This document summarizes research on designing and implementing different audio restoration techniques for removing distortions like clipping, clicks, and broadband noise from audio signals. It presents methods for declipping audio using sparse representations and frame-based reconstruction. Clicks are addressed using an adaptive filtering method, and broadband noise is reduced via spectral subtraction. The performance of these techniques is evaluated using metrics like SNR and algorithms like OMP. Hardware implementation of click removal is done on a TMS320C6713 DSK board using tools like MATLAB and Code Composer Studio.
A New Method for Pitch Tracking and Voicing Decision Based on Spectral Multi-...CSCJournals
This paper proposes a new voicing detection and pitch estimation method that is particularly robust for noisy speech. This method is based on the spectral analysis of the speech multi-scale product. The multi-scale product (MP) consists of making the product of wavelet transform coefficients. The wavelet used is the quadratic spline function. We argue that the spectral of Multi-scale Product Analysis is capable of revealing an estimate of a pitch-harmonic more accurately even in a heavy noisy scenario. We evaluate our approach on the Keele database. The experimental results show the robustness of our method for noisy speech, and the good performance for clean speech in comparison with state-of-the-art algorithms.
Realization and design of a pilot assist decision making system based on spee...csandit
A system based on speech recognition is proposed fo
r pilot assist decision-making. It is based
on a HIL aircraft simulation platform and uses the
microcontroller SPCE061A as the central
processor to achieve better reliability and higher
cost-effect performance. Technologies of
LPCC (linear predictive cepstral coding) and DTW (D
ynamic Time Warping) are applied for
isolated-word speech recognition to gain a smaller
amount of calculation and a better real-time
performance. Besides, we adopt the PWM (Pulse Width
Modulation) regulation technology to
effectively regulate each control surface by speech
, and thus to assist the pilot to make decisions.
By trial and error, it is proved that we have a sat
isfactory accuracy rate of speech recognition
and control effect. More importantly, our paper pro
vides a creative idea for intelligent human-
computer interaction and applications of speech rec
ognition in the field of aviation control. Our
system is also very easy to be extended and applied
International Journal of Engineering Research and Development (IJERD)IJERD Editor
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
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
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
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
A New Method for Pitch Tracking and Voicing Decision Based on Spectral Multi-...CSCJournals
This paper proposes a new voicing detection and pitch estimation method that is particularly robust for noisy speech. This method is based on the spectral analysis of the speech multi-scale product. The multi-scale product (MP) consists of making the product of wavelet transform coefficients. The wavelet used is the quadratic spline function. We argue that the spectral of Multi-scale Product Analysis is capable of revealing an estimate of a pitch-harmonic more accurately even in a heavy noisy scenario. We evaluate our approach on the Keele database. The experimental results show the robustness of our method for noisy speech, and the good performance for clean speech in comparison with state-of-the-art algorithms.
Realization and design of a pilot assist decision making system based on spee...csandit
A system based on speech recognition is proposed fo
r pilot assist decision-making. It is based
on a HIL aircraft simulation platform and uses the
microcontroller SPCE061A as the central
processor to achieve better reliability and higher
cost-effect performance. Technologies of
LPCC (linear predictive cepstral coding) and DTW (D
ynamic Time Warping) are applied for
isolated-word speech recognition to gain a smaller
amount of calculation and a better real-time
performance. Besides, we adopt the PWM (Pulse Width
Modulation) regulation technology to
effectively regulate each control surface by speech
, and thus to assist the pilot to make decisions.
By trial and error, it is proved that we have a sat
isfactory accuracy rate of speech recognition
and control effect. More importantly, our paper pro
vides a creative idea for intelligent human-
computer interaction and applications of speech rec
ognition in the field of aviation control. Our
system is also very easy to be extended and applied
International Journal of Engineering Research and Development (IJERD)IJERD Editor
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
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
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
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
Performance Analysis of Acoustic Echo Cancellation TechniquesIJERA Editor
Mainly, the adaptive filters are implemented in time domain which works efficiently in most of the applications. But in many applications the impulse response becomes too large, which increases the complexity of the adaptive filter beyond a level where it can no longer be implemented efficiently in time domain. An example of where this can happen would be acoustic echo cancellation (AEC) applications. So, there exists an alternative solution i.e. to implement the filters in frequency domain. AEC has so many applications in wide variety of problems in industrial operations, manufacturing and consumer products. Here in this paper, a comparative analysis of different acoustic echo cancellation techniques i.e. Frequency domain adaptive filter (FDAF), Least mean square (LMS), Normalized least mean square (NLMS) &Sign error (SE) is presented. The results are compared with different values of step sizes and the performance of these techniques is measured in terms of Error rate loss enhancement (ERLE), Mean square error (MSE)& Peak signal to noise ratio (PSNR).
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.
Speech Recognition Systems(SRS) have been implemented by various processors including the digital signal processors(DSPs) and field programmable gate arrays(FPGAs) and their performance has been reported in literature. The fundamental purpose of speech is communication, i.e., the transmission of messages.In the case of speech, the fundamental analog form of the message is an acoustic waveform, which we call the speech signal. Speech signals can be converted to an electrical waveform by a microphone, further manipulated by both analog and digital signal processing, and then converted back to acoustic form by a loudspeaker, a telephone handset or headphone, as desired.The recognition of speech requires feature extraction and classification. The systems that use speech as input require a microcontroller to carry out the desired actions. In this paper, Cypress Programmable System on Chip (PSoC) has been studied and used for implementation of SRS. From all the available PSoCs, PSoC5 containing ARM Cortex-M3 as its CPU is used. The noise signals are firstly nullified from the speech signals using LogMMSE filtering. These signals are then sent to the PSoC5 wherein the speech is recognized and desired actions are performed.
Performance Analysis and Simulation of Decimator for Multirate ApplicationsIJEEE
In this paper, a decimator design has been presented for multirate digital signal processing. The decimator design has been analysed and simulated for performance comparison in terms of filter order and ripple factor. Direct form-I with decimation factor 2 have been used for performance and ripple analysis. The decimators have been designed & simulated using MATLAB. It can be observed from the simulated results that as we increase the filter order, ripple factor decreases, for the same filter structure. On the other hand, increasing filter order will increase its area and implementation cost.
Speaker Recognition System using MFCC and Vector Quantization Approachijsrd.com
This paper presents an approach to speaker recognition using frequency spectral information with Mel frequency for the improvement of speech feature representation in a Vector Quantization codebook based recognition approach. The Mel frequency approach extracts the features of the speech signal to get the training and testing vectors. The VQ Codebook approach uses training vectors to form clusters and recognize accurately with the help of LBG algorithm.
FIR Filter Implementation by Systolization using DA-based DecompositionIDES Editor
In this paper we present 1D and 2D systolic
Distributed Arithmetic (DA) based structures that are designed
for the implementation of Finite Impulse Response (FIR) filters.
The paper compares the 1D DA based systolic structure with
1D systolic DA based decomposition method. The filters are
implemented on a Xilinx Virtex II Pro (XC2VP30) FPGA using
HDL and system metrics like Area, Gate Count, Maximum
Usable Frequency and Power consumption are estimated for
different filter orders and address lengths. The 1D systolic
decomposition structure is also compared with the existing
system generator implementation of DA FIR.. Results for an
exemplary implementation are presented.
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
Simulation Study of FIR Filter based on MATLABijsrd.com
First, the rapid design of FIR digital filter was completed by using the Signal Processing Toolbox FDA Tool, the case filter design of a composite signal by filtering, to prove that the content filter designed for filtering. MATLAB and Simulink programs of the filter were used to verify the performance of the filter in MATLAB. Experimental results show that the low-pass filter filters the high frequency component of input signals mixed. Comparison of two types of simulation, the latter method was more convenient quickly, and reduces the workload.
Performance enhancement of dct based speaker recognition using wavelet de noi...eSAT Journals
Abstract Presence of noise in the speech signal is one of the major problems in Speaker Recognition. The speaker recognition performance gradually degrades as the intensity of noise increases. The system gives high accuracy when the speech signal is noise free, but in real life scenario getting a noise free speech signal is challenging. Hence, elimination of the noise from speech signal is an important aspect in speaker recognition process. This work uses wavelet based denoising of the recorded speech signal in order to enhance the performance of speaker recognition. In this paper, wavelet based denoising technique has been applied to the DCT based speaker recognition system which was proposed in our previous work. Additive white Gaussian noise has been added to the speech signal and performance analysis of the system has been done using different SNR value. Keywords: Wavelet denoising, AWGN, Speaker Recognition, Thresholding, DCT, Feature Extraction.
A Novel Approach of Area-Efficient FIR Filter Design Using Distributed Arithm...IOSR Journals
Abstract: In this paper, a highly area-efficient multiplier-less FIR filter is presented. Distributed Arithmetic (DA) has been used to implement a bit-serial scheme of a general asymmetric version of an FIR filter, taking optimal advantage of the 3-input LUT-based structure of FPGAs. The implementation of FIR filters on FPGA based on traditional arithmetic method costs considerable hardware resources, which goes against the decrease of circuit scale and the increase of system speed. This paper presents the realization of area efficient architectures using Distributed Arithmetic (DA) for implementation of Finite Impulse Response (FIR) filter. The performance of the bit-serial and bit parallel DA along with pipelining architecture with different quantized versions are analyzed for FIR filter Design. Distributed Arithmetic structure is used to increase the resource usage while pipeline structure is also used to increase the system speed. In addition, the divided LUT method is also used to decrease the required memory units. However, according to Distributed Arithmetic, we can make a Look-Up-Table (LUT) to conserve the MAC values and callout the values according to the input data if necessary. Therefore, LUT can be created to take the place of MAC units so as to save the hardware resources. The simulation results indicate that FIR filters using Distributed Arithmetic can work stable with high speed and can save almost 50 percent hardware resources to decrease the circuit scale, and can be applied to a variety of areas for its great flexibility and high reliability. This method not only reduces the LUT size, but also modifies the structure of the filter to achieve high speed performance. Keywords: DSP, Digital Filters, FIR , FPGA, MAC, Distributed Arithmetic(DA),Divided LUT, pipeline
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.
Performance Analysis of Acoustic Echo Cancellation TechniquesIJERA Editor
Mainly, the adaptive filters are implemented in time domain which works efficiently in most of the applications. But in many applications the impulse response becomes too large, which increases the complexity of the adaptive filter beyond a level where it can no longer be implemented efficiently in time domain. An example of where this can happen would be acoustic echo cancellation (AEC) applications. So, there exists an alternative solution i.e. to implement the filters in frequency domain. AEC has so many applications in wide variety of problems in industrial operations, manufacturing and consumer products. Here in this paper, a comparative analysis of different acoustic echo cancellation techniques i.e. Frequency domain adaptive filter (FDAF), Least mean square (LMS), Normalized least mean square (NLMS) &Sign error (SE) is presented. The results are compared with different values of step sizes and the performance of these techniques is measured in terms of Error rate loss enhancement (ERLE), Mean square error (MSE)& Peak signal to noise ratio (PSNR).
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.
Speech Recognition Systems(SRS) have been implemented by various processors including the digital signal processors(DSPs) and field programmable gate arrays(FPGAs) and their performance has been reported in literature. The fundamental purpose of speech is communication, i.e., the transmission of messages.In the case of speech, the fundamental analog form of the message is an acoustic waveform, which we call the speech signal. Speech signals can be converted to an electrical waveform by a microphone, further manipulated by both analog and digital signal processing, and then converted back to acoustic form by a loudspeaker, a telephone handset or headphone, as desired.The recognition of speech requires feature extraction and classification. The systems that use speech as input require a microcontroller to carry out the desired actions. In this paper, Cypress Programmable System on Chip (PSoC) has been studied and used for implementation of SRS. From all the available PSoCs, PSoC5 containing ARM Cortex-M3 as its CPU is used. The noise signals are firstly nullified from the speech signals using LogMMSE filtering. These signals are then sent to the PSoC5 wherein the speech is recognized and desired actions are performed.
Performance Analysis and Simulation of Decimator for Multirate ApplicationsIJEEE
In this paper, a decimator design has been presented for multirate digital signal processing. The decimator design has been analysed and simulated for performance comparison in terms of filter order and ripple factor. Direct form-I with decimation factor 2 have been used for performance and ripple analysis. The decimators have been designed & simulated using MATLAB. It can be observed from the simulated results that as we increase the filter order, ripple factor decreases, for the same filter structure. On the other hand, increasing filter order will increase its area and implementation cost.
Speaker Recognition System using MFCC and Vector Quantization Approachijsrd.com
This paper presents an approach to speaker recognition using frequency spectral information with Mel frequency for the improvement of speech feature representation in a Vector Quantization codebook based recognition approach. The Mel frequency approach extracts the features of the speech signal to get the training and testing vectors. The VQ Codebook approach uses training vectors to form clusters and recognize accurately with the help of LBG algorithm.
FIR Filter Implementation by Systolization using DA-based DecompositionIDES Editor
In this paper we present 1D and 2D systolic
Distributed Arithmetic (DA) based structures that are designed
for the implementation of Finite Impulse Response (FIR) filters.
The paper compares the 1D DA based systolic structure with
1D systolic DA based decomposition method. The filters are
implemented on a Xilinx Virtex II Pro (XC2VP30) FPGA using
HDL and system metrics like Area, Gate Count, Maximum
Usable Frequency and Power consumption are estimated for
different filter orders and address lengths. The 1D systolic
decomposition structure is also compared with the existing
system generator implementation of DA FIR.. Results for an
exemplary implementation are presented.
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
Simulation Study of FIR Filter based on MATLABijsrd.com
First, the rapid design of FIR digital filter was completed by using the Signal Processing Toolbox FDA Tool, the case filter design of a composite signal by filtering, to prove that the content filter designed for filtering. MATLAB and Simulink programs of the filter were used to verify the performance of the filter in MATLAB. Experimental results show that the low-pass filter filters the high frequency component of input signals mixed. Comparison of two types of simulation, the latter method was more convenient quickly, and reduces the workload.
Performance enhancement of dct based speaker recognition using wavelet de noi...eSAT Journals
Abstract Presence of noise in the speech signal is one of the major problems in Speaker Recognition. The speaker recognition performance gradually degrades as the intensity of noise increases. The system gives high accuracy when the speech signal is noise free, but in real life scenario getting a noise free speech signal is challenging. Hence, elimination of the noise from speech signal is an important aspect in speaker recognition process. This work uses wavelet based denoising of the recorded speech signal in order to enhance the performance of speaker recognition. In this paper, wavelet based denoising technique has been applied to the DCT based speaker recognition system which was proposed in our previous work. Additive white Gaussian noise has been added to the speech signal and performance analysis of the system has been done using different SNR value. Keywords: Wavelet denoising, AWGN, Speaker Recognition, Thresholding, DCT, Feature Extraction.
A Novel Approach of Area-Efficient FIR Filter Design Using Distributed Arithm...IOSR Journals
Abstract: In this paper, a highly area-efficient multiplier-less FIR filter is presented. Distributed Arithmetic (DA) has been used to implement a bit-serial scheme of a general asymmetric version of an FIR filter, taking optimal advantage of the 3-input LUT-based structure of FPGAs. The implementation of FIR filters on FPGA based on traditional arithmetic method costs considerable hardware resources, which goes against the decrease of circuit scale and the increase of system speed. This paper presents the realization of area efficient architectures using Distributed Arithmetic (DA) for implementation of Finite Impulse Response (FIR) filter. The performance of the bit-serial and bit parallel DA along with pipelining architecture with different quantized versions are analyzed for FIR filter Design. Distributed Arithmetic structure is used to increase the resource usage while pipeline structure is also used to increase the system speed. In addition, the divided LUT method is also used to decrease the required memory units. However, according to Distributed Arithmetic, we can make a Look-Up-Table (LUT) to conserve the MAC values and callout the values according to the input data if necessary. Therefore, LUT can be created to take the place of MAC units so as to save the hardware resources. The simulation results indicate that FIR filters using Distributed Arithmetic can work stable with high speed and can save almost 50 percent hardware resources to decrease the circuit scale, and can be applied to a variety of areas for its great flexibility and high reliability. This method not only reduces the LUT size, but also modifies the structure of the filter to achieve high speed performance. Keywords: DSP, Digital Filters, FIR , FPGA, MAC, Distributed Arithmetic(DA),Divided LUT, pipeline
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.
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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.
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.
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.
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.
Isolated word recognition using lpc & vector quantizationeSAT Journals
Abstract Speech recognition is always looked upon as a fascinating field in human computer interaction. It is one of the fundamental steps towards understanding human recognition and their behavior. This paper explicates the theory and implementation of Speech recognition. This is a speaker-dependent real time isolated word recognizer. The major logic used was to first obtain the feature vectors using LPC which was followed by vector quantization. The quantized vectors were then recognized by measuring the Minimum average distortion. All Speech Recognition systems contain Two Main Phases, namely Training Phase and Testing Phase. In the Training Phase, the Features of the words are extracted and during the recognition phase feature matching Takes place. The feature or the template thus extracted is stored in the data base, during the recognition phase the extracted features are compared with the template in the database. The features of the words are extracted by using LPC analysis. Vector Quantization is used for generating the code books. Finally the recognition decision is made based on the matching score. MATLAB will be used to implement this concept to achieve further understanding. Index Terms: Speech Recognition, LPC, Vector Quantization, and Code Book.
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
Speech Recognition Systems(SRS) have been implemented by various processors including the digital signal processors(DSPs) and field programmable gate arrays(FPGAs) and their performance has been reported in literature. The fundamental purpose of speech is communication, i.e., the transmission of messages.In the case of speech, the fundamental analog form of the message is an acoustic waveform, which we call the speech signal. Speech signals can be converted to an electrical waveform by a microphone, further manipulated by both analog and digital signal processing, and then converted back to acoustic form by a loudspeaker, a telephone handset or headphone, as desired.The recognition of speech requires feature extraction and classification. The systems that use speech as input require a microcontroller to carry out the desired actions. In this paper, Cypress Programmable System on Chip (PSoC) has been studied and used for implementation of SRS. From all the available PSoCs, PSoC5 containing ARM Cortex-M3 as its CPU is used. The noise signals are firstly nullified from the speech signals using LogMMSE filtering. These signals are then sent to the PSoC5 wherein the speech is recognized and desired actions are performed.
Design and Performance Analysis of Convolutional Encoder and Viterbi Decoder ...IJERA Editor
In digital communication forward error correction methods have a great practical importance when channel is
noisy. Convolutional error correction code can correct both type of errors random and burst. Convolution
encoding has been used in digital communication systems including deep space communication and wireless
communication. The error correction capability of convolutional code depends on code rate and constraint
length. The low code rate and high constraint length has more error correction capabilities but that also
introduce large overhead. This paper introduces convolutional encoders for various constraint lengths. By
increasing the constraint length the error correction capability can be increased. The performance and error
correction also depends on the selection of generator polynomial. This paper also introduces a good generator
polynomial which has high performance and error correction capabilities.
DESIGN REALIZATION AND PERFORMANCE EVALUATION OF AN ACOUSTIC ECHO CANCELLATIO...sipij
Nowadays, in the field of communications, AEC (acoustic echo cancellation) is truly essential with respect
to the quality of multimedia transmission. In this paper, we designed and developed an efficient AEC based
on adaptive filters to improve quality of service in telecommunications against the phenomena of acoustic
echo, which is indeed a problem in hands-free communications.The main advantage of the proposed algorithm is its capacity of tracking non-stationary signals such as acoustic echo. In this work the acoustic echo cancellation (AEC) is modeled using a digital signal
processing technique especially Simulink Blocksets. The algorithm’s code is generated in Matlab Simulink
programming environment. At simulation level, results of simulink implementation prove that module
behavior is realistic when it comes to cancellation of echo in hands free communication using adaptive algorithm.Results obtained with our algorithm in terms of ERLE criteria are confronted to IUT-T recommendation
G.168.
Improved performance of scs based spectrum sensing in cognitive radio using d...eSAT Journals
Abstract
Tremendous growth in current wireless networks raises the demand of more frequency spectrum, over the finite availability of spectrum resource. Although, the research has specifies that the available primary users (i.e. licensed user) has not occupying the channel all the time. The most effective technology known as Cognitive radio giving promises for a solution of under utilization of available frequency spectrum in wireless communication. In cognitive radio network two types of wireless user can be define as primary user and secondary user. Primary users have highest priority to utilize the available band of frequency and secondary user can utilize these services only when the channel is vacant by primary user and there will be no any interference. The optimization of this may be implemented by a smart technique such as cognitive radio, which is fully automated intelligent wireless sensor tool having capability to sense, learn & adjust relevant operating parameters dynamically in radio atmosphere. This can be happen if we prefer the appropriate window technique to evaluate system parameter for sensing the availability of vacant band. We show that by comparing the different windows techniques, cognitive radios not only provide better spectrum opportunity but also provide the chance to huge number of wireless users.
Keywords: Primary user, Secondary user, Spectrum Sensing and Window technique etc.
Adaptive Digital Filter Design for Linear Noise Cancellation Using Neural Net...iosrjce
Noise is the most serious issue in the filters and adaptive filters are subjected to this unwanted
component. This paper deals with the problem of the adaptive noise and various adaptive algorithms functions
which when implemented practically shows that the noise is cancelled or removed by the neural network
approach using the exact random basis function. The adaptive filters are used to control the noise and it has a
linear input and output characteristics. This approach is done so as to get the minimum possible error so that to
obtain the error free desired signal. The designed filter will reduce this noise from measured signal by a
reference signal which is highly correlated with the noise signal. This approach gives excellent result for this
signal processing technique that removes or eliminates the linear noise from the different functions. The
simulation results are also mentioned so as to gives a vivid idea of reduced noise using neural networks
algorithm.
Mechanical properties of hybrid fiber reinforced concrete for pavementseSAT Journals
Abstract
The effect of addition of mono fibers and hybrid fibers on the mechanical properties of concrete mixture is studied in the present
investigation. Steel fibers of 1% and polypropylene fibers 0.036% were added individually to the concrete mixture as mono fibers and
then they were added together to form a hybrid fiber reinforced concrete. Mechanical properties such as compressive, split tensile and
flexural strength were determined. The results show that hybrid fibers improve the compressive strength marginally as compared to
mono fibers. Whereas, hybridization improves split tensile strength and flexural strength noticeably.
Keywords:-Hybridization, mono fibers, steel fiber, polypropylene fiber, Improvement in mechanical properties.
Material management in construction – a case studyeSAT Journals
Abstract
The objective of the present study is to understand about all the problems occurring in the company because of improper application
of material management. In construction project operation, often there is a project cost variance in terms of the material, equipments,
manpower, subcontractor, overhead cost, and general condition. Material is the main component in construction projects. Therefore,
if the material management is not properly managed it will create a project cost variance. Project cost can be controlled by taking
corrective actions towards the cost variance. Therefore a methodology is used to diagnose and evaluate the procurement process
involved in material management and launch a continuous improvement was developed and applied. A thorough study was carried
out along with study of cases, surveys and interviews to professionals involved in this area. As a result, a methodology for diagnosis
and improvement was proposed and tested in selected projects. The results obtained show that the main problem of procurement is
related to schedule delays and lack of specified quality for the project. To prevent this situation it is often necessary to dedicate
important resources like money, personnel, time, etc. To monitor and control the process. A great potential for improvement was
detected if state of the art technologies such as, electronic mail, electronic data interchange (EDI), and analysis were applied to the
procurement process. These helped to eliminate the root causes for many types of problems that were detected.
Managing drought short term strategies in semi arid regions a case studyeSAT Journals
Abstract
Drought management needs multidisciplinary action. Interdisciplinary efforts among the experts in various fields of the droughts
prone areas are helpful to achieve tangible and permanent solution for this recurring problem. The Gulbarga district having the total
area around 16, 240 sq.km, and accounts 8.45 per cent of the Karnataka state area. The district has been situated with latitude 17º 19'
60" North and longitude of 76 º 49' 60" east. The district is situated entirely on the Deccan plateau positioned at a height of 300 to
750 m above MSL. Sub-tropical, semi-arid type is one among the drought prone districts of Karnataka State. The drought
management is very important for a district like Gulbarga. In this paper various short term strategies are discussed to mitigate the
drought condition in the district.
Keywords: Drought, South-West monsoon, Semi-Arid, Rainfall, Strategies etc.
Life cycle cost analysis of overlay for an urban road in bangaloreeSAT Journals
Abstract
Pavements are subjected to severe condition of stresses and weathering effects from the day they are constructed and opened to traffic
mainly due to its fatigue behavior and environmental effects. Therefore, pavement rehabilitation is one of the most important
components of entire road systems. This paper highlights the design of concrete pavement with added mono fibers like polypropylene,
steel and hybrid fibres for a widened portion of existing concrete pavement and various overlay alternatives for an existing
bituminous pavement in an urban road in Bangalore. Along with this, Life cycle cost analyses at these sections are done by Net
Present Value (NPV) method to identify the most feasible option. The results show that though the initial cost of construction of
concrete overlay is high, over a period of time it prove to be better than the bituminous overlay considering the whole life cycle cost.
The economic analysis also indicates that, out of the three fibre options, hybrid reinforced concrete would be economical without
compromising the performance of the pavement.
Keywords: - Fatigue, Life cycle cost analysis, Net Present Value method, Overlay, Rehabilitation
Laboratory studies of dense bituminous mixes ii with reclaimed asphalt materialseSAT Journals
Abstract
The issue of growing demand on our nation’s roadways over that past couple of decades, decreasing budgetary funds, and the need to
provide a safe, efficient, and cost effective roadway system has led to a dramatic increase in the need to rehabilitate our existing
pavements and the issue of building sustainable road infrastructure in India. With these emergency of the mentioned needs and this
are today’s burning issue and has become the purpose of the study.
In the present study, the samples of existing bituminous layer materials were collected from NH-48(Devahalli to Hassan) site.The
mixtures were designed by Marshall Method as per Asphalt institute (MS-II) at 20% and 30% Reclaimed Asphalt Pavement (RAP).
RAP material was blended with virgin aggregate such that all specimens tested for the, Dense Bituminous Macadam-II (DBM-II)
gradation as per Ministry of Roads, Transport, and Highways (MoRT&H) and cost analysis were carried out to know the economics.
Laboratory results and analysis showed the use of recycled materials showed significant variability in Marshall Stability, and the
variability increased with the increase in RAP content. The saving can be realized from utilization of recycled materials as per the
methodology, the reduction in the total cost is 19%, 30%, comparing with the virgin mixes.
Keywords: Reclaimed Asphalt Pavement, Marshall Stability, MS-II, Dense Bituminous Macadam-II
Laboratory investigation of expansive soil stabilized with natural inorganic ...eSAT Journals
Abstract
Soil stabilization has proven to be one of the oldest techniques to improve the soil properties. Literature review conducted revealed
that uses of natural inorganic stabilizers are found to be one of the best options for soil stabilization. In this regard an attempt has
been made to evaluate the influence of RBI-81 stabilizer on properties of black cotton soil through laboratory investigations. Black
cotton soil with varying percentages of RBI-81 viz., 0, 0.5, 1, 1.5, 2, and 2.5 percent were studied for moisture density relationships
and strength behaviour of soils. Also the effect of curing period was evaluated as literature review clearly emphasized the strength
gain of soils stabilized with RBI-81 over a period of time. The results obtained shows that the unconfined compressive strength of
specimens treated with RBI-81 increased approximately by 250% for a curing period of 28 days as compared to virgin soil. Further
the CBR value improved approximately by 400%. The studies indicated an increasing trend for soil strength behaviour with
increasing percentage of RBI-81 suggesting its potential applications in soil stabilization.
Influence of reinforcement on the behavior of hollow concrete block masonry p...eSAT Journals
Abstract
Reinforced masonry was developed to exploit the strength potential of masonry and to solve its lack of tensile strength. Experimental
and analytical studies have been carried out to investigate the effect of reinforcement on the behavior of hollow concrete block
masonry prisms under compression and to predict ultimate failure compressive strength. In the numerical program, three dimensional
non-linear finite elements (FE) model based on the micro-modeling approach is developed for both unreinforced and reinforced
masonry prisms using ANSYS (14.5). The proposed FE model uses multi-linear stress-strain relationships to model the non-linear
behavior of hollow concrete block, mortar, and grout. Willam-Warnke’s five parameter failure theory has been adopted to model the
failure of masonry materials. The comparison of the numerical and experimental results indicates that the FE models can successfully
capture the highly nonlinear behavior of the physical specimens and accurately predict their strength and failure mechanisms.
Keywords: Structural masonry, Hollow concrete block prism, grout, Compression failure, Finite element method,
Numerical modeling.
Influence of compaction energy on soil stabilized with chemical stabilizereSAT Journals
Abstract
Increase in traffic along with heavier magnitude of wheel loads cause rapid deterioration in pavements. There is a need to improve
density, strength of soil subgrade and other pavement layers. In this study an attempt is made to improve the properties of locally
available loamy soil using twin approaches viz., i) increasing the compaction of soil and ii) treating the soil with chemical stabilizer.
Laboratory studies are carried out on both untreated and treated soil samples compacted by different compaction efforts. Studies
show that increase in compaction effort results in increase in density of soil. However in soil treated with chemical stabilizer, rate of
increase in density is not significant. The soil treated with chemical stabilizer exhibits improvement in both strength and performance
properties.
Keywords: compaction, density, subgradestabilization, resilient modulus
Geographical information system (gis) for water resources managementeSAT Journals
Abstract
Water resources projects are inherited with overlapping and at times conflicting objectives. These projects are often of varied sizes
ranging from major projects with command areas of millions of hectares to very small projects implemented at the local level. Thus,
in all these projects there is seldom proper coordination which is essential for ensuring collective sustainability.
Integrated watershed development and management is the accepted answer but in turn requires a comprehensive framework that can
enable planning process involving all the stakeholders at different levels and scales is compulsory. Such a unified hydrological
framework is essential to evaluate the cause and effect of all the proposed actions within the drainage basins.
The present paper describes a hydrological framework developed in the form of a Hydrologic Information System (HIS) which is
intended to meet the specific information needs of the various line departments of a typical State connected with water related aspects.
The HIS consist of a hydrologic information database coupled with tools for collating primary and secondary data and tools for
analyzing and visualizing the data and information. The HIS also incorporates hydrological model base for indirect assessment of
various entities of water balance in space and time. The framework would be maintained and updated to reflect fully the most
accurate ground truth data and the infrastructure requirements for planning and management.
Keywords: Hydrological Information System (HIS); WebGIS; Data Model; Web Mapping Services
Forest type mapping of bidar forest division, karnataka using geoinformatics ...eSAT Journals
Abstract
The study demonstrate the potentiality of satellite remote sensing technique for the generation of baseline information on forest types
including tree plantation details in Bidar forest division, Karnataka covering an area of 5814.60Sq.Kms. The Total Area of Bidar
forest division is 5814Sq.Kms analysis of the satellite data in the study area reveals that about 84% of the total area is Covered by
crop land, 1.778% of the area is covered by dry deciduous forest, 1.38 % of mixed plantation, which is very threatening to the
environmental stability of the forest, future plantation site has been mapped. With the use of latest Geo-informatics technology proper
and exact condition of the trees can be observed and necessary precautions can be taken for future plantation works in an appropriate
manner
Keywords:-RS, GIS, GPS, Forest Type, Tree Plantation
Factors influencing compressive strength of geopolymer concreteeSAT Journals
Abstract
To study effects of several factors on the properties of fly ash based geopolymer concrete on the compressive strength and also the
cost comparison with the normal concrete. The test variables were molarities of sodium hydroxide(NaOH) 8M,14M and 16M, ratio of
NaOH to sodium silicate (Na2SiO3) 1, 1.5, 2 and 2.5, alkaline liquid to fly ash ratio 0.35 and 0.40 and replacement of water in
Na2SiO3 solution by 10%, 20% and 30% were used in the present study. The test results indicated that the highest compressive
strength 54 MPa was observed for 16M of NaOH, ratio of NaOH to Na2SiO3 2.5 and alkaline liquid to fly ash ratio of 0.35. Lowest
compressive strength of 27 MPa was observed for 8M of NaOH, ratio of NaOH to Na2SiO3 is 1 and alkaline liquid to fly ash ratio of
0.40. Alkaline liquid to fly ash ratio of 0.35, water replacement of 10% and 30% for 8 and 16 molarity of NaOH and has resulted in
compressive strength of 36 MPa and 20 MPa respectively. Superplasticiser dosage of 2 % by weight of fly ash has given higher
strength in all cases.
Keywords: compressive strength, alkaline liquid, fly ash
Experimental investigation on circular hollow steel columns in filled with li...eSAT Journals
Abstract
Composite Circular hollow Steel tubes with and without GFRP infill for three different grades of Light weight concrete are tested for
ultimate load capacity and axial shortening , under Cyclic loading. Steel tubes are compared for different lengths, cross sections and
thickness. Specimens were tested separately after adopting Taguchi’s L9 (Latin Squares) Orthogonal array in order to save the initial
experimental cost on number of specimens and experimental duration. Analysis was carried out using ANN (Artificial Neural
Network) technique with the assistance of Mini Tab- a statistical soft tool. Comparison for predicted, experimental & ANN output is
obtained from linear regression plots. From this research study, it can be concluded that *Cross sectional area of steel tube has most
significant effect on ultimate load carrying capacity, *as length of steel tube increased- load carrying capacity decreased & *ANN
modeling predicted acceptable results. Thus ANN tool can be utilized for predicting ultimate load carrying capacity for composite
columns.
Keywords: Light weight concrete, GFRP, Artificial Neural Network, Linear Regression, Back propagation, orthogonal
Array, Latin Squares
Experimental behavior of circular hsscfrc filled steel tubular columns under ...eSAT Journals
Abstract
This paper presents an outlook on experimental behavior and a comparison with predicted formula on the behaviour of circular
concentrically loaded self-consolidating fibre reinforced concrete filled steel tube columns (HSSCFRC). Forty-five specimens were
tested. The main parameters varied in the tests are: (1) percentage of fiber (2) tube diameter or width to wall thickness ratio (D/t
from 15 to 25) (3) L/d ratio from 2.97 to 7.04 the results from these predictions were compared with the experimental data. The
experimental results) were also validated in this study.
Keywords: Self-compacting concrete; Concrete-filled steel tube; axial load behavior; Ultimate capacity.
Evaluation of punching shear in flat slabseSAT Journals
Abstract
Flat-slab construction has been widely used in construction today because of many advantages that it offers. The basic philosophy in
the design of flat slab is to consider only gravity forces; this method ignores the effect of punching shear due to unbalanced moments
at the slab column junction which is critical. An attempt has been made to generate generalized design sheets which accounts both
punching shear due to gravity loads and unbalanced moments for cases (a) interior column; (b) edge column (bending perpendicular
to shorter edge); (c) edge column (bending parallel to shorter edge); (d) corner column. These design sheets are prepared as per
codal provisions of IS 456-2000. These design sheets will be helpful in calculating the shear reinforcement to be provided at the
critical section which is ignored in many design offices. Apart from its usefulness in evaluating punching shear and the necessary
shear reinforcement, the design sheets developed will enable the designer to fix the depth of flat slab during the initial phase of the
design.
Keywords: Flat slabs, punching shear, unbalanced moment.
Evaluation of performance of intake tower dam for recent earthquake in indiaeSAT Journals
Abstract
Intake towers are typically tall, hollow, reinforced concrete structures and form entrance to reservoir outlet works. A parametric
study on dynamic behavior of circular cylindrical towers can be carried out to study the effect of depth of submergence, wall thickness
and slenderness ratio, and also effect on tower considering dynamic analysis for time history function of different soil condition and
by Goyal and Chopra accounting interaction effects of added hydrodynamic mass of surrounding and inside water in intake tower of
dam
Key words: Hydrodynamic mass, Depth of submergence, Reservoir, Time history analysis,
Evaluation of operational efficiency of urban road network using travel time ...eSAT Journals
Abstract
Efficiency of the road network system is analyzed by travel time reliability measures. The study overlooks on an important measure of
travel time reliability and prioritizing Tiruchirappalli road network. Traffic volume and travel time were collected using license plate
matching method. Travel time measures were estimated from average travel time and 95th travel time. Effect of non-motorized vehicle
on efficiency of road system was evaluated. Relation between buffer time index and traffic volume was created. Travel time model has
been developed and travel time measure was validated. Then service quality of road sections in network were graded based on
travel time reliability measures.
Keywords: Buffer Time Index (BTI); Average Travel Time (ATT); Travel Time Reliability (TTR); Buffer Time (BT).
Estimation of surface runoff in nallur amanikere watershed using scs cn methodeSAT Journals
Abstract
The development of watershed aims at productive utilization of all the available natural resources in the entire area extending from
ridge line to stream outlet. The per capita availability of land for cultivation has been decreasing over the years. Therefore, water and
the related land resources must be developed, utilized and managed in an integrated and comprehensive manner. Remote sensing and
GIS techniques are being increasingly used for planning, management and development of natural resources. The study area, Nallur
Amanikere watershed geographically lies between 110 38’ and 110 52’ N latitude and 760 30’ and 760 50’ E longitude with an area of
415.68 Sq. km. The thematic layers such as land use/land cover and soil maps were derived from remotely sensed data and overlayed
through ArcGIS software to assign the curve number on polygon wise. The daily rainfall data of six rain gauge stations in and around
the watershed (2001-2011) was used to estimate the daily runoff from the watershed using Soil Conservation Service - Curve Number
(SCS-CN) method. The runoff estimated from the SCS-CN model was then used to know the variation of runoff potential with different
land use/land cover and with different soil conditions.
Keywords: Watershed, Nallur watershed, Surface runoff, Rainfall-Runoff, SCS-CN, Remote Sensing, GIS.
Estimation of morphometric parameters and runoff using rs & gis techniqueseSAT Journals
Abstract
Land and water are the two vital natural resources, the optimal management of these resources with minimum adverse environmental
impact are essential not only for sustainable development but also for human survival. Satellite remote sensing with geographic
information system has a pragmatic approach to map and generate spatial input layers of predicting response behavior and yield of
watershed. Hence, in the present study an attempt has been made to understand the hydrological process of the catchment at the
watershed level by drawing the inferences from moprhometric analysis and runoff. The study area chosen for the present study is
Yagachi catchment situated in Chickamaglur and Hassan district lies geographically at a longitude 75⁰52’08.77”E and
13⁰10’50.77”N latitude. It covers an area of 559.493 Sq.km. Morphometric analysis is carried out to estimate morphometric
parameters at Micro-watershed to understand the hydrological response of the catchment at the Micro-watershed level. Daily runoff
is estimated using USDA SCS curve number model for a period of 10 years from 2001 to 2010. The rainfall runoff relationship of the
study shows there is a positive correlation.
Keywords: morphometric analysis, runoff, remote sensing and GIS, SCS - method
-
Effect of variation of plastic hinge length on the results of non linear anal...eSAT Journals
Abstract The nonlinear Static procedure also well known as pushover analysis is method where in monotonically increasing loads are applied to the structure till the structure is unable to resist any further load. It is a popular tool for seismic performance evaluation of existing and new structures. In literature lot of research has been carried out on conventional pushover analysis and after knowing deficiency efforts have been made to improve it. But actual test results to verify the analytically obtained pushover results are rarely available. It has been found that some amount of variation is always expected to exist in seismic demand prediction of pushover analysis. Initial study is carried out by considering user defined hinge properties and default hinge length. Attempt is being made to assess the variation of pushover analysis results by considering user defined hinge properties and various hinge length formulations available in literature and results compared with experimentally obtained results based on test carried out on a G+2 storied RCC framed structure. For the present study two geometric models viz bare frame and rigid frame model is considered and it is found that the results of pushover analysis are very sensitive to geometric model and hinge length adopted. Keywords: Pushover analysis, Base shear, Displacement, hinge length, moment curvature analysis
Effect of use of recycled materials on indirect tensile strength of asphalt c...eSAT Journals
Abstract
Depletion of natural resources and aggregate quarries for the road construction is a serious problem to procure materials. Hence
recycling or reuse of material is beneficial. On emphasizing development in sustainable construction in the present era, recycling of
asphalt pavements is one of the effective and proven rehabilitation processes. For the laboratory investigations reclaimed asphalt
pavement (RAP) from NH-4 and crumb rubber modified binder (CRMB-55) was used. Foundry waste was used as a replacement to
conventional filler. Laboratory tests were conducted on asphalt concrete mixes with 30, 40, 50, and 60 percent replacement with RAP.
These test results were compared with conventional mixes and asphalt concrete mixes with complete binder extracted RAP
aggregates. Mix design was carried out by Marshall Method. The Marshall Tests indicated highest stability values for asphalt
concrete (AC) mixes with 60% RAP. The optimum binder content (OBC) decreased with increased in RAP in AC mixes. The Indirect
Tensile Strength (ITS) for AC mixes with RAP also was found to be higher when compared to conventional AC mixes at 300C.
Keywords: Reclaimed asphalt pavement, Foundry waste, Recycling, Marshall Stability, Indirect tensile strength.
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.
NO1 Uk best vashikaran specialist in delhi vashikaran baba near me online vas...Amil Baba Dawood bangali
Contact with Dawood Bhai Just call on +92322-6382012 and we'll help you. We'll solve all your problems within 12 to 24 hours and with 101% guarantee and with astrology systematic. If you want to take any personal or professional advice then also you can call us on +92322-6382012 , ONLINE LOVE PROBLEM & Other all types of Daily Life Problem's.Then CALL or WHATSAPP us on +92322-6382012 and Get all these problems solutions here by Amil Baba DAWOOD BANGALI
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Hierarchical Digital Twin of a Naval Power SystemKerry Sado
A hierarchical digital twin of a Naval DC power system has been developed and experimentally verified. Similar to other state-of-the-art digital twins, this technology creates a digital replica of the physical system executed in real-time or faster, which can modify hardware controls. However, its advantage stems from distributing computational efforts by utilizing a hierarchical structure composed of lower-level digital twin blocks and a higher-level system digital twin. Each digital twin block is associated with a physical subsystem of the hardware and communicates with a singular system digital twin, which creates a system-level response. By extracting information from each level of the hierarchy, power system controls of the hardware were reconfigured autonomously. This hierarchical digital twin development offers several advantages over other digital twins, particularly in the field of naval power systems. The hierarchical structure allows for greater computational efficiency and scalability while the ability to autonomously reconfigure hardware controls offers increased flexibility and responsiveness. The hierarchical decomposition and models utilized were well aligned with the physical twin, as indicated by the maximum deviations between the developed digital twin hierarchy and the hardware.
Immunizing Image Classifiers Against Localized Adversary Attacksgerogepatton
This paper addresses the vulnerability of deep learning models, particularly convolutional neural networks
(CNN)s, to adversarial attacks and presents a proactive training technique designed to counter them. We
introduce a novel volumization algorithm, which transforms 2D images into 3D volumetric representations.
When combined with 3D convolution and deep curriculum learning optimization (CLO), itsignificantly improves
the immunity of models against localized universal attacks by up to 40%. We evaluate our proposed approach
using contemporary CNN architectures and the modified Canadian Institute for Advanced Research (CIFAR-10
and CIFAR-100) and ImageNet Large Scale Visual Recognition Challenge (ILSVRC12) datasets, showcasing
accuracy improvements over previous techniques. The results indicate that the combination of the volumetric
input and curriculum learning holds significant promise for mitigating adversarial attacks without necessitating
adversary training.
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
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Water scarcity is the lack of fresh water resources to meet the standard water demand. There are two type of water scarcity. One is physical. The other is economic water scarcity.
Design and implementation of different audio restoration techniques for audio denoising applications
1. IJRET: International Journal of Research in Engineering and Technology eISSN: 2319-1163 | pISSN: 2321-7308
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Volume: 04 Issue: 10 | Oct -2015, Available @ http://www.ijret.org 78
DESIGN AND IMPLEMENTATION OF DIFFERENT AUDIO
RESTORATION TECHNIQUES FOR AUDIO DENOISING
APPLICATIONS
Merin K Mathai1
, Deepa. J2
1
PG Scholar, Department of Electronics and Communication, College of Engineering Chengannur, Kerala, India
2
Associate Professor, Department of Electronics and Communication, College of Engineering Chengannur, Kerala,
India
Abstract
Audio signals are corrupted with many types of distortions. Major audio distortions are categorized into Globalized and
Localized distortions. Localized distortion includes clipping and clicks where only certain samples are affected and globalized
distortions include broadband noise where complete bandwidth is consumed by noise. Audio restoration is a technique for giving
back the audio signals from these distortions. In this paper, audio restoration techniques for removing clipping, clicks and
broadband noise are put forwarded. Recent approaches to solving audio restoration problem is with respect to sparse
representation algorithms. Clipping distortion is addressed with a Sparse representation framework, it is treated as a reverse
problem, where the distorted samples is estimated from the surrounding undistorted samples, they are embedded in frame based
scheme, and reconstructed by using an overlap add method in conjunction with OMP algorithm and Gabor/DCT dictionary for
modelling audio signals. Broadband denoising is done by using spectral subtraction and Click removal is done by using an
adaptive filter method as the first step. Performance measures are done based on perception, average SNR calculation and
defined parameter variations. This paper also targeting towards the software and hardware implementation of the restoration
methods using TMS320C6713 DSK kit with help of tools mainly MATLAB and Code Composer studio.
Key Words: Audio Distortions, OMP algorithm, Gabor/DCT dictionary, TMS320C6713DSK
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1. INTRODUCTION
Audio signals are subjected to a number of degradation
which is common in audio sources. Generally, these
degradations are classified into Localized degradations and
Global degradations. Localized degradations are affect only
certain group of samples; it includes clicks, crackles,
scratches, breakages and clipping. Global degradations are
which affects all samples of the waveform and include
broadband noise and certain types of non-linear distortion
[2].Audio restoration systems are aimed to reduce or
eliminate these types of audible defects present in audio
signals, introduced by the recording or reproduction
mechanisms, or resulting from deterioration of the recording
medium. Most common types of distortions are Clicks,
Clipping and Broadband noise, among which localized
distortions clipping and clicks are very difficult to resolve.
This paper proposes the methods to dig out the effective
ways for eliminating these degradation from audio signals.
1.1 Clipping
Clipping is a waveform distortion that limits a signal to
reach its peak amplitude. It usually occurs in phone sound
transmissions. It may occur in two forms,
[1]. Hard clipping- In cases where the signal is strictly
limited at the threshold, producing a flat cutoff, Fig-1
shows Hard clipping and normal signal waveforms.
[2]. Soft clipping- In cases where the clipped signal
continues to follow the original at a reduced gain. Hard
clipping normally results in high frequency harmonics;
soft clipping results in fewer higher order harmonics
and intermodulation distortion components. A number
of methods are available to restore the clipped signals
[7], [8], [9]. Among which major are frequency
domain methods. Here we are employing a time
domain approach which provides more efficient
restoration of the clipped samples.
Fig -1: Clipping and normal signal waveforms
1.2 Clicks
Clicks in audio signals, shown in Fig-2; also known as
Impulsive noise, consists of relatively short duration noise
pulses, caused by different types of sources, such as
switching noise, various channel environments in a
communication system, dropouts or surface degradation of
audio recordings like CD scratches, clicks from system
keyboards, etc.
2. IJRET: International Journal of Research in Engineering and Technology eISSN: 2319-1163 | pISSN: 2321-7308
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Volume: 04 Issue: 10 | Oct -2015, Available @ http://www.ijret.org 79
Fig -2: Clicks
1.3 Broadband Noise
Broadband noise is a noisy sound signal which has its
energy distributed over a large section of the audible
frequency range, also known as wideband noise. This noise
spectrum consisting of a large number of frequency
components, none of which is individually dominant.
Fig -3: Broadband noise
Fig-3 shows the broadband noisy signal. This type of noise
is often appeared as surface noise in old historical sound
recordings.
2. PROPOSED METHOD
This project proposes the audio denoising methods for audio
signals degraded by clipping, clicks and broadband noise by
addressing these distortions individually. Clipping is
addressed by SR framework leveraging from the concept of
image inpainting, Clicks are addressed by using the basic
principles of adaptive signal processing and Broadband
noise is alleviated by spectral subtraction. All these methods
are briefly described in this section, the Results and
Discussion, Conclusion are presented in Section 3 and 4.
2.1 Declipping framework
Proposed framework is employed in time domain where the
distorted data frames are estimated from the undistorted data
surrounding by it. Sparse representations (SR) model frame
work is used here to model the reliable and corrupted audio
data co-efficients [1]. Currently, signal processing methods
referred as the Sparse Representations are increasingly
popular for solving underdetermined linear equation
systems. Solving the problem of missing samples in signal
processing using SR is termed as Inpainting.
Audio declipping problem can be analysed by using the SR
modelling of the distorted audio signals. The proposed
algorithm for addressing the declipping is defined below in
steps. Here the audio data frames can be viewed as of vector
of co-efficients like c Є RL
.
Proposed Algorithm for Declipping
Step 1. Segment the distorted audio samples into fixed
number of frames of frame length N using analysis window
wa.
Step 2. Each frame is checked to find clipped and
unclipped data.
Step 3. If the frame is corrupted, OMP algorithm used in
conjunction with DCT/Gabor dictionary is used to
reconstruct the corrupted frame using the reliable data in the
respective frame.
Step 4. Restoration of complete audio signal using
Synthesis window ws by OLA method.
The audio declipping problem is defined as the recovery of
the missing co-efficients based on the details of
1) Unclipped data in each frame
2) Clipping level
3) Observed signal properties such as sampling frequency,
no: of bits, length etc.
In linear model, step 2 can be interpreted as
is the windowed frame defined
for .Step 3 followed by step 2 makes the
estimate ( ) of clipped frame co-efficients using reliable
data co-efficients with the help of an inpainting algorithm,
Here Orthogonal Matching Pursuit (OMP) is used. The final
phase, Step 4 the reconstruction of the full signal which is
mathematically modelled as
(1)
where is the synthesis window such that
,
a rectangular window for and a sine window for is
used in this approach?
In the SR modelling, each frame can approximated as
where is the dictionary,
and is the representation vector of the jth
frame. is assumed to be sparse, i.e., to have few nonzero
coefficients compared to in order to recover unknown
samples.
Dictionary D used here is DCT/DGT dictionary. It is used
by OMP algorithm [4] to perform the inpainting of an audio
frame. General steps for OMP algorithm is described below.
This approach emerges from the following optimization
problem in Equation (2):
s.t. (2)
Inorder to solve Equation (2) , OMP algorithm is used here.
3. IJRET: International Journal of Research in Engineering and Technology eISSN: 2319-1163 | pISSN: 2321-7308
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Volume: 04 Issue: 10 | Oct -2015, Available @ http://www.ijret.org 80
OMP algorithm
Input: ɸ ϵ Rk x N
: The sampling matrix
y ϵ Rk
: The measurements vector
m : The sparsity level of the signal x
ԑ0 : OMP threshold
Output: : The estimate of the orginal signal
Procedure :
1. Intialize the residual r0 = y and the index set α0 = {Ø}.
Set iteration counter , i=1
2. Find the index λi = argmaxj=1,..,N | <ri-1 , ɸj> |
3. Augment the index set αi = αi-1 U {λi} and the matrix of
chosen columns
4. Solve a least square problem for new estimate
5. Compute the new residual
6. Increment counter i and return to step2 if i<m or ||ri
2
||
> ԑ0
Algorithm stops when if i>m. or ||ri
2
|| > ԑ0
7. Retrieve the final estimate .
In general audio data can either be waveform samples or
coefficients in transforms like spectrograms. Here
considering single dimensional signals only, this can also be
extended to address formulation above can be used for
multi-dimensional signals by simply considering the
equivalent vectors.
2.2 Click Removal
In order to remove click from a degraded signal, an adaptive
signal processing scheme is used. Therefore, adaptive filters
can be used for noise cancellation in digital systems, in the
same way, this can be applied to a clicked audio signal.
Adaptive algorithms used for this purpose is LMS algorithm
and NLMS algorithm [5]. Fig-4 shows an adaptive noise
canceller designed for removing clicks
Fig-4: Click removal using ANC
where is the clicked noisy signal and is the
clicked noise variant, the error signal (n) is the desired
output signal. Proper values for step size and filter order are
carefully chosen for algorithm to obtain the desired output.
2.3 Broadband Denoising
Broad band denoising using spectral subtraction exploits the
short time amplitude characteristics of speech and
essentially performs a magnitude or power subtraction of the
noise from the noisy speech. In its basic form it is a simple
method that operates in the frequency domain to obtain a
magnitude spectrum estimate of the noise and then use that
estimate to filter the noisy signal[4]. Due to its popularity,
many different variations of it have been developed. The
degraded signal x(n) is modeled as a pure audio signal s(n)
and a superimposed broadband noise d(n). The signal
degradation is processed in the frequency domain using the
short-time Fourier transform. For each frequency bin k and
each frame m, a specific amount of the noise spectrum
b
is subtracted from the short-time spectrum
(b=1 for magnitude subtraction, and b=2 for
power subtraction). The noise spectrum has to be
estimated from a noise only signal segment. For re-synthesis
the denoised signal spectrum is combined with
the original phase spectrum . The spectral
subtraction method can be modeled as time variant non-
linear filter. Its transfer function depends on the signal-to-
noise ratio which is estimated from the short-
time spectrum and the noise spectrum .
The noise variance within a single frame causes the SNR to
be overestimated for some frequency bins. This results in a
residual noise consisting of short sinusoidal impulses whose
frequencies vary from frame to frame. This phenomenon is
known as musical noise. The noise variance can be reduced
by using a time-averaged signal spectrum instead of
. This reduces the musical noise but without
completely eliminating it. Block diagram of the designed
broad denoiser is shown in Fig-5
Fig-5: Broadband denoiser
Here, speech signals are only considered these are free from
musical noise. Signal broken down into frequency bins, then
processed and reconstructed using proper window. Here
either Hamming/Hanning window can be used for obtaining
the frequency bins, because they provide more frequency
resolution than other windows.
3. RESULTS AND DISCUSSIONS
All the three distortions are addressed and the proposed
restoration methods are implemented using MATLAB.
Click removal is implemented as a part of hardware using
TMS320C6713 DSK. Audio databases are downloaded and
4. IJRET: International Journal of Research in Engineering and Technology eISSN: 2319-1163 | pISSN: 2321-7308
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Volume: 04 Issue: 10 | Oct -2015, Available @ http://www.ijret.org 81
also created using audio softwares. Database consisting of
Speech and music signals with a sampling frequencies of
16 KHz, 8 KHz, and 44.1 KHz are tested and verified the
results by perception tests. Performance evaluation of each
methods has been done and obtained results are shown
below.
3.1 Declipping Results
Table-1 shows the average SNR value of the declipped
estimate obtained as results. SNR value is calculated from
the Equation (3). Datasets used are Speech signal at 8 KHz
and Music signals at 16 KHz of 32ms frames are used.
(3)
Table -1: Average SNR of the restored signal as a function
of clipping level
Clipping
level 0.2 0.4 0.6 0.8 0.9
Music
SNR(db.) 3.05 6.85 10.84 12.94 18.61
Speech
SNR(db.) 2.66 4.17 16.07 24.39 20.57
It is clear that SNR of the declipped estimate is improved so
better from and above the clipping level „0.4 in the case of
both speech and music signals. Gabor dictionary is used for
estimating the co-efficients. It is a fixed dictionary, hence
much improved results can be obtained by using a learning
dictionary. For the much better results are got with Speech
signals , only slight variation is observed in the case of
music signals.
Comparison between DCT and DGT dictionaries are also
performed and its results for a speech signal is shown in
Table-2
Table -2: Comparison of DGT and DCT
Clipping
Level
DGT DCT
0.1 1.0571 0.58740
0.2 4.5882 2.43983
0.3 8.63138 3.97389
0.4 7.85308 4.44919
0.5 15.9986 13.091
0.6 14.9121 13.7017
0.7 17.0139 12.8429
0.8 21.5961 12.3272
0.9 17.5716 12.7401
From the results, DGT is more apt to modelling the audio
signals than DCT since it holds amplitude as well as free
phase information. But at lower clipping levels (<0.5) DCT
equalize and comparable with the performance of DGT.
Performance analysis of declipping is also done by varying
the frame lengths depicted as f1, f2 and f3. Its values are
f1=64, f2=128, f3=256. Chart-1 shows the comparative
change in average SNR value of the restored signal on the
basis of clipping level and frame length. Music and speech
signals is tested. Music signal tested results are shown here.
Chart-1: Comparison of frame lengths
It is clear from graph that at lower clipping levels (<=0.5),
frames of small duration are more apt, here the adapted
frame length from the graph is 128. But for higher clipping
levels (>0.5), larger frames are suitable (adapted frame
length=256), but frame length cannot be increased beyond a
value, it increases the complexity level of the method, thus
in turn increases execution time.
3.2 Broadband Denoising Results
From the plotted results of Fig-6 shows that by broadband
denoising using spectral subtraction gives better results.
This can be more improved by proper window placement
spectral calculations. As the noise level increases(Gaussian
noise), spectral subtraction performance also degrades, even
though SNR level of the restored signal approaches about a
constant value .It also maintains 5dB improvement from the
noisy signal.
Fig-6: SNR plot of broadband denoised signal with variance
to noise level
3.3 Adaptive Declicking Results
Mean Square Error - MSE
The aim of an adaptive filter is to reduce the Mean Square
Error. It is calculated using the following equation,
MSE =E (|d(n)-yd(n)|2
) (4)
The values and graph of this quantity will be essential to
evaluate the performance of the adaptive filter. If adaptive
algorithm works well, after convergence time, the value of
MSE should be reduced gradually to zero. Figures below
shows the MSE plot of designed adaptive filter using the
LMS and NLMS algorithm.
5. IJRET: International Journal of Research in Engineering and Technology eISSN: 2319-1163 | pISSN: 2321-7308
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Volume: 04 Issue: 10 | Oct -2015, Available @ http://www.ijret.org 82
0 1 2 3 4 5 6 7
x 10
4
0
0.05
0.1
0.15
0.2
0.25
0.3
0.35
0.4
0.45
sample number
MSE
0 1 2 3 4 5 6 7
x 10
4
0
0.5
1
1.5
2
2.5
sample no:
MSE
Fig-7: MSE plots for LMS and NLMS algorithms
From the results of Fig-7, the NLMS adaptive algorithm has
less mean square error than that of LMS algorithm, since it
has the feature of adapted step size than fixed step size in
LMS Algorithm. More improved algorithms such as RLS
can also be applied for the better adaptation of the signal.
3.4 Hardware Implementation Results
Adaptive filter method opted for removing the clicks is
implemented using DSK [6]. Its implementation results are
shown below in Fig-8 (a), (b), (c). (a) shows the click
variant sent to the right channel of AIC23 codec in DSK as
the input to adaptive filter , (b) shows the clicked noisy
audio signal sent to the left channel and (c) obtained
adaptive output resembles the desired signal which is
observed using CRO. Fig.9 shows the hardware setup used
for implementation purpose.
(a) Click variant
(b) Noisy signal
(c) Output Signal
Fig-8: Hardware implemented waveform results
Fig-9: Hardware Setup
4. CONCLUSION
This paper is focused on audio inpainting algorithm using
sparse representation of audio signals, removal of broadband
noise and click using the proposed methods. The purpose
was to study SR of audio signals and related audio
inpainting problems. Proposed approach is worth studied
and its software implementation is completed using Matlab.
The project also targeting towards the hardware
implementation of restoration methods using
TMS320C6713 DSK .Overview of currently done work
realized that audio inpainting problem is still current
problem worth to study and resolve. Reconstruction of audio
signals of commercial programs is not perfect and Sparse
representation based audio inpainting algorithms can be nice
tool for improving current situation. Spectral subtraction and
adaptive algorithm approaches are also suitable for reducing
the clicks and broadband distortion from the audio signals. It
is still a current research problem. Adaptive click removal
system is implemented in DSK and verified the results. In
future, targeting towards the complete hardware
implementation of complex OMP declipping algorithm
using FPGAs and more implementations based on sparse
signal representations for applications like Image Inpainting,
Image denoising and Image Compression and addressing
other major audio distortions.
REFERENCES
[1]. Adler A. “A Constrained Matching Pursuit Approach
to Audio Declipping”.Acoustics, Speech and Signal
Processing, IEEE International Conference on
(ICASSP 2011), IEEE, Prague, Czech Republic, 2011
[2]. S.J.Godsill. “Digital audio Restoration” Springer-
Verlag, 1998.
[3]. J. Tropp, “Greed is good: Algorithmic results for
sparse approximation,” IEEE Trans. Inf. Theory, Oct.
2004.
[4]. S. Boll, “Suppression of Acoustic Noise in Speech
Using Spectral Subtraction”, IEEE Trans. Acoust.
Speech, Signal Processing , vol ASSP-27, pp 113-120,
1979
[5]. Bernard Widrow and Samuel D. Stearns “Adaptive
Signal Processing” [Book].
[6]. Texas Instruments Inc., TMS320C6713,
TMS320C6713B Floating-Point Digital Signal
Processors, SPRS186I, Dallas, TX, May 2004.
[7]. A.Janssen, R. Veldhuis, and “L. Vries, Adaptive
interpolation of discrete- time signals that can be
modeled as autoregressive processes”, IEEE Trans.
Acoustics, Speech and Signal Process., Apr.1986.
[8]. M. Lagrange and S. Marchand, “Long interpolation of
audio signals using linear prediction in sinusoidal
modeling”, J. Audio Eng. Soc.2005
[9]. R. C. Maher, “A method for extrapolation of missing
digital audio data,” in Proc.95th AES Conv., 1993
6. IJRET: International Journal of Research in Engineering and Technology eISSN: 2319-1163 | pISSN: 2321-7308
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Volume: 04 Issue: 10 | Oct -2015, Available @ http://www.ijret.org 83
BIOGRAPHIES
Merin K Mathai , M-Tech student in
VLSI and Embedded Systems, College of
Engineering Chengannur, Kerala, India.
Current areas of interest is in ASIC
design, Signal Processing , FPGA based
design.
Email id: merinkmathai@gmail.com
Dr. Deepa J, Associate Professor, College
Of Engineering, Chengannur, Kerala,
India. Her Current areas of interests in
Image processing, Signal processing and
Bioinformatics
Email id: deepaj@ceconline.edu