• The Journal of the Acoustical Society of Korea
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• v.4 no.2
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• pp.13-25
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• 1985

광대역 혹은 협대역 잡음이 섞인 음성의 음질을 향상시키기 위하여 빠른 수렴속도를 갖고 잇는 UFBLMS 알고리즘을 음성처리에 응용한다. 광대역 잡음이 섞인 음성인 경우에는, 입력음성의 SNR 이 0 dB에서 10 dB 사이일 때, UFBLMS 알고리즘의 성능이 spectral subtraction 방법이나 Wiener filtering 방법보다도 FWSNR\sub SEG\ 척도로 약 3 dB 더 좋음을 알 수 있다. 협대역 잡음이 섞인 음 성인 경우에는 UFBLMS 알고리즘의 spectral subtraction 방법보다 FWSNR\sub SEG\ 척도로 약 2 dB 정도 성능이 더 좋다. 여러 음질 향상 알고리즘의 계산상의 복잡도와 음질 향상도 및 인식도를 고려해 보면 frequency weighting 기능을 갖고 있는 UFBLMS 알고리즘을 사용하는 것이 바람직함을 알 수 있다.

• 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.

• Journal of the Korean Institute of Telematics and Electronics S
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• v.36S no.9
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• pp.103-112
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• 1999

In this paper, a speaker-independent isolated world recognition system for a car navigation system is implemented using a general digital signal processor. This paper presents a method combining SNR normalization with RAS as a noise processing method. The semi-continuous hidden markov model is adopted and TMS320C31 is used in implementing the real-time system. Recognition word set is composed of 69 command words for a car navigation system. Experimental results showed that the recognition performance has a maximum of 93.62% in case of a combination of SNR normalization and spectral subtraction, and the performance improvement rate of the system is 3.69%, Presented noise processing method showed good speech recognition performance in 5dB SNR in car environment.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.295-298
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• 2004

본 논문에서는 잡음에 오염된 신호의 지각관계를 해석하여 지각 필터 응답 제어를 통한 음성 신호 개선 알고리즘을 제안한다. 음성 신호 개선 기법은 단일 채널환경에서 사람의 청각시스템에서의 주파수 변별력을 나타내는 각각의 임계대역에 대한 전체 에너지를 나타내는 임계대역 에너지의 지각적인 확산의 영향 즉, 마스킹 확산의 영향을 나타내는 자극에너지를 이용하여 신호와 잡음 에너지에 의해 변화하는 잡음에 의한 신호의 마스킹 구간을 검출하여 묵음 구간 추출 잡음 필터응답과 추정 잡음 오차를 보상시킨 필터응답을 통한 지각 필터 응답을 보상하여 신호를 개선하는 방법이다. 실험 결과 제안한 방법을 통해 SNR에 개선과 음질 개선 효과를 얻을 수 있음을 테스트를 통해 확인하였다.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.15 no.6
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• pp.109-114
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• 2015

This paper relates with the eualization performance of Adaptive Varying-MMA (AV-MMA) in order to the minimization of intersymbol interference that is occurs in the nonlinear communication channel. In order to obtain the error signal in the tap coefficient updating process of adaptive equalization algorithm, the present MMA uses the constant modulus. But in AV-MMA, the adaptively varying modulus are used according to the equalizer output, it is possible to reduce the error signal and possbile to improving the overall equalization performance. In order to improved equalization performance of the AV-MMA in the 64-QAM signal, the present MMA performance were compared. For this, the output signal constellation of equalizer, residual isi, maximum distortion, MSE and SER curves are applied. As a result of computer simulation, the AV-MMA has more better performance in the every performance index than MMA, and the SER performance shows that it has more robustness in high SNR environmnet compared to MMA.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.46 no.2
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• pp.18-26
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• 2009

When channel state information (CSI) is available at the transmitter, the system throughput can be enhanced by adaptive transmissions and opportunistic multiuser scheduling. In this paper, we consider multiple-input multiple-output (MIMO) systems employing bit-interleaved coded orthogonal frequency division multiplexing (BIC-OFDM). We first propose a bit-loading algorithm based on the Levin-Campello algorithm for the BIC-OFDM. Then we will apply this algorithm to the MIMO system with a finite set of constellations, by reassigning residual power on each stream Simulation results show that proposed bit-loading scheme which takes the residual power into account improves the system performance especially at high signal-to-noise ratio (SNR) range.

• The Journal of the Acoustical Society of Korea
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• v.23 no.6
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• pp.446-453
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• 2004

In this paper, a new adaptive perceptual filter using variable threshold to enhance audio signals degraded by additively nonstationary noise is proposed. The adaptive perceptual filter updates variable threshold each time according to the power of signal and the effect of noise variation. So the noisy audio signal is enhanced by the method which controls a residual noise effectively. The proposed algorithm uses the perceptual filter which transforms a time domain signal into frequency domain and calculates an intensity energy and an excitation energy in bark domain. In this method. the stage updated the response of filter is decided by threshold. The proposed algorithm using vairable threshold effectively controls a residual noise using the energy difference of audio signals degraded by the additive nonstationary noise. The proposed method is tested with the noisy audio signals degraded by nonstationary noise at various signal -to-noise ratios (SNR). We carry out NMR and MOS test when the input SNR is 15dB. 20dB. 25dB and 30dB. An approximate improvement of 17.4dB. 15.3dB, 12.8dB. 9.8dB in NMR and enhancement of 2.9, 2.5, 2.3, 1.7 in MOS test is achieved with the input signals. respectively.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.13 no.4
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• pp.127-133
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• 2013

This paper compare the performance of CCA (Compact Constellation Algorithm) adaptive equalization algorithm by effect of the compact slice weighting value for minimization of the intersymbol interference in the communication channel. The CCA combines the conventional DDA and RCA algorithm, it uses the constant modulus of the transmission signal and the considering the output of decision device by the power of compact slice weighting value in order to improving the initial convergence characteristics and the equalization noise by misadjustment in the steady state. In this process, it is confirmed by computer simulation that the compact slice weight affects the performance of CCA adaptive equalization algorithm. The performance index includes the output signal constellation, the residual isi and maximum distortion and MSE that is for the convergence characteristics, the SER according to the signal and noise power ratio at the channel is used. As a result of computer, it shows that the large weighting value gives more good in every performance index. But in SER performance, it is known that the small values gives more good in low SNR and the large values gives more good in high SNR.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2010.11a
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• pp.167-170
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• 2010

잡음 분산 값은 SNR(signal-to-noise ratio) 추정이나 MMSE(minimum mean square error) 계산, 채널 임펄스 응답의 추정 등에 사용되는 중요한 파라미터이다. 채널이 시간에 따라 변하는 무선 통신 환경에서, 신호와 섞여 있는 잡음과 간섭 신호의 정확한 추정에는 그 한계가 있으며 이로 인해 발생하는 추정 오차는 수신기의 데이터 검출 성능을 저하시킨다. 훈련열을 이용하여 채널을 추정하였을 경우 추정된 채널 임펄스 응답 신호 중 다중 경로 신호는 소수에 불과하고 나머지 대부분의 계수는 잡음 성분만을 포함하는 신호이다. 이러한 특징을 이용하여 채널의 추정 계수로 잡음 분산을 추정하는 방법이 기존에 제시되어 있다. 여기서 제안하는 알고리즘은 기존 알고리즘인 PSA(partial sample average)와 비교해 연산량에서 차이가 거의 없이 구현되며, 3GPP TDD[1]에서의 모의 실험을 통하여 기존 알고리즘보다 더 정확한 분산 값을 찾아냄을 확인하였다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.11C
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• pp.1161-1168
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• 2003

In this paper, we propose an improved statistical model-based voice activity detection algorithm and threshold update method. We first improve signal-to-noise ratio by using speech enhancement preprocessing algorithm combined power subtraction method and matched filter, then apply it to LLR test optimum decision rule for improving the performance even in low SNR conditions. And we propose an adaptive threshold update method that was not concerned in any papers. We also perform extensive computer simulations to demonstrate the performance improvement of the proposed VAD algorithm employing the proposed speech enhancement preprocessing algorithm and adaptive threshold update method under various background noise environments. Finally we verify our results by comparing ITU-T G.729 Annex B.