• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.10 no.5
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• pp.139-145
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• 2010

In this paper, we propose robust cell ID detection algorithm and SNR estimation algorithm applicable to mobile base station, which can be operated independently. The proposed cell ID estimation uses signal subspace to estimate cell IDs used in cell. The proposed SNR estimation algorithm uses number of noise subspace vectors and the corresponding eigen-vectors. Through the computer simulations, we showed that performance of the proposed cell ID detection and SNR estimation algorithms are superior to existing correlation based algorithms. Also we showed that the proposed algorithm is suitable to fast moving channel in high background noise and strong interference signal.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.36 no.6C
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• pp.391-397
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• 2011

This paper investigates the frequency offset (PO) estimation scheme for the orthogonal frequency division multiplexing (OFDM)-based cognitive radio (CR) systems. In the CR environments, the conventional FO estimation schemes for the OFDM systems experience significant performance degradation due to the effect of the non-Gaussian noise. In this paper, a novel FO estimation scheme based on the maximum likelihood criterion is proposed for the OFDM-based CR systems in non-Gaussian noise environments. The proposed scheme does not require a specific pilot structure and has a better estimation performance compared with that of the conventional scheme.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.46 no.1
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• pp.87-94
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• 2009

Noise estimation and suppression are a crucial factor of many speech communication and recognition systems. In this paper, proposed algorithm is based on the ratio of variance normalized of noisy power spectrum in time-frequency domain. Our proposed algorithm tracks the threshold and controls the trade-off between residual noise and distortion. This algorithm is evaluated by the ITU-T P.835 signal distortion (SIG) and segment signal to noise ratio (SNR), and is superior to the conventional methods.

• Journal of the Institute of Electronics and Information Engineers
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• v.52 no.7
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• pp.119-124
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• 2015

Speech enhancement has been required as a preprocessor for a noise robust speech recognition system. Codebook-based Speech Enhancement (CBSE) is highly robust in nonstationary noise environments compared with conventional noise estimation algorithms. However, its performance is severely degraded for the codevector combinations that have lower correlation with the input signal since CBSE depends on the trained codebook information. To overcome this problem, only the reliable codevector combinations are selected to be used to remove the codevector combinations that have lower correlation with input signal. The proposed method produces the improved performance compared to the conventional CBSE in terms of Log-Spectral Distortion (LSD) and Perceptual Evaluation of Speech Quality (PESQ).

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.15 no.4
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• pp.237-242
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• 2022

Acquiring high-quality images in control and measurement systems is one of the important factors. Among image acquisition technologies, SFF (Shape from Focus) is a technology for recovering a 3D shape by acquiring 2D images with different focus levels by moving an object at a predetermined step size along the optical axis. For SFF, when an object is moved at a constant step size, mechanical vibration, referred as jitter noise, occurs in each step along the optical axis. In this paper, a new state estimation filter is designed and applied for reducing the jitter noise. For the application of the proposed method, the jitter noise and focus curves are modeled as Gaussian function. Experimental results demonstrate the effectiveness of proposed method.

• 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.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.23 no.6
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• pp.563-573
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• 2013

The ship radiated noise level fluctuates by the interference between direct and reflected paths. The effect of sea surface reflection path on interference depends strongly on sea surface roughness. This paper describes error characteristics of ship acoustic signature estimation by sea surface scattering effect. The coherent reflection coefficient which explains a magnitude of sea surface scattering and its resultant interference acoustic field is analyzed quantitatively as a function of a grazing angle, effective surface height, frequency, source-receiver range and depths of source and receiver. Theoretical interference acoustic field is compared with experimental result for two different sea surfaces and five different frequencies by changing source-receiver range. It is found that both matches well each other and a magnitude of interference acoustic field is decreasing by increasing a grazing angle, effective surface height, frequency, and depths of source and receiver and decreasing source-receiver range. For given experimental conditions, the transmission anomaly which is a bias error of ship acoustic signature estimation, is about a range of 1~3 dB. The bias error of an existing ship radiated noise measurement system is also analyzed considering wind speed, source depth and frequency.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.18 no.8
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• pp.823-831
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• 2008

Since it needs the light computational load and small database, sound source localization method using time delay of arrival(TDOA method) is applied at many research fields such as a robot auditory system, teleconferencing and so on. Researches for time delay estimation, which is the most important thing of TDOA method, had been studied broadly. However studies about factors for time delay estimation are insufficient, especially in case of real environment application. In 1997, Brandstein and Silverman announced that performance of time delay estimation deteriorates as reverberant time of room increases. Even though reverberant time of room is same, performance of estimation is different as the specific part of signals. In order to know that reason, we studied and analyzed the factors for time delay estimation using speech signal and room impulse response. In result, we can know that performance of time delay estimation is changed by different R/D ratio and signal characteristics in spite of same reverberant time. Also, we define the performance index(PI) to show a similar tendency to R/D ratio, and propose the method to improve the performance of time delay estimation with PI.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.38A no.2
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• pp.183-190
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• 2013

In this paper, we propose a spatially adaptive noise detection and removal algorithm for a single degraded image. Under the assumption that an observed image is Gaussian-distributed, the noise information is estimated by local statistics of degraded image, and the degree of the additive noise is detected by the local statistics of the estimated noise. In addition, we describe a noise removal method taking a modified Gaussian filter which is adaptively determined by filter parameters and window size. The experimental results demonstrate the capability of the proposed algorithm.

• Journal of Broadcast Engineering
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• v.6 no.1
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• pp.72-81
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• 2001

An adaptive channel estimation technique for OFDM-based DTV receivers is proposed using a new mobility estimator. Sample mean techniques for channel estimation have displayed good performance in slow fading channels, because averaging reduces noise In channel estimation operation. This paper suggests an algorithm which selects the optimal number of symbols within which the sample mean of consecutive pilot data can be obtained. The designed mobility estimator determines the optimal number by comparing mobility variance and estimated noise valiance. The algorithm using the mobility estimator obtains an optimal channel function under time-invariant or time-variant multipath fading channels, thereby making the best BER performance.