• 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.

• International Journal of Contents
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• v.11 no.1
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• pp.7-14
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• 2015

In this paper, we derive the outage probability for M-ary phase shifting keying (MPSK) and M-ary quadrature amplitude modulation (MQAM) burst transmission (BT) of adaptive decode-and-forward (ADF) cooperative relay systems over quasi-static Rayleigh fading channels. Within a burst, there are pilot symbols and data symbols. Pilot symbols are used for channel estimation schemes and each relay node's transmission mode selection schemes. At first, we focus on ADF relay systems in which the probability density function (PDF) is derived on the basis of error events at relay nodes corresponding to channel estimation errors. Next, the average outage probability is derived as an approximate expression for an arbitrary link signal-to-noise ratio (SNR) for different modulation orders. Its accuracy is demonstrated by comparison with simulation results. Further, it is confirmed that BT-ADF relay systems with pilot symbol based channel estimation schemes enables to select correctly decoded relay nodes without additional signaling between relay nodes and the destination node, and it is verified that the ideal performance is achieved with small SNR loss.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.9A
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• pp.678-683
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• 2003

The pseudo-noise (PN) code acquisition based on the sequential estimation (SE) proposed by Ward performs well only at relatively high chip signal-to-noise ratios (SNRs). In this paper, a seed accumulating sequential estimation (SASE) method and a PN code acquisition system based on it are proposed, which perform well at low chip SNR (of practical interest) also. Then, the mean acquisition time performance of the proposed system is investigated. Numerical results show that the system based on the SASE performs dramatically better than that based on the SE at low chip SNR, and the improvement becomes larger as the period of PN code increases.

• Journal of Communications and Networks
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• v.14 no.5
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• pp.481-486
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• 2012

Accurate and efficient estimation of the number of sources is critical for providing the parameter of targets in problems of array signal processing and blind source separation among other such problems. When conventional estimators work in unfavorable scenarios, e.g., at low signal-to-noise ratio (SNR), with a small number of snapshots, or for sources with a different strength, it is challenging to maintain good performance. In this paper, the detection limit of the minimum description length (MDL) estimator and the signal strength required for reliable detection are first discussed. Though a comparison, we analyze the reason that performances of classical estimators deteriorate completely in unfavorable scenarios. After discussing the limiting distribution of eigenvalues of the sample covariance matrix, we propose a new approach for estimating the number of sources which is based on a sequential hypothesis test. The new estimator performs better in unfavorable scenarios and is consistent in the traditional asymptotic sense. Finally, numerical evaluations indicate that the proposed estimator performs well when compared with other traditional estimators at low SNR and in the finite sample size case, especially when weak signals are superimposed on the strong signals.

• The Journal of the Acoustical Society of Korea
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• v.37 no.6
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• pp.457-467
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• 2018

This paper presents performance comparison of several high-resolution frequency estimation algorithms for pure real tone signal. Algorithms are DFT (Discrete Fourier Transform - for reference purpose), Jacobsen, Candan, reassignment and Cedron. Each algorithm is evaluated under various experimental conditions, e.g., different SNR (Signal to Noise Ratio), window function and window length and performance is compared in the perspective of bias, MSE (Mean Square-Error) and variance. Experimental results indicate that Cedron algorithms works well in the most cases. For actual usage in the engineering problem, each algorithm needs additional analysis and modification.

• Journal of Convergence for Information Technology
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• v.10 no.9
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• pp.22-27
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• 2020

In the fifth generation (5G) networks, the mobile services require much faster connections than in the fourth generation (4G) mobile networks. Recently, as one of the promising 5G technologies, non-orthogonal multiple access (NOMA) has been drawing attention. In NOMA, the users share the frequency and time, so that the more users can be served simultaneously. NOMA has several superiorites over orthogonal multiple access (OMA) of long term evolution (LTE), such as higher system capacity and low transmission latency. In this paper, we investigate impact of channel estimation errors on successive interference cancellation (SIC) performance of NOMA. First, the closed-form expression of the bit-error rate (BER) with channel estimation errors is derived, And then the BER with channel estimation errors is compared to that with the perfect channel estimation. In addition, the signal-to-noise (SNR) loss due to channel estimation errors is analyzed.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.291-294
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• 2004

본 논문에서는 잡음의 변화(variance)가 없는 정적인 통계적 특성(Stationary Static Characteristic)을 갖는 환경에서 잡음 추정을 통해 지각 필터의 성능을 개선하는 알고리즘을 제안한다. 제안된 잡음 추정 알고리즘은 입력되는 잡음에 열화 된 신호의 묵음 구간에서 추정된 잡음을 이용하여 입력되는 잡음의 SNR을 추정 후, 대역 별로 smoothing 상수 값으로 잡음 에너지를 제어하여 첨가된 잡음을 추정함으로써 초기 추정 잡음 보다 가까운 추정 잡음을 얻을 수 있게 된다. 이는 신호를 열화 시킨 잡음을 보다 정확한 추정을 제공함으로써, 지각 필터의 응답을 개선할 수 있고 더불어 잡음에 의해 열화 된 신호의 음질을 개선할 수 있다. 또한 저 대역에 영향을 미치는 잡음인 경우 다른 방법들과는 달리 음질의 개선이 뚜렷하다. 기존의 방식과 비교를 위해 다양한 신호 대 잡음 비(signal-to-noise ratio, SNR)에서 열화 된 오디오 신호를 입력으로 사용하였다. 입력 SNR이 5dB, 10dB, 15dB와 20dB의 각각의 경우에 대하여 SSNR(Segmental SNR)과 잡음 대 마스킹 비(Noise-to-mask ratio, NMR), 음질 테스트를 수행한 결과, 청감 테스트(Mean Opinion Score, MOS Test) 결과의 향상과 음질개선의 개선을 확인할 수 있었다.

• Proceedings of the IEEK Conference
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• 1999.06a
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• pp.687-690
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• 1999

Many high-resolution algorithms based on the eigen-decomposition analysis of observed covariance matrix, such as MVE, MUSIC, and EVM, have been proposed. However, the resolution of spectral estimates for these algorithms is severely degraded when Signal-to-Noise Ratio (SNR) is low and arrival angles of signal are close to each other. And EVM and MUSIC is powerful for the characteristic of SNR. But have the limitation that the number of signals presented is known. While MVE is bad the characteristic of SNR. In this study, we propose a modified MVE to enhance the resolution for Direction-Of-Arrival (DOA) estimation of underwater acoustic signal. This is to remove the limitation that existing algorithms should know the information for the number of signals. Because the algorithms founded on the eigen value estimate DOA with only the noise subspace, they have the high-resolution characteristic. And then, with the method reducing the effect of the signal subspace, we are to reduce the degradation because of complementary relationship between the signal subspace and the noise subspace. This paper, with using the simulation data, we have estimated the proposed algorithms, compared it with other high-resolution algorithms. The simulation results show that the modified MVE proposed is accurate and has a better resolution even though SNR is low, under the same condition.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.1A
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• pp.55-61
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• 2005

Mobile communication systems are required not only to support high-data-rate transmissions in favorable channel conditions but also to be able to tolerate hostile environments possibly encountered by cellular communications. This paper compares CDMA(Code Division Multiple Access), OFDM(Orthogonal Frequency Division Multiplexing), and MC-CDMA(Multi-Carrier CDMA)with inaccurate channel estimates and low SNR environments. The equations and simulation results show that the performance losses in CDMA systems due to imprecise channel estimates are not considerable while frequency-hopping of OFDM systems can result in more than l0dB SNR losses. Also, this paper show that frequency-spreading of MC-CDMA can be very helpful for channel compensation performances than frequency-hopping or time-spreading of OFDM.

• Proceedings of the Acoustical Society of Korea Conference
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• 1998.08a
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• pp.262-266
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• 1998

In the speech recognition under the noisy environments, reducing the mismatch introduced between training and testing environments is an important issue, and spectral subtraction is widely used technique because of its simplicity and relatively good performance in noisy environments. In this paper, we introduced histogram method as a reliable noise estimationi approach for spectral subtraction. To deal with the problem of residual noise after spectral subtraction, we proposed a new ove-estimation technique based on distribution characteristics of histogram used for noise estimation. Since the proposed technique decides the degree of over-estimation adaptively according to the measured noise distribution, it can cope with the SNR variations effectively in compared with the conventional over-estimation technique.