• Title/Summary/Keyword: Wideband Filter

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A Study on the Performance Improvement with Subband Overlapping Variation for Overlapped Multicarrier DS-CDMA Systems (중복된 멀티캐리어 DS-CDMA 시스템의 서브밴드 중복율 변화에 따른 성능개선에 관한 연구)

  • O, Jeong-Heon;Park, Gwang-Cheol;Kim, Gi-Du
    • Journal of the Institute of Electronics Engineers of Korea TC
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    • v.37 no.9
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    • pp.11-23
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    • 2000
  • Multicarrier DS-CDMA is an effective approach to realize wideband CDMA system in a multipath fading channel. In this paper, we propose a convolutionally-coded overlapped multicarrier DS-CDMA system, and analyze the performance with subband overlapping variation to determine the overlapping percentage showing best performance. Given a total number of subcarriers M*R, we will show that the BER variation is highly dependent on the rolloff factor P of raised-cosine chip wave-shaping filter irrespective of convolutional encoding rate I/M and repetition coding rate 1/R. We also analyze the possibility of reduction in total MUI by considering both variation of a rolloff factor (0 ($\beta$ :1) and variation of subband overlapping factor (0 ( A :2), and show that the proposed system may outperform the multicarrier DS-CDMA system in [1, 12].

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Real-Time Implementation of Wideband Adaptive Multi Rate (AMR-WB) Speech Codec Using TMS32OC6201 (TMS320C6201을 이용한 적응 다중 전송율을 갖는 광대역 음성부호화기의 실시간 구현)

  • Lee, Seung-Won;Bae, Keun-Sung
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.29 no.9C
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    • pp.1337-1344
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    • 2004
  • This paper deals with analysis and real-time Implementation of a wide band adaptive multirate speech codec (AMR-WB) using a fixed-point DSP of TI's TMS320C6201. In the AMR-WB codec, input speech is divided into two frequency bands, lower and upper bands, and processed independently. The lower band signal is encoded based on the ACELP algorithm and the upper band signal is processed using the random excitation with a linear prediction synthesis filter. The implemented AMR-WB system used 218 kbytes of program memory and 92 kbytes of data memory. And its proper operation was confirmed by comparing a decoded speech signal sample-by-sample with that of PC-based simulation. Maximum required time of 5 75 ms for processing a frame of 20 ms of speech validates real-time operation of the Implemented system.

Development of the Infrared Space Telescope, MIRIS

  • Han, Won-Yong;Lee, Dae-Hee;Park, Young-Sik;Jeong, Woong-Seob;Ree, Chang-Hee;Nam, Uk-Won;Moon, Bon-Kon;Park, Sung-Joon;Cha, Sang-Mok;Pyo, Jeong-Hyun;Park, Jang-Hyun;Ka, Nung-Hyun;Seon, Kwang-Il;Lee, Duk-Hang;Rhee, Seung-Woo;Park, Jong-Oh;Lee, Hyung-Mok;Matsumoto, Toshio
    • The Bulletin of The Korean Astronomical Society
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    • v.36 no.1
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    • pp.64.1-64.1
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    • 2011
  • MIRIS (Multipurpose Infra-Red Imaging System), is a small infrared space telescope which is being developed by KASI, as the main payload of Science and Technology Satellite 3 (STSAT-3). Two wideband filters (I and H) of the MIRIS enables us to study the cosmic infrared background by detecting the absolute background brightness. The narrow band filter for Paschen ${\alpha}$ emission line observation will be employed to survey the Galactic plane for the study of warm ionized medium and interstellar turbulence. The opto-mechanical design of the MIRIS is optimized to operate around 200K for the telescope, and the cryogenic temperature around 90K for the sensor in the orbit, by using passive and active cooling technique, respectively. The engineering and qualification model of the MIRIS has been fabricated and successfully passed various environmental tests, including thermal, vacuum, vibration and shock tests. The flight model was also assembled and is in the process of system optimization to be launched in 2012 by a Russian rocket. The mission operation scenario and the data reduction software is now being developed. After the successful mission of FIMS (the main payload of STSAT-1), MIRIS is the second Korean space telescope, and will be an important step towards the future of Korean space astronomy.

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A Study on Adaptive Interference Canceller of Wireless Repeater for Wideband Code Division Multiple Access System (WCDMA시스템 무선 중계기의 적응간섭제거기에 관한 연구)

  • Han, Yong-Sik;Yang, Woon-Geun
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.13 no.7
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    • pp.1321-1327
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    • 2009
  • In this paper, as the mobile communication service is widely used and the demand for wireless repeaters is rapidly increasing because of the easiness of extending service areas. But a wireless repeater has a problem the oscillation due to feedback signal. We proposed a new hybrid interference canceller using the adaptive filter with CMA(Constant Modulus Algorithm)-Grouped LMS(Least Mean Square) algorithm in the adaptive interference canceller. The proposed interference canceller has better channel adaptive performance and a lower MSE(Mean Square Error) than conventional structure because it uses the cancellation method of Grouped LMS algorithm. The proposed detector uses the LMS algorithms with two different step size to reduce mean square error and to obtain fast convergence. This structure reduces the number of iterations for the same MSE performance and hardware complexity compared to conventional nonlinear interference canceller.

Design of UWB CMOS Low Noise Amplifier Using Inductor Peaking Technique (인덕터 피킹기법을 이용한 초광대역 CMOS 저잡음 증폭기 설계)

  • Sung, Young-Kyu;Yoon, Kyung-Sik
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.17 no.1
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    • pp.158-165
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    • 2013
  • In this paper, a new circuit topology of an ultra-wideband (UWB) 3.1-10.6GHz CMOS low noise amplifier is presented. The proposed UWB low noise amplifier is designed utilizing RC feedback and LC filter networks which can provide good input impedance matching. In this design, the current-reused topology is adopted to reduce the power consumption and the inductor-peaking technique is applied for the purpose of bandwidth extension. The performance results of this UWB low noise amplifier simulated in $0.18-{\mu}m$ CMOS process technology exhibit a power gain of 14-14.9dB, an input matching of better than -10.8dB, gain flatness of 0.9dB, and a noise figure of 2.7-3.3dB in the frequency range of 3.1-10.6GHz. In addition, the input IP3 is -5dBm and the power consumption is 12.5mW.

Analysis and Design of High Efficiency Feedforward Amplifier Using Distributed Element Negative Group Delay Circuit (분산 소자 형태의 마이너스 군지연 회로를 이용한 고효율 피드포워드 증폭기의 분석 및 설계)

  • Choi, Heung-Jae;Kim, Young-Gyu;Shim, Sung-Un;Jeong, Yong-Chae;Kim, Chul-Dong
    • The Journal of Korean Institute of Electromagnetic Engineering and Science
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    • v.21 no.6
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    • pp.681-689
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    • 2010
  • We will demonstrate a novel topology for the feedforward amplifier. This amplifier does not use a delay element thus providing an efficiency enhancement and a size reduction by employing a distributed element negative group delay circuit. The insertion loss of the delay element in the conventional feedforward amplifier seriously degrades the efficiency. Usually, a high power co-axial cable or a delay line filter is utilized for a low loss, but the insertion loss, cost and size of the delay element still acts as a bottleneck. The proposed negative group delay circuit removes the necessity of the delay element required for a broadband signal suppression loop. With the fabricated 2-stage distributed element negative group delay circuit with -9 ns of total group delay, a 0.2 dB of insertion loss, and a 30 MHz of bandwidth for a wideband code division multiple access downlink band, the feedforward amplifier with the proposed topology experimentally achieved a 19.4 % power added efficiency and a -53.2 dBc adjacent channel leakage ratio with a 44 dBm average output power.

Estimation of Medical Ultrasound Attenuation using Adaptive Bandpass Filters (적응 대역필터를 이용한 의료 초음파 감쇠 예측)

  • Heo, Seo-Weon;Yi, Joon-Hwan;Kim, Hyung-Suk
    • Journal of the Institute of Electronics Engineers of Korea SC
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    • v.47 no.5
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    • pp.43-51
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    • 2010
  • Attenuation coefficients of medical ultrasound not only reflect the pathological information of tissues scanned but also provide the quantitative information to compensate the decay of backscattered signals for other medical ultrasound parameters. Based on the frequency-selective attenuation property of human tissues, attenuation estimation methods in spectral domain have difficulties for real-time implementation due to the complexicity while estimation methods in time domain do not achieve the compensation for the diffraction effect effectively. In this paper, we propose the modified VSA method, which compensates the diffraction with reference phantom in time domain, using adaptive bandpass filters with decreasing center frequencies along depths. The adaptive bandpass filtering technique minimizes the distortion of relative echogenicity of wideband transmit pulses and maximizes the signal-to-noise ratio due to the random scattering, especially at deeper depths. Since the filtering center frequencies change according to the accumulated attenuation, the proposed algorithm improves estimation accuracy and precision comparing to the fixed filtering method. Computer simulation and experimental results using tissue-mimicking phantoms demonstrate that the distortion of relative echogenicity is decreased at deeper depths, and the accuracy of attenuation estimation is improved by 5.1% and the standard deviation is decreased by 46.9% for the entire scan depth.

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.