• The Journal of the Acoustical Society of Korea
• /
• v.17 no.1
• /
• pp.86-97
• /
• 1998

본 논문은 스테레오 음향 반향 제거기에 적용되는 고속 Affine Projection 알고리듬 을 제안한다. 최근 스테레오 원격 회의 시스템은 보다 현실감 있는 원격 회의를 가능케 하 는 장점으로 인해 많은 관심을 끌고 있다. 그러나, 회의실의 원단화자와 마이크로폰사이의 상호교차(cross-coupling)로 인해 음향 반향이 발생하게 된다. 만약 이 반향 신호가 제거되 지 않은채 수신 룸으로 전달되면 결국 음성 통화 품질이 저하된다. 이를 방지하기 위하여 추정 반향 신호를 만들어 내고 통신 품질의 손실 없이 이 반향을 제거하는 음향 반향 제거 기가 필수적이다. 단 채널 음향 반향 제거기와 다르게 스테레오 환경하에서의 음향 반향 제 거기는 전송실의 환경변화로 인한 성능 저하와 각 반향 경로를 추정하기 위해 사용하는 각 적응 필터의 임펄스응답이 반향 경로와 일치하지 않는 등의 각종 문제점들이 발생하게 된 다. 본 논문에서는 서로 상관관계 없는 입력신호를 만들어내고 전송실의 환경변화로 인한 성능저하를 보완하기 위해 전처리단(pre-processing block)을 제안하여 일반적인 방법에 대 해 3-10dB정도의 향상된 성능을 보이며 적은 계산량으로 빠른 수렴성능을 갖는 새로운 형 태의 스테레오 음향 반향 제거기를 제안한다.

• Journal of the Korean Institute of Telematics and Electronics
• /
• v.21 no.2
• /
• pp.24-28
• /
• 1984

A new type of digital echo canceler for two-wire digital transmission is presented. The new principle estimates an echo signal by use of the arithmetic means estimate for each transmitted data pattern, which leads to relatively simple hardware. The principle is compared with adaptive digital filter methods through theoretical analysis and computer simulation. The results show that the proposed method has fast convergence property with respect to its hardware simplicity and that the convergence time is independent of echo level. Quantization effects are also analyzed.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.24 no.9A
• /
• pp.1346-1353
• /
• 1999

We propose the MADF(multiplication free adaptive digital filter) algorithm and implement the fractionally spaced equalizer based on it. To evaluate the performance of proposed MADF algorithm, fractionally spaced equalizer(FSE) is used. Especially, we present that this method have the advantages for the condition having the low-frequency and slow speed

• 제어로봇시스템학회:학술대회논문집
• /
• 1988.10a
• /
• pp.172-177
• /
• 1988

In this paper, a new fast algorithm of FIR least squares filter with linear phase is presented. The general unknown statistics case is considered, whereby only sample records of the data are available. Taking advantage of the near-to-Toeplitz+Hankel structure of the resulting normal equation, a fast algorithm which gurantees the linear phase constraint, is developed that recursively produces the filter coefficient of linear phase FIR filter for a single block of data.

• Proceedings of the IEEK Conference
• /
• 2003.07e
• /
• pp.2479-2482
• /
• 2003

능동소음제어 시스템에서 많이 사용되어 온 적응 알고리듬은 filtered-X LMS (FXLMS) 알고리듬으로, 이 알고리듬에서의 수렴속도는 필터링 된 신호에 의해서 좌우되기 때문에 FXLMS 적용시 실제 수렴성능이 저하되거나 수렴이 안 되는 경우도 발생할 수 있다. 본 논문의 목적은 변환영역에서 능동소음제어 시스템의 2차 경로 모델링을 행함으로써 전체 능동소음제어 시스템 동작의 수렴성능을 향상시키려는 것이다

• Journal of Biomedical Engineering Research
• /
• v.13 no.2
• /
• pp.107-114
• /
• 1992

An adaptive filter for noise cancellation of ECG Is proposed. An adaptive noise canceller using the least mean squares algorithm Is used to reduce unwanted noise. An adaptive filter for nolse cancella lion minimizes the mean-square error between a primary input and a reference input. A primary input is the noisy ECG, and a reference input is a noise that Is correlated in some way with the noise in the primary input or a signal that is correlated only with ECG in the primary input.

• Journal of the Korea Institute of Information Security & Cryptology
• /
• v.9 no.4
• /
• pp.11-22
• /
• 1999

Least-mean-square(LMS) adaptive filters have proven to be extremely useful in a number of signal processing tasks. However LMS adaptive filter suffer from a slow rate of convergence for a given steady-state mean square error as compared to the behavior of recursive least squares adaptive filter. In this paper an efficient signal interference control technique is introduced to improve the convergence speed of LMS algorithm with tap weighted vectors updating which were controled by reusing data which was abandoned data in the Adaptive transversal filter in the scheme with data recycling buffers. The computer simulation show that the character of convergence and the value of MSE of proposed algorithm are faster and lower than the existing LMS according to increasing the step-size parameter $\mu$ in the experimentally computed. learning curve. Also we find that convergence speed of proposed algorithm is increased by (B+1) time proportional to B which B is the number of recycled data buffer without complexity of computation. Adaptive transversal filter with proposed data recycling buffer algorithm could efficiently reject ISI of channel and increase speed of convergence in avoidance burden of computational complexity in reality when it was experimented having the same condition of LMS algorithm.

• The Journal of the Acoustical Society of Korea
• /
• v.18 no.5
• /
• pp.3-11
• /
• 1999

The conventional stereophonic acoustic echo cancellers need two adaptive filters to estimate one channel echo signal. Since the two channel signals are strongly correlated, the ESR of the input signals is considerably increased whatever the input signals may be. This causes the slow convergence of the adaptive filter for echo cancellation. To speed up the convergence, the AP algorithm is frequently used for the stereophonic acoustic echo canceller although there isn't a fast version for 2-channel case. The AP algorithm can be approximated with the Gram-Schmidt orthogonalization and a TDL structure. We propose a two channel algorithm for stereophonic acoustic echo canceller with the approximated AP algorithm.

• The Journal of the Korea institute of electronic communication sciences
• /
• v.16 no.4
• /
• pp.625-632
• /
• 2021

This paper studies the effect of the number of filters in the CNN (Convolutional Neural Network) layer on the performance of a noise attenuator. Speech is estimated from a noised speech signal using a 64-neuron, 16-kernel CNN filter and an error back-propagation algorithm. In this study, in order to verify the performance of the noise attenuator with respect to the number of filters, a program using Keras library was written and simulation was performed. As a result of simulation, it can be seen that this system has the smallest MSE (Mean Squared Error) and MAE (Mean Absolute Error) values when the number of filters is 16, and the performance is the lowest when there are 4 filters. And when there are more than 8 filters, it was shown that the MSE and MAE values do not differ significantly depending on the number of filters. From these results, it can be seen that about 8 or more filters must be used to express the characteristics of the speech signal.

• Journal of the Institute of Electronics Engineers of Korea SP
• /
• v.48 no.4
• /
• pp.1-11
• /
• 2011

In order to obtain more clear x-ray images, an antiscatter grid, which can absorb the scattered rays, is employed. The artifacts due to the grid pattern are, however, visible, and thus should be removed by employing digital filters. For over exposed x-ray images, the strength of the grid artifacts are too big to be removed if fixed-bandwidth filters are employed. In this paper, for an efficient grid artifact reduction, we analyze the characteristics of the image formation and image saturation as the x-ray exposure increases. We can notice that, as the saturation begins to occur, the maximum of the artifact component decreases contrary to increasing exposure amount. We propose then an adaptive filtering algorithm for reduction of the grid artifacts, where the significant-signal bandwidth of the artifact component is used to choose appropriate filter bandwidths. The proposed algorithm is tested for real x-ray digital images, and can efficiently remove the grid artifacts.