• The Journal of Korean Institute of Communications and Information Sciences
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• v.26 no.6B
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• pp.760-766
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• 2001

본 논문에서는 차량내 핸즈프리 전화통신을 위한 음향 반향 및 배경 잡음 제거기를 제안한다. 제안한 시스템은 새로운 잔여 반향 제거 기법과 실시간 구현에 적합한 동시통화 검출기를 포함한다. 잔여 반향 제거에서는 근단화자가 없는 구간에 대하여 선형 예측기를 이용하여 잔여 반향 신호의 인접 샘플간의 상관도를 제거하여 잡음 제거기의 입력으로 사용한다. 잔여 반향 신호의 음성특성을 제거함으로써 잡음 제거기를 이용하여 배경 잡음과 더불어 잔여 반향의 전력을 효과적으로 줄일 수 있다. 제안된 시스템에서는 상용 저전송률 음성부호화기와의 결합을 고려하여 IS-127(EVRC)에 포함되어 있는 잡음 제거기를 사용하였다. 90 km/h로 정속 주행하는 차내의 핸즈프리 환경에서 제안된 시스템은 30 dB이상의 간섭신호 제거 성능을 보였다. 제안된 시스템은 16비트 고정 소수점 연산을 하는 저가의 DSP를 이용하여 실시간 구현되었다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.11C
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• pp.917-924
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• 2008

Motion estimation and compensation in the spatial domain check the searching area of specified size in the previous frame and search block to minimize the difference with current block. When we check the searching area, it consumes the most encoding times due to increasing the complexity. We can solve this fault by means of motion estimation using shifting matrix in the transform domain instead of the spatial domain. We derive so the existed shifting matrix to a new recursion equation that we decrease more computations. We modify simply vertical shifting matrix and horizontal shifting matrix in the transform domain for motion estimation of half-pixel accuracy. So, we solve increasing computation due to bilinear interpolation in the spatial domain. Simulation results prove that motion estimation by the proposed algorithm in DCT-based transform domain provides higher PSNR using fewer bits than results in the spatial domain.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.9C
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• pp.1337-1344
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• 2004

This paper deals with analysis and real-time Implementation of a wide band adaptive multirate speech codec (AMR-WB) using a fixed-point DSP of TI's TMS320C6201. In the AMR-WB codec, input speech is divided into two frequency bands, lower and upper bands, and processed independently. The lower band signal is encoded based on the ACELP algorithm and the upper band signal is processed using the random excitation with a linear prediction synthesis filter. The implemented AMR-WB system used 218 kbytes of program memory and 92 kbytes of data memory. And its proper operation was confirmed by comparing a decoded speech signal sample-by-sample with that of PC-based simulation. Maximum required time of 5 75 ms for processing a frame of 20 ms of speech validates real-time operation of the Implemented system.

• The Journal of the Acoustical Society of Korea
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• v.24 no.5
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• pp.248-254
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• 2005

In this Paper, the structure of a split bandwidth wideband speech coder and its highband coder for tone qualify elevation are Proposed. The lowband and highband by the split bandwidth method are encoded independently applying the G.729E and MLT (Modulated Lapped Transform) residual model. In the highband structure which is encoded by low bit rate of 4kbps, the MLT residual signals are distinguished to voice and unvoice signal . The voice signals are applied to MLT peak picking method by lowband pitch period. Because transformed MLT residual signals are represented by periodic signal that have periodic peak. The unvoice signals are applied to MLT which linear prediction spectral response is added and do vector quantization. Performance for proposed 15.8kbps wideband speech coder was verified through subjective listening test.

• The Journal of the Acoustical Society of Korea
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• v.28 no.2
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• pp.148-154
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• 2009

In audio coding, SBR technology synthesizes the high-bands using patched time-frequency information from low-bands and the correction parameters, Since SBR transmits only correction parameters for high-bands, it provides a low-rate coding of high-bands, and is used as a core module of MPEG-4 HE-AAC, SBR was originally designed for audio signal and its performance for speech signal tends to decrease, and the major reason is an excessive noise floor in high-bands which is caused by incorrect tonality computation, In this paper, a new method to determine noise floor level in an adaptive fashion according to the speech characteristics is proposed in order to solve the problem of SBR for speech signal, The proposed method maintains the compatibility with the standard SBR, and the subjective performance evaluation shows that the proposed method improves the SBR performance especially for male speech signal compared with the standard SBR.

• Proceedings of the KOSOMBE Conference
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• v.1992 no.05
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• pp.175-178
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• 1992

This paper proposes a method called linear prediction (a high performant technique in digital speech processing) for analyzing digital ECG signals. There are several significant properties indicating that ECG signals have an important feature in the residual error signal obtained after processing by Durbin's linear prediction algorithm. The ECG signal classification puts an emphasis on the residual error signal. For each ECG's QRS complex. the feature for recognition is obtained from a nonlinear transformation which transforms every residual error signal to set of three states pulse-cord train relative to the original ECG signal. The pulse-cord train has the advantage of easy implementation in digital hardware circuits to achive automated ECG diagnosis. The algorithm performs very well feature extraction in arrythmia detection. Using this method, our studies indicate that the PVC (premature ventricular contration) detection has a at least 90 percent sensityvity for arrythmia data.

• The Journal of the Acoustical Society of Korea
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• v.32 no.6
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• pp.556-562
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• 2013

This paper is aimed to improve performance of Band-Selection speech/audio Coder reconstucted band spectrum that is not sent by the comfort noise. To improve the performance, we use the Spectral Band Replication(SBR) technique instead of substitution of Comfort noise. To synthesize SBR signal, the SBR algorithm is referenced in selected signals and the spectrum synthesized by SBR is injected to non-selected band. Each sub-band spectrum has been energy-weighted by real audio signal. We propose the enhanced the Band-Selection Coder that utilizes synthesized SBR signal from selected signal instead of comfort noise.

• Journal of Biomedical Engineering Research
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• v.12 no.3
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• pp.185-190
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• 1991

An automatized latency calculation method of an e-OAE(evoked otacoustic emission) is proposed. The e-OAE signal measured from a normal adult is averaged 1000 times to remove noises. This averaged signal is converted to digital signal and that is processed by IBM-AT computer for latency calculation. we separate the stimulated and the emitted signal on the time domain by a modified LPC (linear prediction coding) spectrum, and the latency is calculated by cross-correlation method. By proposed latency calculation method the latency is 7.9[ms] for normal adult. The performance of the proposed method is also compared with that of the auto-correlation and cross performance of the proposed method is also compared with that of the auto-correlation and cross-correlation method. The result show that the proposed method has same precision with the conventional methods and can automatically calculate latency without subjective observation.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.14 no.9
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• pp.2091-2098
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• 2010

This paper proposes a hologram data compression scheme that uses the existing image/video compression techniques, in which the existing techniques are modified appropriately to fit to the characteristics of hologram. In this paper we use CGH as the hologram data. The proposed scheme uses the generation characteristics of a CGH to consist of a pre-processing, spatial segmentation of a CGH, frequency-transformation with 2D-DCT (2-dimensional discrete cosine transform), and motion estimation and residual image generation in the frequency-domain. It uses H.264/AVC, the lossless compressor BinHex, and a linear quantizer that we have made. From the experiments the proposed scheme showed the image quality of about 25.4 dB at the compression ratio of 10:1 and about 16.5dB at 90:1 compression ratio.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.12B
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• pp.2094-2108
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• 2000

오류를 수반하는 통신망을 통한 멀티미디어 데이터의 응용은 최근 그 수요가 급증하고 있다. 하지만 그 구현은 많은 문제점들을 야기하는데, 전송된 비디오 데이터에 발생한 오류를 처리하는 문제가 그 중 하나이다. 이는 압축된 비트열에 발생한 오류가 영상의 시-공간 방향으로 심각한 전파 현상을 수반하기 때문이다. 이러한 심각한 오류 전파를 완화하기 위해 본 논문에서는 EREC라 알려진 오류 제한 기법을 적용하고, 적용된 EREC의 오류 전파 특성을 분석하였다. 이를 통해, 압축 부호화된 하나의 기본 블록 (매크로 블록)이 복호시 오류가 생길 확률을 추정하였으며, 추정된 확률의 근사를 통해 양 끝단(전송단과 수신단)에서의 비디오 화질 열화를 예측하였다. 추정 확률의 근사는 매 기본 블록에서 발생된 비트수에 대한 그 기본 블록이 복호시 오류가 생길 확률을 간단한 1차식을 통한 선형 회귀법으로 모델링 되었으며, 따라서 간단한 방법을 통해 양 끝단의 화질 열화를 효과적으로 예측할 수 있었다. 부호화된 비트열이 전송 오류에 보다 강인하게 되도록 하기 위해, 본 논문에서 개발된 화질 열화 모델을 양자화기 선택에 적용함으로써, 새로운 최적 양자화 기법을 제시하였다. 본 논문에서 제안된 최적 양자화 기법은, 기존의 양자기 최적화 기법들과는 달리, 복호단에서의 복원 영상 화질이 주어진 비트율에서 최적이 되도록 양자화를 수행한다. H.263 비디오 압축 규격에 적용한 제안 양자화 기법의 실험 결과를 통해, 제안 기법이 매우 적은 계산상의 부하를 비용으로 객관적 화질은 물론 주관적 화질까지 크게 개선할 수 있음을 확인할 수 있었다.내었다.Lc. lacti ssp. lactis의 젖산과 초산의 생성량은 각각 0.089, 0.003과 0.189, 0.003M이었다. 따라서 corn steep liquor는 L. fermentum와 Lc. lactis ssp, lactis 의 생장을 위해 질소 또는 탄소 공급원으로서 배지에 첨가 될 수 있는 우수한 농업 부산물로 판단되었다.징하며 WLWQ에 적용되는 몇 가지 제약을 관찰하고 이를 일반적인 언어원리로 설명한다. 첫째, XP는 주어로만 해석되는데 그 이유는 XP가 목적어 혹은 부가어 등 다른 기능을 할 경우 생략 부위가 생략의 복원 가능선 원리 (the deletion-up-to recoverability principle)를 위배하기 때문이다. 둘째, WLWQ가 내용 의문문으로만 해석되는데 그 이유는 양의 공리(the maxim of quantity: Grice 1975) 때문이다. 평서문으로 해석될 경우 WP에 들어갈 부분이 XP의 자질의 부분집합에 불과하므로 명제가 아무런 정보제공을 하지 못한다. 반면 의문문 자체는 정보제공을 추구하지 않으므로 앞에서 언급한 양의 공리로부터 자유롭다. 셋째, WLWQ의 XP는 주제어 표지 ‘는/-은’을 취하나 주어표지 ‘가/-이’는 취하지 못한다(XP-는/-은 vs. XP-가/-이). 이는 IP내부 에 비공범주의 존재 여부에 따라 C의 음운형태(PF)가 시성이 정해진다는 가설로 설명하고자 했다. WLWQ에 대한 우리의 논의가 옳다면, 본 논문은 다음과 같은 이론적 함의를 기닌다. 첫째, WLWQ의 존재는 생략에 대한 두 이론 즉 LF 복사 이론과 PF 삭제 이론