2024-03-28T14:33:35Z
http://ijet.pl/index.php/ijet/oai
oai:ojs.ijet.ise.pw.edu.pl:article/1687
2019-09-01T21:32:26Z
ijet:DSP
Localization of Copy-Move Forgery in speech signals through watermarking using DCT-QIM
Lalitha, N.V.
Srinivasa Rao, Ch.
JayaSree, P.V.Y.
watermarking; copy-move forgery; Discrete Cosine Transform; Quantization Index Modulation; Hash bits.
Digital speech copyright protection and forgery identification are the prevalent issues in our advancing digital world. In speech forgery, voiced part of the speech signal is copied and pasted to a specific location which alters the meaning of the speech signal. Watermarking can be used to safe guard the copyrights of the owner. To detect copy-move forgeries a transform domain watermarking method is proposed. In the proposed method, watermarking is achieved through Discrete Cosine Transform (DCT) and Quantization Index Modulation (QIM) rule. Hash bits are also inserted in watermarked voice segments to detect Copy-Move Forgery (CMF) in speech signals. Proposed method is evaluated on two databases and achieved good imperceptibility. It exhibits robustness in detecting the watermark and forgeries against signal processing attacks such as resample, low-pass filtering, jittering, compression and cropping. The proposed work contributes for forensics analysis in speech signals. This proposed work also compared with the some of the state-of-art methods.
Electronics and Telecommunications Committee
2019-09-01
info:eu-repo/semantics/article
info:eu-repo/semantics/publishedVersion
application/pdf
http://ijet.pl/index.php/ijet/article/view/10.24425-ijet.2019.129809
International Journal of Electronics and Telecommunications; Vol 65, No 3 (2019); 527-532
2300-1933
eng
http://ijet.pl/index.php/ijet/article/view/10.24425-ijet.2019.129809/603
Copyright (c) 2019 International Journal of Electronics and Telecommunications
oai:ojs.ijet.ise.pw.edu.pl:article/3381
2022-05-31T02:27:45Z
ijet:DSP
Construction of generalized Rademacher functions in terms of ternary logic: solving the problem of visibility of using Galois fields for digital signal processing
Matrassulova, Dinara
Vitulyova, Elizaveta
Suleimenov, Ibragim
Using generalized Rademacher functions, constructed as a sequence of elements of Galois fields , and intended to find the spectral representation of signals with n levels, the expediency of using the concept of "logical imaginary unit" is substantiated. These functions form a complete basis on the interval corresponding to 3^n -1 discrete time intervals and for n=1 passing into the classical Rademacher functions. The advantage of such spectra obtained using Galois Fields Fourier Transform is that the range of variation of the spectrum amplitudes remains the same as the range of variation of the original signal, which is modeled on discrete time functions taking values in the Galois field.
Electronics and Telecommunications Committee
2022-05-31
info:eu-repo/semantics/article
info:eu-repo/semantics/publishedVersion
application/pdf
http://ijet.pl/index.php/ijet/article/view/10.24425-ijet.2022.139873
International Journal of Electronics and Telecommunications; Vol 68, No 2 (2022); 237-244
2300-1933
eng
http://ijet.pl/index.php/ijet/article/view/10.24425-ijet.2022.139873/960
http://ijet.pl/index.php/ijet/article/downloadSuppFile/10.24425-ijet.2022.139873/2894
http://ijet.pl/index.php/ijet/article/downloadSuppFile/10.24425-ijet.2022.139873/2896
http://ijet.pl/index.php/ijet/article/downloadSuppFile/10.24425-ijet.2022.139873/3086
http://ijet.pl/index.php/ijet/article/downloadSuppFile/10.24425-ijet.2022.139873/3087
Copyright (c) 2022 International Journal of Electronics and Telecommunications
oai:ojs.ijet.ise.pw.edu.pl:article/1446
2018-07-20T19:13:02Z
ijet:DSP
Optimization of the spectrum of digital diagnostic signals to improve the analysis of harmonic parameters using resampling algorithms
Jarmołowicz, Marcin
Kornatowski, Eugeniusz
The paper discusses the influence of resampling distortions on the quality of spectral resolution optimization in diagnostic signals digitally recorded for objects in a steady state. Analysis of harmonic parameters and detection of foreign frequencies, which are most often interpreted as fault results, may be problematic because of the spectral leakage effect. The reason is that some frequencies which are not actually present in the signal can be observed in the DFT (discrete Fourier transform) vector. In addition, actual existing frequencies may be distorted. The use of window functions most often reduces the effect of spectral leakage, but also increases harmonic distortion. When the signal contains only the fundamental frequency and harmonics, it is possible to adjust its spectral resolution to eliminate any distortion for regular frequencies. If the algorithms which perform the analysis of the diagnostic signal require a fixed number samples, high quality resampling should be used.
Electronics and Telecommunications Committee
2018-07-20
info:eu-repo/semantics/article
info:eu-repo/semantics/publishedVersion
application/pdf
http://ijet.pl/index.php/ijet/article/view/10.24425-123527
International Journal of Electronics and Telecommunications; Vol 64, No 3 (2018); 335-341
2300-1933
eng
http://ijet.pl/index.php/ijet/article/view/10.24425-123527/479
http://ijet.pl/index.php/ijet/article/downloadSuppFile/10.24425-123527/1191
Copyright (c) 2018 International Journal of Electronics and Telecommunications
oai:ojs.ijet.ise.pw.edu.pl:article/2501
2020-06-01T14:11:57Z
ijet:DSP
On Derivation of Discrete Time Fourier Transform from Its Continuous Counterpart
Borys, Andrzej Marek
sampling of signals; relation between discrete and continuous time Fourier transforms
This paper is devoted to some problems that appear in derivations of the discrete time Fourier transform from a formula for its continuous time counterpart for transformation from the time into the frequency domain as well as to those regarding transformation in the inverse direction. In particular, the latter ones remained so far an unresolved problem. It is solved for the first time here. Many detailed explanations accompanying the solution found are presented. Finally, it is also worth noting that our derivations do not exploit any of such sophisticated mathematical tools as the so-called Dirac delta and Dirac comb.
Electronics and Telecommunications Committee
2020-06-01
info:eu-repo/semantics/article
info:eu-repo/semantics/publishedVersion
application/pdf
http://ijet.pl/index.php/ijet/article/view/10.24425-ijet.2020.131885
International Journal of Electronics and Telecommunications; Vol 66, No 2 (2020); 355-360
2300-1933
eng
http://ijet.pl/index.php/ijet/article/view/10.24425-ijet.2020.131885/702
http://ijet.pl/index.php/ijet/article/downloadSuppFile/10.24425-ijet.2020.131885/1911
http://ijet.pl/index.php/ijet/article/downloadSuppFile/10.24425-ijet.2020.131885/2216
Copyright (c) 2020 International Journal of Electronics and Telecommunications
oai:ojs.ijet.ise.pw.edu.pl:article/3303
2022-03-08T12:08:34Z
ijet:DSP
Effect of time-domain windowing on isolated speech recognition system performance
Thalengala, Ananthakrishna
Hoblidar, Anitha
Tumkur, Girisha S
Hidden Markov model (HMM); Isolated speech recognition (ISR) system; Kannada language; Mono-phone model; Mel frequency cepstral coefficients (MFCC).
Speech recognition system extract the textual data from the speech signal. The research in speech recognition domain is challenging due to the large variabilities involved with the speech signal. Variety of signal processing and machine learning techniques have been explored to achieve better recognitionaccuracy. Speech is highly non-stationary in nature and therefore analysis is carried out by considering short time-domain window or frame. In the speech recognition task, cepstral (Mel frequency cepstral coefficients (MFCC)) features are commonly used and are extracted for short time-frame. The effectiveness of features depend upon duration of the time-window chosen. The present study is aimed at investigation of optimal time-window duration for extraction of cepstral features in the context of speech recognition task. A speaker independent speech recognition system for the Kannada language has been considered for the analysis. In the current work, speech utterances of Kannada news corpusrecorded from different speakers have been used to create speech database. The hidden Markov tool kit (HTK) has been used to implement the speech recognition system. The MFCC along with their first and second derivative coefficients are considered as feature vectors. Pronunciation dictionary required for the studyhas been built manually for mono-phone system. Experiments have been carried out and results have been analyzed for different time-window lengths. The overlapping Hamming window has been considered in this study. The best average word recognition accuracy of 61.58% has been obtained for a window length of 110 msec duration. This recognition accuracy is comparable with the similar work found in literature. The experiments have shown that best word recognition performance can be achieved by tuning the window length to its optimum value.
Electronics and Telecommunications Committee
2022-02-26
info:eu-repo/semantics/article
info:eu-repo/semantics/publishedVersion
application/pdf
http://ijet.pl/index.php/ijet/article/view/10.24425-ijet.2022.139856
International Journal of Electronics and Telecommunications; Vol 68, No 1 (2022); 161-166
2300-1933
eng
http://ijet.pl/index.php/ijet/article/view/10.24425-ijet.2022.139856/942
http://ijet.pl/index.php/ijet/article/downloadSuppFile/10.24425-ijet.2022.139856/2833
http://ijet.pl/index.php/ijet/article/downloadSuppFile/10.24425-ijet.2022.139856/2836
Copyright (c) 2022 International Journal of Electronics and Telecommunications
oai:ojs.ijet.ise.pw.edu.pl:article/1294
2019-02-16T00:04:57Z
ijet:DSP
PCA Assisted DTCWT Denoising for Improved DOA Estimation of Closely Spaced and Coherent Signals
Ganage, Dharmendra Gokuldas
Ravinder, Yerram
Electronics & Telecommunication; Signal Processing; Mobile Communication
Performance of standard Direction of Arrival (DOA) estimation techniques degraded under real-time signal conditions. The classical algorithms are Multiple Signal Classification (MUSIC), and Estimation of Signal Parameters via Rotational Invariance Technique (ESPRIT). There are many signal conditions hamper on its performance, such as closely spaced and coherent signals caused due to the multipath propagations of signals results in a decrease of the signal to noise ratio (SNR) of the received signal. In this paper, a novel DOA estimation technique named CW-PCA MUSIC is proposed using Principal Component Analysis (PCA) to threshold the nearby correlated wavelet coefficients of Dual-Tree Complex Wavelet transform (DTCWT) for denoising the signals before applying to MUSIC algorithm. The proposed technique improves the detection performance under closely spaced, and coherent signals with relatively low SNR conditions. Also, this method requires fewer snapshots, and less antenna array elements compared with standard MUSIC and wavelet-based DOA estimation algorithms.
Electronics and Telecommunications Committee
2019-02-16
info:eu-repo/semantics/article
info:eu-repo/semantics/publishedVersion
application/pdf
http://ijet.pl/index.php/ijet/article/view/10.24425-ijet.2019.126290
International Journal of Electronics and Telecommunications; Vol 65, No 1 (2019); 111-117
2300-1933
eng
http://ijet.pl/index.php/ijet/article/view/10.24425-ijet.2019.126290/528
Copyright (c) 2019 International Journal of Electronics and Telecommunications
oai:ojs.ijet.ise.pw.edu.pl:article/2532
2020-09-10T21:07:54Z
ijet:DSP
Filtering Property of Signal Sampling in General and Under-Sampling as a Specific Operation of Filtering Connected with Signal Shaping at the Same Time
Borys, Andrzej Marek
signal sampling; filtering; discrete-time Fourier transform
In this paper, we show that signal sampling operation can be considered as a kind of all-pass filtering in the time domain, when the Nyquist frequency is larger or equal to the maximal frequency in the spectrum of a signal sampled. We demonstrate that this seemingly obvious observation has wide-ranging implications. They are discussed here in detail. Furthermore, we discuss also signal shaping effects that occur in the case of signal under-sampling. That is, when the Nyquist frequency is smaller than the maximal frequency in the spectrum of a signal sampled. Further, we explain the mechanism of a specific signal distortion that arises under these circumstances. We call it the signal shaping, not the signal aliasing, because of many reasons discussed throughout this paper. Mainly however because of the fact that the operation behind it, called also the signal shaping here, is not a filtering in a usual sense. And, it is shown that this kind of shaping depends upon the sampling phase. Furthermore, formulated in other words, this operation can be viewed as a one which shapes the signal and performs the low-pass filtering of it at the same time. Also, an interesting relation connecting the Fourier transform of a signal filtered with the use of an ideal low-pass filter having the cut frequency lying in the region of under-sampling with the Fourier transforms of its two under-sampled versions is derived. This relation is presented in the time domain, too.
Electronics and Telecommunications Committee
2020-09-07
info:eu-repo/semantics/article
info:eu-repo/semantics/publishedVersion
application/pdf
http://ijet.pl/index.php/ijet/article/view/10.24425-ijet.2020.134016
International Journal of Electronics and Telecommunications; Vol 66, No 3 (2020); 589-594
2300-1933
eng
http://ijet.pl/index.php/ijet/article/view/10.24425-ijet.2020.134016/736
http://ijet.pl/index.php/ijet/article/downloadSuppFile/10.24425-ijet.2020.134016/2234
Copyright (c) 2020 International Journal of Electronics and Telecommunications
oai:ojs.ijet.ise.pw.edu.pl:article/665
2017-08-14T21:46:22Z
ijet:DSP
State Space-Based Method for the DOA Estimation by the Forward-Backward Data Matrix Using Small Snapshots
Liu, Jianfeng
In this presentation, a new low computational burden method for the direction of arrival (DOA) estimation from noisy signal using small snapshots is presented. The approach introduces State Space-based Method (SSM) to represent the received array signal, and uses small snapshots directly to form the Hankel data matrix. Those Hankel data matrices are then utilized to construct forward-backward data matrix that is used to estimate the state space model parameters from which the DOA of the incident signals can be extracted. In contrast to existing methods, such as MUSIC, Root-MUSIC that use the covariance data matrix to estimate the DOA and the sparse representation (SR) based DOA which is obtained by solving the sparsest representation of the snapshots, the SSM algorithm employs forward-backward data matrix formed only using small snapshots and doesn't need additional spatial smoothing method to process coherent signals. Three numerical experiments are employed to compare the performance among the SSM, Root-MUSIC and SR-based method as well as Cramér–Rao bound (CRB). The simulation results demonstrate that when a small number of snapshots, even a single one, are used, the SSM always performs better than the other two method no matter under the circumstance of uncorrelated or correlated signal. The simulation results also show that the computational burden is reduced significantly and the number of antenna elements is saved greatly.
Electronics and Telecommunications Committee
2017-08-14
info:eu-repo/semantics/article
info:eu-repo/semantics/publishedVersion
application/pdf
http://ijet.pl/index.php/ijet/article/view/10.1515-eletel-2017-0042
International Journal of Electronics and Telecommunications; Vol 63, No 3 (2017); 315-322
2300-1933
eng
http://ijet.pl/index.php/ijet/article/view/10.1515-eletel-2017-0042/404
Copyright (c) 2017 International Journal of Electronics and Telecommunications
oai:ojs.ijet.ise.pw.edu.pl:article/3943
2023-05-18T20:31:22Z
ijet:DSP
Audio Compression using a Modified Vector Quantization algorithm for Mastering Applications
Prince, Shajin
D, Bini
Kirubaraj A, Alfred
Immanuel J, Samson
M, Surya
vector quantization; scalable;perceptual coder; audio mastering; bit stream
Audio data compression is used to reduce the transmission bandwidth and storage requirements of audio data. It is the second stage in the audio mastering process with audio equalization being the first stage. Compression algorithms such as BSAC, MP3 and AAC are used as standards in this paper. The challenge faced in audio compression is compressing the signal at low bit rates. The previous algorithms which work well at low bit rates cannot be dominant at higher bit rates and vice-versa. This paper proposes an altered form of vector quantization algorithm which produces a scalable bit stream which has a number of fine layers of audio fidelity. This modified form of the vector quantization algorithm is used to generate a perceptually audio coder which is scalable and uses the quantization and encoding stages which are responsible for the psychoacoustic and arithmetical terminations that are actually detached as practically all the data detached during the prediction phases at the encoder side is supplemented towards the audio signal at decoder stage. Therefore, clearly the quantization phase which is modified to produce a bit stream which is scalable. This modified algorithm works well at both lower and higher bit rates. Subjective evaluations were done by audio professionals using the MUSHRA test and the mean normalized scores at various bit rates was noted and compared with the previous algorithms.
Electronics and Telecommunications Committee
2023-05-18
info:eu-repo/semantics/article
info:eu-repo/semantics/publishedVersion
application/pdf
http://ijet.pl/index.php/ijet/article/view/10.24425-ijet.2023.144363
International Journal of Electronics and Telecommunications; Vol 69, No 2 (2023); 287-292
2300-1933
eng
http://ijet.pl/index.php/ijet/article/view/10.24425-ijet.2023.144363/1094
http://ijet.pl/index.php/ijet/article/downloadSuppFile/10.24425-ijet.2023.144363/3719
http://ijet.pl/index.php/ijet/article/downloadSuppFile/10.24425-ijet.2023.144363/3720
Copyright (c) 2023 International Journal of Electronics and Telecommunications
oai:ojs.ijet.ise.pw.edu.pl:article/1845
2019-10-21T18:07:30Z
ijet:DSP
Analysis of High-Performance Near-threshold Dual Mode Logic design
Bikki, Pavankumar
CMOS logic; dual mode logic; dynamic mode; high performance; minimum energy point; near-threshold
A novel dual mode logic (DML) model has a superior energy-performance compare to CMOS logic. The DML model has unique feature that allows switching between both modes of operation as per the real-time system requirements. The DML functions in two dissimilar modes (static and dynamic) of operation with its specific features, to selectively obtain either low-energy or high-performance. The sub-threshold region DML achieves minimum-energy. However, sub-threshold region consequence in performance is enormous. In this paper, the working of DML model in the moderate inversion region has been explored. The near-threshold region holds much of the energy saving of sub-threshold designs, along with improved performance. Furthermore, robustness to supply voltage and sensitivity to the process temperature variations are presented. Monte carol analysis shows that the projected near-threshold region has minimum energy along with the moderate performance.
Electronics and Telecommunications Committee
NO
2019-10-07
info:eu-repo/semantics/article
info:eu-repo/semantics/publishedVersion
application/pdf
http://ijet.pl/index.php/ijet/article/view/10.24425-ijet.2019.130253
International Journal of Electronics and Telecommunications; Vol 65, No 4 (2019); 723-729
2300-1933
eng
http://ijet.pl/index.php/ijet/article/view/10.24425-ijet.2019.130253/633
http://ijet.pl/index.php/ijet/article/downloadSuppFile/10.24425-ijet.2019.130253/1448
Copyright (c) 2019 International Journal of Electronics and Telecommunications
oai:ojs.ijet.ise.pw.edu.pl:article/2754
2021-01-31T00:11:06Z
ijet:DSP
Design of sparse FIR filters with low group delay
Konopacki, Jacek
Engineering and technology; Electronic engineering; Telecommunications engineering;
The aim of the work is to present the method for designing sparse FIR filters with very low group delay and approximately linear-phase in the passband. Significant reduction of the group delay, e.g. several times in relation to the linear phase filter, may cause the occurrence of undesirable overshoot in the magnitude frequency response. The method proposed in this work consists of two stages. In the first stage, FIR filter with low group delay is designed using minimax constrained optimization that provides overshoot elimination. In the second stage, the same process is applied iteratively to reach sparse solution. Design examples demonstrate the effectiveness of the proposed method.
Electronics and Telecommunications Committee
Ministry of Science and Higher Education
2021-01-31
info:eu-repo/semantics/article
info:eu-repo/semantics/publishedVersion
application/pdf
http://ijet.pl/index.php/ijet/article/view/10.24425-ijet.2021.135953
International Journal of Electronics and Telecommunications; Vol 67, No 1 (2021); 121-126
2300-1933
eng
http://ijet.pl/index.php/ijet/article/view/10.24425-ijet.2021.135953/787
http://ijet.pl/index.php/ijet/article/downloadSuppFile/10.24425-ijet.2021.135953/2256
http://ijet.pl/index.php/ijet/article/downloadSuppFile/10.24425-ijet.2021.135953/2257
http://ijet.pl/index.php/ijet/article/downloadSuppFile/10.24425-ijet.2021.135953/2259
Copyright (c) 2021 International Journal of Electronics and Telecommunications
oai:ojs.ijet.ise.pw.edu.pl:article/1110
2018-02-01T12:59:01Z
ijet:DSP
The implementation of the parallel scrambler scheme for the IEEE 802.11 standard
Kudinov, Alexey
Antimirov, Yaroslav
Tyshchenko, Igor
Popova, Mariia
Cherepanov, Alexander
This article is devoted to the development of the scrambler circuit. Nowadays, new WiFi standard IEEE 802.11 is being put into operation, so that there is a huge need in modern, energy-efficient algorithms, which will be used in the data transmission. Consequently, some of the scrambler circuits, which could be implemented for the IEEE 802.11 standard are described with its comparison. In addition, an example in Python is given for readers to use it in their researches.
Electronics and Telecommunications Committee
The Russian Ministry of Education and Science
2018-01-31
info:eu-repo/semantics/article
info:eu-repo/semantics/publishedVersion
application/pdf
http://ijet.pl/index.php/ijet/article/view/10.24425-118151
International Journal of Electronics and Telecommunications; Vol 64, No 1 (2018); 91-94
2300-1933
eng
http://ijet.pl/index.php/ijet/article/view/10.24425-118151/442
Copyright (c) 2018 International Journal of Electronics and Telecommunications
oai:ojs.ijet.ise.pw.edu.pl:article/4263
2023-10-28T09:18:15Z
ijet:DSP
Development Of A Monitoring System For Electric Power Substations Based On IoТ And Implementation Of Designs On FPGA
Kalimoldayev, Maksat
Wójcik, Waldemar
Shermantayeva, Zhazira
Абстрактный:В данной статье разработана система мониторинга на основе IoT-технологий электрической системы подстанции в Республике Казахстан. На данный момент работа энергосистем крайне важна для поддержания частоты электрического тока с течением времени. Для приложений управления и мониторинга необходимо учитывать связь в допустимых пределах. Технологии IoT рассматриваются основными функциями в приложениях для мониторинга и управления энергосистемами в режиме реального времени, а также принятия эффективных решений как по техническим, так и по финансовым вопросам системы, для мониторинга основной формы регистрации данных на электрической подстанции в городе Шымкент Республики Казахстан, для последовательного эффективного принятия решений системными операторами. В данной работе была реализована и внедрена система мониторинга на базе интернета вещей для подстанции энергосистемы с использованием специализированного устройства, встроенного в контроллер ПЛИС для быстрой комплексной цифровизации трансформаторных подстанций распределительных электрических сетей реального времени. Платформа IoT также обеспечивает полную удаленную наблюдаемость и повысит надежность работы операторов энергосистемы в режиме реального времени. Эта статья в основном направлена на предоставление практического приложения, которое было реализовано и протестировано.
Electronics and Telecommunications Committee
KAZAKH NATIONAL UNIVERSITY NAMED AFTER AL-FARABI
2023-10-28
info:eu-repo/semantics/article
info:eu-repo/semantics/publishedVersion
application/pdf
http://ijet.pl/index.php/ijet/article/view/10.24425-ijet.2023.147708
International Journal of Electronics and Telecommunications; Vol 69, No 4 (2023); 819-824
2300-1933
eng
http://ijet.pl/index.php/ijet/article/view/10.24425-ijet.2023.147708/1168
http://ijet.pl/index.php/ijet/article/downloadSuppFile/10.24425-ijet.2023.147708/4169
http://ijet.pl/index.php/ijet/article/downloadSuppFile/10.24425-ijet.2023.147708/4287
http://ijet.pl/index.php/ijet/article/downloadSuppFile/10.24425-ijet.2023.147708/4288
http://ijet.pl/index.php/ijet/article/downloadSuppFile/10.24425-ijet.2023.147708/4290
Copyright (c) 2023 International Journal of Electronics and Telecommunications
oai:ojs.ijet.ise.pw.edu.pl:article/1987
2019-09-01T21:32:26Z
ijet:DSP
Performance Evaluation of Audio Coding by Amalgam AAC and FLAC Audio codec using MDCT and INTMDCT Algorithm
Kamala Dhas, M. Davidson
Priyadharsini, R.
The MDCT and IntMDCT Algorithm is widely utilized is Audio coding.By lifting scheme or rounding operation IntegerMDCT is evolved from Modified Discrete Cosine Transform.This method acquire the properties of MDCT and contribute excelling invertiblity and good spectral mean.In this paper we discuss about the audio codec like AAC and FLAC using MDCT and Integer MDCT algorithm and to find which algorithm shows better Compression Ratio(CR).The confines of this task is to hybriding lossy and lossless audio codec with diminished bit rate but with finer sound quality. Certainly the quality of the audio is figure out by Subjective and Objective testing which is in terms of MOS (Mean opinion square) , ABx and some of the hearing aid testing methodology like PEAQ(Perceptual Evaluation Audio Quality) and ODG(Objective Difference Grade)is followed. Execution measure, that is Compression Ratio(CR) and Sound Pressure Level (SPL) is approximated.
Electronics and Telecommunications Committee
2019-09-01
info:eu-repo/semantics/article
info:eu-repo/semantics/publishedVersion
application/pdf
http://ijet.pl/index.php/ijet/article/view/10.24425-ijet.2019.129810
International Journal of Electronics and Telecommunications; Vol 65, No 3 (2019); 533-539
2300-1933
eng
http://ijet.pl/index.php/ijet/article/view/10.24425-ijet.2019.129810/604
http://ijet.pl/index.php/ijet/article/downloadSuppFile/10.24425-ijet.2019.129810/1622
Copyright (c) 2019 International Journal of Electronics and Telecommunications
oai:ojs.ijet.ise.pw.edu.pl:article/2705
2021-12-19T22:28:56Z
ijet:DSP
Detection and localization of audio event for home surveillance using CRNN
Suruthhi, V.S.
Smita, V.
Rolant Gini, J.
Ramachandran, K. I.
Convolutional Recurrent Neural Network (CRNN); Gated Recurrent Unit (GRU); Long Short-Term Memory (LSTM); Sound event localization and detection (SELD)
Safety and security have been a prime priority in people’s lives, and having a surveillance system at home keeps people and their property more secured. In this paper, an audio surveillance system has been proposed that does both the detection and localization of the audio or sound events. The combined task of detecting and localizing the audio events is known as Sound Event Localization and detection (SELD). The SELD in this work is executed through Convolutional Recurrent Neural Network (CRNN) architecture. CRNN is a stacked layer of convolutional neural network (CNN), recurrent neural network (RNN) and fully connected neural network (FNN). The CRNN takes multichannel audio as input, extracts features and does the detection and localization of the input audio events in parallel. The SELD results obtained by CRNN with the gated recurrent unit (GRU) and with long short-term memory (LSTM) unit are compared and discussed in this paper. The SELD results of CRNN with LSTM unit gives 75% F1 score and 82.8% frame recall for one overlapping sound. Therefore, the proposed audio surveillance system that uses LSTM unit produces better detection and overall performance for one overlapping sound.
Electronics and Telecommunications Committee
2021-12-01
info:eu-repo/semantics/article
info:eu-repo/semantics/publishedVersion
application/pdf
http://ijet.pl/index.php/ijet/article/view/0.24425-ijet.2021.139771
International Journal of Electronics and Telecommunications; Vol 67, No 4 (2021); 735-741
2300-1933
eng
http://ijet.pl/index.php/ijet/article/view/0.24425-ijet.2021.139771/906
http://ijet.pl/index.php/ijet/article/downloadSuppFile/0.24425-ijet.2021.139771/2169
Copyright (c) 2021 International Journal of Electronics and Telecommunications
oai:ojs.ijet.ise.pw.edu.pl:article/1225
2018-04-27T08:36:48Z
ijet:DSP
Inertial navigation static calibration
Niespodziany, Slawomir
Signal processing
Inertial navigation is a device, which estimatesits position, based on sensing external conditions (such asacceleration or angular velocity). It is widely used in variuosapplications. Its presence in a drone vehicle for example, allowsflight stabilization, by position estimation and feedback-basedregulation algorithm execution. A smartphone makes a use ofinertial navigation by detecting movement and flipping screenorientation. It is a ubiquitous part of many devices of everydayuse, but before using filters and algorithms allowing to calculatethe position, a calibration must first be applied to the device. Thispaper focuses on a separate calibration of each of the sensors- an accelerometer, gyroscope and magnetometer. The furtherstep requires a cross–sensor calibration, and the third step isimplementation of data filtration algotithm.
Electronics and Telecommunications Committee
2018-04-27
info:eu-repo/semantics/article
info:eu-repo/semantics/publishedVersion
application/pdf
http://ijet.pl/index.php/ijet/article/view/10.24425-119518
International Journal of Electronics and Telecommunications; Vol 64, No 2 (2018); 243-248
2300-1933
eng
http://ijet.pl/index.php/ijet/article/view/10.24425-119518/464
Copyright (c) 2018 International Journal of Electronics and Telecommunications