• The Journal of the Acoustical Society of Korea
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• v.22 no.8
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• pp.687-693
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• 2003

In this paper, a new adaptive audio signal enhancement algorithm is proposed. In order to remove a broadband noise from a noisy signal, a filter is designed and applied adaptively to noisy audio signal. The noisy signal is first transformed to frequency domain and divided into bark domain to calculate excitation energy. A filter will be calculated to eliminate the noise by using the excitation energy and noisy energy which is obtained from a silent area. The filter is adaptively adjusted and continuously applied until the threshold point is met. The algorithm also works well even though the noise's energy change all of a sudden. SNR, NMR comparison and MOS Test are performed to show the effectiveness of the proposed algorithm.

• The Journal of the Acoustical Society of Korea
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• v.29 no.6
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• pp.374-380
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• 2010

It is known that the problem of FIR filtering with noisy input and output data can be solved by a total least squares (TLS) estimation. It is also known that the performance of the TLS estimation is very sensitive to the ratio between the variances of the input and output noises. In this paper, we propose a convex combination algorithm between the ordinary recursive LS based TLS (RTLS) and the ordinary recursive LS (RLS). This combined algorithm is robust to the noise variance ratio and has almost the same complexity as the RTLS. Simulation results show that the proposed algorithm performs near TLS in noise variance ratio ${\gamma}{\approx}1$ and that it outperforms TLS and LS in the rage of 2 < $\gamma$ < 20. Consequently, the practical workability of the TLS method applied to noisy data has been significantly broadened.

• Proceedings of the Korean Vacuum Society Conference
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• 2009.02a
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• pp.254-254
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• 2009

• The Journal of Korean Institute of Communications and Information Sciences
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• v.13 no.4
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• pp.285-301
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• 1988

The symbol error rate equations of multi-level quadrature PRS(QPRS) system have been derived in the individual and composite environment of Gaussian/impulsive noise, cochannel CW interference, carrier offset, phase jitter and fading. And using the derived error rate equations, the probability of error has been evaluated and shown in graphs as functions of carrier to noise power ratio, carrier to interference power ratio, phase error, impulsive index, the ration of Gaussian noise to impulsive noise power component, signal to noise power ration in phase locked loop(PLL), and fading figures. The rseults show that the error rate performances are generally more more degraded by impulsive noise than by Gaussian noise. But on the contrary the erors occurred more frequently by Gaussian noise than impulsive noise in a fading environment.

• The Journal of the Acoustical Society of Korea
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• v.18 no.7
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• pp.45-55
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• 1999

In this paper, we used the WLMS(Wavelet domain Least Mean Square) adaptive filter based on the wavelet transform to cancel grain noise. Usually, grain noise occurs in changes of the crystalline structure of metals in high temperature environment. It makes the detection of flaw difficult. The WLMS adaptive filtering algorithm establishes the faster convergence rate by orthogonalizaing the input vector of adaptive filter as compared with that of LMS adaptive filtering algorithm in time domain. We implemented the WLMS adaptive filter by using the delayed version of the primary input vector as the reference input vector and then implemented the CA-CFAR(Cell Averaging- Constant False Alarm Rate) threshold estimator. CA-CFAR threshold estimator enables to detect the flaw and back echo signals automatically. Here, we used the output signals of adaptive filter as its input signal. To Cow the statistical characteristic of ultrasonic signals corrupted by grain noise, we performed run test. The results showed that ultrasonic signals are nonstationary signal, that is, signals whose statistical properties vary with time. The performance of each filter is appreciated by the signal-to-noise ratio. After LMS adaptive filtering in time domain, SNR improves to about 2-3㏈ but after WLMS adaptive filtering in wavelet domain, SNR improves to about 4-6㏈.

• Proceedings of the IEEK Conference
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• 2000.09a
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• pp.831-834
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• 2000

본 논문은 주파수 전이신호와 시간 전이 신호에 대해서 고조파 잡음 여기 방법과 시간 분리 여기 방법을 적용한 2.4kbps 음성부호화 방법을 제안한다. 혼합 여기 부호화 방법은 주기 신호와 비 주기 신호를 효과적으로 표현하기 위해 하모닉 잡음 모델을 사용한다. 혼합신호에 대한 잡음 성분은 캡스트럴 분석 방법을 사용함으로써 추출되고, AR(Autoregressive Model) 모델에 의해 표현된다. 시간 전이구간 신호에서의 모호한 음성을 효과적으로 제거하기 위한 또 다른 방법이 제안된다. 제안된 시간 분리 방법은 시간 에너지 변화정도를 관찰함으로써 전이 시점을 감지하고 다른 시간 길이를 가지는 두 블록으로 분리하여 분석한다. 시간 분리 방법은 분석을 위한 비대칭 윈도우와 합성에서의 위상 합성 방법을 포함한다. 제안된 방법을 사용한 2.4kbps 음성부호화 방법은 주관적 음질 평가에서 전이구간에서의 지각적 음질의 향상을 보여주었으며, 원본 음성 스펙트럼과의 고조파 비 매칭에 의한 윙윙거리는 기계적인 잡음을 감소시킨다.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.26 no.12
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• pp.1926-1933
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• 2022

Parameterized multichannel Wiener filter (PMWF) is a linear filter that can control the trade-off between residual noise and signal distortion using the embedded parameter. To apply the PMWF to noisy inputs, accurate noise estimation is important and multichannel minima-controlled recursive averaging (MMCRA) is widely used. However, in the case of the MMCRA, the accuracy of noise estimation decreases when a directional interference is involved into the array inputs. Consequently, the performance of the PMWF is degraded. Therefore, we propose a noise power spectral density (PSD) estimation method for the PMWF in this paper. The proposed method is based on a consecutive process of eigenvalue decomposition on noisy input PSD, estimation of the target component contribution using directional information, and exponential weighting for improved estimation of the target contribution. For evaluation, four objective measures were compared with the MMCRA and we verify that the PMWF with the proposed noise estimation method can improve performance in environments where directional interfereces exist.

• Journal of the Korean Institute of Intelligent Systems
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• v.15 no.2
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• pp.230-235
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• 2005

In this paper, we propose an algorithm that improves a tone quality of a noisy audio signal in order to enhance a performance of perceptual filter using intelligent estimation of noise pattern from a band degraded by additive noise. The proposed method doesn't use the estimated noise which is obtained from silent range. Instead new estimated noise according to the power of signal and effect of noise variation is considered for each frame. So the noisy audio signal is enhanced by the method which controls a estimation of noise Pattern effectively in a noise corruption band. To show the performance of the proposed algorithm, various input signals which had a different signal-to-noise ratio(SNR) such as $5\cal{dB},\;10\cal{dB},\;15\cal{dB}\;and\;20\cal{dB}$ were used to test the proposed algorithm. we carry out SSNR and NMR of objective measurement and MOS test of subjective measurement. An approximate improvement of $7.4\cal{dB},\;6.8\cal{dB},\;5.7\cal{dB},\;5.1\cal{dB}$ in SSNR and $15.7\cal{dB},\;15.5\cal{dB},\;15.2\cal{dB},\;14.8\cal{dB}$ in NMR is achieved with the input signals, respectively. And we confirm the enhancement of tone quality in terms of mean opinion score(MOS) test which is result of subjective measurement.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.15 no.11
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• pp.955-961
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• 1990

Test statistics are obtained for detection of weak signals in signal-dependent noise using rank statistics. A generalized model is used in this paper in order to consider non-additivenoise as well as purely-additive noise. Locally optimum rank detectors for the model are shown to have similarity to locally optimum detectors and to be generalizations of these for the purely-additive noise model. A similar result is obtained for multi-input cases.

• Proceedings of the Korea Institute of Convergence Signal Processing
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• 2000.12a
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• pp.185-188
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• 2000

현대의 잡음이 많은 환경에서 적응 음향반향제거기는 배경잡음의 영향으로 원활한 통화환경을 제공할 수 없다. 이러한 문제점을 해결하기 위하여 음향반향 제거와 더불어 배경잡음을 제거하는 결합구조의 적응 음향반향제거기가 제안되었다. 본 논문에서는 기존의 결합구조가 가지는 단점을 보완하여 적응 음향반향제거기의 성능을 향상시켰다. 제안한 결합구조는 적응 음향잡음제거기의 기준입력 신호를 적응 음향잡음제거기의 오차신호와 같게 구성함으로서 배경잡음 신호뿐만 아니라 잔여반향 신호도 효율적으로 제거할 수 있다. 성능 평가를 위한 실험결과, 제안한 방법이 기존의 방법에 비하여 ERLE 성능이 수렴 구간에서 3㏈ 이상 향상되었음을 확인하였다.