• 한국정보통신학회논문지
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• 제17권7호
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• pp.1571-1576
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• 2013

본 논문에서는 잡음 환경 하에서 음성신호에 대한 신호대잡음비(SNR)를 개선하기 위한 알고리즘을 제안한다. 본 논문에서 제안하는 알고리즘은 백색잡음 및 자동차잡음 등과 같은 배경잡음으로부터 음성신호의 SNR을 개선할 목적으로 먼저 저역, 중역, 고역 SNR 대역에서 SNR을 추정한다. 다음으로 본 알고리즘은 각 대역에서 스펙트럼을 강조함으로써 잡음으로 오염된 음성신호 속에서 잡음신호를 차감한다. 백색잡음, 자동차잡음에 의하여 오염된 음성에 대하여 본 논문에서 제안한 알고리즘이 스펙트럼 차감 방법과 비교하여 양호한 신호대잡음비 값을 구하였다. 실험결과로부터 스펙트럼 차감 방법과 비교하여 백색잡음에 대하여 최대 4.2 dB, 자동차잡음에 대하여 최대 3.7 dB의 출력 신호대잡음비가 개선된 것을 확인할 수 있었다.

• 대한의용생체공학회:의공학회지
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• 제34권1호
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• pp.24-33
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• 2013

To enhance the speech perception of hearing aid users in noisy environment, most hearing aid devices adopt various beamforming algorithms such as the first-order differential microphone (DM1) and the two-stage directional microphone (DM2) algorithms that maintain sounds from the direction of the interlocutor and reduce the ambient sounds from the other directions. However, these conventional algorithms represent poor directionality ability in low frequency area. Therefore, to enhance the speech perception of hearing aid uses in low frequency range, our group had suggested a fractional delay subtraction and integration (FDSI) algorithm and estimated its theoretical performance using computer simulation in previous article. In this study, we performed a KEMAR test in non-reverberant room that compares the performance of DM1, DM2, broadband beamforming (BBF), and proposed FDSI algorithms using several objective indices such as a signal-to-noise ratio (SNR) improvement, a segmental SNR (seg-SNR) improvement, a perceptual evaluation of speech quality (PESQ), and an Itakura-Saito measure (IS). Experimental results showed that the performance of the FDSI algorithm was -3.26-7.16 dB in SNR improvement, -1.94-5.41 dB in segSNR improvement, 1.49-2.79 in PESQ, and 0.79-3.59 in IS, which demonstrated that the FDSI algorithm showed the highest improvement of SNR and segSNR, and the lowest IS. We believe that the proposed FDSI algorithm has a potential as a beamformer for digital hearing aid devices.

• ETRI Journal
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• 제33권6호
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• pp.841-848
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• 2011

A signal-to-noise ratio (SNR) enhancement algorithm using multiple chirp symbols with clock drift is proposed for accurate ranging. Improvement of the ranging performance can be achieved by using the multiple chirp symbols according to Cramer-Rao lower bound; however, distortion caused by clock drift is inevitable practically. The distortion induced by the clock drift is approximated as a linear phase term, caused by carrier frequency offset, sampling time offset, and symbol time offset. SNR of the averaged chirp symbol obtained from the proposed algorithm based on the phase derotation and the symbol averaging is enhanced. Hence, the ranging performance is improved. The mathematical analysis of the SNR enhancement agrees with the simulations.

• 한국음향학회지
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• 제24권1호
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• pp.43-51
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• 2005

본 논문에서는 잡음 에너지 제어를 통한 지각 필터의 성능을 향상시킴으로써 잡음에 의해 열화 된 오디오 신호의 음질을 개선하는 알고리즘을 제안한다. 기존의 방식에서는 묵음 구간에서 획득한 잡음 에너지를 사용하여 필터를 구성하여 사용하지만, 신호 구간마다 달라지는 신호의 세기 및 잡음의 환경 정도에 많은 영향을 받아 잡음의 에너지가 급격하게 변화한다면 음질의 개선률이 감소함을 알 수 있다. 그러나 제안하는 방식에서는 묵음 구간에서 추정한 잡음의 에너지 제어를 통해 초기 추정 잡음보다 가까운 추정 잡음을 얻음으로써 잡음 에너지가 급격하게 변화하여도 음질 개선률에는 변화가 적음을 알 수 있었다. 또한 저 대역에 영향을 미치는 잡음의 경우에도 다른 방법들과는 달리 음질의 개선이 뚜렷하였다. 기존 방식과의 비교를 위해 다양한 신호 대 잡음 비 (signal-to-noise ratio, SNR)에서 열화 된 오디오 신호를 입력으로 사용하였다. 입력 SNR이 5dB, l0dE, 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 음질의 개선을 확인할 수 있었다.

• 대한의용생체공학회:의공학회지
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• 제26권3호
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• pp.177-184
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• 2005

In this paper, we proposed a new hearing aid algorithm to improve SNR(signal to noise ratio) of noisy speech signal and speech perception. The proposed hearing aid algorithm is a multi-band loudness compensation based independent component analysis (ICA). The proposed algorithm was compared with a conventional spectral subtraction algorithm on behind-the-ear type hearing aid. The proposed algorithm successfully separated a target speech signal from background noise and from a mixture of the speech signals. The algorithms were compared each other by means of SNR. The average improvement of SNR by ICA based algorithm was 16.64dB, whereas spectral subtraction algorithm was 8.67dB. From the clinical tests, we concluded that our proposed algorithm would help hearing aid user to hear clearly a target speech in noisy conditions.

• ETRI Journal
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• 제35권3호
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• pp.371-377
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• 2013

An effective signal detection algorithm with low complexity is presented for multiple-input multiple-output orthogonal frequency division multiplexing systems. The proposed technique, QR-MLD, combines the conventional maximum likelihood detection (MLD) algorithm and the QR algorithm, resulting in much lower complexity compared to MLD. The proposed technique is compared with a similar algorithm, showing that the complexity of the proposed technique with T=1 is a 95% improvement over that of MLD, at the expense of about a 2-dB signal-to-noise-ratio (SNR) degradation for a bit error rate (BER) of $10^{-3}$. Additionally, with T=2, the proposed technique reduces the complexity by 73% for multiplications and 80% for additions and enhances the SNR performance about 1 dB for a BER of $10^{-3}$.

• 한국통신학회논문지
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• 제28권11C호
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• pp.1161-1168
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• 2003

본 논문에서는 음성검출의 성능을 향상시킬 목적으로 정합 필터를 이용한 음성향상 전처리 과정을 통하여 SNR을 개선한 후, 이를 LLR(Log Likelihood Ratio) 검사에 의한 최적 결정방법을 적용하여 확률적인 모델을 기준으로 하는 향상된 음성검출 방법을 제안한다. 또한 기존의 음성검출 방법들에서는 제시되지 않았던 문턱값 갱신 알고리즘을 제안하며, 이 방법을 통해서 기존의 방법들에서 성능이 좋지 않았던 낮은 SNR 환경에서도 음성검출을 할 수 있게 되었다. 마지막으로 컴퓨터 시뮬레이션을 통하여 이미 상용화되어 널리 이용중인 G.729B(ITU-TG.729 Annex B)의 음성검출 결과와 비교를 통해서 제안한 음성검출 방법의 성능의 우수성을 검증하며, 실제적인 환경에도 적용이 가능함을 보인다.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 1999년도 하계종합학술대회 논문집
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• pp.61-64
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• 1999

The turbo code appropriate to IMT-2000 is known to have a good performance whenever the size of frame increases. But it is not appropriate to a sort of video service to need real time because of decoding complexity and long delay time by the size of frame. Therefore this paper proposes decoding decision algorithm of short frame in which soft output is weighted according to iteration number in turbo decoder. Performance of the proposed algorithm is analysed in the AWGN channel when short length of frame is 100, 256, 640. As the result. it is appeared that the proposed decoding decision algorithm has improved in BER other than in the existing MAP decoding algorithm.

• Investigative Magnetic Resonance Imaging
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• 제20권4호
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• pp.215-223
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• 2016

Purpose: The management of metal-induced field inhomogeneities is one of the major concerns of distortion-free magnetic resonance images near metallic implants. The recently proposed method called "Slice Encoding for Metal Artifact Correction (SEMAC)" is an effective spin echo pulse sequence of magnetic resonance imaging (MRI) near metallic implants. However, as SEMAC uses the noisy resolved data elements, SEMAC images can have a major problem for improving the signal-to-noise ratio (SNR) without compromising the correction of metal artifacts. To address that issue, this paper presents a novel reconstruction technique for providing an improvement of the SNR in SEMAC images without sacrificing the correction of metal artifacts. Materials and Methods: Low-rank approximation in each coil image is first performed to suppress the noise in the slice direction, because the signal is highly correlated between SEMAC-encoded slices. Secondly, SEMAC images are reconstructed by the best linear unbiased estimator (BLUE), also known as Gauss-Markov or weighted least squares. Noise levels and correlation in the receiver channels are considered for the sake of SNR optimization. To this end, since distorted excitation profiles are sparse, $l_1$ minimization performs well in recovering the sparse distorted excitation profiles and the sparse modeling of our approach offers excellent correction of metal-induced distortions. Results: Three images reconstructed using SEMAC, SEMAC with the conventional two-step noise reduction, and the proposed image denoising for metal MRI exploiting sparsity and low rank approximation algorithm were compared. The proposed algorithm outperformed two methods and produced 119% SNR better than SEMAC and 89% SNR better than SEMAC with the conventional two-step noise reduction. Conclusion: We successfully demonstrated that the proposed, novel algorithm for SEMAC, if compared with conventional de-noising methods, substantially improves SNR and reduces artifacts.

• 대한전자공학회논문지SD
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• 제44권10호
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• pp.67-74
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• 2007

터보부호는 디지털 이동통신 시스템에서 사용되는 채널부호화 기법의 일종으로서 다양한 채널 환경에서 반복 횟수가 증가하면 복호하는데 필요한 지연시간과 계산량이 증가하는 단점을 가진다. 본 논문에서는 터보 복호기의 현재 복호 과정에서 첫번째 복호기와 두 번째 복호기의 연판정 출력값의 차이값에 대한 절대평균값을 중단조건으로 이용하여 BER 성능의 손실없이 모든 SNR 영역에서 평균 반복복호 횟수를 크게 감소시킬 수 있는 효율적인 반복중단 알고리즘을 제안한다. 모의실험 결과, 제안된 알고리즘의 평균 반복복호 횟수는 낮은 SNR 영역에서 SDR 알고리즘과 비교하여 약 $18.25%{\sim}20.58%$ 정도의 감소효과를 나타냈으며, 높은 SNR 영역에서 외부정보 값에 대한 분산값을 이용한 방법과 비교하여 약 $22.96%{\sim}28.74%$ 정도의 감소효과를 나타내었다.