• IEIE Transactions on Smart Processing and Computing
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• 제3권2호
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• pp.41-51
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• 2014

This paper describes a method to estimate the noise power using the minimum statistics approach, which was originally proposed for audio processing. The proposed minimum statistics-based method separates a noisy image into multiple frequency bands using the three-level discrete wavelet transform. By assuming that the output of the high-pass filter contains both signal detail and noise, the proposed algorithm extracts the region of pure noise from the high frequency band using an appropriate threshold. The region of pure noise, which is free from the signal detail part and the DC component, is well suited for minimum statistics condition, where the noise power can be extracted easily. The proposed algorithm reduces the computational load significantly through the use of a simple processing architecture without iteration with an estimation accuracy greater than 90% for strong noise at 0 to 40dB SNR of the input image. Furthermore, the well restored image can be obtained using the estimated noise power information in parametric image restoration algorithms, such as the classical parametric Wiener or ForWaRD image restoration filters. The experimental results show that the proposed algorithm can estimate the noise power accurately, and is particularly suitable for fast, low-cost image restoration or enhancement applications.

• 전기전자학회논문지
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• 제24권4호
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• pp.1122-1128
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• 2020

대역 확장(Bandwidth Extension)이란 채널 용량 부족 혹은 이동통신 기기에 탑재된 코덱의 특성으로 인해 부호화 및 복호화 과정에서 대역 제한(band limited)되거나 손상된 협대역 신호(NB, Narrow Band)를 복원, 확장하여 광대역 신호(WB, Wide Band)로 전환 시켜주는 것을 의미한다. 대역 확장 연구는 주로 음성 신호 위주로 대역 복제(SBR, Spectral Band Replication), IGF(Intelligent Gap Filling)과 같이 고대역을 주파수 영역으로 변환하여 복잡한 특징 추출 과정을 거쳐 이를 바탕으로 사라지거나 손상된 고대역을 복원한다. 본 논문에서는 딥러닝 모델 중 오토인코더(Autoencoder)를 바탕으로 1차원 합성곱 신경망(CNN, Convolutional Neural Network)들의 잔차 연결을 활용하여 복잡한 사전 전처리 과정 없이 일정한 길이의 시간 영역 신호를 입력시켜 대역 확장 시킨 음향 신호를 출력하는 모델을 제안한다. 또한 음성 영역에 제한되지 않는 음악을 포함한 여러 종류의 음원을 포함하는 데이터셋에 훈련시켜도 손상된 고대역을 복원할 수 있음을 확인하였다.

• 한국멀티미디어학회논문지
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• 제21권3호
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• pp.391-399
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• 2018

In this paper, we construct and compare various guitar chord classification systems using perceptron neural network and convolutional neural network without pre-processing other than Fourier transform to identify the optimal chord classification system. Conventional guitar chord classification schemes use, for better feature extraction, computationally demanding pre-processing techniques such as stochastic analysis employing a hidden markov model or an acoustic data filtering and hence are burdensome for real-time chord classifications. For this reason, we construct various perceptron neural networks and convolutional neural networks that use only Fourier tranform for data pre-processing and compare them with dataset obtained by playing an electric guitar. According to our comparison, convolutional neural networks provide optimal performance considering both chord classification acurracy and fast processing time. In particular, convolutional neural networks exhibit robust performance even when only small fraction of low frequency components of the data are used.

• 한국지능시스템학회:학술대회논문집
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• 한국퍼지및지능시스템학회 2001년도 추계학술대회 학술발표 논문집
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• pp.5-9
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• 2001

In this paper, we propose the new audio watermarking method and can be used on line processing. Instead of the wavelet transform, we use the integer wavelet transform for the reduction of the computational load. The watermark associated with the chi-square distribution is inserted into the signal on the integer wavelet domain. When extracting the watermark, the spread spectrum methods are used with the coefficients associated with the covariance sequence. We show that the chi-square distribution is a good tool for the spread spectrum method on the wavelet domain. This watermarking technique may be used for the control of the electrical product which can be controlled with the hidden signals and can be moved according to the audible signals simultaneously.

• 한국음향학회:학술대회논문집
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• 한국음향학회 1998년도 학술발표대회 논문집 제17권 2호
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• pp.39-42
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• 1998

본 논문에서는 방송용 오디오 기기가 갖는 다채널의 특성과 각 채널에 대한 다양한 신호처리 기능의 특성을 고려하여 다채널 디지털 오디오 신호 처리기의 구조를 제안하고 범용 DSP를 이용하여 실시간 병렬 처리 시스템을 구현하였다. 구현된 시스템은 32비트 부동수소점 DSP를 이용하였으며 스테레오 채널의 48KHz 표본화 주파수를 지원하고 20비트 해상도를 갖는 시스템이다. 다채널 디지털 오디오 신호 처리 시스템의 구조는 디지털 신호 처리 과정을 수행하는 디지털 오디오 데이터 처리 부분과 시스템을 제어하기 위한 제어 정보 처리 부분으로 제안하였다. 이러한 구조에 적합한 실시간 시스템을 구현하기 위해 전체 시스템은 4부분의 모듈로 구성된다.

• 대한의용생체공학회:의공학회지
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• 제7권1호
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• pp.75-79
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• 1986

In this paper, a multigate pulsed doppler system with parallel signal processing which enables the transcutaneous assesment of the blood-flow velocities of five adjacent sample volumes is developed. The outputs of the system are audio signals, velocity curves and velocity profiles of 5 adjacent sample volumes. We have applied to the major superficial arteries and obtained the detailed information about the time dependent blood-flow behavior.

• 대한전기학회:학술대회논문집
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• 대한전기학회 2006년 학술대회 논문집 정보 및 제어부문
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• pp.419-421
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• 2006

The importance of automatic discrimination image data has evolved as a research topic over recent years. We have used forward neural network as a classifier using sound data features within image data, our initial tests have shown encouraging results that indicate the viability of our approach.

• 한국정보통신설비학회:학술대회논문집
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• 한국정보통신설비학회 2009년도 정보통신설비 학술대회
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• pp.131-135
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• 2009

This paper is the introduction of our work on distributed load scheduling in single-level tree network. In this paper, we derive a new calculation model in single-level tree network and show a closed-form formulation of the time for computation system. There are so many examples of the application of this technology such as distributed database, biology computation on genus, grid computing, numerical computing, video and audio signal processing, etc.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2002년도 하계종합학술대회 논문집(2)
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• pp.149-152
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• 2002

The performance of a 1-bit DAC depends on that of the analog circuits. The mixed SC-CT (switched capacitor-continuous time) architecture is an effective design methodology for the analog circuits. This paper Proposes a new buffer scheme for the 1-bit digital-to-analog subconverter and a new SF-DSC(smoothing filter and differential-to-sig le converter) which performs both the smoothing filter and the differential-to-single convertor simultaneously.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2002년도 하계종합학술대회 논문집(2)
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• pp.157-160
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• 2002

This paper proposed a guideline for selecting the arithmetic circuit architecture. The guideline incorpo-rates the new concept of PDSP (power-delay-size product) and the weighting method. HSPICE simulations havc been performed to several full adders in order to prove the validity of the proposed guideline. We applied this guideline to select an optimized FA (full adder) architecture and successfully implemented the DAC's digital blocks.