This document discusses speech compression analysis using MATLAB. It begins with an introduction to speech compression, noting its importance for efficient storage and transmission of audio data. It then discusses various speech compression techniques, including lossy and lossless compression as well as standards like MPEG. It focuses on using the discrete cosine transform and MATLAB commands to analyze speech signals, including reading wav files, applying windowing functions and the DCT, and playing/viewing the output. The document concludes by discussing current applications of speech compression technologies like MPEG.
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.
Speech Emotion Recognition is a recent research topic in the Human Computer Interaction (HCI) field. The need has risen for a more natural communication interface between humans and computer, as computers have become an integral part of our lives. A lot of work currently going on to improve the interaction between humans and computers. To achieve this goal, a computer would have to be able to distinguish its present situation and respond differently depending on that observation. Part of this process involves understanding a user‟s emotional state. To make the human computer interaction more natural, the objective is that computer should be able to recognize emotional states in the same as human does. The efficiency of emotion recognition system depends on type of features extracted and classifier used for detection of emotions. The proposed system aims at identification of basic emotional states such as anger, joy, neutral and sadness from human speech. While classifying different emotions, features like MFCC (Mel Frequency Cepstral Coefficient) and Energy is used. In this paper, Standard Emotional Database i.e. English Database is used which gives the satisfactory detection of emotions than recorded samples of emotions. This methodology describes and compares the performances of Learning Vector Quantization Neural Network (LVQ NN), Multiclass Support Vector Machine (SVM) and their combination for emotion recognition.
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.
Financial Transactions in ATM Machines using Speech SignalsIJERA Editor
Speech is the natural and simplest way of communication and Speech Recognition is a fascinating application of Digital Signal Processing which has many real-world applications. In this paper, a speech recognition system is developed for Automated Teller Machines (ATMs) using Wavelet Packet Decomposition (WPD) and Artificial Neural Networks (ANN). Speech signals are one-dimensional and are random in nature. ATM machines communicate with the customers using the stored speech samples and the user communicates with the machine using spoken digits. Daubechies wavelets are employed here. A multilayer neural network trained with back propagation training algorithm is used for classification purpose. The proposed method is implemented for 500 speakers uttering 10 spoken digits in English. The experimental results show good recognition accuracy of 87.38% and the efficiency of combining these two 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.
Speech Emotion Recognition is a recent research topic in the Human Computer Interaction (HCI) field. The need has risen for a more natural communication interface between humans and computer, as computers have become an integral part of our lives. A lot of work currently going on to improve the interaction between humans and computers. To achieve this goal, a computer would have to be able to distinguish its present situation and respond differently depending on that observation. Part of this process involves understanding a user‟s emotional state. To make the human computer interaction more natural, the objective is that computer should be able to recognize emotional states in the same as human does. The efficiency of emotion recognition system depends on type of features extracted and classifier used for detection of emotions. The proposed system aims at identification of basic emotional states such as anger, joy, neutral and sadness from human speech. While classifying different emotions, features like MFCC (Mel Frequency Cepstral Coefficient) and Energy is used. In this paper, Standard Emotional Database i.e. English Database is used which gives the satisfactory detection of emotions than recorded samples of emotions. This methodology describes and compares the performances of Learning Vector Quantization Neural Network (LVQ NN), Multiclass Support Vector Machine (SVM) and their combination for emotion recognition.
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.
Financial Transactions in ATM Machines using Speech SignalsIJERA Editor
Speech is the natural and simplest way of communication and Speech Recognition is a fascinating application of Digital Signal Processing which has many real-world applications. In this paper, a speech recognition system is developed for Automated Teller Machines (ATMs) using Wavelet Packet Decomposition (WPD) and Artificial Neural Networks (ANN). Speech signals are one-dimensional and are random in nature. ATM machines communicate with the customers using the stored speech samples and the user communicates with the machine using spoken digits. Daubechies wavelets are employed here. A multilayer neural network trained with back propagation training algorithm is used for classification purpose. The proposed method is implemented for 500 speakers uttering 10 spoken digits in English. The experimental results show good recognition accuracy of 87.38% and the efficiency of combining these two techniques
EFFECT OF DYNAMIC TIME WARPING ON ALIGNMENT OF PHRASES AND PHONEMESkevig
Speech synthesis and recognition are the basic techniques used for man-machine communication. This type
of communication is valuable when our hands and eyes are busy in some other task such as driving a
vehicle, performing surgery, or firing weapons at the enemy. Dynamic time warping (DTW) is mostly used
for aligning two given multidimensional sequences. It finds an optimal match between the given sequences.
The distance between the aligned sequences should be relatively lesser as compared to unaligned
sequences. The improvement in the alignment may be estimated from the corresponding distances. This
technique has applications in speech recognition, speech synthesis, and speaker transformation. The
objective of this research is to investigate the amount of improvement in the alignment corresponding to the
sentence based and phoneme based manually aligned phrases. The speech signals in the form of twenty five
phrases were recorded from each of six speakers (3 males and 3 females). The recorded material was
segmented manually and aligned at sentence and phoneme level. The aligned sentences of different speaker
pairs were analyzed using HNM and the HNM parameters were further aligned at frame level using DTW.
Mahalanobis distances were computed for each pair of sentences. The investigations have shown more than
20 % reduction in the average Mahalanobis distances.
DATA HIDING IN AUDIO SIGNALS USING WAVELET TRANSFORM WITH ENHANCED SECURITYcsandit
Rapid increase in data transmission over internet results in emphasis on information security.
Audio steganography is used for secure transmission of secret data with audio signal as the
carrier. In the proposed method, cover audio file is transformed from space domain to wavelet
domain using lifting scheme, leading to secure data hiding. Text message is encrypted using
dynamic encryption algorithm. Cipher text is then hidden in wavelet coefficients of cover audio
signal. Signal to Noise Ratio (SNR) and Squared Pearson Correlation Coefficient (SPCC)
values are computed to judge the quality of the stego audio signal. Results show that stego
audio signal is perceptually indistinguishable from the cover audio signal. Stego audio signal is
robust even in presence of external noise. Proposed method provides secure and least error
data extraction.
Novel Approach of Implementing Psychoacoustic model for MPEG-1 Audioinventy
Research Inventy : International Journal of Engineering and Science is published by the group of young academic and industrial researchers with 12 Issues per year. It is an online as well as print version open access journal that provides rapid publication (monthly) of articles in all areas of the subject such as: civil, mechanical, chemical, electronic and computer engineering as well as production and information technology. The Journal welcomes the submission of manuscripts that meet the general criteria of significance and scientific excellence. Papers will be published by rapid process within 20 days after acceptance and peer review process takes only 7 days. All articles published in Research Inventy will be peer-reviewed.
Implementation of Digital Hearing AID for Sensory Neural Impairmentijtsrd
Hearing impairment is a chronicle disability affecting on people in world. The hearing aid is to amplify sound to overcome a hearing loss or impairment. The hearing aid picks up the sound signal with the microphone and amplifies all frequency sound signals but the sensory neural impairment person cannot hear particular frequency of sound in a noisy environment, since the auditory nerve is damaged. In this paper we are using MATLAB to design an adaptive filter for noise removal and filter banks for amplifies the particular frequency that a person with hearing loss can listen. Navya Bharathi K S | Bindu Shree C | Dr. V Udayashankara "Implementation of Digital Hearing AID for Sensory Neural Impairment" Published in International Journal of Trend in Scientific Research and Development (ijtsrd), ISSN: 2456-6470, Volume-4 | Issue-4 , June 2020, URL: https://www.ijtsrd.com/papers/ijtsrd31133.pdf Paper Url :https://www.ijtsrd.com/engineering/bio-mechanicaland-biomedical-engineering/31133/implementation-of-digital-hearing-aid-for-sensory-neural-impairment/navya-bharathi-k-s
Audio Steganography Coding Using the Discreet Wavelet TransformsCSCJournals
The performance of audio steganography compression system using discreet wavelet transform (DWT) is investigated. Audio steganography coding is the technology of transforming stego-speech into efficiently encoded version that can be decoded in the receiver side to produce a close representation of the initial signal (non compressed). Experimental results prove the efficiency of the used compression technique since the compressed stego-speech are perceptually intelligible and indistinguishable from the equivalent initial signal, while being able to recover the initial stego-speech with slight degradation in the quality .
GENDER RECOGNITION SYSTEM USING SPEECH SIGNALIJCSEIT Journal
In this paper, a system, developed for speech encoding, analysis, synthesis and gender identification is
presented. A typical gender recognition system can be divided into front-end system and back-end system.
The task of the front-end system is to extract the gender related information from a speech signal and
represents it by a set of vectors called feature. Features like power spectrum density, frequency at
maximum power carry speaker information. The feature is extracted using First Fourier Transform (FFT)
algorithm. The task of the back-end system (also called classifier) is to create a gender model to recognize
the gender from his/her speech signal in recognition phase. This paper also presents the digital processing
of a speech signals (pronounced “A” and “B”) which are taken from 10 persons, 5 of them are Male and
the rest of them are Female. Power Spectrum Estimation of the signal is examined .The frequency at
maximum power of the English Phonemes is extracted from the estimated power spectrum. The system uses
threshold technique as identification tool. The recognition accuracy of this system is 80% on average.
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
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 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
Hiding voice data in center density of speech spectrum for secure transmissioneSAT Journals
Abstract
Speech data is becoming an effective and indispensable way for fast information transmission but associated with it are number of
unprecedented threats. The idea of this paper is to present a robust speech watermarking method to realize secure voice data
transmission. In this method carrier is transformed into frequency domain. Pre-processed normalized covert voice data
undergoes exponential transformation which is then substituted in center density of high frequency, high energy subband of
carrier. High frequency components are chosen for embedding watermark for two reasons. First, during transmission carrier
might get contaminated with noise and filteration generally suppress low frequency components so embedding in high frequency
component will keep watermark intact. Second reason is human ears are less sensitive to high frequencies so slight change in
amplitude of high frequency components is imperceptible.. Technique uses frequency masking, invisible, and blind approach also.
By applying reverse approach sensitive message can be extracted from the watermarked carrier. For embedding and retrieving
watermark secret key have been used. Experimental results have shown that proposed method does not change the size of the
cover signal even after embedding, does not degrade the quality of carrier and exhibit vigorous voice data hiding performance.
Proposed work can be used in those applications where maintaining integrity and secrecy about the information against
intentional or unintentional access is given utmost importance.
KeyWords: Index Terms- Exponential, Frequency Masking, Musical sequence, Speech watermarking, secure voice
data, signal to noise ratio
Performance Analysis of Cognitive Radio Networks (IEEE 802.22) for Various Ne...rahulmonikasharma
In nowadays the number of wireless users and applications increases, it has become more and more difficult for the proper spectrum utilization by allocate frequencies. However measurements have shown that there is no spectrum scarcity; rather, there is inefficient utilization only. Cognitive Radio (CR) to facilitate efficient utilization of the radio spectrum in a fair-minded way and to provide highly reliable communication for all users of the networks. In this paper, a simulation framework based on NetSim simulator is proposed. This framework can be used to investigate and evaluate the impact of lower layers, i.e., data link layer and physical layer. Due to the importance of packet drop probability, delay and throughput as QoS requirements in real-time reliable applications, these metrics are evaluated over Cognitive Radio Networks (CRNs) through NetSim simulator. Our simulations demonstrate that the design of new networks over CRNs should be considered based on CR-related parameters such as activity model of Primary Users(PU), Secondary Users(SU),sensing time ,spectral efficiency, throughput, delay and Interference. An Analysis of the result shows that, the CBR traffic is the best in terms of throughput and spectral efficiency when the different conditions of PUs and SUs.
Data Compression using Multiple Transformation Techniques for Audio Applicati...iosrjce
IOSR Journal of Computer Engineering (IOSR-JCE) is a double blind peer reviewed International Journal that provides rapid publication (within a month) of articles in all areas of computer engineering and its applications. The journal welcomes publications of high quality papers on theoretical developments and practical applications in computer technology. Original research papers, state-of-the-art reviews, and high quality technical notes are invited for publications.
EFFECT OF DYNAMIC TIME WARPING ON ALIGNMENT OF PHRASES AND PHONEMESkevig
Speech synthesis and recognition are the basic techniques used for man-machine communication. This type
of communication is valuable when our hands and eyes are busy in some other task such as driving a
vehicle, performing surgery, or firing weapons at the enemy. Dynamic time warping (DTW) is mostly used
for aligning two given multidimensional sequences. It finds an optimal match between the given sequences.
The distance between the aligned sequences should be relatively lesser as compared to unaligned
sequences. The improvement in the alignment may be estimated from the corresponding distances. This
technique has applications in speech recognition, speech synthesis, and speaker transformation. The
objective of this research is to investigate the amount of improvement in the alignment corresponding to the
sentence based and phoneme based manually aligned phrases. The speech signals in the form of twenty five
phrases were recorded from each of six speakers (3 males and 3 females). The recorded material was
segmented manually and aligned at sentence and phoneme level. The aligned sentences of different speaker
pairs were analyzed using HNM and the HNM parameters were further aligned at frame level using DTW.
Mahalanobis distances were computed for each pair of sentences. The investigations have shown more than
20 % reduction in the average Mahalanobis distances.
DATA HIDING IN AUDIO SIGNALS USING WAVELET TRANSFORM WITH ENHANCED SECURITYcsandit
Rapid increase in data transmission over internet results in emphasis on information security.
Audio steganography is used for secure transmission of secret data with audio signal as the
carrier. In the proposed method, cover audio file is transformed from space domain to wavelet
domain using lifting scheme, leading to secure data hiding. Text message is encrypted using
dynamic encryption algorithm. Cipher text is then hidden in wavelet coefficients of cover audio
signal. Signal to Noise Ratio (SNR) and Squared Pearson Correlation Coefficient (SPCC)
values are computed to judge the quality of the stego audio signal. Results show that stego
audio signal is perceptually indistinguishable from the cover audio signal. Stego audio signal is
robust even in presence of external noise. Proposed method provides secure and least error
data extraction.
Novel Approach of Implementing Psychoacoustic model for MPEG-1 Audioinventy
Research Inventy : International Journal of Engineering and Science is published by the group of young academic and industrial researchers with 12 Issues per year. It is an online as well as print version open access journal that provides rapid publication (monthly) of articles in all areas of the subject such as: civil, mechanical, chemical, electronic and computer engineering as well as production and information technology. The Journal welcomes the submission of manuscripts that meet the general criteria of significance and scientific excellence. Papers will be published by rapid process within 20 days after acceptance and peer review process takes only 7 days. All articles published in Research Inventy will be peer-reviewed.
Implementation of Digital Hearing AID for Sensory Neural Impairmentijtsrd
Hearing impairment is a chronicle disability affecting on people in world. The hearing aid is to amplify sound to overcome a hearing loss or impairment. The hearing aid picks up the sound signal with the microphone and amplifies all frequency sound signals but the sensory neural impairment person cannot hear particular frequency of sound in a noisy environment, since the auditory nerve is damaged. In this paper we are using MATLAB to design an adaptive filter for noise removal and filter banks for amplifies the particular frequency that a person with hearing loss can listen. Navya Bharathi K S | Bindu Shree C | Dr. V Udayashankara "Implementation of Digital Hearing AID for Sensory Neural Impairment" Published in International Journal of Trend in Scientific Research and Development (ijtsrd), ISSN: 2456-6470, Volume-4 | Issue-4 , June 2020, URL: https://www.ijtsrd.com/papers/ijtsrd31133.pdf Paper Url :https://www.ijtsrd.com/engineering/bio-mechanicaland-biomedical-engineering/31133/implementation-of-digital-hearing-aid-for-sensory-neural-impairment/navya-bharathi-k-s
Audio Steganography Coding Using the Discreet Wavelet TransformsCSCJournals
The performance of audio steganography compression system using discreet wavelet transform (DWT) is investigated. Audio steganography coding is the technology of transforming stego-speech into efficiently encoded version that can be decoded in the receiver side to produce a close representation of the initial signal (non compressed). Experimental results prove the efficiency of the used compression technique since the compressed stego-speech are perceptually intelligible and indistinguishable from the equivalent initial signal, while being able to recover the initial stego-speech with slight degradation in the quality .
GENDER RECOGNITION SYSTEM USING SPEECH SIGNALIJCSEIT Journal
In this paper, a system, developed for speech encoding, analysis, synthesis and gender identification is
presented. A typical gender recognition system can be divided into front-end system and back-end system.
The task of the front-end system is to extract the gender related information from a speech signal and
represents it by a set of vectors called feature. Features like power spectrum density, frequency at
maximum power carry speaker information. The feature is extracted using First Fourier Transform (FFT)
algorithm. The task of the back-end system (also called classifier) is to create a gender model to recognize
the gender from his/her speech signal in recognition phase. This paper also presents the digital processing
of a speech signals (pronounced “A” and “B”) which are taken from 10 persons, 5 of them are Male and
the rest of them are Female. Power Spectrum Estimation of the signal is examined .The frequency at
maximum power of the English Phonemes is extracted from the estimated power spectrum. The system uses
threshold technique as identification tool. The recognition accuracy of this system is 80% on average.
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
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 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
Hiding voice data in center density of speech spectrum for secure transmissioneSAT Journals
Abstract
Speech data is becoming an effective and indispensable way for fast information transmission but associated with it are number of
unprecedented threats. The idea of this paper is to present a robust speech watermarking method to realize secure voice data
transmission. In this method carrier is transformed into frequency domain. Pre-processed normalized covert voice data
undergoes exponential transformation which is then substituted in center density of high frequency, high energy subband of
carrier. High frequency components are chosen for embedding watermark for two reasons. First, during transmission carrier
might get contaminated with noise and filteration generally suppress low frequency components so embedding in high frequency
component will keep watermark intact. Second reason is human ears are less sensitive to high frequencies so slight change in
amplitude of high frequency components is imperceptible.. Technique uses frequency masking, invisible, and blind approach also.
By applying reverse approach sensitive message can be extracted from the watermarked carrier. For embedding and retrieving
watermark secret key have been used. Experimental results have shown that proposed method does not change the size of the
cover signal even after embedding, does not degrade the quality of carrier and exhibit vigorous voice data hiding performance.
Proposed work can be used in those applications where maintaining integrity and secrecy about the information against
intentional or unintentional access is given utmost importance.
KeyWords: Index Terms- Exponential, Frequency Masking, Musical sequence, Speech watermarking, secure voice
data, signal to noise ratio
Performance Analysis of Cognitive Radio Networks (IEEE 802.22) for Various Ne...rahulmonikasharma
In nowadays the number of wireless users and applications increases, it has become more and more difficult for the proper spectrum utilization by allocate frequencies. However measurements have shown that there is no spectrum scarcity; rather, there is inefficient utilization only. Cognitive Radio (CR) to facilitate efficient utilization of the radio spectrum in a fair-minded way and to provide highly reliable communication for all users of the networks. In this paper, a simulation framework based on NetSim simulator is proposed. This framework can be used to investigate and evaluate the impact of lower layers, i.e., data link layer and physical layer. Due to the importance of packet drop probability, delay and throughput as QoS requirements in real-time reliable applications, these metrics are evaluated over Cognitive Radio Networks (CRNs) through NetSim simulator. Our simulations demonstrate that the design of new networks over CRNs should be considered based on CR-related parameters such as activity model of Primary Users(PU), Secondary Users(SU),sensing time ,spectral efficiency, throughput, delay and Interference. An Analysis of the result shows that, the CBR traffic is the best in terms of throughput and spectral efficiency when the different conditions of PUs and SUs.
Data Compression using Multiple Transformation Techniques for Audio Applicati...iosrjce
IOSR Journal of Computer Engineering (IOSR-JCE) is a double blind peer reviewed International Journal that provides rapid publication (within a month) of articles in all areas of computer engineering and its applications. The journal welcomes publications of high quality papers on theoretical developments and practical applications in computer technology. Original research papers, state-of-the-art reviews, and high quality technical notes are invited for publications.
Speech signal compression and encryption based on sudoku, fuzzy C-means and t...IJECEIAES
Compression and encryption of speech signals are essential multimedia technologies. In the field of speech, these technologies are needed to meet the security and confidentiality of information requirements for transferring huge speech signals via a network, and for decreasing storage space for rapid retrieval. In this paper, we propose an algorithm that includes hybrid transformation in order to analyses the speech signal frequencies. The speech signal is then compressed, after removing low and less intense frequencies, to produce a well compressed speech signal and ensure the quality of the speech. The resulting compressed speech is then used as an input in a scrambling algorithm that was proposed on two levels. One of these is an external scramble that works on mixing up the segments of speech that were divided using Fuzzy C-Means and changing their locations. The internal scramble scatters the values of each block internally based on the pattern of a Sudoku puzzle and quadratic map so that the resulting speech is an input to a proposed encryption algorithm using the threefish algorithm. The proposed algorithm proved to be highly efficient in the compression and encryption of the speech signal based on approved statistical measures.
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 implementation of different audio restoration techniques for audio...eSAT Journals
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
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.
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
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.
In the recent years, large scale information transfer by remote computing and the development
of massive storage and retrieval systems have witnessed a tremendous growth. To cope up with the
growth in the size of databases, additional storage devices need to be installed and the modems and
multiplexers have to be continuously upgraded in order to permit large amounts of data transfer between
computers and remote terminals. This leads to an increase in the cost as well as equipment. One solution
to these problems is “COMPRESSION” where the database and the transmission sequence can be
encoded efficiently. In this we investigated for optimum wavelet, optimum level, and optimum scaling
factor.
Data detection with a progressive parallel ici canceller in mimo ofdmeSAT 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.
A multilevel security scheme using chaos based encryption and steganography f...eSAT Journals
Abstract Steganography is the practice of encoding secret information in indiscernible way. Audio steganography is the science of hiding some secret text or audio information in a host audio. This paper contains audio stegnography using bit modification of time domain audio samples which is a well known simple technique for multimedia data embedding with potential for large payload But before applying stegnography or hiding secret audio in cover audio a chaos based encryption is performed on secret audio data in order to increase the security against steganalysis. And the bit modification technique is also not the simple LSB (least significant bit) modification technique. It contains two approaches which make the steganalysis more difficult as compare to simple LSB technique. First approach is with selection of bits of sample for embedding secret audio while the other approach is to use the compliment of the secret audio before hiding it in the host message. The improvised proposed approach works against steganalysis and decreases the probability of secret audio being extracted by an intruder. Chaos based encryption is used to secure secret audio in case the stegnography technique breaks. Keywords: Audio steganography, Chaos based encryption, steganalysis, and LSB modification technique
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.
In the present-day communications speech signals get contaminated due to
various sorts of noises that degrade the speech quality and adversely impacts
speech recognition performance. To overcome these issues, a novel approach
for speech enhancement using Modified Wiener filtering is developed and
power spectrum computation is applied for degraded signal to obtain the
noise characteristics from a noisy spectrum. In next phase, MMSE technique
is applied where Gaussian distribution of each signal i.e. original and noisy
signal is analyzed. The Gaussian distribution provides spectrum estimation
and spectral coefficient parameters which can be used for probabilistic model
formulation. Moreover, a-priori-SNR computation is also incorporated for
coefficient updation and noise presence estimation which operates similar to
the conventional VAD. However, conventional VAD scheme is based on the
hard threshold which is not capable to derive satisfactory performance and a
soft-decision based threshold is developed for improving the performance of
speech enhancement. An extensive simulation study is carried out using
MATLAB simulation tool on NOIZEUS speech database and a comparative
study is presented where proposed approach is proved better in comparison
with existing technique.
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%.
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.
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.
CFD Simulation of By-pass Flow in a HRSG module by R&R Consult.pptxR&R Consult
CFD analysis is incredibly effective at solving mysteries and improving the performance of complex systems!
Here's a great example: At a large natural gas-fired power plant, where they use waste heat to generate steam and energy, they were puzzled that their boiler wasn't producing as much steam as expected.
R&R and Tetra Engineering Group Inc. were asked to solve the issue with reduced steam production.
An inspection had shown that a significant amount of hot flue gas was bypassing the boiler tubes, where the heat was supposed to be transferred.
R&R Consult conducted a CFD analysis, which revealed that 6.3% of the flue gas was bypassing the boiler tubes without transferring heat. The analysis also showed that the flue gas was instead being directed along the sides of the boiler and between the modules that were supposed to capture the heat. This was the cause of the reduced performance.
Based on our results, Tetra Engineering installed covering plates to reduce the bypass flow. This improved the boiler's performance and increased electricity production.
It is always satisfying when we can help solve complex challenges like this. Do your systems also need a check-up or optimization? Give us a call!
Work done in cooperation with James Malloy and David Moelling from Tetra Engineering.
More examples of our work https://www.r-r-consult.dk/en/cases-en/
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.
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1. IJRET: International Journal of Research in Engineering and Technology eISSN: 2319-1163 | pISSN: 2321-7308
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Volume: 03 Issue: 01 | Jan-2014, Available @ http://www.ijret.org 538
SPEECH COMPRESSION ANALYSIS USING MATLAB
Manas Arora1
, Neha Maurya2
, Poonam Pathak3
, Vartika Singh4
1, 2
M.Tech Student, Digital Communication, BBDNIIT, Uttar Pradesh, India,
3
Associate Professor, 4
Sr. Lecturer, Electronics & Comm., BBDNIIT, Uttar Pradesh, India
Abstract
The growth of the cellular technology and wireless networks all over the world has increased the demand for digital information by
manifold. This massive demand poses difficulties for handling huge amounts of data that need to be stored and transferred. To
overcome this problem we can compress the information by removing the redundancies present in it. Redundancies are the major
source of generating errors and noisy signals. Coding in MATLAB helps in analyzing compression of speech signals with varying bit
rate and remove errors and noisy signals from the speech signals. Speech signal’s bit rate can also be reduced to remove error and
noisy signals which is suitable for remote broadcast lines, studio links, satellite transmission of high quality audio and voice over
internet This paper focuses on speech compression process and its analysis through MATLAB by which processed speech signal can
be heard with clarity and in noiseless mode at the receiver end .
Keywords: Speech compression, bit rate, filter, MPEG, DCT.
----------------------------------------------------------------------***----------------------------------------------------------------------
1. INTRODUCTION
Data compression is a technique in which data content of the
input signal to system is compressed so that original signal is
obtained as output and unwanted or undesired signals are
removed. Therefore when speech signals are used in the form
of data it is termed as SPEECH COMPRESSION.
Speech is a very basic way for humans to convey information
to one another. Speech has a small bandwidth of 4 kHz. Speech
compression involves coding of real-time audio signals at the
lowest possible bit rates. The compression of speech signals has
many practical applications. It is used in digital cellular
technology where many users share the same frequency
bandwidth. Compression allows more users to share the system
at a particular time. It can also be used for digital voice storage
that are used for answering machines and pre-recorded
telephone calls that are used for purpose of providing any kind
of information to user or advertising. For a given memory size,
compression allows longer messages to be stored than
otherwise [1].
2. SPEECH COMPRESSION
Speech compression enables efficient storage and transmission
of data. There may be varying amounts of compression in data
according to the sampling rate used. This gives different levels
of system complexity and compressed quality of speech data.
The recorded waveform which is compressed can be
transmitted with or without loss. Therefore, there are two types
of compression: lossy and lossless. The digital audio data is
processed through mixing, filtering and equalization. The
speech signal is fed into an encoder that uses fewer bits than
original audio data bit rate. This results in reducing the
transmission bandwidth of digital audio streams and also
reduces storage size of audio files. [15] Compression may be
lossy or lossless. Lossy compression is transparent to human
audibility but lossless being have a compressing factor from 6
to 1. [15].An uncoded speech signal is [17]
Fig 1: Uncoded speech signal
At the decoder, bit stream decoding bit is done which is
followed by frequency sample reconstruction. Finally
frequency-to-time mapping produces decoded speech signal.
DCT is used to compress speech signals. [15].
Discrete Cosine Transform (DCT) is a main transform or
mapping method used which maps the time domain into
frequency domain. A DCT expresses a finite sequence of data
points in terms of a sum of cosine functions oscillating at
different frequencies. The discrete cosine transform is a linear,
2. IJRET: International Journal of Research in Engineering and Technology eISSN: 2319-1163 | pISSN: 2321-7308
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Volume: 03 Issue: 01 | Jan-2014, Available @ http://www.ijret.org 539
invertible function defined as F: RN RN (where R denotes
the set of real numbers), or equivalently an
invertible N × N square matrix. Mathematically DCT is
calculated from (1) [2]
1,.......0
2
1
cos
1
0
Nk
N
kn
xX
N
n
nk
(1)
The cosine graphs define the nature of frequency used in DCT
transform and define the frequency of the speech signal. The
cosine graphs define the window function. The window
function has cosine function as (2).
(2)
Basic cosine graphs used in the discrete cosine transform are:
Fig 2: Cosine graphs used in DCT
The speech compression using DCT method is an efficient
method by which we can modify the frequency of the signal
and analyze the signals in different frequency ranges in a fixed
frequency domain. DCT also effectively compresses the speech
signals and maintain the audibility feature of speech signals.
3. NEED OF SPEECH COMPRESSION
A basic question arises what is the need of speech
compression? Answer lies within the requirement of
appropriate bit rate for high quality speech signal. Bit rate is
defined as number of sample per second which is given by
Rbits calculates from (3) where fs is the sampling frequency
and n describes the number of bits.
nfR Sbits (3)
The requirement of achieving high bit rate and clarity of audio
signals basically defines the need of speech compression[14]. A
given frequency bandwidth can be used by a number of users at
a time and moreover, larger size speech data files can be stored.
4. SPEECH COMPRESSION TECHNIQUE IN
MPEG TECHNOLOGY
A speech signal is a complex signal which has a wide spectrum
and MPEG helps in perceptual phenomena for our ears. The
large frequencies and smaller frequencies below masking
threshold are inaudible to human ears. MPEG standard helps in
defining standard and filtering audio signals in the available
bandwidth and maintains signal to quantization ratio. In
MPEG-1 audio compression is performed as given in figure 1
and involves two process. Filter bank which uses filters divides
spectrum of incoming signals in sub bands. The quantizer
quantizes the sub bands. [10]
Fig 3: Block Diagram of MPEG-1 Encoder
The basic method by which speech compression in MPEG
technology is done is by utilizing an MPEG encoder and
decompression is performed by MPEG decoder. Figure 3 shows
block diagrams of the MPEG/speech encoder and decoder. [11,
12]
In MPEG, compression of the input stream passes through a
filter bank which divides the input into multiple sub bands. The
input is simultaneously also passed through a psychoacoustic
model which is used to determine the signal-to-mask ratio of
each sub band. The bit or noise allocation block utilizes the
output of psychoacoustic model and a decision is made about
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Volume: 03 Issue: 01 | Jan-2014, Available @ http://www.ijret.org 540
the total number of code bits which are available for
quantization process [3, 13].
Code bits also help in removing quantization noise. The output
of compressor is quantized audio samples and these formats of
the data are transformed into a decodable bit stream [3, 13].
In MPEG, decompression of the speech signals simply reverses
the formatting and then reconstruction of quantized code bits is
done then reconstructs the quantized sub band values, which
are finally transformed into the set of sub band values into a
time-domain audio signal [3, 13].
SPEECH ENCODER
Fig 4: Speech Encoder block diagram
SPEECH DECODER
Fig 5: Speech Decoder block diagram
The MPEG speech standard has three distinct layers for
compression.
Layer I is the basic algorithm
Layers II and III are enhancements of Layer I. Each
successive layer improves the compression performance
at the increasing cost of encoder and decoder complexity
[14].
4.1 Layer I:
The Layer I algorithm uses the basic filter bank which is found
in all layers. This filter bank divides the audio signal into 32
constant-width frequency bands. The filters are simple and
provide time and frequency resolutions by which it is easily
perceived by human ears. The design of Layer I has three
concessions. First, the 32 constant width bands which do not
accurately reflect the ear’s critical frequency bands. Second, the
filter bank and its inverse are not lossless transformations.
Thirdly, adjacent filter bands have a significant frequency
overlap [4, 5, 6, 7, 8, 15]
4.2 Layer II:
The Layer II algorithm is a simple enhancement of Layer I. It
improves compression performance by coding data in larger
groups. By the Layer II encoder forms frames of 3 by 12 by 32
that is total of 1152 samples per audio channel as compared to
Layer I codes data in single groups of 12 samples for each sub
band .Layer II removes stereo redundancy coding and there is
one bit allocation The encoder encodes with a unique scale
factor for each group of 12 samples to avoid audible distortion.
The Layer II algorithm improves performance as it uses
efficient code for representing the bit allocation, the scale factor
values, and the quantized samples. [4, 5, 6, 7, 8, 15]
4.3 Layer III:
The Layer III algorithm is a much more refined
approach.[16,17] though it is based on the same filter bank
found in Layers I and II. Layer III compensates for some filter
bank deficiencies by processing the filter outputs with a
modified discrete cosine transform (MDCT) [4, 5, 6, 7, 8, 15].
These all the three layers completely design the methods of
audio compression in MPEG technology.
5. IMPLEMENTATION IN MATLAB
MATLAB is a useful tool which is used to analyze speech
signals which are read in >wav format. Following commands
are used to analyze.
wavread: it reads speech signal
windowsize: defines window function of transformation
wavplay: it produces speech signal after transformation
length: defines length of speech to be processed by
transforming principle
dct: performs discrete cosine transform
idct: performs inverse discrete cosine transform
6. OUTPUT IN MATLAB
In speech signal analyzing through MATLAB we obtain
following spectrums
Speech Signals with different Amplitudes(as input data)
Portion of signal according to length and window-size
Speech spectrograms
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Volume: 03 Issue: 01 | Jan-2014, Available @ http://www.ijret.org 541
Fig 6: Speech Signals with different Amplitudes
Fig 7: Portion of speech signal to be processed
Fig8: Speech Spectrograms
7. PRESENT SCENARIO
MPEG technology is an ever developing technology in the field
of communication in which speech compression is finding its
space and utility. Currently speech compression is used in
satellite communications, DBS TV’s, TATA SKY, dish
television network which uses MPEG-4,MPEG-1,MPEG-2 and
MPEG-DASH. MPEG-D is being used in audio technologies
like MPEG SURROUND, SAOC (Spatial Audio Object
Coding) and USAC (UNIFIED SPEECH AND AUDIO
CODING)..Speech compression has dynamic adaptive
streaming over HTTP (DASH) technology in the current era of
communication. It is also being widely used in cellular
technology that includes mobile telephony and Voice over IP.
[16]
CONCLUSIONS
Speech compression is a standard for designing and
compressing audio and speech signals which are transmitted to
the receiver end. MPEG technology and its standards define the
compression techniques and provide means for filtering noisy
or undesired signals. Hence, we obtain noise-free speech signal
after decompression of compressed speech signal.
ACKNOWLEDGEMENTS
We are thankful to our college and guides Mrs.Poonam Pathak
and Miss.Vartika Singh respectively for providing time to time
help in studying the topic and providing us a background to
understand it in deeper details
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Volume: 03 Issue: 01 | Jan-2014, Available @ http://www.ijret.org 542
REFERENCES
[1] Sinha D.,Tewhik A.H, ”Low Bit Rate Transparent
Audio Compression using Adapted Wavelets”, IEEE
TRANSACTIONS ON SIGNAL PROCESSING.
VOL. 41. NO. 12. DECEMBER 1993
[2] Salomon D, "Video Compression: Data compression”
Springer ISBN 978-1-84628-602-5,2007
[3] http://mpeg.chiariglione.org/
[4] ISO/IEC TR 23000-1:2007 – Information technology
– Multimedia application format (MPEG-A) – Part 1:
Purpose for multimedia application formats
[5] ISO. "ISO/IEC 23002-1:2006 – Information
technology – MPEG video technologies – Part 1:
Accuracy requirements for implementation of integer-
output 8x8 inverse discrete cosine transform
[6] ISO. "ISO/IEC 23003-1:2007 – Information
technology – MPEG audio technologies – Part 1:
MPEG Surround
[7] ISO. "ISO/IEC 23004-1:2007 – Information
technology – Multimedia Middleware – Part 1:
Architecture
[8] ISO. "ISO/IEC 29116-1:2008 – Information
technology – Supplemental media technologies – Part
1: Media streaming application format protocols
[9] http://mpeg.chiariglione.org/working_documents.htm#
MPEG-V
[10] Satellite Communications,Seventh Edition
2009,Agarwal D.C,Khana Publications,p-p 368-369
[11] Davis Yan Pan,”Digital Audio Compression”, Digital
Technical Journal Vol. 5 No. 2, Spring 1993
[12] K. Brandenburg and J. D. Johnston, "Second
Generation Perceptual Audio Coding: The Hybrid
Coder," Preprint 2937, 88th Audio Engineering
Society Convention, Montreaux (1990).
[13] K. Brandenburg, J. Herre, J. D. Johnston, Y. Mahieux,
and E. Schroeder, "ASPEC: Adaptive Spectral
Perceptual Entropy Coding of High Quality Music
Signals," Preprint 3011, 90th Audio Engineering
Society Convention, Paris (1991)
[14] Pan, D “A tutorial on MPEG/audio compression”
MultiMedia, IEEE (Volume:2 , Issue: 2 ),2002
[15] Gersho, A. “Advances in speech and audio
compression” Proceedings of the IEEE Volume:82
, Issue: 6 ,August 2002
[16] M. Arjona Ramírez,M. Minami, “Technology and
standards for low-bit-rate vocoding methods,” in The
Handbook of Computer Networks, H. Bidgoli, Ed.,
New York: Wiley, 2008, vol. 2, pp. 447–467
[17] Joshi Anil Kumar,mehta Ashish, “Audio
Compression using Logarithmic Approach for PSNR
enhancement”
BIOGRAPHIES
Mr. Manas Arora, is a M.TECH 2nd
year
student of BBDNIIT college and completed my
B.TECH from SRMCEM,LKO with 77.4%
(HONOURS ) in the year 2012.
Ms. Neha Maurya is a M.TECH 2nd
year
student of BBDNIIT college and completed
B.TECH from SRMCEM,LKO with 74.16%
in the year 2012.
Mrs. Poonam Pathak,is presently Head of
Department of Electronics & Communication
Department at BBDNIIT.She has done her
MTECH in DIGITAL ELECTRONICS &
SYSTEMS.
Ms Vartika Singh is presently working in
BBDNIIT as a senior lecturer. She has done
her Bachelors and Masters Degree in
Electronics and Communication Engineering
from Amity University, Uttar Pradesh.
Presently, she has focused her working area within the various
aspects of the image processing using some suitable simulation
software in the vicinity of implementation of some highly
efficient and accurate algorithms for digital watermarking.Her
research interests also include cooperative communications,
free-space optical communications and wireless
communications