The document presents a voice recognition system that uses the Mel Frequency Cepstral Coefficients (MFCC) algorithm to extract voice features from a user's speech and activate a device if the voice matches a stored sample. The system operates in two phases: 1) recording and extracting voice features to create a database, and 2) comparing input voice samples to those in the database. If a match is found, the system activates a testing device (Arduino Uno). The MFCC algorithm frames the voice, applies windowing and FFT, filters to a triangular shape, and uses DCT to extract features. Vector quantization is used to match input features to the database for authentication and device activation. Simulation results demonstrate the system's
International Journal of Engineering Research and Applications (IJERA) is an open access online peer reviewed international journal that publishes research and review articles in the fields of Computer Science, Neural Networks, Electrical Engineering, Software Engineering, Information Technology, Mechanical Engineering, Chemical Engineering, Plastic Engineering, Food Technology, Textile Engineering, Nano Technology & science, Power Electronics, Electronics & Communication Engineering, Computational mathematics, Image processing, Civil Engineering, Structural Engineering, Environmental Engineering, VLSI Testing & Low Power VLSI Design etc.International Journal of Engineering Research and Applications (IJERA) is an open access online peer reviewed international journal that publishes research and review articles in the fields of Computer Science, Neural Networks, Electrical Engineering, Software Engineering, Information Technology, Mechanical Engineering, Chemical Engineering, Plastic Engineering, Food Technology, Textile Engineering, Nano Technology & science, Power Electronics, Electronics & Communication Engineering, Computational mathematics, Image processing, Civil Engineering, Structural Engineering, Environmental Engineering, VLSI Testing & Low Power VLSI Design etc.
Wavelet Based Noise Robust Features for Speaker RecognitionCSCJournals
Extraction and selection of the best parametric representation of acoustic signal is the most important task in designing any speaker recognition system. A wide range of possibilities exists for parametrically representing the speech signal such as Linear Prediction Coding (LPC) ,Mel frequency Cepstrum coefficients (MFCC) and others. MFCC are currently the most popular choice for any speaker recognition system, though one of the shortcomings of MFCC is that the signal is assumed to be stationary within the given time frame and is therefore unable to analyze the non-stationary signal. Therefore it is not suitable for noisy speech signals. To overcome this problem several researchers used different types of AM-FM modulation/demodulation techniques for extracting features from speech signal. In some approaches it is proposed to use the wavelet filterbanks for extracting the features. In this paper a technique for extracting the features by combining the above mentioned approaches is proposed. Features are extracted from the envelope of the signal and then passed through wavelet filterbank. It is found that the proposed method outperforms the existing feature extraction techniques.
International Journal of Engineering Research and Applications (IJERA) is an open access online peer reviewed international journal that publishes research and review articles in the fields of Computer Science, Neural Networks, Electrical Engineering, Software Engineering, Information Technology, Mechanical Engineering, Chemical Engineering, Plastic Engineering, Food Technology, Textile Engineering, Nano Technology & science, Power Electronics, Electronics & Communication Engineering, Computational mathematics, Image processing, Civil Engineering, Structural Engineering, Environmental Engineering, VLSI Testing & Low Power VLSI Design etc.
AN ANALYSIS OF SPEECH RECOGNITION PERFORMANCE BASED UPON NETWORK LAYERS AND T...IJCSEA Journal
Speech is the most natural way of information exchange. It provides an efficient means of means of manmachine communication using speech interfacing. Speech interfacing involves speech synthesis and speech recognition. Speech recognition allows a computer to identify the words that a person speaks to a microphone or telephone. The two main components, normally used in speech recognition, are signal processing component at front-end and pattern matching component at back-end. In this paper, a setup that uses Mel frequency cepstral coefficients at front-end and artificial neural networks at back-end has been developed to perform the experiments for analyzing the speech recognition performance. Various experiments have been performed by varying the number of layers and type of network transfer function, which helps in deciding the network architecture to be used for acoustic modelling at back end.
Transmitting audio via fiber optics under nonlinear effects and optimized tun...IJECEIAES
The ability of fiber optic to overcome the signal transmission problems is making it a dominant transmission medium. Despite of this major positive attribute of optic fibers, there is still a downside for using the fiber optic communication; that is the nonlinearity problem. For the first time, a design of an audio signal is suggested and executed in MATLAB with integration with OptiSystem TM Software .The audio signal then transmitted in different shapes of modulation signals (NRZ, RZ, & RC) for different distances (100 km & 75 km) via a fiber optic media to be received in a receiving part of the simulated system. Three tests are used to do so. The first is the Quality-facto (Q-Factor) against the received power, second test is eye diagram performance and finally is the measuring of the amplitude of output (received) signal for each modulation signal shape using the Oscilloscope Visualizer. The NZR modulation signal was found to be the best one of the three used signals’ types in all three tests. The Q-factor for NRZ pulse shape (=12) was higher than that for RZ (=10) and RC (=8) for a 100 km distance at the same received power level.
International Journal of Engineering Research and Applications (IJERA) is an open access online peer reviewed international journal that publishes research and review articles in the fields of Computer Science, Neural Networks, Electrical Engineering, Software Engineering, Information Technology, Mechanical Engineering, Chemical Engineering, Plastic Engineering, Food Technology, Textile Engineering, Nano Technology & science, Power Electronics, Electronics & Communication Engineering, Computational mathematics, Image processing, Civil Engineering, Structural Engineering, Environmental Engineering, VLSI Testing & Low Power VLSI Design etc.International Journal of Engineering Research and Applications (IJERA) is an open access online peer reviewed international journal that publishes research and review articles in the fields of Computer Science, Neural Networks, Electrical Engineering, Software Engineering, Information Technology, Mechanical Engineering, Chemical Engineering, Plastic Engineering, Food Technology, Textile Engineering, Nano Technology & science, Power Electronics, Electronics & Communication Engineering, Computational mathematics, Image processing, Civil Engineering, Structural Engineering, Environmental Engineering, VLSI Testing & Low Power VLSI Design etc.
Wavelet Based Noise Robust Features for Speaker RecognitionCSCJournals
Extraction and selection of the best parametric representation of acoustic signal is the most important task in designing any speaker recognition system. A wide range of possibilities exists for parametrically representing the speech signal such as Linear Prediction Coding (LPC) ,Mel frequency Cepstrum coefficients (MFCC) and others. MFCC are currently the most popular choice for any speaker recognition system, though one of the shortcomings of MFCC is that the signal is assumed to be stationary within the given time frame and is therefore unable to analyze the non-stationary signal. Therefore it is not suitable for noisy speech signals. To overcome this problem several researchers used different types of AM-FM modulation/demodulation techniques for extracting features from speech signal. In some approaches it is proposed to use the wavelet filterbanks for extracting the features. In this paper a technique for extracting the features by combining the above mentioned approaches is proposed. Features are extracted from the envelope of the signal and then passed through wavelet filterbank. It is found that the proposed method outperforms the existing feature extraction techniques.
International Journal of Engineering Research and Applications (IJERA) is an open access online peer reviewed international journal that publishes research and review articles in the fields of Computer Science, Neural Networks, Electrical Engineering, Software Engineering, Information Technology, Mechanical Engineering, Chemical Engineering, Plastic Engineering, Food Technology, Textile Engineering, Nano Technology & science, Power Electronics, Electronics & Communication Engineering, Computational mathematics, Image processing, Civil Engineering, Structural Engineering, Environmental Engineering, VLSI Testing & Low Power VLSI Design etc.
AN ANALYSIS OF SPEECH RECOGNITION PERFORMANCE BASED UPON NETWORK LAYERS AND T...IJCSEA Journal
Speech is the most natural way of information exchange. It provides an efficient means of means of manmachine communication using speech interfacing. Speech interfacing involves speech synthesis and speech recognition. Speech recognition allows a computer to identify the words that a person speaks to a microphone or telephone. The two main components, normally used in speech recognition, are signal processing component at front-end and pattern matching component at back-end. In this paper, a setup that uses Mel frequency cepstral coefficients at front-end and artificial neural networks at back-end has been developed to perform the experiments for analyzing the speech recognition performance. Various experiments have been performed by varying the number of layers and type of network transfer function, which helps in deciding the network architecture to be used for acoustic modelling at back end.
Transmitting audio via fiber optics under nonlinear effects and optimized tun...IJECEIAES
The ability of fiber optic to overcome the signal transmission problems is making it a dominant transmission medium. Despite of this major positive attribute of optic fibers, there is still a downside for using the fiber optic communication; that is the nonlinearity problem. For the first time, a design of an audio signal is suggested and executed in MATLAB with integration with OptiSystem TM Software .The audio signal then transmitted in different shapes of modulation signals (NRZ, RZ, & RC) for different distances (100 km & 75 km) via a fiber optic media to be received in a receiving part of the simulated system. Three tests are used to do so. The first is the Quality-facto (Q-Factor) against the received power, second test is eye diagram performance and finally is the measuring of the amplitude of output (received) signal for each modulation signal shape using the Oscilloscope Visualizer. The NZR modulation signal was found to be the best one of the three used signals’ types in all three tests. The Q-factor for NRZ pulse shape (=12) was higher than that for RZ (=10) and RC (=8) for a 100 km distance at the same received power level.
Limited Data Speaker Verification: Fusion of FeaturesIJECEIAES
The present work demonstrates experimental evaluation of speaker verification for dif- ferent speech feature extraction techniques with the constraints of limited data (less than 15 seconds). The state-of-the-art speaker verification techniques provide good performance for sufficient data (greater than 1 minutes). It is a challenging task to develop techniques which perform well for speaker verification under limited data condition. In this work different features like Mel Frequency Cepstral Coefficients (MFCC), Linear Prediction Cepstral Coefficients (LPCC), Delta (4), Delta-Delta (44), Linear Prediction Residual (LPR) and Linear Prediction Residual Phase (LPRP) are considered. The performance of individual features is studied and for better verification performance, combination of these features is attempted. A comparative study is made between Gaussian mixture model (GMM) and GMM-universal background model (GMM-UBM) through experimental evaluation. The experiments are conducted using NIST-2003 database. The experimental results show that, the combination of features provides better performance compared to the individual features. Further GMM-UBM modeling gives reduced equal error rate (EER) as compared to GMM.
Adaptive wavelet thresholding with robust hybrid features for text-independe...IJECEIAES
The robustness of speaker identification system over additive noise channel is crucial for real-world applications. In speaker identification (SID) systems, the extracted features from each speech frame are an essential factor for building a reliable identification system. For clean environments, the identification system works well; in noisy environments, there is an additive noise, which is affect the system. To eliminate the problem of additive noise and to achieve a high accuracy in speaker identification system a proposed algorithm for feature extraction based on speech enhancement and a combined features is presents. In this paper, a wavelet thresholding pre-processing stage, and feature warping (FW) techniques are used with two combined features named power normalized cepstral coefficients (PNCC) and gammatone frequency cepstral coefficients (GFCC) to improve the identification system robustness against different types of additive noises. Universal background model Gaussian mixture model (UBM-GMM) is used for features matching between the claim and actual speakers. The results showed performance improvement for the proposed feature extraction algorithm of identification system comparing with conventional features over most types of noises and different SNR ratios.
Area Efficient and high-speed fir filter implementation using divided LUT methodIJMER
Traditional method of implementing FIR filters costs considerable hardware resourses,
which goes against the decrease of circuit scale and the increase of system speed. A new design and
implementation of FIR filters using Distributed Arithmetic is provided in this paper to slove this
problem. Distributed Arithmetic structure is used to increase the resourse useage while pipeline
structure is also used to increase the system speed. In addition, the devided LUT method is also used to
decrease the required memory units. The simulation results indicate that FIR filters using Distributed
Arithmetic can work stable with high speed and can save almost 50 percent hardware resourses to
decrease the circuit scale, and can be applied to a variety of areas for its great flexibility and high
reliability
Implementation Cost Analysis of the Interpolator for the Wimax Technologyiosrjce
The design of the multirate filter (programmable) has been proposed which can be used in digital
transceivers that meets 802.16d/e (wimax) standard in the wireless communication system. Wimax is a
technology emerging in the wireless communication system in order to increase the broadband wireless internet
access. As there is wide spread need of the digital representation of the signal for the transmission and storage
which create the challenges in DSP [1]. In this paper, analysis of the implementation cost of interpolator for the
wimax technology, and cost of interpolator is analyzed on the basis of number of adders and multiplier. The
Filters are designed using the FDA (filters design and analysis) tool in MATLAB.
Design and implementation of DA FIR filter for bio-inspired computing archite...IJECEIAES
This paper elucidates the system construct of DA-FIR filter optimized for design of distributed arithmetic (DA) finite impulse response (FIR) filter and is based on architecture with tightly coupled co-processor based data processing units. With a series of look-up-table (LUT) accesses in order to emulate multiply and accumulate operations the constructed DA based FIR filter is implemented on FPGA. The very high speed integrated circuit hardware description language (VHDL) is used implement the proposed filter and the design is verified using simulation. This paper discusses two optimization algorithms and resulting optimizations are incorporated into LUT layer and architecture extractions. The proposed method offers an optimized design in the form of offers average miminimizations of the number of LUT, reduction in populated slices and gate minimization for DAfinite impulse response filter. This research paves a direction towards development of bio inspired computing architectures developed without logically intensive operations, obtaining the desired specifications with respect to performance, timing, and reliability.
Performance Analysis of Fractional Sample Rate Converter Using Audio Applicat...iosrjce
Fractional rate converters which are generally used for many applications with different frequencies
and are an essential part of communication systems. In this paper fractional rate converter with use of both
FIR and Nyquist FIR have been compared and analyzed. Its implementation can be easily found in the
developing communication systems, but here results are taken for audio applications. The proposed design and
analysis have been developed with the help of MATLAB with order 50 for FIR and 71 for Nyquist, sampling
frequency 48000Hz. The filters are then interpolated by an interpolation factor 2 and decimated by a decimation
factor of 3. The cost implementation of both has been taken into consideration and a result is drawn which
concludes that fractional rate converter for Nyquist FIR filter much more cost effective as compared to the
fractional rate converter for FIR filter
Power efficient and high throughput of fir filter using block least mean squa...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
FPGA based Efficient Interpolator design using DALUT Algorithmcscpconf
Interpolator is an important sampling device used for multirate filtering to provide signal processing in wireless communication system. There are many applications in which sampling rate must be changed. Interpolators and decimators are utilized to increase or decrease the sampling rate. In this paper an efficient method has been presented to implement high speed and area efficient interpolator for wireless communication systems. A multiplier less technique is used which substitutes multiplyand-accumulate operations with look up table (LUT) accesses. Interpolator has been implemented using Partitioned distributed arithmetic look up table (DALUT) technique. This technique has been used to take an optimal advantage of embedded LUTs of the target FPGA. This method is useful to enhance the system performance in terms of speed and area. The proposed interpolator has been designed using half band poly phase FIR structure with Matlab, simulated with ISE, synthesized with Xilinx Synthesis Tools (XST) and implemented on Spartan-3E and Virtex2pro device. The proposed LUT based multiplier less approach has shown a maximum operating frequency of 92.859 MHz with Virtex Pro and 61.6 MHz with Spartan 3E by consuming considerably less resources to provide cost effective solution for wireless communication systems.
IJERA (International journal of Engineering Research and Applications) is International online, ... peer reviewed journal. For more detail or submit your article, please visit www.ijera.com
Analysis of Impact of Channel Error Rate on Average PSNR in Multimedia TrafficIOSR Journals
Abstract : The performance of the multimedia traffic in Ad-Hoc networks is highly impacted with the Signal to Noise Ratio. The Average PSNR (Peak Signal to Noise Ratio) is an important parameter for the evaluation of multimedia traffic in Ad-Hoc Networks. With the increase of bandwidth of the channels, it becomes necessary to take care of other network parameters like PSNR and ASNR( Average Signal to Noise Ratio) .Enhanced bandwidth with higher channel error rates demand a careful analysis of signal to noise ratio for optimum performance. In this paper, we have evaluated the effect of channel error rate on Average PSNR for the MPEG-4 traffic in Ad-hoc Networks. Keywords: MANETs, Evalvid, MPEG-4, Fragmentation, PSNR
Intelligent Arabic letters speech recognition system based on mel frequency c...IJECEIAES
Speech recognition is one of the important applications of artificial intelligence (AI). Speech recognition aims to recognize spoken words regardless of who is speaking to them. The process of voice recognition involves extracting meaningful features from spoken words and then classifying these features into their classes. This paper presents a neural network classification system for Arabic letters. The paper will study the effect of changing the multi-layer perceptron (MLP) artificial neural network (ANN) properties to obtain an optimized performance. The proposed system consists of two main stages; first, the recorded spoken letters are transformed from the time domain into the frequency domain using fast Fourier transform (FFT), and features are extracted using mel frequency cepstral coefficients (MFCC). Second, the extracted features are then classified using the MLP ANN with back-propagation (BP) learning algorithm. The obtained results show that the proposed system along with the extracted features can classify Arabic spoken letters using two neural network hidden layers with an accuracy of around 86%.
Limited Data Speaker Verification: Fusion of FeaturesIJECEIAES
The present work demonstrates experimental evaluation of speaker verification for dif- ferent speech feature extraction techniques with the constraints of limited data (less than 15 seconds). The state-of-the-art speaker verification techniques provide good performance for sufficient data (greater than 1 minutes). It is a challenging task to develop techniques which perform well for speaker verification under limited data condition. In this work different features like Mel Frequency Cepstral Coefficients (MFCC), Linear Prediction Cepstral Coefficients (LPCC), Delta (4), Delta-Delta (44), Linear Prediction Residual (LPR) and Linear Prediction Residual Phase (LPRP) are considered. The performance of individual features is studied and for better verification performance, combination of these features is attempted. A comparative study is made between Gaussian mixture model (GMM) and GMM-universal background model (GMM-UBM) through experimental evaluation. The experiments are conducted using NIST-2003 database. The experimental results show that, the combination of features provides better performance compared to the individual features. Further GMM-UBM modeling gives reduced equal error rate (EER) as compared to GMM.
Adaptive wavelet thresholding with robust hybrid features for text-independe...IJECEIAES
The robustness of speaker identification system over additive noise channel is crucial for real-world applications. In speaker identification (SID) systems, the extracted features from each speech frame are an essential factor for building a reliable identification system. For clean environments, the identification system works well; in noisy environments, there is an additive noise, which is affect the system. To eliminate the problem of additive noise and to achieve a high accuracy in speaker identification system a proposed algorithm for feature extraction based on speech enhancement and a combined features is presents. In this paper, a wavelet thresholding pre-processing stage, and feature warping (FW) techniques are used with two combined features named power normalized cepstral coefficients (PNCC) and gammatone frequency cepstral coefficients (GFCC) to improve the identification system robustness against different types of additive noises. Universal background model Gaussian mixture model (UBM-GMM) is used for features matching between the claim and actual speakers. The results showed performance improvement for the proposed feature extraction algorithm of identification system comparing with conventional features over most types of noises and different SNR ratios.
Area Efficient and high-speed fir filter implementation using divided LUT methodIJMER
Traditional method of implementing FIR filters costs considerable hardware resourses,
which goes against the decrease of circuit scale and the increase of system speed. A new design and
implementation of FIR filters using Distributed Arithmetic is provided in this paper to slove this
problem. Distributed Arithmetic structure is used to increase the resourse useage while pipeline
structure is also used to increase the system speed. In addition, the devided LUT method is also used to
decrease the required memory units. The simulation results indicate that FIR filters using Distributed
Arithmetic can work stable with high speed and can save almost 50 percent hardware resourses to
decrease the circuit scale, and can be applied to a variety of areas for its great flexibility and high
reliability
Implementation Cost Analysis of the Interpolator for the Wimax Technologyiosrjce
The design of the multirate filter (programmable) has been proposed which can be used in digital
transceivers that meets 802.16d/e (wimax) standard in the wireless communication system. Wimax is a
technology emerging in the wireless communication system in order to increase the broadband wireless internet
access. As there is wide spread need of the digital representation of the signal for the transmission and storage
which create the challenges in DSP [1]. In this paper, analysis of the implementation cost of interpolator for the
wimax technology, and cost of interpolator is analyzed on the basis of number of adders and multiplier. The
Filters are designed using the FDA (filters design and analysis) tool in MATLAB.
Design and implementation of DA FIR filter for bio-inspired computing archite...IJECEIAES
This paper elucidates the system construct of DA-FIR filter optimized for design of distributed arithmetic (DA) finite impulse response (FIR) filter and is based on architecture with tightly coupled co-processor based data processing units. With a series of look-up-table (LUT) accesses in order to emulate multiply and accumulate operations the constructed DA based FIR filter is implemented on FPGA. The very high speed integrated circuit hardware description language (VHDL) is used implement the proposed filter and the design is verified using simulation. This paper discusses two optimization algorithms and resulting optimizations are incorporated into LUT layer and architecture extractions. The proposed method offers an optimized design in the form of offers average miminimizations of the number of LUT, reduction in populated slices and gate minimization for DAfinite impulse response filter. This research paves a direction towards development of bio inspired computing architectures developed without logically intensive operations, obtaining the desired specifications with respect to performance, timing, and reliability.
Performance Analysis of Fractional Sample Rate Converter Using Audio Applicat...iosrjce
Fractional rate converters which are generally used for many applications with different frequencies
and are an essential part of communication systems. In this paper fractional rate converter with use of both
FIR and Nyquist FIR have been compared and analyzed. Its implementation can be easily found in the
developing communication systems, but here results are taken for audio applications. The proposed design and
analysis have been developed with the help of MATLAB with order 50 for FIR and 71 for Nyquist, sampling
frequency 48000Hz. The filters are then interpolated by an interpolation factor 2 and decimated by a decimation
factor of 3. The cost implementation of both has been taken into consideration and a result is drawn which
concludes that fractional rate converter for Nyquist FIR filter much more cost effective as compared to the
fractional rate converter for FIR filter
Power efficient and high throughput of fir filter using block least mean squa...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
FPGA based Efficient Interpolator design using DALUT Algorithmcscpconf
Interpolator is an important sampling device used for multirate filtering to provide signal processing in wireless communication system. There are many applications in which sampling rate must be changed. Interpolators and decimators are utilized to increase or decrease the sampling rate. In this paper an efficient method has been presented to implement high speed and area efficient interpolator for wireless communication systems. A multiplier less technique is used which substitutes multiplyand-accumulate operations with look up table (LUT) accesses. Interpolator has been implemented using Partitioned distributed arithmetic look up table (DALUT) technique. This technique has been used to take an optimal advantage of embedded LUTs of the target FPGA. This method is useful to enhance the system performance in terms of speed and area. The proposed interpolator has been designed using half band poly phase FIR structure with Matlab, simulated with ISE, synthesized with Xilinx Synthesis Tools (XST) and implemented on Spartan-3E and Virtex2pro device. The proposed LUT based multiplier less approach has shown a maximum operating frequency of 92.859 MHz with Virtex Pro and 61.6 MHz with Spartan 3E by consuming considerably less resources to provide cost effective solution for wireless communication systems.
IJERA (International journal of Engineering Research and Applications) is International online, ... peer reviewed journal. For more detail or submit your article, please visit www.ijera.com
Analysis of Impact of Channel Error Rate on Average PSNR in Multimedia TrafficIOSR Journals
Abstract : The performance of the multimedia traffic in Ad-Hoc networks is highly impacted with the Signal to Noise Ratio. The Average PSNR (Peak Signal to Noise Ratio) is an important parameter for the evaluation of multimedia traffic in Ad-Hoc Networks. With the increase of bandwidth of the channels, it becomes necessary to take care of other network parameters like PSNR and ASNR( Average Signal to Noise Ratio) .Enhanced bandwidth with higher channel error rates demand a careful analysis of signal to noise ratio for optimum performance. In this paper, we have evaluated the effect of channel error rate on Average PSNR for the MPEG-4 traffic in Ad-hoc Networks. Keywords: MANETs, Evalvid, MPEG-4, Fragmentation, PSNR
Intelligent Arabic letters speech recognition system based on mel frequency c...IJECEIAES
Speech recognition is one of the important applications of artificial intelligence (AI). Speech recognition aims to recognize spoken words regardless of who is speaking to them. The process of voice recognition involves extracting meaningful features from spoken words and then classifying these features into their classes. This paper presents a neural network classification system for Arabic letters. The paper will study the effect of changing the multi-layer perceptron (MLP) artificial neural network (ANN) properties to obtain an optimized performance. The proposed system consists of two main stages; first, the recorded spoken letters are transformed from the time domain into the frequency domain using fast Fourier transform (FFT), and features are extracted using mel frequency cepstral coefficients (MFCC). Second, the extracted features are then classified using the MLP ANN with back-propagation (BP) learning algorithm. The obtained results show that the proposed system along with the extracted features can classify Arabic spoken letters using two neural network hidden layers with an accuracy of around 86%.
Speech Recognized Automation System Using Speaker Identification through Wire...IOSR Journals
Abstract : This paper discusses the methodology for a project named “Speech Recognized Automation System using Speaker Identification through wireless communication”. This project gives the design of Automation system using wireless communication and speaker recognition using Matlab code. Straightforward programming interface of Matlab makes it an ideal tool for speech analysis in project. This automation system is useful for home appliances as well as in industry. This paper discusses the overall design of a wireless automation system which is built and implemented. The speech recognition centers on recognition of speech commands stored in data base of Matlab and it is matched with incoming voice command of speaker. Mel Frequency Cepstral Coefficient (MFCC) algorithm is used to recognize the speech of speaker and to extract features of speech. It uses low-power RF ZigBee transceiver wireless communication modules which are relatively cheap. This automation system is intended to control lights, fans and other electrical appliances in a home or office using speech commands like Light, Fan etc. Further, if security is not big issue then Speech processor is used to control the appliances without speaker identification. Keywords — Automation system, MATLAB code, MFCC, speaker identification, ZigBee transceiver.
Speech Recognized Automation System Using Speaker Identification through Wire...IOSR Journals
This paper discusses the methodology for a project named “Speech Recognized Automation System
using Speaker Identification through wireless communication”. This project gives the design of Automation
system using wireless communication and speaker recognition using Matlab code. Straightforward
programming interface of Matlab makes it an ideal tool for speech analysis in project. This automation system
is useful for home appliances as well as in industry. This paper discusses the overall design of a wireless
automation system which is built and implemented. The speech recognition centers on recognition of speech
commands stored in data base of Matlab and it is matched with incoming voice command of speaker. Mel
Frequency Cepstral Coefficient (MFCC) algorithm is used to recognize the speech of speaker and to extract
features of speech. It uses low-power RF ZigBee transceiver wireless communication modules which are
relatively cheap. This automation system is intended to control lights, fans and other electrical appliances in a
home or office using speech commands like Light, Fan etc. Further, if security is not big issue then Speech
processor is used to control the appliances without speaker identification
Comparative Study of Different Techniques in Speaker Recognition: ReviewIJAEMSJORNAL
The speech is most basic and essential method of communication used by person.On the basis of individual information included in speech signals the speaker is recognized. Speaker recognition (SR) is useful to identify the person who is speaking. In recent years speaker recognition is used for security system. In this paper we have discussed the feature extraction techniques like Mel frequency cepstral coefficient (MFCC), Linear predictive coding (LPC), Dynamic time wrapping (DTW), and for classification Gaussian Mixture Models (GMM), Artificial neural network (ANN)& Support vector machine (SVM).
ADAPTIVE WATERMARKING TECHNIQUE FOR SPEECH SIGNAL AUTHENTICATION ijcsit
Biometrics data recently has become a major role in determining the identity of the person. With such
importance for the use of biometrics data, there are many attacks that threaten the security and integrity of
biometrics data itself. Therefore, it becomes necessary to protect the originality of biometrics data against
manipulation and fraud. This paper presents an authentication technique to achieve the authenticity of
speech signals based on adaptive watermarking technique. The basic idea is depends on extracting the
speech features from the speech signal initially and then using these features as a watermark. The
watermark information embeds into the same speech signal. The short time energy technique is used to
identifying the suitable positions for embedding the watermark in order to avoid the regions that used in
the speech recognition system. After exclusion the important areas that used in speech recognition the
Genetic Algorithm (GA) is used to generate random locations to hide the watermark information in an
intelligent manner. The experimental results have achieved high efficiency in establishing the authenticity
of speech signal and the process of embedding
Comparative study to realize an automatic speaker recognition system IJECEIAES
In this research, we present an automatic speaker recognition system based on adaptive orthogonal transformations. To obtain the informative features with a minimum dimension from the input signals, we created an adaptive operator, which helped to identify the speaker’s voice in a fast and efficient manner. We test the efficiency and the performance of our method by comparing it with another approach, mel-frequency cepstral coefficients (MFCCs), which is widely used by researchers as their feature extraction method. The experimental results show the importance of creating the adaptive operator, which gives added value to the proposed approach. The performance of the system achieved 96.8% accuracy using Fourier transform as a compression method and 98.1% using Correlation as a compression method.
Machine Learning Based Session Drop Prediction in LTE Networks and its SON As...Ericsson
Abnormal bearer session release (i.e. bearer session drop) in cellular telecommunication networks may seriously impact the quality of experience of mobile users. The latest mobile technologies enable high granularity real-time reporting of all conditions of individual sessions, which gives rise to use data analytics methods to process and monetize this data for network optimization. One such example for analytics is Machine Learning (ML) to predict session drops well before the end of session.
Similar to IRJET- Device Activation based on Voice Recognition using Mel Frequency Cepstral Coefficients (MFCC’s) Algorithm (20)
Harnessing WebAssembly for Real-time Stateless Streaming PipelinesChristina Lin
Traditionally, dealing with real-time data pipelines has involved significant overhead, even for straightforward tasks like data transformation or masking. However, in this talk, we’ll venture into the dynamic realm of WebAssembly (WASM) and discover how it can revolutionize the creation of stateless streaming pipelines within a Kafka (Redpanda) broker. These pipelines are adept at managing low-latency, high-data-volume scenarios.
Using recycled concrete aggregates (RCA) for pavements is crucial to achieving sustainability. Implementing RCA for new pavement can minimize carbon footprint, conserve natural resources, reduce harmful emissions, and lower life cycle costs. Compared to natural aggregate (NA), RCA pavement has fewer comprehensive studies and sustainability assessments.
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.
Literature Review Basics and Understanding Reference Management.pptxDr Ramhari Poudyal
Three-day training on academic research focuses on analytical tools at United Technical College, supported by the University Grant Commission, Nepal. 24-26 May 2024
Saudi Arabia stands as a titan in the global energy landscape, renowned for its abundant oil and gas resources. It's the largest exporter of petroleum and holds some of the world's most significant reserves. Let's delve into the top 10 oil and gas projects shaping Saudi Arabia's energy future in 2024.
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.
Industrial Training at Shahjalal Fertilizer Company Limited (SFCL)MdTanvirMahtab2
This presentation is about the working procedure of Shahjalal Fertilizer Company Limited (SFCL). A Govt. owned Company of Bangladesh Chemical Industries Corporation under Ministry of Industries.
6th International Conference on Machine Learning & Applications (CMLA 2024)ClaraZara1
6th International Conference on Machine Learning & Applications (CMLA 2024) will provide an excellent international forum for sharing knowledge and results in theory, methodology and applications of on Machine Learning & Applications.
Welcome to WIPAC Monthly the magazine brought to you by the LinkedIn Group Water Industry Process Automation & Control.
In this month's edition, along with this month's industry news to celebrate the 13 years since the group was created we have articles including
A case study of the used of Advanced Process Control at the Wastewater Treatment works at Lleida in Spain
A look back on an article on smart wastewater networks in order to see how the industry has measured up in the interim around the adoption of Digital Transformation in the Water Industry.
Cosmetic shop management system project report.pdfKamal Acharya
Buying new cosmetic products is difficult. It can even be scary for those who have sensitive skin and are prone to skin trouble. The information needed to alleviate this problem is on the back of each product, but it's thought to interpret those ingredient lists unless you have a background in chemistry.
Instead of buying and hoping for the best, we can use data science to help us predict which products may be good fits for us. It includes various function programs to do the above mentioned tasks.
Data file handling has been effectively used in the program.
The automated cosmetic shop management system should deal with the automation of general workflow and administration process of the shop. The main processes of the system focus on customer's request where the system is able to search the most appropriate products and deliver it to the customers. It should help the employees to quickly identify the list of cosmetic product that have reached the minimum quantity and also keep a track of expired date for each cosmetic product. It should help the employees to find the rack number in which the product is placed.It is also Faster and more efficient way.
Hybrid optimization of pumped hydro system and solar- Engr. Abdul-Azeez.pdffxintegritypublishin
Advancements in technology unveil a myriad of electrical and electronic breakthroughs geared towards efficiently harnessing limited resources to meet human energy demands. The optimization of hybrid solar PV panels and pumped hydro energy supply systems plays a pivotal role in utilizing natural resources effectively. This initiative not only benefits humanity but also fosters environmental sustainability. The study investigated the design optimization of these hybrid systems, focusing on understanding solar radiation patterns, identifying geographical influences on solar radiation, formulating a mathematical model for system optimization, and determining the optimal configuration of PV panels and pumped hydro storage. Through a comparative analysis approach and eight weeks of data collection, the study addressed key research questions related to solar radiation patterns and optimal system design. The findings highlighted regions with heightened solar radiation levels, showcasing substantial potential for power generation and emphasizing the system's efficiency. Optimizing system design significantly boosted power generation, promoted renewable energy utilization, and enhanced energy storage capacity. The study underscored the benefits of optimizing hybrid solar PV panels and pumped hydro energy supply systems for sustainable energy usage. Optimizing the design of solar PV panels and pumped hydro energy supply systems as examined across diverse climatic conditions in a developing country, not only enhances power generation but also improves the integration of renewable energy sources and boosts energy storage capacities, particularly beneficial for less economically prosperous regions. Additionally, the study provides valuable insights for advancing energy research in economically viable areas. Recommendations included conducting site-specific assessments, utilizing advanced modeling tools, implementing regular maintenance protocols, and enhancing communication among system components.