• Title/Summary/Keyword: Narrowband

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A New Wideband Speech/Audio Coder Interoperable with ITU-T G.729/G.729E (ITU-T G.729/G.729E와 호환성을 갖는 광대역 음성/오디오 부호화기)

  • Kim, Kyung-Tae;Lee, Min-Ki;Youn, Dae-Hee
    • Journal of the Institute of Electronics Engineers of Korea SP
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    • v.45 no.2
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    • pp.81-89
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    • 2008
  • Wideband speech, characterized by a bandwidth of about 7 kHz (50-7000 Hz), provides a substantial quality improvement in terms of naturalness and intelligibility. Although higher data rates are required, it has extended its application to audio and video conferencing, high-quality multimedia communications in mobile links or packet-switched transmissions, and digital AM broadcasting. In this paper, we present a new bandwidth-scalable coder for wideband speech and audio signals. The proposed coder spits 8kHz signal bandwidth into two narrow bands, and different coding schemes are applied to each band. The lower-band signal is coded using the ITU-T G.729/G.729E coder, and the higher-band signal is compressed using a new algorithm based on the gammatone filter bank with an invertible auditory model. Due to the split-band architecture and completely independent coding schemes for each band, the output speech of the decoder can be selected to be a narrowband or wideband according to the channel condition. Subjective tests showed that, for wideband speech and audio signals, the proposed coder at 14.2/18 kbit/s produces superior quality to ITU-T 24 kbit/s G.722.1 with the shorter algorithmic delay.

The Study on the Performance of DS/CDMA with a Suppressed Pilot Channel in Mobile Satellite Communication System (이동위성 통신 시스템에서 억압 파일롯트 채널을 이용한 DS / CDMA의 성능 분석)

  • Chung, Boo-Young;Choi, Bong-Keun;Kang, Young-Heung;Lee, Jin
    • The Journal of Korean Institute of Electromagnetic Engineering and Science
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    • v.8 no.2
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    • pp.151-160
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    • 1997
  • In this paper, we have carried out the DS/CDMA with a suppressed pilot channel, which is used in receiving coherently with Rake diversity and in synchronizing the chip timing, in the mobile satellite communication. Also, we have investigated the envelope variation of a shadowed Rician fading simulator, and analyzed the error performences of DS/CDMA in the mobile satellite communication. The results showed that the error performance in the Heavy shadowing environment might be degraded more than in the Rayleigh fading environment since the fading envelopes in the former environment are varied randomly compared with those in the latter environment. And the performence of DS/CDMA system could be improved about 10 dB compared with that of narrowband QPSK system. In conclusion, DS/CDMA with a suppressed pilot channel had the best performance in the case of the suppressed pilot channel to transmission power ratio $\beta$=-8 dB, the number of complex delay profiles $N_{profile}$=32, and using these values, the error performance of DS/CDMA in Light shadowing environment was identical to the ideal QPSK error performance.

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An Algorithm for Submarine Passive Sonar Simulator (잠수함 수동소나 시뮬레이터 알고리즘)

  • Jung, Young-Cheol;Kim, Byoung-Uk;An, Sang-Kyum;Seong, Woo-Jae;Lee, Keun-Hwa;Hahn, Joo-Young
    • The Journal of the Acoustical Society of Korea
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    • v.32 no.6
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    • pp.472-483
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    • 2013
  • Actual maritime exercise for improving the capability of submarine sonar operator leads to a lot of cost and constraints. Sonar simulator maximizes the capability of sonar operator and training effect by solving these problems and simulating a realistic battlefield environment. In this study, a passive sonar simulator algorithm is suggested, where the simulator is divided into three modules: maneuvering module, noise source module, and sound propagation module. Maneuvering module is implemented in three-dimensional coordinate system and time interval is set as the rate of vessel changing course. Noise source module consists of target noise, ocean ambient noise, and self noise. Target noise is divided into modulated/unmodulated and narrowband/broadband signals as their frequency characteristics, and they are applied to ship radiated noise level depending on the vessel tonnage and velocity. Ocean ambient noise is simulated depending on the wind noise considering the waveguide effect and other ambient noise. Self noise is also simulated for flow noise and insertion loss of sonar-dome. The sound propagation module is based on ray propagation, where summation of amplitude, phase, and time delay for each eigen-ray is multiplied by target noise in the frequency domain. Finally, simulated results based on various scenarios are in good agreement with generated noise in the real ocean.

Total Degradation Performance Evaluation of the Time- and Frequency-Domain Clipping in OFDM Systems (OFDM 시스템에서 시간 및 주파수 영역 클리핑의 Total Degradation 성능평가)

  • Han, Chang-Sik;Seo, Man-Jung;Im, Sung-Bin
    • Journal of the Institute of Electronics Engineers of Korea TC
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    • v.44 no.7 s.361
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    • pp.17-22
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    • 2007
  • OFDM (Orthogonal Frequency Division Multiplexing) is a special case of multicarrier transmission, where a single data stream is transmitted over a number of lower-rate subcarrier. One of the main reasons to use OFDM is to increase robustness against frequency-selective fading or narrowband interference. Unfortunately, an OFDM signal consists of a number of independently modulated subcarriers, which can give a large PAPR (Peak-to-Average Power Ratio) when added up coherently. In this paper, we investigate the performance of a simple PAPR reduction scheme, which requires no change of a receiver structure or no additional information transmission. The approach we employed is clipping in the time and frequency domains. The time-domain clipping is carried out with a predetermined clipping level while the frequency-domain clipping is done within EVM (Error Vector Magnitude). This approach is suboptimal with lower computational complexity compared to the optimal method. This evaluation is carried out on the OFDM system with an nonlinear amplifier. The simulation results demonstrated that the PAPR reduction algorithm is one of ways to reduce the effects of the nonlinear distortion of an HPA (High Power Amplifier).

A VHF/UHF-Band Variable Gain Low Noise Amplifier for Mobile TV Tuners (모바일 TV 튜너용 VHF대역 및 UHF 대역 가변 이득 저잡음 증폭기)

  • Nam, Ilku;Lee, Ockgoo;Kwon, Kuduck
    • Journal of the Institute of Electronics and Information Engineers
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    • v.51 no.12
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    • pp.90-95
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    • 2014
  • This paper presents a VHF/UHF-band variable gain low noise amplifier for multi-standard mobile TV tuners. A proposed VHF-band variable gain amplifier is composed of a resistive shunt-feedback low noise amplifier to remove external matching components, a single-to-differential amplifier with input PMOS transcoductors to improve low frequency noise performance, a variable shunt-feedback resistor and an attenuator to control variable gain range. A proposed UHF-band variable gain amplifier consists of a narrowband low noise amplifier with capacitive tuning to improve noise performance and interference rejection performance, a single-to-differential with gm gain control and an attenuator to adjust gain control range. The proposed VHF-band and UHF-band variable gain amplifier were designed in a $0.18{\mu}m$ RF CMOS technology and draws 22 mA and 17 mA from a 1.8 V supply voltage, respectively. The designed VHF-band and UHF-band variable gain amplifier show a voltage gain of 27 dB and 27 dB, a noise figure of 1.6-1.7 dB and 1.3-1.7 dB, OIP3 of 13.5 dBm and 16 dBm, respectively.

An acoustic sensor fault detection method based on root-mean-square crossing-rate analysis for passive sonar systems (수동 소나 시스템을 위한 실효치교차율 분석 기반 음향센서 결함 탐지 기법)

  • Kim, Yong Guk;Park, Jeong Won;Kim, Young Shin;Lee, Sang Hyuck;Kim, Hong Kook
    • The Journal of the Acoustical Society of Korea
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    • v.36 no.1
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    • pp.30-38
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    • 2017
  • In this paper, we propose an underwater acoustic sensor fault detection method for passive sonar systems. In general, a passive sonar system displays processed results of array signals obtained from tens of the acoustic sensors as a two-dimensional image such as displays for broadband or narrowband analysis. Since detection result display in the operation software is to display the accumulated result through the array signal processing, it is difficult to determine the possibility where signal may be contaminated by the fault or failure of a single channel sensor. In this paper, accordingly, we propose a detection method based on the analysis of RMSCR (Root Mean Square Crossing-Rate), and the processing techniques for the faulty sensors are analyzed. In order to evaluate the performance of the proposed method, the precision of detecting fault sensors is measured by using signals acquired from real array being operated in several coastal areas. Besides, we compare performance of fault processing techniques. From the experiments, it is shown that the proposed method works well in underwater environments with high average RMS, and mute (set to zero) shows the best performance with regard to fault processing techniques.

Improvement of Microphone Away Performance in the Low Frequencies Using Modulation Technique (변조 기법을 이용한 마이크로폰 어레이의 저주파 대역 특성 개선)

  • Kim, Gi-Bak;Cho, Nam-Ik
    • Journal of the Institute of Electronics Engineers of Korea SP
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    • v.42 no.4 s.304
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    • pp.111-118
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    • 2005
  • In this paper, we employ the modulation technique for improving the characteristics of beamformer in the low frequencies and thus improving the overall noise reduction performance. In the 1-dimensional uniform linear microphone arrays, we can suppress the narrowband noise component using the delay-and-sum beamforming. But, for the wideband noise signal, the delay-and-sum beamformer does not work well for the reduction of low frequency component because the inter-element spacing is usually set to avoid spatial aliasing at high frequencies. Hence, the beamwidth is not uniform with respect to each frequency and it is usually wider at the low frequencies. In order to obtain the beamwidth independent of frequencies, subarray systems[1][2][3][4] and multi-beamforming[5] have been proposed. However these algorithms need large space and more microphones since they are based on the theory that the size of the array is proportional to the wavelength of the input signal. In the proposed beamformer, we reduce the low frequency noise by using modulation technique that does not need additional sensors or non-uniform spacing. More Precisely, the array signals are split into subbands, and the low frequency components are shifted to high frequencies by modulation and reduced by the delay-and-sum beamforming techniques with small size microphone array. Experimental results show that the proposed technique Provides better performance than the conventional ones, especially in the low frequency band.

Analysis of echolocation click signals of Indo-Pacific bottlenose dolphin (Tursiops aduncus) in Jeju Island (제주 남방큰돌고래 반향정위 클릭음 특성 분석)

  • Choi, Kang-Hoon;Yoon, Young Geul;Kim, Sunhyo;Kim, Hyeonsu;Choi, Jee Woong;Bae, Ho Seuk;Park, Kyeongju
    • The Journal of the Acoustical Society of Korea
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    • v.38 no.1
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    • pp.56-65
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    • 2019
  • The Indo-Pacific bottlenose dolphins (Tursiops aduncus) are a toothed whale using echolocation clicks and have been studied continuously abroad. However, most studies on this whale's clicks were performed about captive animals and, in Korea, only the studies of the whistle sounds of this animal were done. In this paper, acoustic characteristics of clicks were analyzed about the free-ranging Indo-Pacific bottlenose dolphins living in the coast of the Jeju Island. Acoustic parameters such as signal duration, 1st and 2nd peak frequency, 3 dB and 10 dB bandwidth for acoustic characteristics were calculated and compared with those of Australian species. As a result, the signal durations had average of $38{\mu}s$ and most clicks were within range of $20{\mu}s-60{\mu}s$. The two types of bandwidths showed both narrowband and broadband characteristics, and bimodal signal characteristics were confirmed through the 1st peak frequencies(average of $96kHz{\pm}18kHz$) and the 2nd peak frequencies(average of $69kHz{\pm}19kHz$).

Implementation of Heat Control System using NB-IoT (NB-IoT를 활용한 발열 제어 시스템 구현)

  • Shin, DongKeun;Kim, HyungJin
    • The Journal of Korea Institute of Information, Electronics, and Communication Technology
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    • v.12 no.2
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    • pp.135-141
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    • 2019
  • Internet of thing becomes more active, many sensor devices are increasing. Sensors can use network wired network or use mobile communication network. From the viewpoint of the transmission rate, the mobile communication network can be roughly divided into two types of high-speed communication and low-speed communication. In the case of hundreds of millions of sensors in the mobile communication network, resources are wasted to use high-speed communication. Communication is required to reduce the transmission rate and appropriately allocate resources without wasting such resources. As the Internet of Thing has been activated, Narrowband Internet of Thing(NB-IoT), which is one of the low-power technologies in recent mobile communications, is in the spotlight from various companies. Currently, it can be seen that only NB-IoT or other low power consumption communication has the potential to be able to connect to the Internet with rapidly increasing sensor devices. In this paper, we designed and implemented a heater controller using Huawei NB-IoT communication Module, a server that collects controller information, and an application that allows default settings for devices. The main function of this system is to collect temperature and heater status and give it to the server, control the heater from the server, and set parameters for the heater to operate automatically. The system can be applied to places where wired communication is not established, such as road information, smart agriculture, and small reservoirs as well as heaters.

Geoacoustic Inversion and Source Localization with an L-Shaped Receiver Array (L-자형 선배열을 이용한 지음향학적 인자 역산 및 음원 위치 추정)

  • Kim, Kyung-Seop;Lee, Keun-Hwa;Kim, Seong-Il;Kim, Young-Gyu;Seong, Woo-Jae
    • The Journal of the Acoustical Society of Korea
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    • v.25 no.7
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    • pp.346-355
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    • 2006
  • Acoustic data from a shallow water experiment in the East Sea of Korea (MAPLE IV) is Processed to investigate the Performance of matched-field geo-acoustic inversion and source localization. The receiver array consists of two legs as in an L-shape. one vertical and the other horizontal lying on the seabed. Narrowband multi-tone CW source was towed along a slightly inclined bathymetry track. The matched-field geo-acoustic inversion includes comparisons between three processing techniques. all based on the Bartlett processor as; (1) the coherent processing of the data from the full array, (2) the incoherent Product of each output from both the horizontal and vertical arrays, and (3) the cross correlation between the horizontal and vertical arrays. as well as processing each array leg separately. To verify the inversion results. matched-field source localization for low level source signal components were performed using the same Processors used at the inversion stage.