• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2004년도 ICEIC The International Conference on Electronics Informations and Communications
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• pp.233-236
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• 2004

This paper proposes new method for the enhancement of nonexistent high frequency spectral contents from low sample rate audio signal. For example, Due to the protocol constraint, the audio bandwidth of MP3 is restricted to 16Khz. Although band-restricted MP3 audio provide savings of storage space and network bandwidth, it suffers a major problem of a loss in high frequency fidelity such as localization, ambient information, and bright nature of audio. This paper provides a new mathematical analysis for the adaptive estimation of the high frequency contents based on the nature of the input low sample rate audio. Proposed method can be worked globally to any kind of audio such as speech and music that are restricted by sampling rate and bandwidth.

• International journal of advanced smart convergence
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• 제11권4호
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• pp.81-87
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• 2022

In this paper, we propose a watershed algorithm that applies a high-frequency enhancement filter to emphasize the boundary and a local adaptive threshold to search for minimum points. The previous method causes the problem of over-segmentation, and over- segmentation appears around the boundary of the object, creating an inaccurate boundary of the region. The proposed method applies a high-frequency enhancement filter that emphasizes the high-frequency region while preserving the low-frequency region, and performs a minimum point search to consider local characteristics. When merging regions, a fixed threshold is applied. As a result of the experiment, the proposed method reduced the number of segmented regions by about 58% while preserving the boundaries of the regions compared to when high frequency emphasis filters were not used.

• 한국음향학회:학술대회논문집
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• 한국음향학회 1984년도 추계학술발표회 논문집
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• pp.70-72
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• 1984

본 논문에서는 음성신호에 섞인 잡음을 효과적으로 제거하기 위해 Two-sided Transversal Filter를 이용하는 방법을 제시하였다. 이 방법은 기존의 방법과는 달리 신호의 단기자기상관관계(short-term autocorrelation)을 충분히 이용하기 위해 적응 필터의 입력으로, 지연된 신호뿐만 아니라 미래의 신호도 이용하도록 하였다. 이 결과 음성신호에 섞인 잡음(백색잡음, 유색잡음)을 아주 효과적으로 제거함을 알 수 있다.

• 전자공학회논문지B
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• 제31B권7호
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• pp.33-41
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• 1994

A single channel adaptive noise canceling (ANC) technique with a recursive time delay estimator (RTDE) is presented for removing effects of additive noise on the speech signal. While the conventional method makes a reference signal for the adaptive filter using the pitch estimated on a frame basis from the input speech, the proposed method makes the reference signal using the delay estimated recursively on a sample-by-sample basis. As the RTDEs, the recursion formulae of autocorrelation function (ACF) and average magnitude difference function (AMDF) are derived. The normalized least mean square (NLMS) and recursive least square (RLS) algorithms are applied for adaptation of filter coefficients. Experimental results with noisy speech demonstrate that the proposed method improves the perceived speech quality as well as the signal-to-noise ratio and cepstral distance when compared with the conventional method.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2004년도 하계종합학술대회 논문집(1)
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• pp.19-22
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• 2004

This paper implemented 2.14GHz band Adaptive Hybrid Doherty (AHD) amplifier, as well as, wished to improve the high power characteristics and efficiency by composing bias adjustment circuit with LDMOS. Finally, through CAE, confirmed that AHD amplifier have superior performance than hybrid balanced amplifier. Superior characteristics of AHD amplifier is expected to affect immensely in amplifier field hereafter.

• 대한전기학회논문지
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• 제36권5호
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• pp.367-376
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• 1987

In this paper we have disigned and implemented a real time ALE (adaptive line enhancer) using a high speed digital processor,NEC 7720. For the ALE system, we have employed an adaptive LMS(least mean square) algorithm proposed by Widrow and Hoff and a 32-order FIR(finite impulse response) filter. Extensive computer simulations have been performed to investigate the peformance of the ALE and to determine necessary parameters for hardware design. The developed software for an NEC 7720 was tested in real time operation using an NEC7720 hardware emulator. The ALE has been tested by sinusoidal waves and real CW (continuous wave) signals. It was found that the experimental results were well agreed with the computer simulation results. Thus it can be concluded that the ALE is useful for detection and enhancement of a sinusoidal signal which is corrupted by an additive Gaussian noise.

• 전자공학회논문지S
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• 제35S권2호
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• pp.114-125
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• 1998

In this study, for the first time, we propose the ADI(Adaptive De-Interlacing) algorithm, which improves visually and subjectively, horizontal and vertical edges on the image processed by the ELA (Edge Based Line Average) method. The proposed ADI algorithm enlargesthe window size to 5*3 in order to utilize the feature of the continuity of edges, and the adaptive interpolator is employed to decide adaptiely horizontal, diagonal, and vertical edges. Based on the results of the compter simulation, it is confimed that the new ADI algorithm improve the PSNR by 0.5dB in the Lena image with 512*512 size and by 0.4dB in the sequence image of a salesman, respectively. For the horizontal and vertial edges on the still and salesman sequence images, the proposed ADI algorithm has better visulal improvement than the conventional ELA algorithm.

• 한국음향학회:학술대회논문집
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• 한국음향학회 2004년도 추계학술발표대회논문집 제23권 2호
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• pp.101-104
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• 2004

본 논문에서는 보상기를 채용하여 안정성을 확보한 적응순환필터인 ACHARF(Adaptive Compensated Hyperstable Adaptive Recursive Filter)를 사용하여 잡음제거를 통한 음성의 음질개선을 DSP 프로세서를 통하여 구현하였다. 실험에서는 TI사의 최신 DSP 프로세서인 TMS320C6413와 스테레오 오디오 코덱인 TLV320AIC23을 탑재한 Evaluation board를 사용하였다. 2개의 입력마이크를 이용하여 음성신호와 기준 잡음신호를 별도로 수집하여 알고리즘을 수행하였으며, 실험 결과로 음질개선 효과를 확인할 수 있었다. 본 연구를 통해서 시스템의 성능개선의 핵심은 입력으로 들어오는 음성신호와의 상관도가 가능한 적은 잡음신호를 수집하는 방법이라 생각되며 앞으로 이에 대한 연구가 필요하겠다.

• International Journal of Internet, Broadcasting and Communication
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• 제12권2호
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• pp.21-29
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• 2020

This paper presents an effective single image haze removal using edge-preserving and adaptive transmission estimation to enhance the visibility of outdoor images vulnerable to weather and environmental conditions with computational complexity reduction. The conventional methods involve the time-consuming refinement process. The proposed transmission estimation however does not require the refinement, since it preserves the edges effectively, which selects one between the pixel-based dark channel and the patch-based dark channel in the vicinity of edges. Moreover, we propose an adaptive transmission estimation to improve the visual quality particularly in bright areas like sky. Experimental results with various hazy images represent that the proposed method is superior to the conventional methods in both subjective visual quality and computational complexity. The proposed method can be adopted to compose a haze removal module for realtime devices such as mobile devices, digital cameras, autonomous vehicles, and so on as well as PCs that have enough processing resources.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2003년도 신호처리소사이어티 추계학술대회 논문집
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• pp.533-536
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• 2003

In this paper we propose AODC(Adaptive Over Driving Circuit) to reduce motion blurring and compensate color accuracy on LCD monitor. The proposed method has been experimented on adaptive ODC by motion grade after detection of moving objects. Also, we discriminated motion grade by mean value's degree of differential value between current frame and previous frame to detect moving objects. It has been found that the proposed method provides better performance than the conventional one from the point of view on the blurring of movement objects through the experiment. Its results show that the proposed method is very suitable for a motion blurring reduction on LCD monitor.