• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.55-58
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• 2000

본 논문에서는 화자 확인 시스템의 등록과 확인 과정의 채널 환경 불일치로 성능이 저하되는 문제를 해결하기 위한 새로운 정규화 방법에 대해 설명한다. 제안된 방법은 첫째, 입력 음성으로부터 효과적으로 채널을 추정$\cdot$보상하고 둘째, 스코어 정규화 과정에서 사칭자 모델로서 사용되는 world모델과의 차이를 채널 추정 및 화자 모델 생성에 효과적으로 사용하는 것을 목표로 한다. 이를 위해 입력 음성의 켑스트럼과 HMM world 모델의 파라메터인 평균 켑스트럼과의 차이를 통해 음소열에 종속적인 채널 켑스트럼인 Phone-Dependent Difference Cepstrum을 추정한다. 한편 입력 음성의 음소열은 world모델의 스코어를 얻는 과정에서 함께 얻어질 수 있다. 채널 추정 실험 결과를 통해서 가장 일반적인 채널 정규화방법인 CMS에 의해 추정된 채널에 비해 실제 채널과 유사하며 화자 고유의 특성을 왜곡시키지 않는 채널 추정이 가능함을 확인할 수 있었다.

• Proceedings of the IEEK Conference
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• 2003.07e
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• pp.1715-1718
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• 2003

본 논문에서 우리는 정규화 된 혼합 노름(norm)을 이용한 다중 채널 영상 복원 알고리즘을 제안한다. 채널 내부와 채널 사이의 결정론적 정보를 이용하는 다중채널 복원 문제를 고려한다. 각 채널에서, LMS(Least Mean Square), LMF(Least Mean Fourth), 평탄 함수가 결합된 함수가 제안되었다. LMS와 LMF 사이의 적절한 분배를 제어하는 혼합 노를 매개변수와 해의 평탄 정도를 정의하는 정규화 매개 변수를 소개하며, 두 매개 변수는 각 채널의 잡음 특성에 따라 매번 반복적으로 갱신된다. 제안된 알고리즘은 각 채널의 잡음분포에 대한 지식이 필요하지 앉고 앞에서 언급된 매개 변수는 부분적으로 복원된 영상에 기반을 두고 조절하게 된다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.2C
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• pp.272-282
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• 2004

This paper introduces a regularized mixed norm multi-channel image restoration algorithm using both within-and between- channel deterministic information. For each channel a functional which combines the least mean squares (LMS), the least mean fourth (LMF), and a smoothing functional is proposed. We introduce a mixed norm parameter that controls the relative contribution between the LMS and the LMF, and a regularization parameter defining the degree of smoothness of the solution, where both parameters are updated at each iteration according to the noise characteristics of each channel. The novelty of the proposed algorithm is that no knowledge of the noise distribution for each channel is required and that the parameters mentioned above are adjusted based on the partially restored image.

• Journal of KIISE:Computer Systems and Theory
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• v.30 no.10
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• pp.523-532
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• 2003

Video on Demand(VOD) services cause high resource consumption in a video seuer, because multimedia are characterized by continuous playback, a high bandwidth requirement, and long playback duration. Patching has been proposed to save the network I/O bandwidth of a video server. To achieve true VOD, patching uses multicasting to share video streams, thereby providing immediate VOD services to users without any service latency. A communication channel is used to either multicast the entire video as a regular channel or multicast only the leading portion of a video as a Patching channel. This paper Proposes a buffering patching technique that divides regular channels, as used in patching, into sub-regular channels and regular channels to shorten the holding time of the channels. In the proposed technique, the last portion of video data, corresponding to the size of the buffering window, is not transferred by sub-regular channels, but rather downloaded and buffered in the user buffer from the latest regular channel. When simulations were performed to compare the performance of the proposed technique with that of conventional patching, the results indicated that the proposed technique was superior in terms of the defection rate, average service latency, and fairness, although the amount of video data buffered at each user station was higher than that with patching.

• The Journal of the Acoustical Society of Korea
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• v.23 no.1
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• pp.61-74
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• 2004

In this paper, a contort- and speaker-dependent cepstrum extraction method and a channel normalization method for minimizing the loss of speaker characteristics in the cepstrum were proposed for a robust speaker recognition system over the channel. The proposed extraction method creates a cepstrum based on the pitch synchronous analysis using the inherent pitch of the speaker. Therefore, the cepstrum called the 〃pitch synchronous cepstrum〃 (PSC) represents the impulse response of the vocal tract more accurately in voiced speech. And the PSC can compensate for channel distortion because the pitch is more robust in a channel environment than the spectrum of speech. And the proposed channel normalization method, the 〃formant-broadened pitch synchronous CMS〃 (FBPSCMS), applies the Formant-Broadened CMS to the PSC and improves the accuracy of the intraframe processing. We compared the text-independent closed-set speaker identification on 56 females and 112 males using TIMIT and NTIMIT database, respectively. The results show that pitch synchronous km improves the error reduction rate by up to 7.7% in comparison with conventional short-time cepstrum and the error rates of the FBPSCMS are more stable and lower than those of pole-filtered CMS.

• The Journal of the Acoustical Society of Korea
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• v.21 no.4
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• pp.361-373
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• 2002

In this paper, we propose a channel normalization method to improve the performance of CMS (cepstral mean subtraction) which is widely adopted to normalize a channel variation for speech and speaker recognition. CMS which estimates the channel effects by averaging long-term cepstrum has a weak point that the estimated channel is biased by the formants of voiced speech which include a useful speech information. The proposed Formant-broadened Cepstral Mean Subtraction (FBCMS) is based on the facts that the formants can be found easily in log spectrum which is transformed from the cepstrum by fourier transform and the formants correspond to the dominant poles of all-pole model which is usually modeled vocal tract. The FBCMS evaluates only poles to be broadened from the log spectrum without polynomial factorization and makes a formant-broadened cepstrum by broadening the bandwidths of formant poles. We can estimate the channel cepstrum effectively by averaging formant-broadened cepstral coefficients. We performed the experiments to compare FBCMS with CMS, PFCMS using 4 simulated telephone channels. In the experiment of channel estimation, we evaluated the distance cepstrum of real channel from the cepstrum of estimated channel and found that we were able to get the mean cepstrum closer to the channel cepstrum due to an softening the bias of mean cepstrum to speech. In the experiment of text-independent speaker identification, we showed the result that the proposed method was superior than the conventional CMS and comparable to the pole-filtered CMS. Consequently, we showed the proposed method was efficiently able to normalize the channel variation based on the conventional CMS.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.26 no.6A
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• pp.948-956
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• 2001

이 논문에서는 직접수열 부호분할 다중접속 시스템의 여러 경로 감쇄 환경에서 MMSE 검파기의 성능을 분석한다. 1부에서 MMSE 검파기의 성능을 유도하기 위하여, 필터의 출력단에서 신호를 제외한 나머지 간섭과 배경 잡음 성분을 정규 분포와 근사한 것을 이용하여 여기서는 채널 추정에 오류가 있을 때의 영향을 검토한다. 채널변수에 정규분포의 오류가 있을 때와 구체적인 채널 추정 시스템을 사용한 두 가지 경우에 대해서 채널 추정의 영향을 분석한다. 감쇄 환경, 채널 추정 오류의 분산, 도플러 주파수 등을 변화시켜 얻은 수식적인 결과로 성능 분석의 정확도를 보였다.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2002.11a
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• pp.49-55
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• 2002

기존의 영상 획득 시스템들이 어느 정도의 엘리어싱을 허용하도록 제작되어왔음에도 불구하고, 고해상도 영상에 대한 요구는 점점 더 증가하고 있다. 본 논문에서는 부정확한 부화소 단위의 위치추정 오류를 고려한 고해상도 재구성 알고리즘을 제안한다. 부정확한 부화소 위치 추정 오류로 인해 생기는 불량위치문제(ill-posedness)를 해결하기 위해 정규화된 반복 연산법을 적용하였다. 특히 여러장의 저해상도 영상들을 개별적으로 고려하기에 적합한 다중채널 영상 재구성 방법을 도입하였다. 각 저해상도 영상에서 발생하는 움직임 추정오류는 서로 다른 경향성을 나타내므로, 정규화 파라미터들은 각 채널에 맞게 결정되어야 한다. 이를 위채 정규화 파라미터들을 자동으로 결정하는 방법을 제안한다. 제안한 알고리즘은 움직임 추정 오류에 매우 안정하며, 원 영상과 잡음에 대한 사전정보를 필요로 하지 않는다. 또한 주관적인 측면과 객관적인 측면에서 모두 우수한 결과를 실험적으로 보인다.

• Journal of IKEEE
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• v.25 no.3
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• pp.420-424
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• 2021

The massive MIMO system, which is a core technology of 5G communication systems, has a problem that it is difficult to implement in a frequency division duplex system based on limited channel feedback because a large amount of channel information is required at the transmitting end. In order to solve this problem, the Joint Spatial Division and Multiplexing (JSDM) technique that dramatically reduces the channel information requirement by removing interference between the user groups using channel correlation information that does not change for a long time has been proposed. Recently, a regularized JSDM technique has been proposed to further improve performance by allowing residual interference between the user groups. However, such JSDM-related studies were mainly designed to focus on inter-group interference cancellation, and thus performance analysis was not performed in a more realistic environment assuming limited feedback in the intra-group interference cancellation phase. In this paper, we analyze the performance of the JSDM and regularized JSDM techniques according to the number of groups and users in a limited feedback environment, and through the simulation results, demonstrate that the regularized JSDM technique shows a more remarkable advantage compared to the existing JSDM in a limited feedback environments.

• The Journal of the Acoustical Society of Korea
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• v.35 no.4
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• pp.280-287
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• 2016

In this paper, we propose a normalization algorithm that can be applied to adaptive filters for multi-channel active noise control. The FxLMS (Filtered-x Least Mean Square) algorithm for the single-channel active noise control can be normalized in the same way as the NLMS (Normalized Least Mean Square) algorithm, whereas in case of the multi-channel active noise control, the single-channel normalization for the FxLMS algorithm cannot be extended to the normalization for the multi-channel FxLMS algorithm straightforwardly. First, we adopt a generalized normalization algorithm for the multi-channel FxLMS algorithm based on the principle of minimal disturbance and then, proposed a normalized algorithm considering only diagonal elements to avoid computation for matrix inversion. We carried out performance comparisons of the proposed algorithm with other algorithms without normalization. It is shown that the proposed algorithm presents better convergence characteristics under non-stationary environments.