• The Journal of the Acoustical Society of Korea
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• v.15 no.1E
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• pp.84-88
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• 1996

In this paper, we propose a novel adaptive digital filter for tonal noise cancellation, with a frequency tracking capability. The proposed algorithm not only estimates the magnitude and phase of the tonal disturbance but also tracks its frequency, which changes in quasi-static manner. The algorithm uses the steepest descent method and the instantaneous frequency approach for the phase/magnitude estimation and frequency tracking, respectively. A number of computer simulations have been carried out in order to demonstrate the feasibility of the proposed ANC algorithm under various conditions.

• MALSORI
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• no.64
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• pp.105-119
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• 2007

Noise reduction technologies are indispensable to achieve acceptable speech quality in VoIP systems. This paper proposes a Wiener filter optimized to the estimated SNR of noisy speech for the noise reduction in VoIP environments. The proposed noise canceller is applied as a pre-processor before speech encoding. The performance of the proposed method is evaluated by the PESQ in various noisy conditions. In this paper, the proposed algorithm is applied to G.711, G.723.1, and G.729A which are all VoIP speech codecs. The PESQ results show that the performance of our proposed noise reduction scheme outperforms those of the noise suppression in the IS-127 EVRC and the ETSI standard for the advanced distributed speech recognition front-end.

• JSTS:Journal of Semiconductor Technology and Science
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• v.12 no.3
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• pp.278-285
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• 2012

A digital readout IC for capacitive sensors is presented. Digital capacitance readout circuits suffer from static capacitance of sensors, especially single-ended sensors, and require large passive elements to cancel such DC offset signal. For this reason, to maximize a dynamic range with a small die area, the proposed circuit features digital filters having a coarse and fine compensation steps. Moreover, by employing switched-capacitor circuit for the front-end, correlated double sampling (CDS) technique can be adopted to minimize low-frequency device noise. The proposed circuit targeted 8-kHz signal bandwidth and oversampling ratio (OSR) of 64, thus a $3^{rd}$-order ${\Delta}{\Sigma}$ modulator operating at 1 MH was used for pulse-density-modulated (PDM) output. The proposed IC was designed in a 0.18-${\mu}m$ CMOS mixed-mode process, and occupied $0.86{\times}1.33mm^2$. The measurement results shows suppressed DC power under about -30 dBFS with minimized device flicker noise.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.37 no.8A
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• pp.706-711
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• 2012

Multi-level cell NAND flash is a flash memory technology using multiple levels per cell to allow more bits to be stored. Currently, most multi-level cell NAND stores 2 bits of information per cell. This reduces the amount of margin separating the states and results in the possibility of more errors. The most error cause is coupling noise. Thus, in this paper, we studied coupling noise cancellation scheme for reduction memory on the 16-level cell NAND flash memory channel. Also, we compared the performance threshold detection and proposed scheme.

• Proceedings of the Korea Multimedia Society Conference
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• 2001.11a
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• pp.46-50
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• 2001

Adaptive signal processing is used to acoustic echo canceller. adaptive noise canceller and adaptive algorithm among adaptive algorithms is mainly used because the structure is simple and computa LMS algorithm has trade-off between the converge speed and the steady state error. In this paper, step-size of adaptive algorithm is varied with orthogonality Principles of optimal filter to get fasts though small steady state error. Time varying step-size is determined proportional to the maximum vector of LMS algorithm. As results of simulations, the adaptive algorithm with proposed time-v compared with conventional ones.

• Speech Sciences
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• v.13 no.1
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• pp.117-128
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• 2006

The two-channel Generalized Sidelobe Canceller (GSC) scheme suffers from the presence of leakage signal in the reference channel. The leakage signal is caused by the dissimilar impulse responses between microphones, and different paths from speech source to microphones. Such leakage is detrimental to speech enhancement of the GSC since the desired reference signal becomes corrupted. In order to suppress the signal leakage, two matrix injection methods are proposed. In the first method, a simple gain compensation matrix is used. In the second, a projection matrix for reducing the error between the actual and the ideal primary and reference signals, is used. This paper describes the performance degradation resulting from leakage, and proposes effective methods to resolve the problem. Representative experiments were conducted to demonstrate the effectiveness of the proposed methods on recorded speech and noise in an actual automobile environment.

• Journal of the Institute of Convergence Signal Processing
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• v.2 no.4
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• pp.65-70
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• 2001

In this paper a new scheme combining the turbo coder with parallel interference canceller, which effectively mitigates the effects of multiple access interferences and Ralyeigh fades in the DS-CDAM mobile communication systems is proposed Using the Monte-Carlo simulation, the performance of this scheme in terms of the number of users and signal to noise ration under AWGN and Ralyeigh fading environment is analyzed. The results of simulations show that the proposed scheme outperforms conventional CDMA receiver systems over Rayleigh fading as well as AWGN.

• Proceedings of the Korea Institute of Convergence Signal Processing
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• 2000.12a
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• pp.21-24
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• 2000

In this paper a new scheme combining the turbo coder with parallel interference canceller, which effectively mitigates the effects of multiple access interferences and Rayleigh fades in the BS-CDMA mobile communication systems, is proposed. Using the Monte-Carlo simulation, the performance of this scheme in terms of the number of users and signal to noise ratio under AWGN and Rayleigh fading environment is analyzed.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.5 no.4
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• pp.639-644
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• 2001

Sequence Code Division Multiple Access) and TDMA(Time Division Multiple Access) system. In this system, the narrowband TDMA signals can cause intolerant interferences to CDMA signals. In this paper, DS-CDMA/TDMA frequency sharing system have been analyzed in AWGN(Additive white Gaussian Noise), MAI (Multi Access Interference) and NI(Narrowband Interference) environment. Also, performance improvement has been obtained by adopting an adaptive notch filtering scheme using complex filter bank and CCI canceller.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.21 no.5
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• pp.1172-1179
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• 1996

General adaptive noise canceller(ANC) suffers from the misadjustment of adaptive filter weights, because of the gradient-estimate noise at steady state. In this paper, an adaptive noise cancellation algorithm with speech detector which is distinguishing speech from silence and adaptation-transient region is proposed. The speech detector uses property of adaptive prediction-error filter which can filter the highly correlated speech. To detect speech region, estimation error which is the output of the adaptive filter is applied to the adaptive prediction-error filter. When speech signal apears at the input of the adaptive prediction-error filter. The ratio of input and output energy of adaptive prediction-error filter becomes relatively lower. The ratio becomes large when the white noise appears at the input. So the region of speech is detected by the ratio. Sign algorithm is applied at speech region to prevent the weights from perturbing by output speech of ANC. As results of computer simulation, the proposed algorithm improves segmental SNR and SNR up to about 4 dBand 11 dB, respectively.