• The Journal of the Acoustical Society of Korea
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• v.26 no.4
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• pp.153-158
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• 2007

This paper presents a new approach to noise estimation to improve speech enhancement in non-stationary noisy environments. The proposed method combines the two separate noise power estimates provided by the minimum statistics (MS) for speech presence and soft decision (SD) for speech absence in accordance with SAP (Speech Absence Probability) on a separate frequency bin. The performance of the proposed algorithm is evaluated by the subjective test under various noise environments and yields better results compared with the conventional MS or SD-based schemes.

• The Journal of the Acoustical Society of Korea
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• v.28 no.7
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• pp.680-685
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• 2009

In this paper, we propose a efficient noise power estimation technique for an integrated acoustic echo and noise suppression system in a frequency domain. The proposed method uses speech absence probability (SAP) derived from the microphone input signal as the smoothing parameter updating noise power to reduce the noise power estimation error resulted from the distortions in the unified structure where the noise suppression (NS) operation is placed after the acoustic echo suppression (AES) algorithm. Therefore, in the proposed approach, the smoothing parameter based on SAP derived from the input signal instead of echo-suppressed signal should stop updating noise power estimates during the distorted noise spectrum periods. The performance of the proposed algorithm is evaluated by the objective test under various environments and yields better results compared with the conventional scheme.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.49 no.3
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• pp.89-97
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• 2012

In this paper, we propose a novel approach to noise power estimation for speech enhancement in noisy environments. The method based on IMCRA (improved minima controlled recursive averaging) which is widely used in speech enhancement utilizes a rough VAD (voice activity detection) algorithm which excludes speech components during speech periods in order to improves the performance of the noise power estimation by reducing the speech distortion caused by the conventional algorithm based on the minimum power spectrum derived from the noisy speech. However, since the VAD algorithm is not sufficient to distinguish speech from noise at non-stationary noise and low SNRs (signal-to-noise ratios), the speech distortion resulted from the minimum tracking during speech periods still remained. In the proposed method, minimum power estimate obtained by IMCRA is modified by SMT (spectral minima tracking) to reduce the speech distortion derived from the bias of the estimated minimum power. In addition, in order to effectively estimate minimum power by considering the distribution characteristic of the speech and noise spectrum, the presented method combines the minimum estimates provided by IMCRA and SMT depending on the weighting factor based on the subband. Performance of the proposed algorithm is evaluated by subjective and objective quality tests under various environments and better results compared with the conventional method are obtained.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.47 no.5
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• pp.222-228
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• 2010

In this paper, we propose a new approach to noise estimation incorporating spectral deviation with soft decision scheme to enhance the intelligibility of the degraded speech signal in non-stationary noisy environments. Since the conventional noise estimation technique based on soft decision scheme estimates and updates the noise power spectrum using a fixed smoothing parameter which was assumed in stationary noisy environments, it is difficult to obtain the robust estimates of noise power spectrum in non-stationary noisy environments that spectral characteristics of noise signal such as restaurant constantly change. In this paper, once we first classify the stationary noise and non-stationary noise environments based on the analysis of spectral deviation of noise signal, we adaptively estimate and update the noise power spectrum according to the classified noise types. The performances of the proposed algorithm are evaluated by ITU-T P. 862 perceptual evaluation of speech quality (PESQ) under various ambient noise environments and show better performances compared with the conventional method.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.26 no.12A
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• pp.1971-1977
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• 2001

Multipath Rayleigh fading channels for mobile radio transmission can be represented by the linear filter model, and depending on the delay path characteristics, only a selected number of taps may have significance in the receiver structure design. By using tap-selective equalization, reduction in both processing complexity and power consumption can be obtained. In this paper, we present an optimal tap selection method for a given channel model, and demonstrate the performance improvement over an existing method. We show the method performs the CFAR (Constant False Alarm Rate) detection when the noise power information is available, and derive exact expressions of the error probability for the case of noise power estimation. Using the derived formulas and simulation results, it is demonstrated that the error probability quickly approaches to the optimal performance as the number samples used for the noise power estimation increases.

• The Journal of the Acoustical Society of Korea
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• v.28 no.1
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• pp.77-83
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• 2009

This paper proposes a new noise-biased compensation of minimum statistics(MS) method using a nonlinear function and a priori speech absence probability(SAP) for speech enhancement in non-stationary noisy environments. The minimum statistics(MS) method is well known technique for noise power estimation in non-stationary noisy environments. It tends to bias the noise estimate below that of true noise level. The proposed method is combined with an adaptive parameter based on a sigmoid function and a priori speech absence probability (SAP) for biased compensation. Specifically. we apply the adaptive parameter according to the a posteriori SNR. In addition, when the a priori SAP equals unity, the adaptive biased compensation factor separately increases ${\delta}_{max}$ each frequency bin, and vice versa. We evaluate the estimation of noise power capability in highly non-stationary and various noise environments, the improvement in the segmental signal-to-noise ratio (SNR), and the Itakura-Saito Distortion Measure (ISDM) integrated into a spectral subtraction (SS). The results shows that our proposed method is superior to the conventional MS approach.

• Proceedings of the Acoustical Society of Korea Conference
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• 1998.08a
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• pp.432-437
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• 1998

NLMS 알고리즘을 채용한 음향반향제거기는 주변잡음에 대해서 적응필터의 계수가 오조정되어 반향제거기의 성능이 저하된다. 본 논문에서 음향반향제거기의 마이크 입력신호와 추정 오차신호의 상관도를 이용해서 주변 잡음신호에 의한 계수 오조정이 작은 적응 알고리즘과 잔여반향을 제거하기 위한 후처리기로 구성된 음향 반향 제거기를 제안한다. 기존의 NLMS 알고리즘이 입력신호의전력으로 적응상수를 정규화하지만 제안하는 알고리즘은 마이크 입력신호와 추정 오차신호의상관도와 입력신호 전력의 합으로 정규화한다. 적응필터가 반향 경로를 추정한 경우, 추정 오차신호에는 근단화자 신호가 대부분을 차지한다. 따라서 근단화자 신호가 있는 경우에는 상관도 값이 커져서 적응 상수가 작아지고 근단화자 신호에 의한 계수의 오조정을 줄일 수 있다. 후처리기도 마이크 입력신호와 추정 오차신호의 상관도를 마이크 입력신호의 전력으로 정규화한 값으로 추정 오차신호를 감쇠시킴으로써 근단화자 신호는 감쇠를 적게 하고 잔여반향을 감쇠시킨다. 멀티미디어 PC를 이용한 실험을 통해서 제안하는 알고리즘이 기존의 알고리즘에 비해서 우수한 성능을 보임을 확인했다.

• Journal of Broadcast Engineering
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• v.11 no.2 s.31
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• pp.222-228
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• 2006

In this paper, we propose a receive diversity method for orthogonal frequency division multiplexing (OFDM) systems with cochannel interference. In the method, combining is done in the frequency domain by using the subcarrier based maximum ratio combining (MRC) method. For MRC, we exploit the power of cochannel interference as well as the power of channel noise. The accuracy of the power estimate of interference plus noise is enhanced by averaging the initial estimates over the correlated subchannels where the coherency between the subchannel gains comes from the limited delay spread of the channel. Simulation results show that the proposed method yields 2-3.5dB gain of signal to noise ratio compared to the conventional MRC method and less than 1 dB difference to the ideal case.

• Proceedings of the KIEE Conference
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• 2006.07d
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• pp.2025-2026
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• 2006

충격 잡음은 전력선 통신 시스템의 성능을 열하시키는 대표적인 채널 특성 중 하나이다. 이는 배경잡음에 비해 많게는 수십 dB 이상의 레벨을 가지며, 임의 시간적으로 발생하기 때문에 추정 및 제어가 어렵다. 최근 전력선 통신 시스템이 고속화 되면서 한정된 주파수 대역상에서 한정된 신호 전력으로 대용량의 데이터를 전송하기 위해서 대역효율을 높이기 위한 변복조 기법과 오류정정 코드가 적용되고 있으나, 충격 잡음에 의해서 그 성능이 현저하게 열하된다. 일반적으로 충격잡음은 수신단에서 레벨 제한에 의해 그 영향 정도를 줄일 수 있다. 본 논문에서는 전력선 통신 시스템의 성능 향상을 위해 제안되고 있는 LDPC 부호화 OFDM 시스템에 대해 충격 잡음 수신 레벨 제한에 의한 성능 열하의 완화 정도에 대해 연구하였으며, 시뮬레이션을 통해 최적 성능을 위한 수신 레벨 제한 지표를 제시하였다.

• Journal of Advanced Navigation Technology
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• v.25 no.5
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• pp.357-362
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• 2021

UWB(ultra wide band) communication systems employ short pulses to transmit information which spreads the signal energy over a very wide frequency spectrum. Received signal-to-noise power ratio of UWB signals is an important factor in determining the accuracy of a positioning system. As the signal to noise power ratio gets higher, positioning errors decrease since noise becomes less effective. Calculation of signal to noise power ratio as a function of communication distance provides important guidelines for the system design. And the performance of a positioning system also depends heavily on the channel model. As a result of the analysis, it was found that the performance of the received signal to noise power ratio according to the communication distance was better in the LOS channel environment than in the Non LOS(line of sight) channel environment. And as the symbol interval of the preamble signal increases at a specific communication distance, the channel capacity of the UWB system increases.