This document presents a novel speech enhancement evaluation approach called SR (Signal to Residual spectrum ratio). SR aims to improve speech intelligibility for hearing impaired individuals in non-stationary noisy environments. The approach segments noisy speech into pure, quasi, and non-speech frames using threshold conditions on the signal and estimated noise spectra. Noise power is estimated differently for each frame type. SR and LLR (log likelihood ratio) are used to measure distortions and compare the proposed approach to weighted averaging techniques. Results show the proposed SR approach achieves better segmental SNR and LLR scores than weighted averaging, indicating it enhances speech quality and intelligibility more effectively in car, airport, and train noise environments.
Improving the Efficiency of Spectral Subtraction Method by Combining it with ...IJORCS
In the field of speech signal processing, Spectral subtraction method (SSM) has been successfully implemented to suppress the noise that is added acoustically. SSM does reduce the noise at satisfactory level but musical noise is a major drawback of this method. To implement spectral subtraction method, transformation of speech signal from time domain to frequency domain is required. On the other hand, Wavelet transform displays another aspect of speech signal. In this paper we have applied a new approach in which SSM is cascaded with wavelet thresholding technique (WTT) for improving the quality of speech signal by removing the problem of musical noise to a great extent. Results of this proposed system have been simulated on MATLAB.
Speech Enhancement Using Spectral Flatness Measure Based Spectral SubtractionIOSRJVSP
This paper is aimed to reduce background noise introduced in speech signal during capture, storage, transmission and processing using Spectral Subtraction algorithm. To consider the fact that colored noise corrupts the speech signal non-uniformly over different frequency bands, Multi-Band Spectral Subtraction (MBSS) approach is exploited wherein amount of noise subtracted from noisy speech signal is decided by a weighting factor. Choice of optimal values of weights decides the performance of the speech enhancement system. In this paper weights are decided based on SFM (Spectral Flatness Measure) than conventional SNR (Signal to Noise Ratio) based rule. Since SFM is able to provide true distinction between speech signal and noise signal. Spectrogram, Mean Opinion Score show that speech enhanced from proposed SFM based MBSS possess better perceptual quality and improved intelligibility than existing SNR based MBSS
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.
Improving the Efficiency of Spectral Subtraction Method by Combining it with ...IJORCS
In the field of speech signal processing, Spectral subtraction method (SSM) has been successfully implemented to suppress the noise that is added acoustically. SSM does reduce the noise at satisfactory level but musical noise is a major drawback of this method. To implement spectral subtraction method, transformation of speech signal from time domain to frequency domain is required. On the other hand, Wavelet transform displays another aspect of speech signal. In this paper we have applied a new approach in which SSM is cascaded with wavelet thresholding technique (WTT) for improving the quality of speech signal by removing the problem of musical noise to a great extent. Results of this proposed system have been simulated on MATLAB.
Speech Enhancement Using Spectral Flatness Measure Based Spectral SubtractionIOSRJVSP
This paper is aimed to reduce background noise introduced in speech signal during capture, storage, transmission and processing using Spectral Subtraction algorithm. To consider the fact that colored noise corrupts the speech signal non-uniformly over different frequency bands, Multi-Band Spectral Subtraction (MBSS) approach is exploited wherein amount of noise subtracted from noisy speech signal is decided by a weighting factor. Choice of optimal values of weights decides the performance of the speech enhancement system. In this paper weights are decided based on SFM (Spectral Flatness Measure) than conventional SNR (Signal to Noise Ratio) based rule. Since SFM is able to provide true distinction between speech signal and noise signal. Spectrogram, Mean Opinion Score show that speech enhanced from proposed SFM based MBSS possess better perceptual quality and improved intelligibility than existing SNR based MBSS
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.
This is my presentation on a Journal Club. It's based on the article: "Auditory inspired machine learning techniques can improve speech intelligibility and quality for hearing-impaired listeners". You can find all the references in the slide at the end of the article. I review very basic techniques in noise reduction, and how the techniques are implemented in the area of deep neural-network.
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.
Comparative performance analysis of channel normalization techniqueseSAT Journals
Abstract A major part of the interaction between humans takes place via speech communication. The speech signal carries both useful and unwanted information. Processing of such signals involve enhancing the useful information. The intelligibility of speech signals is significantly reduced due to the presence of unwanted information such as noise. Channel normalization algorithms suppress such additive noise introduced in the speech signals by transmission channel or by recording environment conditions. Enhancing the quality and intelligibility of speech signals improve the performance of speech systems such as Automatic speech recognition (ASR) , voice communication and hearing aids to name the few. Based on the experimental results the comparative analysis of channel normalization techniques have been presented in this paper to find out the most suitable algorithm for enhancing the speech signals. Keywords: Cepstral Mean Normalization, Spectral Subtraction, Weiner filter, Signal to Noise Ratio
International Journal of Engineering Research and Development (IJERD)IJERD Editor
call for paper 2012, hard copy of journal, research paper publishing, where to publish research paper,
journal publishing, how to publish research paper, Call For research paper, international journal, publishing a paper, IJERD, journal of science and technology, how to get a research paper published, publishing a paper, publishing of journal, publishing of research paper, reserach and review articles, IJERD Journal, How to publish your research paper, publish research paper, open access engineering journal, Engineering journal, Mathemetics journal, Physics journal, Chemistry journal, Computer Engineering, Computer Science journal, how to submit your paper, peer reviw journal, indexed journal, reserach and review articles, engineering journal, www.ijerd.com, research journals,
yahoo journals, bing journals, International Journal of Engineering Research and Development, google journals, hard copy of journal
Analysis of PEAQ Model using Wavelet Decomposition Techniquesidescitation
Digital broadcasting, internet audio and music database make use of audio
compression and coding techniques to reduce high quality audio signal without impairing its
perceptual quality. Audio signal compression is the lossy compression
technique, It
converts original converting audio signal into compressed bitstream. The compressed audio
bitstream is decoded at the decoder to produce a close approximation of the original signal.
For the purpose of improving the coding this work attempts to verify the perceptual
evaluation of audio quality (PEAQ) model in BS.1387 using wavelet decomposition
techniques. Finally the comparison of masking threshold for sub-bands using Wavelet
techniques and Fast Fourier transform (FFT) will be done
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.
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.
IJERA (International journal of Engineering Research and Applications) is International online, ... peer reviewed journal. For more detail or submit your article, please visit www.ijera.com
Deep Learning Based Voice Activity Detection and Speech EnhancementNAVER Engineering
발표자: 김준태 (KAIST 박사과정)
발표일: 2018.10
Voice activity detection (VAD) and speech enhancement (SE) are important front-end technologies for noise robust speech recognition system.
From incoming noisy signal, VAD detects the speech signal only and SE removes the noise signal while conserving the speech signal.
For VAD and SE, this presentation will cover the traditional methods, deep learning based methods, and our papers as follows:
1. J. Kim and M. Hahn, "Voice Activity Detection Using an Adaptive Context Attention Model," in IEEE Signal Processing Letters, vol. 25, no. 8, pp. 1181-1185, Aug. 2018.
2. J. Kim and M. Hahn, "Speech Enhancement Using a Two Step Network," submitted to IEEE Signal Processing Letters, 2018.
Also, this presentation will briefly introduce some experimental results in real-world environment (far-field, noisy environment), conducted on the embedded board.
For VAD,
Traditional VAD methods.
Deep learning based VAD methods.
Paper presentation: J. Kim and M. Hahn, "Voice Activity Detection Using an Adaptive Context Attention Model," in IEEE Signal Processing Letters, vol. 25, no. 8, pp. 1181-1185, Aug. 2018.
End point detection based on VAD.
Experimental results of DNN-EPD on embedded board in real-world environment.
For SE,
Traditional SE methods.
Deep learning based SE methods.
Paper presentation: J. Kim and M. Hahn, "Speech Enhancement Using a Two Step Network," submitted to IEEE Signal Processing Letters, 2018.
Experimental results in real-world environment.
A New Speech Enhancement Technique to Reduce Residual Noise Using Perceptual ...IOSR Journals
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
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
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.
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.
Cancellation of Noise from Speech Signal using Voice Activity Detection Metho...ijsrd.com
Speech Enhancement by suppressing uncorrelated acoustically added noise has been a challenging topic of research for many years. These are the primary choice for real time applications due to the simplicity and comparatively low computational load. This paper shows VAD (Voice activity detection) technique that can detect the non speech segment from the speech signal. It is also shown that it can work powerfully in an unpredictable noise ambience. The technique is mostly done in microprocessors or DSP processors because of their flexibility. But there are several advantages of FPGA over DSP processors like high cost per logic element related to these processors makes them improper for large scale use. From the experimental results, VAD method is implemented on the FPGA chip.
An intensity based medical image registration using genetic algorithmsipij
Medical imaging plays a vital role to create images of human body for clinical purposes. Biomedical
imaging has taken a leap by entering into the field of image registration. Image registration integrates the
large amount of medical information embedded in the images taken at different time intervals and images
at different orientations. In this paper, an intensity-based real-coded genetic algorithm is used for
registering two MRI images. To demonstrate the efficiency of the algorithm developed, the alignment of the
image is altered and algorithm is tested for better performance. Also the work involves the comparison of
two similarity metrics, and based on the outcome the best metric suited for genetic algorithm is studied.
Parallax Effect Free Mosaicing of Underwater Video Sequence Based on Texture ...sipij
In this paper, we present feature-based technique for construction of mosaic image from underwater video
sequence, which suffers from parallax distortion due to propagation properties of light in the underwater
environment. The most of the available mosaic tools and underwater image mosaicing techniques yields
final result with some artifacts such as blurring, ghosting and seam due to presence of parallax in the input
images. The removal of parallax from input images may not reduce its effects instead it must be corrected
in successive steps of mosaicing. Thus, our approach minimizes the parallax effects by adopting an efficient
local alignment technique after global registration. We extract texture features using Centre Symmetric
Local Binary Pattern (CS-LBP) descriptor in order to find feature correspondences, which are used further
for estimation of homography through RANSAC. In order to increase the accuracy of global registration,
we perform preprocessing such as colour alignment between two selected frames based on colour
distribution adjustment. Because of existence of 100% overlap in consecutive frames of underwater video,
we select frames with minimum overlap based on mutual offset in order to reduce the computation cost
during mosaicing. Our approach minimizes the parallax effects considerably in final mosaic constructed
using our own underwater video sequences.
Feature selection approach in animal classificationsipij
In this paper, we propose a model for automatic classification of Animals using different classifiers Nearest
Neighbour, Probabilistic Neural Network and Symbolic. Animal images are segmented using maximal
region merging segmentation. The Gabor features are extracted from segmented animal images.
Discriminative texture features are then selected using the different feature selection algorithm like
Sequential Forward Selection, Sequential Floating Forward Selection, Sequential Backward Selection and
Sequential Floating Backward Selection. To corroborate the efficacy of the proposed method, an
experiment was conducted on our own data set of 25 classes of animals, containing 2500 samples. The
data set has different animal species with similar appearance (small inter-class variations) across different
classes and varying appearance (large intra-class variations) within a class. In addition, the images of
flowers are of different poses, with cluttered background under different lighting and climatic conditions.
Experiment results reveal that Symbolic classifier outperforms Nearest Neighbour and Probabilistic Neural
Network classifiers.
This is my presentation on a Journal Club. It's based on the article: "Auditory inspired machine learning techniques can improve speech intelligibility and quality for hearing-impaired listeners". You can find all the references in the slide at the end of the article. I review very basic techniques in noise reduction, and how the techniques are implemented in the area of deep neural-network.
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.
Comparative performance analysis of channel normalization techniqueseSAT Journals
Abstract A major part of the interaction between humans takes place via speech communication. The speech signal carries both useful and unwanted information. Processing of such signals involve enhancing the useful information. The intelligibility of speech signals is significantly reduced due to the presence of unwanted information such as noise. Channel normalization algorithms suppress such additive noise introduced in the speech signals by transmission channel or by recording environment conditions. Enhancing the quality and intelligibility of speech signals improve the performance of speech systems such as Automatic speech recognition (ASR) , voice communication and hearing aids to name the few. Based on the experimental results the comparative analysis of channel normalization techniques have been presented in this paper to find out the most suitable algorithm for enhancing the speech signals. Keywords: Cepstral Mean Normalization, Spectral Subtraction, Weiner filter, Signal to Noise Ratio
International Journal of Engineering Research and Development (IJERD)IJERD Editor
call for paper 2012, hard copy of journal, research paper publishing, where to publish research paper,
journal publishing, how to publish research paper, Call For research paper, international journal, publishing a paper, IJERD, journal of science and technology, how to get a research paper published, publishing a paper, publishing of journal, publishing of research paper, reserach and review articles, IJERD Journal, How to publish your research paper, publish research paper, open access engineering journal, Engineering journal, Mathemetics journal, Physics journal, Chemistry journal, Computer Engineering, Computer Science journal, how to submit your paper, peer reviw journal, indexed journal, reserach and review articles, engineering journal, www.ijerd.com, research journals,
yahoo journals, bing journals, International Journal of Engineering Research and Development, google journals, hard copy of journal
Analysis of PEAQ Model using Wavelet Decomposition Techniquesidescitation
Digital broadcasting, internet audio and music database make use of audio
compression and coding techniques to reduce high quality audio signal without impairing its
perceptual quality. Audio signal compression is the lossy compression
technique, It
converts original converting audio signal into compressed bitstream. The compressed audio
bitstream is decoded at the decoder to produce a close approximation of the original signal.
For the purpose of improving the coding this work attempts to verify the perceptual
evaluation of audio quality (PEAQ) model in BS.1387 using wavelet decomposition
techniques. Finally the comparison of masking threshold for sub-bands using Wavelet
techniques and Fast Fourier transform (FFT) will be done
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.
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.
IJERA (International journal of Engineering Research and Applications) is International online, ... peer reviewed journal. For more detail or submit your article, please visit www.ijera.com
Deep Learning Based Voice Activity Detection and Speech EnhancementNAVER Engineering
발표자: 김준태 (KAIST 박사과정)
발표일: 2018.10
Voice activity detection (VAD) and speech enhancement (SE) are important front-end technologies for noise robust speech recognition system.
From incoming noisy signal, VAD detects the speech signal only and SE removes the noise signal while conserving the speech signal.
For VAD and SE, this presentation will cover the traditional methods, deep learning based methods, and our papers as follows:
1. J. Kim and M. Hahn, "Voice Activity Detection Using an Adaptive Context Attention Model," in IEEE Signal Processing Letters, vol. 25, no. 8, pp. 1181-1185, Aug. 2018.
2. J. Kim and M. Hahn, "Speech Enhancement Using a Two Step Network," submitted to IEEE Signal Processing Letters, 2018.
Also, this presentation will briefly introduce some experimental results in real-world environment (far-field, noisy environment), conducted on the embedded board.
For VAD,
Traditional VAD methods.
Deep learning based VAD methods.
Paper presentation: J. Kim and M. Hahn, "Voice Activity Detection Using an Adaptive Context Attention Model," in IEEE Signal Processing Letters, vol. 25, no. 8, pp. 1181-1185, Aug. 2018.
End point detection based on VAD.
Experimental results of DNN-EPD on embedded board in real-world environment.
For SE,
Traditional SE methods.
Deep learning based SE methods.
Paper presentation: J. Kim and M. Hahn, "Speech Enhancement Using a Two Step Network," submitted to IEEE Signal Processing Letters, 2018.
Experimental results in real-world environment.
A New Speech Enhancement Technique to Reduce Residual Noise Using Perceptual ...IOSR Journals
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
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
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.
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.
Cancellation of Noise from Speech Signal using Voice Activity Detection Metho...ijsrd.com
Speech Enhancement by suppressing uncorrelated acoustically added noise has been a challenging topic of research for many years. These are the primary choice for real time applications due to the simplicity and comparatively low computational load. This paper shows VAD (Voice activity detection) technique that can detect the non speech segment from the speech signal. It is also shown that it can work powerfully in an unpredictable noise ambience. The technique is mostly done in microprocessors or DSP processors because of their flexibility. But there are several advantages of FPGA over DSP processors like high cost per logic element related to these processors makes them improper for large scale use. From the experimental results, VAD method is implemented on the FPGA chip.
An intensity based medical image registration using genetic algorithmsipij
Medical imaging plays a vital role to create images of human body for clinical purposes. Biomedical
imaging has taken a leap by entering into the field of image registration. Image registration integrates the
large amount of medical information embedded in the images taken at different time intervals and images
at different orientations. In this paper, an intensity-based real-coded genetic algorithm is used for
registering two MRI images. To demonstrate the efficiency of the algorithm developed, the alignment of the
image is altered and algorithm is tested for better performance. Also the work involves the comparison of
two similarity metrics, and based on the outcome the best metric suited for genetic algorithm is studied.
Parallax Effect Free Mosaicing of Underwater Video Sequence Based on Texture ...sipij
In this paper, we present feature-based technique for construction of mosaic image from underwater video
sequence, which suffers from parallax distortion due to propagation properties of light in the underwater
environment. The most of the available mosaic tools and underwater image mosaicing techniques yields
final result with some artifacts such as blurring, ghosting and seam due to presence of parallax in the input
images. The removal of parallax from input images may not reduce its effects instead it must be corrected
in successive steps of mosaicing. Thus, our approach minimizes the parallax effects by adopting an efficient
local alignment technique after global registration. We extract texture features using Centre Symmetric
Local Binary Pattern (CS-LBP) descriptor in order to find feature correspondences, which are used further
for estimation of homography through RANSAC. In order to increase the accuracy of global registration,
we perform preprocessing such as colour alignment between two selected frames based on colour
distribution adjustment. Because of existence of 100% overlap in consecutive frames of underwater video,
we select frames with minimum overlap based on mutual offset in order to reduce the computation cost
during mosaicing. Our approach minimizes the parallax effects considerably in final mosaic constructed
using our own underwater video sequences.
Feature selection approach in animal classificationsipij
In this paper, we propose a model for automatic classification of Animals using different classifiers Nearest
Neighbour, Probabilistic Neural Network and Symbolic. Animal images are segmented using maximal
region merging segmentation. The Gabor features are extracted from segmented animal images.
Discriminative texture features are then selected using the different feature selection algorithm like
Sequential Forward Selection, Sequential Floating Forward Selection, Sequential Backward Selection and
Sequential Floating Backward Selection. To corroborate the efficacy of the proposed method, an
experiment was conducted on our own data set of 25 classes of animals, containing 2500 samples. The
data set has different animal species with similar appearance (small inter-class variations) across different
classes and varying appearance (large intra-class variations) within a class. In addition, the images of
flowers are of different poses, with cluttered background under different lighting and climatic conditions.
Experiment results reveal that Symbolic classifier outperforms Nearest Neighbour and Probabilistic Neural
Network classifiers.
Review of ocr techniques used in automatic mail sorting of postal envelopessipij
This paper presents a review of various OCR techniq
ues used in the automatic mail sorting process. A
complete description on various existing methods fo
r address block extraction and digit recognition th
at
were used in the literature is discussed. The objec
tive of this study is to provide a complete overvie
w about
the methods and techniques used by many researchers
for automating the mail sorting process in postal
service in various countries. The significance of Z
ip code or Pincode recognition is discussed.
Robust content based watermarking algorithm using singular value decompositio...sipij
Nowadays, image content is frequently subject to different malicious manipulations. To protect images
from this illegal manipulations computer science community have recourse to watermarking techniques. To
protect digital multimedia content we need just to embed an invisible watermark into images which
facilitate the detection of different manipulations, duplication, illegitimate distributions of these images. In
this work a robust watermarking technique is presented that embedding invisible watermarks into colour
images the singular value decomposition bloc by bloc of a robust transform of images that is the Radial
symmetry transform. Each bit of the watermark is inserted in a bloc of eight pixels large of the blue
channel a high singular value of the corresponding bloc into the radial symmetry map. We justified the
insertion in the blue channel by our feeble sensibility to perturbations in this colour channel of images. We
present also results obtained with different tests. We had tested the imperceptibility of the mark using this
approach and also its robustness face to several attacks.
Application of parallel algorithm approach for performance optimization of oi...sipij
This paper gives a detailed study on the performance of image filter algorithm with various parameters
applied on an image of RGB model. There are various popular image filters, which consumes large amount
of computing resources for processing. Oil paint image filter is one of the very interesting filters, which is
very performance hungry. Current research tries to find improvement in oil paint image filter algorithm by
using parallel pattern library. With increasing kernel-size, the processing time of oil paint image filter
algorithm increases exponentially. I have also observed in various blogs and forums, the questions for
faster oil paint have been asked repeatedly.
Lossless image compression using new biorthogonal waveletssipij
Even though a large number of wavelets exist, one needs new wavelets for their specific applications. One
of the basic wavelet categories is orthogonal wavelets. But it was hard to find orthogonal and symmetric
wavelets. Symmetricity is required for perfect reconstruction. Hence, a need for orthogonal and symmetric
arises. The solution was in the form of biorthogonal wavelets which preserves perfect reconstruction
condition. Though a number of biorthogonal wavelets are proposed in the literature, in this paper four new
biorthogonal wavelets are proposed which gives better compression performance. The new wavelets are
compared with traditional wavelets by using the design metrics Peak Signal to Noise Ratio (PSNR) and
Compression Ratio (CR). Set Partitioning in Hierarchical Trees (SPIHT) coding algorithm was utilized to
incorporate compression of images.
Speaker Identification From Youtube Obtained Datasipij
An efficient, and intuitive algorithm is presented for the identification of speakers from a long dataset (like
YouTube long discussion, Cocktail party recorded audio or video).The goal of automatic speaker
identification is to identify the number of different speakers and prepare a model for that speaker by
extraction, characterization and speaker-specific information contained in the speech signal. It has many
diverse application specially in the field of Surveillance , Immigrations at Airport , cyber security ,
transcription in multi-source of similar sound source, where it is difficult to assign transcription arbitrary.
The most commonly speech parameterization used in speaker verification, K-mean, cepstral analysis, is
detailed. Gaussian mixture modeling, which is the speaker modeling technique is then explained. Gaussian
mixture models (GMM), perhaps the most robust machine learning algorithm has been introduced to
examine and judge carefully speaker identification in text independent. The application or employment of
Gaussian mixture models for monitoring & Analysing speaker identity is encouraged by the familiarity,
awareness, or understanding gained through experience that Gaussian spectrum depict the characteristics
of speaker's spectral conformational pattern and remarkable ability of GMM to construct capricious
densities after that we illustrate 'Expectation maximization' an iterative algorithm which takes some
arbitrary value in initial estimation and carry on the iterative process until the convergence of value is
observed We have tried to obtained 85 ~ 95% of accuracy using speaker modeling of vector quantization
and Gaussian Mixture model ,so by doing various number of experiments we are able to obtain 79 ~ 82%
of identification rate using Vector quantization and 85 ~ 92.6% of identification rate using GMM modeling
by Expectation maximization parameter estimation depending on variation of parameter.
IDENTIFICATION OF SUITED QUALITY METRICS FOR NATURAL AND MEDICAL IMAGESsipij
To assess quality of the denoised image is one of the important task in image denoising application.
Numerous quality metrics are proposed by researchers with their particular characteristics till today. In
practice, image acquisition system is different for natural and medical images. Hence noise introduced in
these images is also different in nature. Considering this fact, authors in this paper tried to identify the
suited quality metrics for Gaussian, speckle and Poisson corrupted natural, ultrasound and X-ray images
respectively. In this paper, sixteen different quality metrics from full reference category are evaluated with
respect to noise variance and suited quality metric for particular type of noise is identified. Strong need to
develop noise dependent quality metric is also identified in this work.
Global threshold and region based active contour model for accurate image seg...sipij
In this contribution, we develop a novel global threshold-based active contour model. This model deploys a new
edge-stopping function to control the direction of the evolution and to stop the evolving contour at weak or
blurred edges. An implementation of the model requires the use of selective binary and Gaussian filtering
regularized level set (SBGFRLS) method. The method uses either a selective local or global segmentation
property. It penalizes the level set function to force it to become a binary function. This procedure is followed by
using a regularisation Gaussian. The Gaussian filters smooth the level set function and stabilises the evolution
process. One of the merits of our proposed model stems from the ability to initialise the contour anywhere inside
the image to extract object boundaries. The proposed method is found to perform well, notably when the
intensities inside and outside the object are homogenous. Our method is applied with satisfactory results on
various types of images, including synthetic, medical and Arabic-characters images.
Beamforming with per antenna power constraint and transmit antenna selection ...sipij
In this paper, transmit beamforming and antenna selection techniques are presented for the Cooperative
Distributed Antenna System. Beamforming technique with minimum total weighted transmit power
satisfying threshold SINR and Per-Antenna Power constraints is formulated as a convex optimization
problem for the efficient performance of Distributed Antenna System (DAS). Antenna Selection technique is
implemented in this paper to select the optimum Remote Antenna Units from all the available ones. This
achieves the best compromise between capacity and system complexity. Dual polarized and Triple
Polarized systems are considered. Simulation results prove that by integrating Beamforming with DAS
enhances its performance. Also by using convex optimization in Antenna Selection enhances the
performance of multi polarized systems.
A voting based approach to detect recursive order number of photocopy documen...sipij
Photocopy documents are very common in our normal life. People are permitted to carry and present
photocopied documents to avoid damages to the original documents. But this provision is misused for
temporary benefits by fabricating fake photocopied documents. Fabrication of fake photocopied document
is possible only in 2nd and higher order recursive order of photocopies. Whenever a photocopied document
is submitted, it may be required to check its originality. When the document is 1st order photocopy, chances
of fabrication may be ignored. On the other hand when the photocopy order is 2nd or above, probability of
fabrication may be suspected. Hence when a photocopy document is presented, the recursive order number
of photocopy is to be estimated to ascertain the originality. This requirement demands to investigate
methods to estimate order number of photocopy. In this work, a voting based approach is used to detect the
recursive order number of the photocopy document using probability distributions exponential, extreme
values and lognormal distributions is proposed. A detailed experimentation is performed on a generated
data set and the method exhibits efficiency close to 89%.
Offline handwritten signature identification using adaptive window positionin...sipij
The paper presents to address this challenge, we have proposed the use of Adaptive Window Positioning
technique which focuses on not just the meaning of the handwritten signature but also on the individuality
of the writer. This innovative technique divides the handwritten signature into 13 small windows of size nxn
(13x13). This size should be large enough to contain ample information about the style of the author and
small enough to ensure a good identification performance. The process was tested with a GPDS dataset
containing 4870 signature samples from 90 different writers by comparing the robust features of the test
signature with that of the user’s signature using an appropriate classifier. Experimental results reveal that
adaptive window positioning technique proved to be the efficient and reliable method for accurate
signature feature extraction for the identification of offline handwritten signatures .The contribution of this
technique can be used to detect signatures signed under emotional duress
Contrast enhancement using various statistical operations and neighborhood pr...sipij
Histogram Equalization is a simple and effective contrast enhancement technique. In spite of its popularity
Histogram Equalization still have some limitations –produces artifacts, unnatural images and the local
details are not considered, therefore due to these limitations many other Equalization techniques have been
derived from it with some up gradation. In this proposed method statistics play an important role in image
processing, where statistical operations is applied to the image to get the desired result such as
manipulation of brightness and contrast. Thus, a novel algorithm using statistical operations and
neighborhood processing has been proposed in this paper where the algorithm has proven to be effective in
contrast enhancement based on the theory and experiment.
A combined method of fractal and glcm features for mri and ct scan images cla...sipij
Fractal analysis has been shown to be useful in image processing for characterizing shape and gray-scale
complexity. The fractal feature is a compact descriptor used to give a numerical measure of the degree of
irregularity of the medical images. This descriptor property does not give ownership of the local image
structure. In this paper, we present a combination of this parameter based on Box Counting with GLCM
Features. This powerful combination has proved good results especially in classification of medical texture
from MRI and CT Scan images of trabecular bone. This method has the potential to improve clinical
diagnostics tests for osteoporosis pathologies.
Image retrieval and re ranking techniques - a surveysipij
There is a huge amount of research work focusing on the searching, retrieval and re-ranking of images in
the image database. The diverse and scattered work in this domain needs to be collected and organized for
easy and quick reference.
Relating to the above context, this paper gives a brief overview of various image retrieval and re-ranking
techniques. Starting with the introduction to existing system the paper proceeds through the core
architecture of image harvesting and retrieval system to the different Re-ranking techniques. These
techniques are discussed in terms of approaches, methodologies and findings and are listed in tabular form
for quick review.
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.
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.
A New Approach for Speech Enhancement Based On Eigenvalue Spectral SubtractionCSCJournals
In this paper, a phase space reconstruction-based method is proposed for speech enhancement. The method embeds the noisy signal into a high dimensional reconstructed phase space and uses Spectral Subtraction idea. The advantages of the proposed method are fast performance, high SNR and good MOS. In order to evaluate the proposed method, ten signals of TIMIT database mixed with the white additive Gaussian noise and then the method was implemented. The efficiency of the proposed method was evaluated by using qualitative and quantitative criteria.
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.
Broad Phoneme Classification Using Signal Based Features ijsc
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.
International Journal of Computational Engineering Research(IJCER)ijceronline
International Journal of Computational Engineering Research(IJCER) is an intentional online Journal in English monthly publishing journal. This Journal publish original research work that contributes significantly to further the scientific knowledge in engineering and Technology.
Suppression of noise in noisy speech signal is required in many speech enhancement applications like signal recording and transmission from one place to other. In this paper a novel single line noise cancellation system is proposed using derivative of normalized least mean spare algorithm. The proposed system has two phases. The first phase is generation of secondary reference signal from incoming primary signal itself at initial silence period and pause between two words, which is essential while adaptive filter using as noise canceller. Second phase is noise cancellation using proposed modified error data normalized step size (EDNSS) algorithm. The performance of the proposed algorithm is compared with normalized least mean square (NLMS) algorithm and original EDNSS algorithm using standard IEEE sentence (SP23) of Noizeus data base with different types of real-world noise at different level of signal to noise ratio (SNR). The output of proposed, NLMS and EDNSS algorithm are measured with output SNR, excessive mean square error (EMSE) and misadjustment (M). The results clearly illustrates that the proposed algorithm gives improved result over conventional NLMS and EDNSS algorithm. The speed of convergence is also maintained as same conventional NLMS algorithm.
A Combined Voice Activity Detector Based On Singular Value Decomposition and ...CSCJournals
voice activity detector (VAD) is used to separate the speech data included parts from silence parts of the signal. In this paper a new VAD algorithm is represented on the basis of singular value decomposition. There are two sections to perform the feature vector extraction. In first section voiced frames are separated from unvoiced and silence frames. In second section unvoiced frames are silence frames. To perform the above sections, first, windowing the noisy signal then Hankel’s matrix is formed for each frame. The basis of statistical feature extraction of purposed system is slope of singular value curve related to each frame by using linear regression. It is shown that the slope of singular values curve per different SNRs in voiced frames is more than the other types and this property can be to achieve the goal the first part can be used. High similarity between feature vector of unvoiced and silence frame caused to approach for separation of the two categories above cannot be used. So in the second part, the frequency characteristics for identification of unvoiced frames from silent frames have been used. Simulation results show that high speed and accuracy are the advantages of the proposed system.
Compressive Sensing in Speech from LPC using Gradient Projection for Sparse R...IJERA Editor
This paper presents compressive sensing technique used for speech reconstruction using linear predictive coding because the
speech is more sparse in LPC. DCT of a speech is taken and the DCT points of sparse speech are thrown away arbitrarily.
This is achieved by making some point in DCT domain to be zero by multiplying with mask functions. From the incomplete
points in DCT domain, the original speech is reconstructed using compressive sensing and the tool used is Gradient
Projection for Sparse Reconstruction. The performance of the result is compared with direct IDCT subjectively. The
experiment is done and it is observed that the performance is better for compressive sensing than the DCT.
We present a causal speech enhancement model working on the
raw waveform that runs in real-time on a laptop CPU. The proposed model is based on an encoder-decoder architecture with
skip-connections. It is optimized on both time and frequency
domains, using multiple loss functions. Empirical evidence
shows that it is capable of removing various kinds of background noise including stationary and non-stationary noises,
as well as room reverb. Additionally, we suggest a set of
data augmentation techniques applied directly on the raw waveform which further improve model performance and its generalization abilities. We perform evaluations on several standard
benchmarks, both using objective metrics and human judgements. The proposed model matches state-of-the-art performance of both causal and non causal methods while working
directly on the raw waveform.
Index Terms: Speech enhancement, speech denoising, neural
networks, raw waveform
Vaccine management system project report documentation..pdfKamal Acharya
The Division of Vaccine and Immunization is facing increasing difficulty monitoring vaccines and other commodities distribution once they have been distributed from the national stores. With the introduction of new vaccines, more challenges have been anticipated with this additions posing serious threat to the already over strained vaccine supply chain system in Kenya.
Quality defects in TMT Bars, Possible causes and Potential Solutions.PrashantGoswami42
Maintaining high-quality standards in the production of TMT bars is crucial for ensuring structural integrity in construction. Addressing common defects through careful monitoring, standardized processes, and advanced technology can significantly improve the quality of TMT bars. Continuous training and adherence to quality control measures will also play a pivotal role in minimizing these defects.
Saudi Arabia stands as a titan in the global energy landscape, renowned for its abundant oil and gas resources. It's the largest exporter of petroleum and holds some of the world's most significant reserves. Let's delve into the top 10 oil and gas projects shaping Saudi Arabia's energy future in 2024.
Industrial Training at Shahjalal Fertilizer Company Limited (SFCL)MdTanvirMahtab2
This presentation is about the working procedure of Shahjalal Fertilizer Company Limited (SFCL). A Govt. owned Company of Bangladesh Chemical Industries Corporation under Ministry of Industries.
Final project report on grocery store management system..pdfKamal Acharya
In today’s fast-changing business environment, it’s extremely important to be able to respond to client needs in the most effective and timely manner. If your customers wish to see your business online and have instant access to your products or services.
Online Grocery Store is an e-commerce website, which retails various grocery products. This project allows viewing various products available enables registered users to purchase desired products instantly using Paytm, UPI payment processor (Instant Pay) and also can place order by using Cash on Delivery (Pay Later) option. This project provides an easy access to Administrators and Managers to view orders placed using Pay Later and Instant Pay options.
In order to develop an e-commerce website, a number of Technologies must be studied and understood. These include multi-tiered architecture, server and client-side scripting techniques, implementation technologies, programming language (such as PHP, HTML, CSS, JavaScript) and MySQL relational databases. This is a project with the objective to develop a basic website where a consumer is provided with a shopping cart website and also to know about the technologies used to develop such a website.
This document will discuss each of the underlying technologies to create and implement an e- commerce website.
TECHNICAL TRAINING MANUAL GENERAL FAMILIARIZATION COURSEDuvanRamosGarzon1
AIRCRAFT GENERAL
The Single Aisle is the most advanced family aircraft in service today, with fly-by-wire flight controls.
The A318, A319, A320 and A321 are twin-engine subsonic medium range aircraft.
The family offers a choice of engines
Student information management system project report ii.pdfKamal Acharya
Our project explains about the student management. This project mainly explains the various actions related to student details. This project shows some ease in adding, editing and deleting the student details. It also provides a less time consuming process for viewing, adding, editing and deleting the marks of the students.
Overview of the fundamental roles in Hydropower generation and the components involved in wider Electrical Engineering.
This paper presents the design and construction of hydroelectric dams from the hydrologist’s survey of the valley before construction, all aspects and involved disciplines, fluid dynamics, structural engineering, generation and mains frequency regulation to the very transmission of power through the network in the United Kingdom.
Author: Robbie Edward Sayers
Collaborators and co editors: Charlie Sims and Connor Healey.
(C) 2024 Robbie E. Sayers
Democratizing Fuzzing at Scale by Abhishek Aryaabh.arya
Presented at NUS: Fuzzing and Software Security Summer School 2024
This keynote talks about the democratization of fuzzing at scale, highlighting the collaboration between open source communities, academia, and industry to advance the field of fuzzing. It delves into the history of fuzzing, the development of scalable fuzzing platforms, and the empowerment of community-driven research. The talk will further discuss recent advancements leveraging AI/ML and offer insights into the future evolution of the fuzzing landscape.
A Novel Uncertainty Parameter SR ( Signal to Residual Spectrum Ratio ) Evaluation Approach for Speech Enhancement
1. Signal & Image Processing : An International Journal (SIPIJ) Vol.5, No.5, October 2014
A NOVEL UNCERTAINTY PARAMETER SR
(SIGNAL TO RESIDUAL SPECTRUM RATIO)
EVALUATION APPROACH FOR SPEECH
ENHANCEMENT
M. Ravichandra Kumar1 and B. Ravi Teja2
1Department of Electronics and Communication,
M-tech, Gudlavalleru Engineering College, A.P, India
2Department of Electronics and Communication Engineering, Assistant professor,
Gudlavalleru Engineering College, A.P, India
ABSTRACT
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.
KEYWORDS
Noise Estimation, Voice Activity Detector (VAD), Speech Enhancement, SR (Signal to Residual spectrum
ratio) parameter, Speech Intelligibility Improvement.
1. INTRODUCTION
The major problem arises in speech enhancement background noise and it is affected by speech
signal. There are many applications which are speech recognition, hearing aid, VOIP (Voice over
Internet Protocol), teleconferencing systems and mobile phones of reduces background noise. The
noise present in the both analogy and digital systems. An unwanted signal as noise and it
degrades the speech intelligibility and speech quality. Vehicle noise and background noise are the
different types of noises. In speech enhancement mainly considered as noise estimation it requires
to estimate of noise from noisy speech signal. Speech enhancement main objective is to give
better performance of speech quality and speech intelligibility by using various algorithms and
based on these algorithms to minimise the MSE (Man Square Error) [5]. The effect of various
distortions (attenuation and amplification distortions) present in the speech signal so these
distortions are proper control to improve the speech intelligibility. The negative difference
DOI : 10.5121/sipij.2014.5501 1
2. Signal & Image Processing : An International Journal (SIPIJ) Vol.5, No.5, October 2014
between clean and enhanced spectrum would be amplification distortion, while a positive
difference would be attenuation distortion. Speech enhancement for noise reduction can be
categorised into three fundamental classes and those are model based, spectral restoration and
filtering technique methods. All the methods are common feature is clean speech power spectrum
estimation from noisy environment spectrum.
The presence or absence of human speech detected is called VAD (Voice Activity Detector). In
speech processing technique used VAD and also called as speech detection or speech active
detector as well as VAD used in noise reduction also. Multimedia application VAD allows
simultaneously voice and data. Here consider another application cellular based system (GSM,
CDMA) in discontinuous transmission mode. Speech intelligibility and speech quality both are
correlated highly by measuring frequency domain of segmental SNR so for this measure is refer
to residual spectrum ratio [14].
2
2. RELATED WORK
In obtainable algorithms are not suitable for estimate of background noise but VAD (voice
activity detector) good background noise estimation algorithm for stationary environment [13].
Speech presence or speech absence of human speech is detected by VAD (voice activity detector)
by using this algorithm to estimate noise in speech pauses only. Every algorithm makes to give
speech quality but not speech intelligibility and this drawback occurred in present existing
algorithms [3]. Wiener and MMSE (minimum mean square error) algorithms are used to
minimize the error in between of enhanced and clean spectrum so these algorithms are based in
spectral principals.
Most of the algorithms were proposed speech recognized application to estimate the noise in non
stationary environments VAD did not estimate the noise accurately. The lack of intelligibility in
present algorithms is not proper to estimation of noise. These problems can be reduces by using
the propose algorithm SR (signal to residual spectrum ratio) for improve speech quality and
speech intelligibility in noisy environment.
3. PROPOSED WORK
Consider, here P(n) and Q(n) are the clean speech, noise and then noisy speech denoted as
follows,
X(n) = P(n) + Q(n) (1)
Time domain of noisy speech is segmented by frames by using of windowed technique let it be
consider hamming window and represented equation as follows
2 (n-1)
Nw-1
W[n,] = 0.54 – 0.4 cos for 0nNw-1 (2)
The short time Fourier transforms is give equation for wave form of windowed speech signal
, =
,
4. Signal Image Processing : An International Journal (SIPIJ) Vol.5, No.5, October 2014
3.1 Determination of Threshold Condition
Speech intelligibility and speech quality both are correlated highly so to measure using segmental
SNR (Signal to Noise Ratio) in consider frequency domain version and this measure to mention
as signal to residual spectrum.
SNRESI(k) = (4)
S2(k)
(S(k)-S(k))2
S(k) is speech enhancement algorithm of estimated spectrum and S(k) is clean speech magnitude
spectrum. To improve the speech intelligibility by proper control of distortions using regions are
constraint and it has follows
3
a) S(k) S(k), suggested only attenuation distortion
b) S(k) 2. S(k), suggested greater or 6.02 db of amplification distortion
c) S(k) S(k) 2. S(k) , suggested up to 6.02 db of amplification distortion
Reason (a) and Reason (b) from that we constraint to this reason S(k) S(k) and it is used in
speech enhancement algorithms.
S(k) 2. S(k) (5)
This after squaring on both sides becomes
S2(k) 4. S2(k) (6)
So assume S(k)= X(k) it is not enhance noisy speech by algorithms and then
S2(k) = X2(k) = S2(k) + Q2(k) and reduces to S2(k) 1/3 Q2(k)
SNR(k) 1/3 (7)
3.2. SR (Signal to Residual spectrum ratio)
Figure.1 represents the SR algorithm and the noisy signal is segmented using windowed
technique eq.(2) later FFT is performed on the segmented frames with the help of with the help
of eq.(3). Noisy speech has different frames so we can calculate SNR (Signal To Noise Ratio)
based on threshold determination.
3.3. Noise power estimation method
Here focused on noise estimation and it has different approaches so the fundamental component
of speech enhancement is noise power estimation. It required estimating of noise from noisy
speech spectrum by using different algorithms based on classification of speech into quasi speech,
original speech and noise speech [11].
3.3.1. Non-Speech
It has to be occurred in speech absence or speech pauses only and to estimate noise power of
these frames for the following proposed condition
if S(k)2.S(k) then
A(m,k) = Â(m-1,k)+(1-) |Â(m ,k)|2 (8)
Where is called as smoothing factor and typically set to =0.98 and lies in 01.
3.3.2. Quasi-Speech
To estimate noise power for quasi speech is both noise and speech on each frame and the
proposed condition
5. Signal Image Processing : An International Journal (SIPIJ) Vol.5, No.5, October 2014
4
It S(k) S(k) 2. S(k) then
Â(m, k) = B(m, k)Â(m-1, k)+(1-B(m, k)) (9)
Where Â(m, k) is non speech frame of noise spectrum estimation
Figure 1: SR Algorithm.
3.4. Tracking the Minimum of Noisy Speech
To tracking of noisy speech by regularly averaging precedent spectral values, here used rule non
linear in different approach [10]
if Bmin (m − 1, k) B (m, k) then
1-
1-
Bmin (m, k )= Bmin (m − 1, k) + ( (B (m, k) – B( m − 1, k )) (10)
If Bmin( m − 1, k ) B (m, k) then
Bmin (m, k) = B(m, k) (11)
To determine the values of , and by experiment, in practical implementation smoothing
parameter in (11) whose maximum value is 0.98 to avoid deadlock for r(m,k)=1.
3.5. Speech Presence Probability
To measure how much speech present probability in noisy speech by following equation
|A(m,k)|2
Bmin(m,k)
Bsp(m,k) = (12)
6. Signal Image Processing : An International Journal (SIPIJ)
Vol.5, No.5, October 2014
where Bmin(m,k) and |A(m,k)|2 are represented as local minimum and power spectrum of noisy
speech. Speech present and speech absent are dependent on the ratio of speech present probability
if it is grater to threshold then consider as speech present otherwise it
gives speech absent.
3.6. Computing Logistic Function
Logistic function is one of special case in the form of mathematical and it is also called as
sigmoid function or sigmoid curve as given function
g(x)=1/(1-e-x)
Figure 2: sigmoid curve
3.7. Calculating Frequency Dependent Smoothing Constant
To compute smoothing factor need the time
B (m, k) = s+ (1- s) Bsp (m, k)
Where, s is denotes as constant.
time-frequency domain as to follow this equation
To updating of minimum noisy spectrum is B
B( m, k) = B(m − 1, k) + (1 − )
Bmin (m, k) and given equation
Where, B (l, k) is average noise spectrum and
, is known as smoothing factor
b(, k) = b b(
-1,k)+(1- b) I (, k)
Where b (, k) is a smoothing constant, the above recursive absolutely utilize the correlation for
speech presence in adjacent frames.
For r (m, k) = 1.
r(m, k) = N (m − 1, k)/N
2(m, k)
Posterior SNR of smoothed version is represented by eq. (16)
The Wiener filter solves the signal estimation problem for stationary signals. A major
contribution was the use of a statistical model for the estimated signal the filter is best in the
intellect of the MMSE [16]. . We shall focus here on the discrete-discrete
time version of the Wiener filter
and it is used to generate estimated pure signal from a given noise speech signal.
5
(13)
[9].
(14)
(15)
(16)
(17)
he
7. Signal Image Processing : An International Journal (SIPIJ) Vol.5, No.5, October 2014
6
4. IMPLEMENTATION AND RESULTS
Speech enhancement algorithms are tested on MATLAB for Non-stationary environment of
speech database [2]. The unvoiced speech regions to detected correctly by observed the results of
proposed algorithm and speech activity region also accurately measured even noise is present.
The table gives classification of results and performance of the algorithms in which segmental
SNR and LLR (Log Like hood Ratio) are compared to proposed algorithm SR (Signal to Residual
spectrum ratio). Spectrogram is way to visualize the speech signal in the domain time-frequency
representation. In speech signal through several intermediate levels which are linguistic message
and paralinguistic information including emotion is effectively visualized based on the
spectrogram. Now we can concludes the variations in the noisy speech signal of Spectrogram
represented in different areas those are trains, cars, and airport.
Table1: comparison of weighted average technique and proposed SR technique using LLR
and segmental SNR methods
(i)
Type of Noise
LLR Segmental SNR
SNR in
db
Weighted
Average
Technique
Proposed
(SR)
Technique
Weighted
Average
Technique
Proposed
(SR)
Technique
CAR
0 1.687827 1.500914 -6.806391 -6.716270
5 1.842711 1.596159 -5.668619 5.485975
10 1.976017 1.602708 -4.866237 -3.861581
15 1.831509 1.580956 -4.335797 -3.537122
AIRPORT
0 1.237398 1.057377 -3.802483 -3.440414
5 1.124488 0.934859 -2.781458 -2.526855
10 0.919158 0.736983 -0.731036 -0.083965
15 0.910468 0.549913 1.310788 3.080826
TRAIN
0 2.091845 1.798190 -6.486321 -6.296185
5 2.322675 1.845213 -5.559169 -4.970945
10 2.036162 1.759774 -5.251629 -4.206358
15 2.230337 1.827800 -3.211548 4.284449
8. Signal Image Processing : An International Journal (SIPIJ) Vol.5, No.5, October 2014
7
(ii)
(iii)
(iv)
Figure 3: timing wave form and spectrogram of (i) pure speech signal (ii) noisy speech signal and enhanced
signal with (iii) weighted average technique (iv) proposed SR technique in car noise with different SNR
levels.
(i)
9. Signal Image Processing : An International Journal (SIPIJ) Vol.5, No.5, October 2014
8
(ii)
(iii)
(iv)
Figure 4: timing wave form and spectrogram of (i) pure speech signal (ii) noisy speech signal and enhanced
signal with (iii) weighted average technique (iv) proposed SR technique in airport noise with different SNR
levels.
(i)
10. Signal Image Processing : An International Journal (SIPIJ) Vol.5, No.5, October 2014
9
(ii)
(iii)
(iv)
Figure 5: Timing wave form and spectrogram of (i) pure speech signal (ii) noisy speech signal and
enhanced signal with (iii) weighted average technique (iv) proposed SR technique in Train noise with
different SNR levels.
5. CONCLUSION
This paper focused on the issue of noise estimation for enhancement of noisy speech. The noise
estimate was updated continuously in every frame using time–frequency smoothing factors
calculated based on speech-presence probability in each frequency bin of the noisy speech
spectrum [1].The main achievements of speech enhancement algorithms are speech intelligibility
and speech quality. Here to reduce the amplification distortion and attenuation distortion by using
proposed method SR (Signal to Residual spectrum ratio) [5]. The proper control of these
distortions to improve speech intelligibility it is main drawback of speech enhancement
algorithms. The proposed method SR it gives better performance when compared to the previous
existing methods are LLR (log like hood ratio) and segmental SNR.
12. Signal Image Processing : An International Journal (SIPIJ) Vol.5, No.5, October 2014
11
Authors
Ravichandra Kumar Manike pursuing M.Tech in the branch of Digital
Electronics and Communication Systems at Gudlavalleru Engineering College and
B.Tech degree in Electronics and Communication Engineering received from
Prakasam Engineering College in the year of 2011.Gate qualified in the year 2012
13.
Ravi Teja Ballikura received the B.Tech and M.Tech degree in Electronics and
Communication Engineering in 2010 from Bapatla Engineering College, Digital
electronics and communication systems in 2012 from Gudlavalleru Engineering
College affiliated by JNTUK, Kakinada respectively. Working as a assistant
professor in Gudlavalleru Engineering College from 2012 to till date. Research
interests in speech processing and more especially in enhancement of speech signal.