• Proceedings of the IEEK Conference
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• 2000.09a
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• pp.123-126
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• 2000

STC(Sinusoidal Transform Coding) 방식은 음성신호의 주파수 영역에서 스펙트럼 피크치들을 정현파로 모델링하여 합성하는 방식을 말한다. 저전송률 STC 방식에서는 전송되는 정보량을 줄이기 위해 스펙트럼 피크를 대신해 음성신호의 스펙트럼 포락선 정보와, 피치정보를 이용하여 얻어지는 고조파 성분들을 정현파로 모델링하여 음성을 합성한다. 본 논문에서는 음성신호의 정현파 모델에 기반하여 2.4kbps 전송속도를 갖는 음성부호화 알고리즘을 제안하였으며, 실험결과로 합성음의 파형과 스펙트럼 특성, 위상특성, 그리고 MOS(Mean Opinion Score) 테스트를 이용한 합성음의 음질을 비교/분석 하였다.

• The Journal of the Acoustical Society of Korea
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• v.24 no.1
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• pp.43-51
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• 2005

In this paper, we propose an algorithm that improves a tone quality of a noisy audio signal in order to enhance a Performance of perceptual filter using noise energy control. Most of the algorithms which were proposed by the other researchers usually applied a filter using the noise energy acquired from a silent range. In this case. the improvement rate of tone quality decreases if the noise energy is changed by the magnitude or environment variation in a signal frame. But the Proposed method Provides the means to find a food estimated noise through energy control of the estimated noise which is obtained from a silent range. Also we can get the enhancement of tone qualify in low frequency band unlike other methods. To show the performance of the Proposed algorithm, various input signals which had a different signal-to-noise ratio (SNR) such as 5dB, l0dB, 15dB and 20dB were used to test the proposed algorithm. With the proposed algorithm, we could confirm the enhancement of tone quality in terms of segmental SNR (SSNR). noise-to-mask ration (NMR) and mean opinion score (MOS) test.

• Journal of the Institute of Electronics Engineers of Korea SD
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• v.46 no.3
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• pp.1-9
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• 2009

In this paper proposed a new method which adaptively eliminates comparator offsets using the threshold voltage shift by the Fowler-Nordheim stress. The method evaluates the sign of comparator offset and gives the FN stress to the stronger MOSFETs of the comparator, leading to offset reduction. We have used an appropriate stressing operation, named 'stress-packet', in order to converge the offset value to zero. We applied the method to the latch-type comparator which is prevalently used for DRAM bitline sense amplifier, and verified through experiments that offsets of the latch-type comparators are nearly eliminated with the stress-packet operations. We also discuss about the reliability issues that must be guaranteed for field application of this method.

• Journal of the Korean Institute of Intelligent Systems
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• v.15 no.2
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• pp.230-235
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• 2005

In this paper, we propose an algorithm that improves a tone quality of a noisy audio signal in order to enhance a performance of perceptual filter using intelligent estimation of noise pattern from a band degraded by additive noise. The proposed method doesn't use the estimated noise which is obtained from silent range. Instead new estimated noise according to the power of signal and effect of noise variation is considered for each frame. So the noisy audio signal is enhanced by the method which controls a estimation of noise Pattern effectively in a noise corruption band. To show the performance of the proposed algorithm, various input signals which had a different signal-to-noise ratio(SNR) such as $5\cal{dB},\;10\cal{dB},\;15\cal{dB}\;and\;20\cal{dB}$ were used to test the proposed algorithm. we carry out SSNR and NMR of objective measurement and MOS test of subjective measurement. An approximate improvement of $7.4\cal{dB},\;6.8\cal{dB},\;5.7\cal{dB},\;5.1\cal{dB}$ in SSNR and $15.7\cal{dB},\;15.5\cal{dB},\;15.2\cal{dB},\;14.8\cal{dB}$ in NMR is achieved with the input signals, respectively. And we confirm the enhancement of tone quality in terms of mean opinion score(MOS) test which is result of subjective measurement.

• The Journal of the Acoustical Society of Korea
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• v.23 no.6
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• pp.446-453
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• 2004

In this paper, a new adaptive perceptual filter using variable threshold to enhance audio signals degraded by additively nonstationary noise is proposed. The adaptive perceptual filter updates variable threshold each time according to the power of signal and the effect of noise variation. So the noisy audio signal is enhanced by the method which controls a residual noise effectively. The proposed algorithm uses the perceptual filter which transforms a time domain signal into frequency domain and calculates an intensity energy and an excitation energy in bark domain. In this method. the stage updated the response of filter is decided by threshold. The proposed algorithm using vairable threshold effectively controls a residual noise using the energy difference of audio signals degraded by the additive nonstationary noise. The proposed method is tested with the noisy audio signals degraded by nonstationary noise at various signal -to-noise ratios (SNR). We carry out NMR and MOS test when the input SNR is 15dB. 20dB. 25dB and 30dB. An approximate improvement of 17.4dB. 15.3dB, 12.8dB. 9.8dB in NMR and enhancement of 2.9, 2.5, 2.3, 1.7 in MOS test is achieved with the input signals. respectively.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.291-294
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• 2004

본 논문에서는 잡음의 변화(variance)가 없는 정적인 통계적 특성(Stationary Static Characteristic)을 갖는 환경에서 잡음 추정을 통해 지각 필터의 성능을 개선하는 알고리즘을 제안한다. 제안된 잡음 추정 알고리즘은 입력되는 잡음에 열화 된 신호의 묵음 구간에서 추정된 잡음을 이용하여 입력되는 잡음의 SNR을 추정 후, 대역 별로 smoothing 상수 값으로 잡음 에너지를 제어하여 첨가된 잡음을 추정함으로써 초기 추정 잡음 보다 가까운 추정 잡음을 얻을 수 있게 된다. 이는 신호를 열화 시킨 잡음을 보다 정확한 추정을 제공함으로써, 지각 필터의 응답을 개선할 수 있고 더불어 잡음에 의해 열화 된 신호의 음질을 개선할 수 있다. 또한 저 대역에 영향을 미치는 잡음인 경우 다른 방법들과는 달리 음질의 개선이 뚜렷하다. 기존의 방식과 비교를 위해 다양한 신호 대 잡음 비(signal-to-noise ratio, SNR)에서 열화 된 오디오 신호를 입력으로 사용하였다. 입력 SNR이 5dB, 10dB, 15dB와 20dB의 각각의 경우에 대하여 SSNR(Segmental SNR)과 잡음 대 마스킹 비(Noise-to-mask ratio, NMR), 음질 테스트를 수행한 결과, 청감 테스트(Mean Opinion Score, MOS Test) 결과의 향상과 음질개선의 개선을 확인할 수 있었다.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.40 no.6
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• pp.84-89
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• 2003

In this paper, we real-time implemented to the TMS320C5416 the vocoder of variable bit rate applied the SOLA-B algorithm by Henja to the ITU-T G.729A vocoder of 8kbps transmission rate. This proposed method using the SOLA-B algorithm is that it is reduced the duration of the speech in encoding and is played at the speed of normal by extending the duration of the speech in decoding. At this time, we bandied that the interval of cross correlation function if skipped every 3 sample for decreasing the computational amount of SOLA-B algorithm. The real-time implemented vocoder of C.729A and SOLA-B algorithm is represented the complexity of maximum that is 10.2MIPS in encoder and 2.8MIPS in decoder of 8kbps transmission rate. Also, it is represented the complexity of maximum that is 18.5MIPS in encoder and 13.1MIPS in decoder of 6kbps, it is 18.5MIPS in encoder and 13.1MIPS in decoder of 4kbps. The used memory is about program ROM 9.7kwords, table ROM 4.5kwords, RAM 5.1 kwords. The waveform of output is showed by the result of C simulator and Bit Exact. Also, for evaluation of speech quality of the vocoder of real-time implemented variable bit rate, it is estimated the MOS score of 3.69 in 4kbps.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.71-74
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• 2000

전보(Telegram)는 우리의 생활에서 필요한 정보통신 시스템이다. 지금까지 전보의 기능은 긴급하거나 짧은 정보의 내용을 전화국의 메신저를 통해 문자위주로 변환하여 수신자에게 전달하고있다. 이와는 달리 음성 전보 시스템(Voice Telegram System)은 기존의 문자전달방식에 음성을 가미한 것으로서 발신자의 목소리도 함께 기록하여 수신자에게 전달하는 방식이다. 본 논문에서 음성 전보에 적합한 알고리즘을 새로이 제안하다. 유/무성음 판별 후 유성음에 대해 피치주기별로 유사도를 측정하여 유사도가 낮은 파형은 전송하고 유사도가 높은 파형은 피치와 에너지 정보만 전송하는 방법이다. 실험결과 전체 음성에 대해서 $41\%$의 압축율과 MOS 테스트 결과 4.1을 얻을 수 있었다.

• Proceedings of the Acoustical Society of Korea Conference
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• 1998.06c
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• pp.151-154
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• 1998

이중 전송율(5.3 과 6.3kbit/s)을 제공하는 G.723.1 음성 코더는 공중망을 통한 H.324 POTS 영상 회의 규격의 음성 코더로 채택된 것으로, MPMLQ, ACELP 알고리즘에 근거한다. 본 논문에서는 Annex A를 포함한 G.723.1 음성 코더 알고리즘을 C&S Technology에서 개발한 음성 신호 처리를 위한 범용 DSP인 CSD17C00 칩을 이용하여 실시간 응용이 가능하도록 구현하였다. G.723.1 에 대한 양방향 평가가 Codec loopback을 통해 수행되었으며, ITU에서 제공한 테스트 절차에 따라 평가되었다. 또한, 본 논문에서 구현된 G.723.1 음성 코더는 27MIPS의 계산 속도를 갖으며, 프로그램 ROM의 크기는 8.85K Words이고, 10K 데이터 ROM과 4K 데이터 RAM을 필요로 하고 있다. 경쟁 제품과의 MOS 측정 음질 평가를 실시한 결과, CSD17C00에서의 음질 성능이 더 우수함을 입증 함으로써, 본 논문에서 보여준 CSD17C00을 기반으로 구현된 G.723.1 알고리즘의 실시간 구현기술의 타당성을 검증하게 되었다.

• Journal of the Korean Institute of Telematics and Electronics C
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• v.34C no.8
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• pp.8-21
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• 1997

In this paper, we consider resistive shorts on gate-source, gate-drain, and drain-source as well as opens in MOS FETs included in typical memory cell of VLSI SRAM and analyze behavior of memory by using PSPICE simulation. Using conventional fault models and this behavioral analysis, we propose linear testing algorithm of complexity O(N) which can be applied to both functional testing and IDDQ (quiescent power supply current) testing simultaneously to improve functionality and reliability of memory. Finally, we implement BIST (built-in self tsst) circuit and BICS(built-in current sensor), which are embedded on memory chip, to carry out functional testing efficiently and to detect various defects at high-speed respectively.