• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.15 no.5
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• pp.335-342
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• 2022

In this paper, a single-bit 3rd-order feedforward delta sigma modulator is proposed for audio applications. The proposed modulator is based on a class-C inverter for low voltage and power applications. For the high-precision requirement, the class-C inverter with regulated cascode structure increases its DC gain and acts as a low-voltage subthreshold amplifier. The proposed Class-C inverter-based modulator is designed and simulated in 180-nm CMOS process. With no performance loss and a low supply voltage compatibility, the proposed class-C inverter-based switched-capacitor modulator achieves high power efficiency. This design achieves an signal-to-noise-and-distortion ratio (SNDR) of 93.9 dB, an signal-to-noise ratio (SNR) of 108 dB, an spurious-free dynamic range (SFDR) of 102 dB, and a dynamic range (DR) of 102 dB at a signal bandwidth of 20 kHz and a sampling frequency of 4 MHz, while only using 280 μW of power consumption from a 0.8-V power supply.

• The Journal of the Acoustical Society of Korea
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• v.17 no.2E
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• pp.38-46
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• 1998

In this paper, we propose a remote speech input device, a new method of user-friendly speech input in spontaneous speech recognition system. We focus the user friendliness on hands-free and microphone independence in speech recognition applications. Our method adopts two algorithms, the automatic speech detection and the microphone array delay-and-sum beamforming (DSBF)-based speech enhancement. The automatic speech detection algorithm is composed of two stages; the detection of speech and nonspeech using the pitch information for the detected speech portion candidate. The DSBF algorithm adopts the time domain cross-correlation method as its time delay estimation. In the performance evaluation, the speech detection algorithm shows within-200 ms start point accuracy of 93%, 99% under 15dB, 20dB, and 25dB signal-to-noise ratio (SNR) environments, respectively and those for the end point are 72%, 89%, and 93% for the corresponding environments, respectively. The classification of speech and nonspeech for the start point detected region of input signal is performed by the pitch information-base method. The percentages of correct classification for speech and nonspeech input are 99% and 90%, respectively. The eight microphone array-based speech enhancement using the DSBF algorithm shows the maximum SNR gaing of 6dB over a single microphone and the error reductin of more than 15% in the spontaneous speech recognition domain.

• The Journal of the Korea institute of electronic communication sciences
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• v.8 no.4
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• pp.555-560
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• 2013

This paper first proposes a classification algorithm which detects a voiced, unvoiced, and silence signal using distance measure, logarithm power and root mean square methods at each frame, then a spectrum subtraction algorithm based on a subband filter. The proposed algorithm subtracts spectrums of white noise and street noise from noisy signal based on the subband filter at each frame. In this experiment, experimental results of the proposed spectrum subtraction algorithm demonstrate using the speech and noise data of Aurora-2 database. Based on measuring the speech-to-noise ratio (SNR), experiments confirm that the proposed algorithm is effective for the speech by contaminated the noise. From the experiments, the improvement in the output SNR values was approximately 2.1 dB and 1.91 dB better for white noise and street noise, respectively.

• Journal of Navigation and Port Research
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• v.29 no.3 s.99
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• pp.245-250
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• 2005

In this paper, considering PAR of transmitter which is pointed out OFDM system's problem, we designed Coded OFDM (COFDM) system and estimated BER and SNR using PAR reduction method In order to evaluate performance, we compared M-ary PSK (M-ary Phase Shift Keying) with M-ary QAM (M-ary Quadrature Amplitude Modulation). In result, performance of 16-PSK and 16-QAM came to good Moreover, 16-QAM showed better performance of about 2 dB in 10-3 error probability and performance of about 5 dB in Peak power clipping than that of 16-PSK.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.11 no.6
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• pp.411-420
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• 1986

Quantization noise of nonadaptive Linear Delta Modulation(LDM) and adaptive Constant Factor Delta Modulation(CFDM) systems is studied. The formulas for quantization noise of CFDM system are derived on the basis of the rusults of LDM. And the output signal-to-quantization noise ratios(SNR) in LDM and CFDM systems are calculated in the range of bit rates from 16[Kb/s] to 96[Kb/s]. By comparing LDM and CFDM, it is known that the adaptive DM is superior to non-adaptive DM by 8[dB] when bit rate is 20[Kb/s] and SNR advantage increases to 14[dB] when bit rate is 56[Kb/s]. All the theoretical results agree well with the experimental results.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.7C
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• pp.665-671
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• 2006

In this paper, we propose and observe a system that adopts Independent-MCS (Modulation and Coding Scheme) level for each layer in the combined AMC-V-BLAST (Adaptive Modulation and Coding-Vertical-Bell-lab Layered Space-Time) system. Also, comparing with the combined system using Common-MCS level, we observe throughput performance improvement. As a result of simulation, Independent-MCS level case adapts modulation and coding scheme for maximum throughput to each channel condition in separate layer, resulting in improved throughput compared to Common-MCS level case. Especially, the results show that the combined AMC-V-BLAST system with Independent-MCS level achieves a gain of 700kbps in $7dB{\sim}9dB$ SNR (Signal-to-Noise Ratio) range.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.15 no.12 s.91
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• pp.1207-1215
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• 2004

A Gigabit Ethernet upstream transmission over a WDM-PON employing remotely wavelength-locked Fabry-Perot lasers has been demonstrated. We have successfully demonstrated a WDM transmission of four Gigabit Ethernet channels with 100 GHz channel spacing over 30 km conventional single mode fiber. The measured f-factor was larger than 17.1 dB. We have also investigated the beating noise characteristics of a wavelength-locked Fabry-Perot laser and showed the remotely wavelength-locked Fabry-Perot laser suppresses the intensity noise of the incoherent light injected, which cause a 6.3 dB SNR improvement compared with that of the conventional spectrum-sliced light source.

• The Journal of the Acoustical Society of Korea
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• v.16 no.1
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• pp.10-15
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• 1997

In this paper, an efficient LSF quantizer, named improved PSVQ(IPSVQ), is proposed to apply in the 8 kbps QCELP speech coder. By using 27 bits IPSVQ instead of 40 bits DPCM quantizer per frame, we can save 13 bits/frame and allocate those bits to the codebook gain and the pitch gain parameters. Hence we improve the overall performance of the QCELP codec. The enhanced QCELP shows the performance improvement of 0.9 dB SNR and 0.4 dB SEGSNR. Informal listening tests also confirm the improvement in the speech quality.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.1C
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• pp.96-102
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• 2010

This paper proposes a new adaptive algorithm to cancel the acoustic feedback and noise signals in the digital hearing aids. The proposed algorithm improves its convergence performances by canceling the speech signal from the residual signal using two microphones. The feedback canceller firstly cancels the feedback signal among the mic signal, and then it is reduced the noise using the beamforming method. To verify the performances of the proposed algorithm, the simulations were carried out for some cases. As the results of simulations, it was proved that the feedback canceller and the noise canceller advance about 14.43 dB for SFR, 10.19 dB for SNR respectively during speech, in the case of using the new algorithm.

• Journal of the Korean Society of Radiology
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• v.12 no.4
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• pp.541-547
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• 2018

To evaluate the effect of dose and image quality for Chest Digital Tomosynthesis(CDT) using sensitivity and tube voltage(kV). CDT images of the phantom were acquired varying sensitivity 200, 320, 400 according to set tube voltage of 125 kV and 135 kV. The dose and Dose Area Product(DAP) according to change of sensitivity and kV were evaluated and Image quality was evaluated by PSNR, CNR, SNR using Image J. Dose were lowered 14~23% less than sensitivity 200, 125 kV and DAP were lowered 13~26% less than sensitivity 200, 125 kV. PSNR were over 27 dB, which were significant value and CNR, SNR were better as sensitivity value was lower. But there were different statistical significant to each item. CNR and SNR were not statistically significant at sensitivity 320, 135 kV(P>0.05). CDT can improve image quality with lower radiation dose using better than quality and correction power at digital radiography system.