• Journal of electromagnetic engineering and science
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• v.9 no.2
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• pp.71-77
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• 2009

OFDM(Orthogonal Frequency Division Multiplexing) signals sutler serious nonlinear distortion in the nonlinear HPA(High Power Amplifier) because of high PAPR(Peak Average Power Ratio). Nonlinear distortion can be improved by a pre-distorter, but this pre-distorter is insufficient when the PAPR is very high in an OPFDM system. In this paper, a DFT(Discrete Fourier Transform) transform technique is introduced for PAPR reduction. It is especially important to consider the memory effect of HPA for more precise predistortion. Therefore, in this paper, we consider two models, the TWTA(Traveling-Wave Tube Amplifier) model of Saleh without a memory effect and the HPA memory polynomial model that has a memory effect. We design a pre-distorter and an adaptive pre-distorter that uses the NLMS(Normalized Least Mean Square) algorithm for the compensation of this nonlinear distortion. Without the consideration of a memory effect, the system performance would be degraded, even if the pre-distorter is used for the compensation of the nonlinear distortion. From the simulation results, we can confirm that the proposed system shows an improvement in performance.

• Journal of the Institute of Electronics Engineers of Korea TE
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• v.37 no.5
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• pp.36-47
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• 2000

The CDMA digital cellular systems provide better voice Quality than analog systems, however there exists inherent delays due to speech coding and transmission processing, which brings echoes returned by the BSC and PSTN interface. In this paper, we show the performance improvement of a proposed echo canceller by real time implementation, where Block Update NLMS algorithm is applied into the TMS320C54X DSP. By applying the proposed method into the practical mobile phone, we verify that various types of echoes (LE, ESE, AE) may be removed more precisely. We also cope with echo path change resulting from change of delay length after taking VAD to find echo path delay.

• Proceedings of the Korean Institute of Navigation and Port Research Conference
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• v.2
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• pp.353-357
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• 2006

One-sided and two-sided ATF for GNSS receiver are deigned, implemented and evaluated in this paper. The difference f filter characteristics such as the location of zeros and the frequency response is reviewed and examined with experiments. NLMS adaptation algorithm is adopted for updating the weighting coefficients of the 12-tap FIR filter. he performance of ATF is evaluated using real signals consisting of the signals from GPS simulator and the signal generator. The output of ATF is fed into the SDR to evaluate SNR and the position accuracy. The complexity of implementation is also compared and the effects of the time delay and the phase delay are examined. The experimental results show that one-sided and two-sided ATF give similar performance against single tone CWI.

• Proceedings of the Acoustical Society of Korea Conference
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• 1997.06a
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• pp.23-26
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• 1997

스테레오 반향시호 제거기에 있어서 두 입력 스테레요 신호 사이에는 높은 상호상관으로 이하여 적응 필터의 수렴특성이 저한된다. 실제 확성통화 회의시스템인 경우, 입력 스테레요 신호사이에는 환경의 변화등에 의하여 미소하게 상호상관이 변하게 되지만, 기존의 적응 알고리즘으로는 이러한 미소 변화에 빠르게 적응하지 못하여 반향신호 제거량이 충분하지 못하다. 본 연구에서는 스테레요 반향신호 제거기에 상호상관의 미소한 변화를 강조할 수 있는 Projection 알고리즘을 이용하였다. 실제 확성통화 회의시스템을 구성하여 녹음한 음성신호에 대하여 시뮬레이션을 행하여 LMS , NLMS 알고리즘과 비교.평가하였다.

• Proceedings of the IEEK Conference
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• 2002.06d
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• pp.413-416
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• 2002

In this paper, a speech enhancement system using microphone array with adaptive Post-Processing is implemented in real-lime with TMS320C6201 DSP. It consists of delay-and-sum beamformer and adaptive post-processing filters with NLMS (Normalized Least Mean Square) algorithm. THS1206 ADC is used for collection of 4-channel microphone signals. Sizes of program memory, data ROM and data RAM of the implemented system are 15,744, 748 and 47,540 bytes, respectively. Finally 21.839${\times}$106 clocks per second is required for real-time operation.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.2C
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• pp.222-228
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• 2004

This paper describes a new adaptive algorithm which can reduce the required computation quantities in the adaptive filter. The proposed adaptive algorithm uses only the signs of the normalized input signal rather than the input signals when coefficients of the filter are adapted. By doing so, there is no need for the multiplications and divisions which are mostly responsible for the computation quantities. To analyze the convergence characteristics of the proposed algorithm, the condition and speed of the convergence are derived mathematically. Also, we simulate an echo canceller adopting this algorithm and compare the performances of convergence for this algorithm with the ones for the other algorithm. As the results of simulations, it is proved that the echo canceller adopting this algorithm shows almost the same performances of convergence as the echo canceller adopting the SIA algorithm.

• The Journal of the Acoustical Society of Korea
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• v.23 no.3E
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• pp.79-84
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• 2004

In this paper, the AEC(acoustic echo canceller) is designed and implemented using VHDL(VHSIC hardware description language). The designed Echo Canceller employs the pipeline and the master-slave structure, and is realized with FPGA. As an adaptive algorithm, the Normalized LMS algorithm is used. For the coefficient adjustment, the Stochastic Iteration Algorithm(SIA) which uses only current residual values is used and the number of registers are evidently reduced and convergence speed is also much improved comparing to existing methods by using EAB of FPGA for FIR filter structure of transceiver. The designed Echo Canceller is verified with the test board implemented for this paper. From the timing simulation echo signals at about 1500 sampling data are converged and ERLE is improved by about 42-dB.

• Journal of Korea Multimedia Society
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• v.3 no.3
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• pp.263-271
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• 2000

In this paper, an acoustic echo canceller(AEC) is implemented by ADSP-2181. This AEC uses a noise robust adaptive algorithm and a postprocessing method which attenuates residual echo using cross-correlation between estimated error signal and microphone input signal. We propose new postprocessing method that uses two thresholds to prevent signal distortion after postprocessing and to improve the performance of AEC without extra computational burden. Through experiments using a 16 bit fixed-point DSP board (ADSP-2181 EZ-KIT Lite board), it is shown that the noise robust adaptive algorithm performs well in the double-talk situations and the convergence speed is comparable to NLMS. Using the postprocessor, ERLE is improved about 20 dB. As a result, the AEC with a postprocessor shows better performance than conventional ones.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.22 no.4
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• pp.430-439
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• 2011

In this paper, we design a digital predistortion(DPD) for linearity enhancement of RF power amplifier operating in 900 MHz band. We verify improvement of linearity by comparing the proposed DPD using tanh as a nonlinear indexing function and the DPD using linear indexing function based on signal amplitude. The digital predistortion is realized by look-up table(LUT) method, and the Saleh model is employed for power amplifier modeling, then a commercial power amplifier module is used for measurement. The LUT has 256 tables, and the NLMS(Normalized Least Mean Square) algorithm was utilized for an adaptive algorithm for estimation. As a result, we improve the ACLR(Adjacent Channel Leakage Ratio) by around 15 dB.

• The Journal of the Acoustical Society of Korea
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• v.15 no.2E
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• pp.39-44
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• 1996

The performance of two-channel adaptive noise canceller is ofter degraded by the weights perturbation due to the speech signal. In this paper, an adaptive noise canceller employing a speech detector and two adaptation algorithms which are switched according to the speech detector is proposed. When highly correlated speech signal is detected, the tap weights of the adaptive filter are adapted by the sign algorithm. On the other hand, the weights are adapted by the NLMS algorithm when silence is detected or when the characteristics of the noise propagation channel is changed. The employed speech detector utilizes the power ratio of the input and the output of an adaptive linear prediction-error filter. According to the computer simulation, the proposed method yields better performance than conventional ones.