive digital filter (ADF) to achieve noise reduction in a speech and image degraded by additive background noise is proposed. Since a speech signal can be represented as the stationary signal over a short interval of time, most of speech signal can be predicted by the LPEF. This estimation is performed by the ADF which is used as system identification. Noise reduction is achieved by subtracting the reconstructed noise from the speech degraded by additive background noise. Most of the MR image accelerating methods suffers from degradation of acquired images, which is often correlated with the degree of acceleration. However, Wideband MRI is a novel technique that transcends such flaws.In this paper we proposed LPEF and ADF for reducing the noise in speech and also we demonstrate that Wideband MRI is capable of obtaining images with identical quality as conventional MR images in terms of SNR in wireless LAN.
Improvement of minimum tracking in Minimum Statistics noise estimation methodCSCJournals
Noise spectrum estimation is a fundamental component of speech enhancement and speech recognition systems. In this paper we propose a new method for minimum tracking in Minimum Statistics (MS) noise estimation method. This noise estimation algorithm is proposed for highly nonstationary noise environments. This was confirmed with formal listening tests which indicated that the proposed noise estimation algorithm when integrated in speech enhancement was preferred over other noise estimation algorithms.
Development of Algorithm for Voice Operated Switch for Digital Audio Control ...IJMER
International Journal of Modern Engineering Research (IJMER) is Peer reviewed, online Journal. It serves as an international archival forum of scholarly research related to engineering and science education.
Single Channel Speech Enhancement using Wiener Filter and Compressive Sensing IJECEIAES
The speech enhancement algorithms are utilized to overcome multiple limitation factors in recent applications such as mobile phone and communication channel. The challenges focus on corrupted speech solution between noise reduction and signal distortion. We used a modified Wiener filter and compressive sensing (CS) to investigate and evaluate the improvement of speech quality. This new method adapted noise estimation and Wiener filter gain function in which to increase weight amplitude spectrum and improve mitigation of interested signals. The CS is then applied using the gradient projection for sparse reconstruction (GPSR) technique as a study system to empirically investigate the interactive effects of the corrupted noise and obtain better perceptual improvement aspects to listener fatigue with noiseless reduction conditions. The proposed algorithm shows an enhancement in testing performance evaluation of objective assessment tests outperform compared to other conventional algorithms at various noise type conditions of 0, 5, 10, 15 dB SNRs. Therefore, the proposed algorithm significantly achieved the speech quality improvement and efficiently obtained higher performance resulting in better noise reduction compare to other conventional algorithms.
Speech Enhancement for Nonstationary Noise Environmentssipij
In this paper, we present a simultaneous detection and estimation approach for speech enhancement in nonstationary noise environments. A detector for speech presence in the short-time Fourier transform domain is combined with an estimator, which jointly minimizes a cost function that takes into account both detection and estimation errors. Under speech-presence, the cost is proportional to a quadratic spectral amplitude error, while under speech-absence, the distortion depends on a certain attenuation factor. Experimental results demonstrate the advantage of using the proposed simultaneous detection and estimation approach which facilitate suppression of nonstationary noise with a controlled level of speech distortion.
ROBUST FEATURE EXTRACTION USING AUTOCORRELATION DOMAIN FOR NOISY SPEECH RECOG...sipij
Previous research has found autocorrelation domain as an appropriate domain for signal and noise
separation. This paper discusses a simple and effective method for decreasing the effect of noise on the
autocorrelation of the clean signal. This could later be used in extracting mel cepstral parameters for
speech recognition. Two different methods are proposed to deal with the effect of error introduced by
considering speech and noise completely uncorrelated. The basic approach deals with reducing the effect
of noise via estimation and subtraction of its effect from the noisy speech signal autocorrelation. In order
to improve this method, we consider inserting a speech/noise cross correlation term into the equations used
for the estimation of clean speech autocorrelation, using an estimate of it, found through Kernel method.
Alternatively, we used an estimate of the cross correlation term using an averaging approach. A further
improvement was obtained through introduction of an overestimation parameter in the basic method. We
tested our proposed methods on the Aurora 2 task. The Basic method has shown considerable improvement
over the standard features and some other robust autocorrelation-based features. The proposed techniques
have further increased the robustness of the basic autocorrelation-based method.
Speech enhancement using spectral subtraction technique with minimized cross ...eSAT Journals
Abstract The aim of speech enhancement is to get significant reduction of noise and enhanced speech from noisy speech. There are several
approaches for speech enhancement .earlier approaches didn’t consider cross spectral terms into account. Cross spectral terms
become prominent when processing window size becomes small i.e. 20ms-30ms. In this paper, an enhancement method is
proposed for significant reduction of noise, and improvement in the quality and perceptibility of speech degraded by correlated
additive background noise. The proposed method is based on the spectral subtraction technique. The simple spectral subtraction
technique results in poor reduction of noise. One of the main reasons for this is neglecting the cross spectral terms of speech and
noise, based on the appropriation that clean speech and noise signals are completely uncorrelated to each other, which is not true
on short time basis. In this paper an improvement in reduction of the noise is achieved as compared to the earlier methods. This
fact is mainly attributed to the cross spectral terms between speech and noise. This algorithm can be implemented and used in
hearing aids for the benefit of hearing impaired people. Objective speech quality measures, spectrogram analyses and subjective
listening tests conforms the proposed method is more effective in comparison with earlier speech enhancement techniques.
Keywords: Spectral Subtaction,Cross Spectral Components
Attention gated encoder-decoder for ultrasonic signal denoisingIAESIJAI
Ultrasound imaging is one of the most widely used non-destructive testing methods. The transducer emits pulses that travel through the imaged samples and are reflected by echo-forming impedance. The resulting ultrasonic signals usually contain noise. Most of the traditional noise reduction algorithms require high skills and prior knowledge of noise distribution, which has a crucial impact on their performances. As a result, these methods generally yield a loss of information, significantly influencing the final data and deeply limiting both sensitivity and resolution of imaging devices in medical and industrial applications. In the present study, a denoising method based on an attention-gated convolutional autoencoder is proposed to fill this gap. To evaluate its performance, the suggested protocol is compared to widely used methods such as butterworth filtering (BF), discrete wavelet transforms (DWT), principal component analysis (PCA), and convolutional autoencoder (CAE) methods. Results proved that better denoising can be achieved especially when the original signal-to-noise ratio (SNR) is very low and the sound waves’ traces are distorted by noise. Moreover, the initial SNR was improved by up to 30 dB and the resulting Pearson correlation coefficient was maintained over 99% even for ultrasonic signals with poor initial SNR.
A Novel Uncertainty Parameter SR ( Signal to Residual Spectrum Ratio ) Evalua...sipij
Usually, hearing impaired people use hearing aids which are implemented with speech enhancement
algorithms. Estimation of speech and estimation of nose are the components in single channel speech
enhancement system. The main objective of any speech enhancement algorithm is estimation of noise power
spectrum for non stationary environment. VAD (Voice Activity Detector) is used to identify speech pauses
and during these pauses only estimation of noise. MMSE (Minimum Mean Square Error) speech
enhancement algorithm did not enhance the intelligibility, quality and listener fatigues are the perceptual
aspects of speech. Novel evaluation approach SR (Signal to Residual spectrum ratio) based on uncertainty
parameter introduced for the benefits of hearing impaired people in non stationary environments to control
distortions. By estimation and updating of noise based on division of original pure signal into three parts
such as pure speech, quasi speech and non speech frames based on multiple threshold conditions. Different
values of SR and LLR demonstrate the amount of attenuation and amplification distortions. The proposed
method will compared with any one method WAT(Weighted Average Technique) Hence by using
parameters SR (signal to residual spectrum ratio) and LLR (log like hood ratio), MMSE (Minim Mean
Square Error) in terms of segmented SNR and LLR.
Наше приложение (iOs, Android) и драйвер(Windows) представлены в независимом исследовании устройств для обучения студентов с ослабленным слухом.
The article by Saketh Sharma, Nitya Tiwari, and Prem C. Pandey in the "Int. Conf. on Intelligent Human Computer Interaction" proceedings (www.petralex.pro)
Implementation of a Digital Hearing Aid with User-Settable Frequency Response...IAMCP MENTORING
The article by Saketh Sharma, Nitya Tiwari, and Prem C. Pandey in the "Int. Conf. on Intelligent Human Computer Interaction" proceedings describes among others the Petralex smartphone App as a hearing aid - the product of the IAMCP members - IT4You (www.petralex.pro)
Similar to IJERD (www.ijerd.com) International Journal of Engineering Research and Development (20)
A Novel Method for Prevention of Bandwidth Distributed Denial of Service AttacksIJERD Editor
Distributed Denial of Service (DDoS) Attacks became a massive threat to the Internet. Traditional
Architecture of internet is vulnerable to the attacks like DDoS. Attacker primarily acquire his army of Zombies,
then that army will be instructed by the Attacker that when to start an attack and on whom the attack should be
done. In this paper, different techniques which are used to perform DDoS Attacks, Tools that were used to
perform Attacks and Countermeasures in order to detect the attackers and eliminate the Bandwidth Distributed
Denial of Service attacks (B-DDoS) are reviewed. DDoS Attacks were done by using various Flooding
techniques which are used in DDoS attack.
The main purpose of this paper is to design an architecture which can reduce the Bandwidth
Distributed Denial of service Attack and make the victim site or server available for the normal users by
eliminating the zombie machines. Our Primary focus of this paper is to dispute how normal machines are
turning into zombies (Bots), how attack is been initiated, DDoS attack procedure and how an organization can
save their server from being a DDoS victim. In order to present this we implemented a simulated environment
with Cisco switches, Routers, Firewall, some virtual machines and some Attack tools to display a real DDoS
attack. By using Time scheduling, Resource Limiting, System log, Access Control List and some Modular
policy Framework we stopped the attack and identified the Attacker (Bot) machines
Hearing loss is one of the most common human impairments. It is estimated that by year 2015 more
than 700 million people will suffer mild deafness. Most can be helped by hearing aid devices depending on the
severity of their hearing loss. This paper describes the implementation and characterization details of a dual
channel transmitter front end (TFE) for digital hearing aid (DHA) applications that use novel micro
electromechanical- systems (MEMS) audio transducers and ultra-low power-scalable analog-to-digital
converters (ADCs), which enable a very-low form factor, energy-efficient implementation for next-generation
DHA. The contribution of the design is the implementation of the dual channel MEMS microphones and powerscalable
ADC system.
Influence of tensile behaviour of slab on the structural Behaviour of shear c...IJERD Editor
-A composite beam is composed of a steel beam and a slab connected by means of shear connectors
like studs installed on the top flange of the steel beam to form a structure behaving monolithically. This study
analyzes the effects of the tensile behavior of the slab on the structural behavior of the shear connection like slip
stiffness and maximum shear force in composite beams subjected to hogging moment. The results show that the
shear studs located in the crack-concentration zones due to large hogging moments sustain significantly smaller
shear force and slip stiffness than the other zones. Moreover, the reduction of the slip stiffness in the shear
connection appears also to be closely related to the change in the tensile strain of rebar according to the increase
of the load. Further experimental and analytical studies shall be conducted considering variables such as the
reinforcement ratio and the arrangement of shear connectors to achieve efficient design of the shear connection
in composite beams subjected to hogging moment.
Gold prospecting using Remote Sensing ‘A case study of Sudan’IJERD Editor
Gold has been extracted from northeast Africa for more than 5000 years, and this may be the first
place where the metal was extracted. The Arabian-Nubian Shield (ANS) is an exposure of Precambrian
crystalline rocks on the flanks of the Red Sea. The crystalline rocks are mostly Neoproterozoic in age. ANS
includes the nations of Israel, Jordan. Egypt, Saudi Arabia, Sudan, Eritrea, Ethiopia, Yemen, and Somalia.
Arabian Nubian Shield Consists of juvenile continental crest that formed between 900 550 Ma, when intra
oceanic arc welded together along ophiolite decorated arc. Primary Au mineralization probably developed in
association with the growth of intra oceanic arc and evolution of back arc. Multiple episodes of deformation
have obscured the primary metallogenic setting, but at least some of the deposits preserve evidence that they
originate as sea floor massive sulphide deposits.
The Red Sea Hills Region is a vast span of rugged, harsh and inhospitable sector of the Earth with
inimical moon-like terrain, nevertheless since ancient times it is famed to be an abode of gold and was a major
source of wealth for the Pharaohs of ancient Egypt. The Pharaohs old workings have been periodically
rediscovered through time. Recent endeavours by the Geological Research Authority of Sudan led to the
discovery of a score of occurrences with gold and massive sulphide mineralizations. In the nineties of the
previous century the Geological Research Authority of Sudan (GRAS) in cooperation with BRGM utilized
satellite data of Landsat TM using spectral ratio technique to map possible mineralized zones in the Red Sea
Hills of Sudan. The outcome of the study mapped a gossan type gold mineralization. Band ratio technique was
applied to Arbaat area and a signature of alteration zone was detected. The alteration zones are commonly
associated with mineralization. The alteration zones are commonly associated with mineralization. A filed check
confirmed the existence of stock work of gold bearing quartz in the alteration zone. Another type of gold
mineralization that was discovered using remote sensing is the gold associated with metachert in the Atmur
Desert.
Reducing Corrosion Rate by Welding DesignIJERD Editor
The paper addresses the importance of welding design to prevent corrosion at steel. Welding is
used to join pipe, profiles at bridges, spindle, and a lot more part of engineering construction. The
problems happened associated with welding are common issues in these fields, especially corrosion.
Corrosion can be reduced with many methods, they are painting, controlling humidity, and also good
welding design. In the research, it can be found that reducing residual stress on the welding can be
solved in corrosion rate reduction problem.
Preheating on 500oC and 600oC give better condition to reduce corosion rate than condition after
preheating 400oC. For all welding groove type, material with 500oC and 600oC preheating after 14 days
corrosion test is 0,5%-0,69% lost. Material with 400oC preheating after 14 days corrosion test is 0,57%-0,76%
lost.
Welding groove also influence corrosion rate. X and V type welding groove give better condition to reduce
corrosion rate than use 1/2V and 1/2 X welding groove. After 14 days corrosion test, the samples with
X welding groove type is 0,5%-0,57% lost. The samples with V welding groove after 14 days corrosion test is
0,51%-0,59% lost. The samples with 1/2V and 1/2X welding groove after 14 days corrosion test is 0,58%-
0,71% lost.
Router 1X3 – RTL Design and VerificationIJERD Editor
Routing is the process of moving a packet of data from source to destination and enables messages
to pass from one computer to another and eventually reach the target machine. A router is a networking device
that forwards data packets between computer networks. It is connected to two or more data lines from different
networks (as opposed to a network switch, which connects data lines from one single network). This paper,
mainly emphasizes upon the study of router device, it‟s top level architecture, and how various sub-modules of
router i.e. Register, FIFO, FSM and Synchronizer are synthesized, and simulated and finally connected to its top
module.
Active Power Exchange in Distributed Power-Flow Controller (DPFC) At Third Ha...IJERD Editor
This paper presents a component within the flexible ac-transmission system (FACTS) family, called
distributed power-flow controller (DPFC). The DPFC is derived from the unified power-flow controller (UPFC)
with an eliminated common dc link. The DPFC has the same control capabilities as the UPFC, which comprise
the adjustment of the line impedance, the transmission angle, and the bus voltage. The active power exchange
between the shunt and series converters, which is through the common dc link in the UPFC, is now through the
transmission lines at the third-harmonic frequency. DPFC multiple small-size single-phase converters which
reduces the cost of equipment, no voltage isolation between phases, increases redundancy and there by
reliability increases. The principle and analysis of the DPFC are presented in this paper and the corresponding
simulation results that are carried out on a scaled prototype are also shown.
Mitigation of Voltage Sag/Swell with Fuzzy Control Reduced Rating DVRIJERD Editor
Power quality has been an issue that is becoming increasingly pivotal in industrial electricity
consumers point of view in recent times. Modern industries employ Sensitive power electronic equipments,
control devices and non-linear loads as part of automated processes to increase energy efficiency and
productivity. Voltage disturbances are the most common power quality problem due to this the use of a large
numbers of sophisticated and sensitive electronic equipment in industrial systems is increased. This paper
discusses the design and simulation of dynamic voltage restorer for improvement of power quality and
reduce the harmonics distortion of sensitive loads. Power quality problem is occurring at non-standard
voltage, current and frequency. Electronic devices are very sensitive loads. In power system voltage sag,
swell, flicker and harmonics are some of the problem to the sensitive load. The compensation capability
of a DVR depends primarily on the maximum voltage injection ability and the amount of stored
energy available within the restorer. This device is connected in series with the distribution feeder at
medium voltage. A fuzzy logic control is used to produce the gate pulses for control circuit of DVR and the
circuit is simulated by using MATLAB/SIMULINK software.
Study on the Fused Deposition Modelling In Additive ManufacturingIJERD Editor
Additive manufacturing process, also popularly known as 3-D printing, is a process where a product
is created in a succession of layers. It is based on a novel materials incremental manufacturing philosophy.
Unlike conventional manufacturing processes where material is removed from a given work price to derive the
final shape of a product, 3-D printing develops the product from scratch thus obviating the necessity to cut away
materials. This prevents wastage of raw materials. Commonly used raw materials for the process are ABS
plastic, PLA and nylon. Recently the use of gold, bronze and wood has also been implemented. The complexity
factor of this process is 0% as in any object of any shape and size can be manufactured.
Spyware triggering system by particular string valueIJERD Editor
This computer programme can be used for good and bad purpose in hacking or in any general
purpose. We can say it is next step for hacking techniques such as keylogger and spyware. Once in this system if
user or hacker store particular string as a input after that software continually compare typing activity of user
with that stored string and if it is match then launch spyware programme.
A Blind Steganalysis on JPEG Gray Level Image Based on Statistical Features a...IJERD Editor
This paper presents a blind steganalysis technique to effectively attack the JPEG steganographic
schemes i.e. Jsteg, F5, Outguess and DWT Based. The proposed method exploits the correlations between
block-DCTcoefficients from intra-block and inter-block relation and the statistical moments of characteristic
functions of the test image is selected as features. The features are extracted from the BDCT JPEG 2-array.
Support Vector Machine with cross-validation is implemented for the classification.The proposed scheme gives
improved outcome in attacking.
Secure Image Transmission for Cloud Storage System Using Hybrid SchemeIJERD Editor
- Data over the cloud is transferred or transmitted between servers and users. Privacy of that
data is very important as it belongs to personal information. If data get hacked by the hacker, can be
used to defame a person’s social data. Sometimes delay are held during data transmission. i.e. Mobile
communication, bandwidth is low. Hence compression algorithms are proposed for fast and efficient
transmission, encryption is used for security purposes and blurring is used by providing additional
layers of security. These algorithms are hybridized for having a robust and efficient security and
transmission over cloud storage system.
Application of Buckley-Leverett Equation in Modeling the Radius of Invasion i...IJERD Editor
A thorough review of existing literature indicates that the Buckley-Leverett equation only analyzes
waterflood practices directly without any adjustments on real reservoir scenarios. By doing so, quite a number
of errors are introduced into these analyses. Also, for most waterflood scenarios, a radial investigation is more
appropriate than a simplified linear system. This study investigates the adoption of the Buckley-Leverett
equation to estimate the radius invasion of the displacing fluid during waterflooding. The model is also adopted
for a Microbial flood and a comparative analysis is conducted for both waterflooding and microbial flooding.
Results shown from the analysis doesn’t only records a success in determining the radial distance of the leading
edge of water during the flooding process, but also gives a clearer understanding of the applicability of
microbes to enhance oil production through in-situ production of bio-products like bio surfactans, biogenic
gases, bio acids etc.
Gesture Gaming on the World Wide Web Using an Ordinary Web CameraIJERD Editor
- Gesture gaming is a method by which users having a laptop/pc/x-box play games using natural or
bodily gestures. This paper presents a way of playing free flash games on the internet using an ordinary webcam
with the help of open source technologies. Emphasis in human activity recognition is given on the pose
estimation and the consistency in the pose of the player. These are estimated with the help of an ordinary web
camera having different resolutions from VGA to 20mps. Our work involved giving a 10 second documentary to
the user on how to play a particular game using gestures and what are the various kinds of gestures that can be
performed in front of the system. The initial inputs of the RGB values for the gesture component is obtained by
instructing the user to place his component in a red box in about 10 seconds after the short documentary before
the game is finished. Later the system opens the concerned game on the internet on popular flash game sites like
miniclip, games arcade, GameStop etc and loads the game clicking at various places and brings the state to a
place where the user is to perform only gestures to start playing the game. At any point of time the user can call
off the game by hitting the esc key and the program will release all of the controls and return to the desktop. It
was noted that the results obtained using an ordinary webcam matched that of the Kinect and the users could
relive the gaming experience of the free flash games on the net. Therefore effective in game advertising could
also be achieved thus resulting in a disruptive growth to the advertising firms.
Hardware Analysis of Resonant Frequency Converter Using Isolated Circuits And...IJERD Editor
-LLC resonant frequency converter is basically a combo of series as well as parallel resonant ckt. For
LCC resonant converter it is associated with a disadvantage that, though it has two resonant frequencies, the
lower resonant frequency is in ZCS region[5]. For this application, we are not able to design the converter
working at this resonant frequency. LLC resonant converter existed for a very long time but because of
unknown characteristic of this converter it was used as a series resonant converter with basically a passive
(resistive) load. . Here, it was designed to operate in switching frequency higher than resonant frequency of the
series resonant tank of Lr and Cr converter acts very similar to Series Resonant Converter. The benefit of LLC
resonant converter is narrow switching frequency range with light load[6] . Basically, the control ckt plays a
very imp. role and hence 555 Timer used here provides a perfect square wave as the control ckt provides no
slew rate which makes the square wave really strong and impenetrable. The dead band circuit provides the
exclusive dead band in micro seconds so as to avoid the simultaneous firing of two pairs of IGBT’s where one
pair switches off and the other on for a slightest period of time. Hence, the isolator ckt here is associated with
each and every ckt used because it acts as a driver and an isolation to each of the IGBT is provided with one
exclusive transformer supply[3]. The IGBT’s are fired using the appropriate signal using the previous boards
and hence at last a high frequency rectifier ckt with a filtering capacitor is used to get an exact dc
waveform .The basic goal of this particular analysis is to observe the wave forms and characteristics of
converters with differently positioned passive elements in the form of tank circuits.
Simulated Analysis of Resonant Frequency Converter Using Different Tank Circu...IJERD Editor
LLC resonant frequency converter is basically a combo of series as well as parallel resonant ckt. For
LCC resonant converter it is associated with a disadvantage that, though it has two resonant frequencies, the
lower resonant frequency is in ZCS region [5]. For this application, we are not able to design the converter
working at this resonant frequency. LLC resonant converter existed for a very long time but because of
unknown characteristic of this converter it was used as a series resonant converter with basically a passive
(resistive) load. . Here, it was designed to operate in switching frequency higher than resonant frequency of the
series resonant tank of Lr and Cr converter acts very similar to Series Resonant Converter. The benefit of LLC
resonant converter is narrow switching frequency range with light load[6] . Basically, the control ckt plays a
very imp. role and hence 555 Timer used here provides a perfect square wave as the control ckt provides no
slew rate which makes the square wave really strong and impenetrable. The dead band circuit provides the
exclusive dead band in micro seconds so as to avoid the simultaneous firing of two pairs of IGBT’s where one
pair switches off and the other on for a slightest period of time. Hence, the isolator ckt here is associated with
each and every ckt used because it acts as a driver and an isolation to each of the IGBT is provided with one
exclusive transformer supply[3]. The IGBT’s are fired using the appropriate signal using the previous boards
and hence at last a high frequency rectifier ckt with a filtering capacitor is used to get an exact dc
waveform .The basic goal of this particular analysis is to observe the wave forms and characteristics of
converters with differently positioned passive elements in the form of tank circuits. The supported simulation
is done through PSIM 6.0 software tool
Amateurs Radio operator, also known as HAM communicates with other HAMs through Radio
waves. Wireless communication in which Moon is used as natural satellite is called Moon-bounce or EME
(Earth -Moon-Earth) technique. Long distance communication (DXing) using Very High Frequency (VHF)
operated amateur HAM radio was difficult. Even with the modest setup having good transceiver, power
amplifier and high gain antenna with high directivity, VHF DXing is possible. Generally 2X11 YAGI antenna
along with rotor to set horizontal and vertical angle is used. Moon tracking software gives exact location,
visibility of Moon at both the stations and other vital data to acquire real time position of moon.
“MS-Extractor: An Innovative Approach to Extract Microsatellites on „Y‟ Chrom...IJERD Editor
Simple Sequence Repeats (SSR), also known as Microsatellites, have been extensively used as
molecular markers due to their abundance and high degree of polymorphism. The nucleotide sequences of
polymorphic forms of the same gene should be 99.9% identical. So, Microsatellites extraction from the Gene is
crucial. However, Microsatellites repeat count is compared, if they differ largely, he has some disorder. The Y
chromosome likely contains 50 to 60 genes that provide instructions for making proteins. Because only males
have the Y chromosome, the genes on this chromosome tend to be involved in male sex determination and
development. Several Microsatellite Extractors exist and they fail to extract microsatellites on large data sets of
giga bytes and tera bytes in size. The proposed tool “MS-Extractor: An Innovative Approach to extract
Microsatellites on „Y‟ Chromosome” can extract both Perfect as well as Imperfect Microsatellites from large
data sets of human genome „Y‟. The proposed system uses string matching with sliding window approach to
locate Microsatellites and extracts them.
Importance of Measurements in Smart GridIJERD Editor
- The need to get reliable supply, independence from fossil fuels, and capability to provide clean
energy at a fixed and lower cost, the existing power grid structure is transforming into Smart Grid. The
development of a smart energy distribution grid is a current goal of many nations. A Smart Grid should have
new capabilities such as self-healing, high reliability, energy management, and real-time pricing. This new era
of smart future grid will lead to major changes in existing technologies at generation, transmission and
distribution levels. The incorporation of renewable energy resources and distribution generators in the existing
grid will increase the complexity, optimization problems and instability of the system. This will lead to a
paradigm shift in the instrumentation and control requirements for Smart Grids for high quality, stable and
reliable electricity supply of power. The monitoring of the grid system state and stability relies on the
availability of reliable measurement of data. In this paper the measurement areas that highlight new
measurement challenges, development of the Smart Meters and the critical parameters of electric energy to be
monitored for improving the reliability of power systems has been discussed.
Study of Macro level Properties of SCC using GGBS and Lime stone powderIJERD Editor
One of the major environmental concerns is the disposal of the waste materials and utilization of
industrial by products. Lime stone quarries will produce millions of tons waste dust powder every year. Having
considerable high degree of fineness in comparision to cement this material may be utilized as a partial
replacement to cement. For this purpose an experiment is conducted to investigate the possibility of using lime
stone powder in the production of SCC with combined use GGBS and how it affects the fresh and mechanical
properties of SCC. First SCC is made by replacing cement with GGBS in percentages like 10, 20, 30, 40, 50 and
by taking the optimum mix with GGBS lime stone powder is blended to mix in percentages like 5, 10, 15, 20 as
a partial replacement to cement. Test results shows that the SCC mix with combination of 30% GGBS and 15%
limestone powder gives maximum compressive strength and fresh properties are also in the limits prescribed by
the EFNARC.
UiPath Test Automation using UiPath Test Suite series, part 4DianaGray10
Welcome to UiPath Test Automation using UiPath Test Suite series part 4. In this session, we will cover Test Manager overview along with SAP heatmap.
The UiPath Test Manager overview with SAP heatmap webinar offers a concise yet comprehensive exploration of the role of a Test Manager within SAP environments, coupled with the utilization of heatmaps for effective testing strategies.
Participants will gain insights into the responsibilities, challenges, and best practices associated with test management in SAP projects. Additionally, the webinar delves into the significance of heatmaps as a visual aid for identifying testing priorities, areas of risk, and resource allocation within SAP landscapes. Through this session, attendees can expect to enhance their understanding of test management principles while learning practical approaches to optimize testing processes in SAP environments using heatmap visualization techniques
What will you get from this session?
1. Insights into SAP testing best practices
2. Heatmap utilization for testing
3. Optimization of testing processes
4. Demo
Topics covered:
Execution from the test manager
Orchestrator execution result
Defect reporting
SAP heatmap example with demo
Speaker:
Deepak Rai, Automation Practice Lead, Boundaryless Group and UiPath MVP
Key Trends Shaping the Future of Infrastructure.pdfCheryl Hung
Keynote at DIGIT West Expo, Glasgow on 29 May 2024.
Cheryl Hung, ochery.com
Sr Director, Infrastructure Ecosystem, Arm.
The key trends across hardware, cloud and open-source; exploring how these areas are likely to mature and develop over the short and long-term, and then considering how organisations can position themselves to adapt and thrive.
State of ICS and IoT Cyber Threat Landscape Report 2024 previewPrayukth K V
The IoT and OT threat landscape report has been prepared by the Threat Research Team at Sectrio using data from Sectrio, cyber threat intelligence farming facilities spread across over 85 cities around the world. In addition, Sectrio also runs AI-based advanced threat and payload engagement facilities that serve as sinks to attract and engage sophisticated threat actors, and newer malware including new variants and latent threats that are at an earlier stage of development.
The latest edition of the OT/ICS and IoT security Threat Landscape Report 2024 also covers:
State of global ICS asset and network exposure
Sectoral targets and attacks as well as the cost of ransom
Global APT activity, AI usage, actor and tactic profiles, and implications
Rise in volumes of AI-powered cyberattacks
Major cyber events in 2024
Malware and malicious payload trends
Cyberattack types and targets
Vulnerability exploit attempts on CVEs
Attacks on counties – USA
Expansion of bot farms – how, where, and why
In-depth analysis of the cyber threat landscape across North America, South America, Europe, APAC, and the Middle East
Why are attacks on smart factories rising?
Cyber risk predictions
Axis of attacks – Europe
Systemic attacks in the Middle East
Download the full report from here:
https://sectrio.com/resources/ot-threat-landscape-reports/sectrio-releases-ot-ics-and-iot-security-threat-landscape-report-2024/
The Art of the Pitch: WordPress Relationships and SalesLaura Byrne
Clients don’t know what they don’t know. What web solutions are right for them? How does WordPress come into the picture? How do you make sure you understand scope and timeline? What do you do if sometime changes?
All these questions and more will be explored as we talk about matching clients’ needs with what your agency offers without pulling teeth or pulling your hair out. Practical tips, and strategies for successful relationship building that leads to closing the deal.
Dev Dives: Train smarter, not harder – active learning and UiPath LLMs for do...UiPathCommunity
💥 Speed, accuracy, and scaling – discover the superpowers of GenAI in action with UiPath Document Understanding and Communications Mining™:
See how to accelerate model training and optimize model performance with active learning
Learn about the latest enhancements to out-of-the-box document processing – with little to no training required
Get an exclusive demo of the new family of UiPath LLMs – GenAI models specialized for processing different types of documents and messages
This is a hands-on session specifically designed for automation developers and AI enthusiasts seeking to enhance their knowledge in leveraging the latest intelligent document processing capabilities offered by UiPath.
Speakers:
👨🏫 Andras Palfi, Senior Product Manager, UiPath
👩🏫 Lenka Dulovicova, Product Program Manager, UiPath
Search and Society: Reimagining Information Access for Radical FuturesBhaskar Mitra
The field of Information retrieval (IR) is currently undergoing a transformative shift, at least partly due to the emerging applications of generative AI to information access. In this talk, we will deliberate on the sociotechnical implications of generative AI for information access. We will argue that there is both a critical necessity and an exciting opportunity for the IR community to re-center our research agendas on societal needs while dismantling the artificial separation between the work on fairness, accountability, transparency, and ethics in IR and the rest of IR research. Instead of adopting a reactionary strategy of trying to mitigate potential social harms from emerging technologies, the community should aim to proactively set the research agenda for the kinds of systems we should build inspired by diverse explicitly stated sociotechnical imaginaries. The sociotechnical imaginaries that underpin the design and development of information access technologies needs to be explicitly articulated, and we need to develop theories of change in context of these diverse perspectives. Our guiding future imaginaries must be informed by other academic fields, such as democratic theory and critical theory, and should be co-developed with social science scholars, legal scholars, civil rights and social justice activists, and artists, among others.
Connector Corner: Automate dynamic content and events by pushing a buttonDianaGray10
Here is something new! In our next Connector Corner webinar, we will demonstrate how you can use a single workflow to:
Create a campaign using Mailchimp with merge tags/fields
Send an interactive Slack channel message (using buttons)
Have the message received by managers and peers along with a test email for review
But there’s more:
In a second workflow supporting the same use case, you’ll see:
Your campaign sent to target colleagues for approval
If the “Approve” button is clicked, a Jira/Zendesk ticket is created for the marketing design team
But—if the “Reject” button is pushed, colleagues will be alerted via Slack message
Join us to learn more about this new, human-in-the-loop capability, brought to you by Integration Service connectors.
And...
Speakers:
Akshay Agnihotri, Product Manager
Charlie Greenberg, Host
Neuro-symbolic is not enough, we need neuro-*semantic*Frank van Harmelen
Neuro-symbolic (NeSy) AI is on the rise. However, simply machine learning on just any symbolic structure is not sufficient to really harvest the gains of NeSy. These will only be gained when the symbolic structures have an actual semantics. I give an operational definition of semantics as “predictable inference”.
All of this illustrated with link prediction over knowledge graphs, but the argument is general.
UiPath Test Automation using UiPath Test Suite series, part 3DianaGray10
Welcome to UiPath Test Automation using UiPath Test Suite series part 3. In this session, we will cover desktop automation along with UI automation.
Topics covered:
UI automation Introduction,
UI automation Sample
Desktop automation flow
Pradeep Chinnala, Senior Consultant Automation Developer @WonderBotz and UiPath MVP
Deepak Rai, Automation Practice Lead, Boundaryless Group and UiPath MVP
Transcript: Selling digital books in 2024: Insights from industry leaders - T...BookNet Canada
The publishing industry has been selling digital audiobooks and ebooks for over a decade and has found its groove. What’s changed? What has stayed the same? Where do we go from here? Join a group of leading sales peers from across the industry for a conversation about the lessons learned since the popularization of digital books, best practices, digital book supply chain management, and more.
Link to video recording: https://bnctechforum.ca/sessions/selling-digital-books-in-2024-insights-from-industry-leaders/
Presented by BookNet Canada on May 28, 2024, with support from the Department of Canadian Heritage.
"Impact of front-end architecture on development cost", Viktor TurskyiFwdays
I have heard many times that architecture is not important for the front-end. Also, many times I have seen how developers implement features on the front-end just following the standard rules for a framework and think that this is enough to successfully launch the project, and then the project fails. How to prevent this and what approach to choose? I have launched dozens of complex projects and during the talk we will analyze which approaches have worked for me and which have not.
"Impact of front-end architecture on development cost", Viktor Turskyi
IJERD (www.ijerd.com) International Journal of Engineering Research and Development
1. International Journal of Engineering Research and Development
eISSN : 2278-067X, pISSN : 2278-800X, www.ijerd.com
Volume 2, Issue 2 (July 2012), PP. 57-64
Different Techniques for the Enhancement of the Intelligibility of
a Speech Signal
Pankaj Bactor1, Anil Garg 2
1
Department of ECE, M.M.E.C-M.M. University, Mullana-Ambala-Haryana-India
2
Department of ECE, M.M.E.C-M.M. University, Mullana-Ambala-Haryana-India
Abstract––Speech enhancement is a popular method for making ASR systems more robust. Spectral subtraction is
performed by subtracting the average magnitude of the noise spectrum from the spectrum of the noisy speech to estimate
the magnitude of the enhanced speech spectrum. In this paper, adaptive techniques have been explored for the
enhancement of the speech such as ANC, ANFIS, DFNN, MDFNN and EDFNN. By using these techniques, it has been
observed that the root mean square error (RMSE) and number of epochs are less and at the same time the membership
functions are also less.
Keywords—ANC, ANFIS, DFNN, EDFNN, RMSE, SNR
I. INTRODUCTION
In this paper, a comparative performance analysis of single channel, dual-channel and multi-channel (using
microphone arrays) speech enhancement techniques, with different types of noise at different SNR‘S have been studied.
Single channel spectral subtraction was originally designed to improve human speech intelligibility and many attempts have
been made to maximized signal-to-noise Ratio (SNR) or minimized speech distortion. Speech enhancement refers to the
improvement in the quality or intelligibility of a speech signal and the reversal of degradations that have corrupted it. Quality
is a subjective measure which reflects on the pleasantness of the speech or on the amount of effort needed to understand the
speech material. Intelligibility is an objective measure which signifies the amount of speech material correctly understood.
The main objective of Speech Enhancement is to enhance the speech signal to obtain a clean signal with higher quality. Such
system has been widely used in long distance telephony applications.
Artificial intelligence is the science and engineering of making intelligent machines, especially intelligent
computer programs. It is related to the similar task of using computers to understand human intelligence. Fuzzy logic deals
with reasoning that is approximate rather than fixed and exact. Fuzzy logic is a form of many-valued logic; it deals with
reasoning that is approximate rather than fixed and exact. In contrast with traditional logic theory, where binary sets have
two-valued logic: true or false, fuzzy logic variables may have a truth value that ranges in degree between 0 and 1. Artificial
Neural Network includes training, learning and generalization. Training is the process by which these connection weights
(W) are assigned. These weights determine the output of the neural network. The weights are adjusted based on how valid
the neural network performed. This process is repeated until the validation error is within an acceptable limit. Training a
network involves presenting input patterns in a way so that the system minimizes its error and improves its performance. The
weights are adjusted based on how valid the neural network performed. This process is repeated until the validation error is
within an acceptable limit. Learning is necessary when the information about inputs/ outputs is unknown or incompl ete a
priori. Learning process is based on comparison, between networks computed output and the correct expected output,
generating ‗error‘. The ‗error‘ generated is used to change network parameter that result improved performance. The NN
possesses the capability to generalize. They can predict new outcomes from past trends. The NN is said to generalize well
when it sensibly interpolates input patterns that are new to network. When input-output mapping computed by network is
correct.
Some of them are as follows: SCSET (Single-Channel Speech Enhancement Techniques) – Spectral Subtraction
Method, Spectral Subtraction with over subtraction model and Non-linear spectral subtraction, MCSET (Multi-Channel
Speech Enhancement Techniques) – Adaptive Noise cancellation and Multisensory Beam forming, SSUAA (Spectral
Subtraction Using Adaptive Averaging), NREWF (Noise Reduction using Enhanced Wiener Filtering), CSSFG (Cepstral
Smoothing of Spectral Filter Gains), SSSTMD (Spectral Subtraction in the Short Time Modulation Domain), Adaptive
Noise Cancellation In The Wavelet Domain (ANCWD), Speech Enhancement Using Adaptive Neuro-Fuzzy Filtering
(SEANFF) and Adaptive Noise Cancellation Using Cascaded Correlation Neural Networks (ANCCCNN).
II. SINGLE CHANNEL SPEECH ENHANCEMENT TECHNIQUES
Single-channel speech enhancement techniques apply to situations in which .a unique acquisition channel is
available. This may be imposed by the system used (as telephone-based applications) or by the availability of the desired
signal (as prerecorded applications). When the noise process is stationary and speech activity can be detected, spectral
subtraction (SS) is a direct way to enhance the noisy speech.
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2. Different Techniques for the Enhancement of the Intelligibility of a Speech Signal
A. Spectral Subtraction Method: (SS)
The spectral subtraction method can lead to negative values, resulting from differences among the noise estimator
and the actual noise frame. To cope with this problem, negative values must be set to zero, Producing Spectral spikes, well-
known as ―musical noise‖. This effect causes an annoying perception of enhanced speech and, therefore, it must be
corrected.
B. Spectral Subtraction with Oversubtraction Model: (SSOM)
SSOM procedure was introduced in order to compensate for the ―musical noise‖ effect. It reduces the perception of
musical noise.
C. Non-Linear Spectral Subtraction: (NSS)
NSS approach is based on combining two different ideas: i) The use of an extended noise and an over subtraction
model ii) Non-linear implementation of the subtraction process, taking into account that the subtraction process must depend
on the SNR of the frame, in order to apply less subtraction with high SNRs and vice versa.
III. MULTI CHANNEL SPEECH ENHANCEMENT TECHNIQUES
Multi-channel speech enhancement techniques take advantage of the availability of multiple signal input to our
system, making possible the use of noise references in an adaptive noise cancellation device, the use of phase alignment to
reject undesired noise components, or even the use of phase alignment and noise cancellation stages into a combined
scheme.
A. Adaptive Noise Cancellation: (ANC)
Adaptive noise cancellation is a powerful speech enhancement technique based in the availability of an auxiliary
channel, known as reference path, where a correlated sample or reference of the contaminating noise is present. This
reference input will be filtered following an adaptive algorithm, in order to subtract the output of this filtering process from
the main path, where noisy speech is present.
B. Multisensor Beamforming
Multisensor beamforming through microphone arrays, derived from radar and sonar applications, can be
implemented in a variety of ways, being delay-and-sum beamforming the most direct approach. The underlaying idea of this
scheme is based on the assumption that the contribution of the reflex ion is small, and that we know the direction of arrival
of the desired signal. Then, through a correct alignment of the phase function in each sensor, the desired signal can be
enhanced, rejecting all the noisy components not aligned in phase.
IV. SPECTRAL SUBTRACTION USING ADAPTIVE AVERAGING (SSUAA)
This method provides a noise reduction procedure which functions well with arbitrary frame lengths, gives low
residual noise, high-quality speech, and low background noise artifacts, and introduces only a short delay. These are
important properties when the noise reduction methods are integrated together with other speech enhancement methods and
speech coders in real-time communication systems. The method reduces the variability of the gain function—in this case, a
complex function—in two ways. First, the variance of the current block‘s spectrum estimate is reduced using the Bartlett
method by trading frequency resolution for variance reduction. Second, an adaptive averaging of the gain function is used
which is dependent on the discrepancy between the estimated noise spectrum and the current input signal spectrum estimate.
V. NOISE REDUCTION USING ENHANCED WIENER FILTERING (NREWF)
The problem of noise reduction has attracted a considerable amount of research attention over the past several
decades. Among the numerous techniques that were developed, the optimal Wiener filter can be considered as one of the
most fundamental noise reduction approaches, which has been delineated in different forms and adopted in various
applications. Although it is not a secret that the Wiener filter may cause some detrimental effects to the speech signal
(appreciable or even significant degradation in quality or intelligibility), few efforts have been reported to show the inher ent
relationship between noise reduction and speech distortion. When no a priori knowledge is available, we can still achieve a
better control of noise reduction and speech distortion by properly manipulating the Wiener filter, resulting in a suboptimal
Wiener filter. In case that we have multiple microphone sensors, the multiple observations of the speech signal can be used
to reduce noise with less or even no speech distortion.
VI. CEPSTRAL SMOOTHING OF SPECTRAL FILTER GAINS (CSSFG)
Cepstral smoothing is a useful amendment to speech enhancement filters operating in real noise environments.
Annoying noise fluctuations are prevented even in the case of babble noise. As opposed to conventional methods, Cepstral
smoothing allows for a selective smoothing of different spectral structures represented by the respective Cepstral
coefficients. This makes the protection of the characteristics of speech possible while musical noise is suppressed. Cepstral
smoothing preserves speech onsets, plosives, and quasi-stationary narrowband structures like voiced speech. The proposed
recursive temporal smoothing is applied to higher Cepstral coefficients only, excluding those representing the pitch
information. As the higher Cepstral coefficients describe the finer spectral structure of the Fourier spectrum, smoothing them
along time prevents single coefficients of the filter function from changing excessively and independently of their
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3. Different Techniques for the Enhancement of the Intelligibility of a Speech Signal
neighboring bins, thus suppressing musical noise. The proposed Cepstral smoothing technique is very effective in
nonstationary noise.
VII. SPECTRAL SUBTRACTION IN THE SHORT TIME MODULATION DOMAIN
(SSSTMD)
Speech enhancement aims at improving the quality of noisy speech. This is normally accomplished by reducing
the noise (in such a way that the residual noise is not annoying to the listener), while minimizing the speech distortion
introduced during the enhancement process. In this technique, the modulation domain has been investigated as an alternative
to the acoustic domain for speech enhancement. More specifically, it determines how competitive the modulation domain is
for spectral subtraction as compared to the acoustic domain. For this purpose, the traditional analysis, modification, synthesis
and framework to include modulation domain processing has been extended. Then it compensates the noisy modulation
spectrum for additive noise distortion by applying the spectral subtraction algorithm in the modulation domain. Using an
objective speech quality measure as well as formal subjective listening tests, it has been showed that the proposed method
results in improved speech quality. Furthermore, the proposed method achieves better noise suppression than the MMSE
method.
VIII. ADAPTIVE NOISE CANCELLATION IN THE WAVELET DOMAIN (ANCWD)
Adaptive filtering has been used for speech denoising in the time domain. During the last decade, wavelet transform has
been developed for speech enhancement. In this paper we are proposing to use adaptive filtering in the Wavelet transform
domain. We propose a hybrid method of using adaptive filters on the lower scales of a wavelet transformed speech together
with the conventional methods (Thresholding, Spectral Subtraction, and Wiener filtering) on the higher scale coefficients.
The results demonstrate that the suggested approach is computationally efficient and has a good performance.
IX. SPEECH ENHANCEMENT USING ADAPTIVE NEURO-FUZZY FILTERING
(SEANFF)
It presents an adaptive neuro-fuzzy filtering scheme using the artificial neuro-fuzzy inference system (ANFIS) for noise
reduction in speech. The measurable output noisy speech with 5dB SNR level is taken as the contaminated version of the
interference to compare with the output data of the filter. This function returns the initial FIS structure that contains a set of
fuzzy rules to cover the feature space. Finally, the ANFIS hybrid learning algorithm that combines the recursive least-
squares estimation (RLSE) method and the back propagation gradient descent (BP/GD) is applied to determine the premise
and the consequent parameters. After training, the ANFIS output (i.e. estimated interference) was determined. Then the
estimated information signal is calculated as the difference between the measured signal and the estimated interference.
X. ADAPTIVE NOISE CANCELLATION USING CASCADED CORRELATION
NEURAL NETWORKS (ANCCCNN)
The main objective of Speech Enhancement is to enhance the speech signal to obtain a clean signal with higher
quality. Such system has been widely used in long distance telephony applications. A novel adaptive noise cancellation
algorithm using cascaded correlation neural networks is described. In the proposed algorithm the objective is to filter out an
interference component by identifying the non-linear model between a measurable noise source and the corresponding
immeasurable interference. The cascaded correlation neural network algorithm has the powerful capabilities of learning and
adaptation.
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4. Different Techniques for the Enhancement of the Intelligibility of a Speech Signal
XI. WORKING ALGORITHM
Fig- 1 Algorithm for SNR estimation
XII. SIMULATION STUDIES
MATLAB Simulations has been carried out to estimate the SNR. Various steps for estimation of the SNR are as follows:
Fig- 2 Voice signal
Fig- 3 Noise source signal
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5. Different Techniques for the Enhancement of the Intelligibility of a Speech Signal
Fig- 4 Mixed signal
Fig- 5 Unknown channel characteristics
Fig- 6 Random and channel Noise
Fig- 7 Measured Signal
61
6. Different Techniques for the Enhancement of the Intelligibility of a Speech Signal
Fig- 8 After notch filter
Fig- 9 Number of membership functions
Fig- 10 Unknown and estimated Noise
Fig- 11 original and estimated signal
62
7. Different Techniques for the Enhancement of the Intelligibility of a Speech Signal
XIII. RESULT ANALYSIS
Fig- 12 RMSE v/s Number of epochs
Fig- 13 Coherence between original and estimated signal
XIV. CONCLUSION
In this paper, some speech enhancement techniques are reviewed and after that it has been concluded that the fuzzy
and neural algorithms used for minimizing noise from a set of sound files has given the best optimal results. The root mean
square error (RMSE) and number of epochs are less and at the same time the membership functions are also less. Since a
database consist of multiple voice files of one subject recorded with different words, the fuzzy tool require some fine tunin g
of parameters to arrive at best possible results where noise effect is minimum and RMSE is also close to zero and estimated
SNR is very close to the original signal SNR. It can further be extended on images and videos. Therefore it is suggested that
one should try to minimize noise using similar methodology for reducing noise form images and videos also as fuzzy
algorithm has given us very promising results.
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