• The Journal of Korean Institute of Communications and Information Sciences
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• v.26 no.6A
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• pp.923-928
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• 2001

본 논문은 레이크 수신기의 성능 개선에 관한 것으로 다중 경로 페이딩에 의한 시간 지연으로 나타나는 자기 간섭이 제거된 DS-CDMA/QPSK 신호의 성능을 분석하였다. 자기 간섭 제거를 위해 새로운 레이크 수신기를 제안하고 제안된 수신기의 시스템 성능은 차량 이동통신 환경, 실내 이동통신 환경과 실외에서 실내로 이동중인 환경으로 구분하여 라이시안과 레일리 페이딩 환경에서 분석하였다. 제안된 레이크 수신기는 적용한 DS-CDMA 시스템의 합성 수신 SNR 증가로 오율 성능이 개선되었으며 오율 $10^{-3}$에서 K=6[dB]일 경우에 1[dB]의 성능 개선 효과가 나타났으며, K=3.7[dB]일 경우에는 1.5[dB], K=0[dB]일 경우에는 2[dB]의 성능 개선 효과를 얻을 수 있었다.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.46 no.1
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• pp.89-93
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• 2009

In this paper, we investigated effect of sensing time on the performance of enhanced spectrum sensing method, which is the energy detector with multiples of verification using time delay, under Suzuki channel. We assumed that SNR is 1dB, $P_{FA}=0.1$ and various mobile speed such as 3, 60, 110 km/h. The performance is investigated by simulation and compared to that of conventional energy detector.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2008.05a
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• pp.784-788
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• 2008

In this research, we propose a remote speech input unit, a new method of user-friendly speech input in speech recognition system. We focused the user friendliness on hands-free and microphone independence in speech recognition applications. Our module adopts two algorithms, the automatic speech detection and speech enhancement based on the microphone array-based beamforming method. In the performance evaluation of speech detection, within-200msec accuracy with respect to the manually detected positions is about 97percent under the noise environments of 25dB of the SNR. The microphone array-based speech enhancement using the delay-and-sum beamforming algorithm shows about 6dB of maximum SNR gain over a single microphone and more than 12% of error reduction rate in speech recognition.

• ETRI Journal
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• v.35 no.3
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• pp.371-377
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• 2013

An effective signal detection algorithm with low complexity is presented for multiple-input multiple-output orthogonal frequency division multiplexing systems. The proposed technique, QR-MLD, combines the conventional maximum likelihood detection (MLD) algorithm and the QR algorithm, resulting in much lower complexity compared to MLD. The proposed technique is compared with a similar algorithm, showing that the complexity of the proposed technique with T=1 is a 95% improvement over that of MLD, at the expense of about a 2-dB signal-to-noise-ratio (SNR) degradation for a bit error rate (BER) of $10^{-3}$. Additionally, with T=2, the proposed technique reduces the complexity by 73% for multiplications and 80% for additions and enhances the SNR performance about 1 dB for a BER of $10^{-3}$.

• Journal of Digital Convergence
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• v.10 no.10
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• pp.277-282
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• 2012

The speech recognition system can not quickly adapt to varied environmental noise factors that degrade the performance of recognition. In this paper, the echo noise robust HMM learning model using average estimator LMS algorithm is proposed. To be able to adapt to the changing echo noise HMM learning model consists of the recognition performance is evaluated. As a results, SNR of speech obtained by removing Changing environment noise is improved as average 3.1dB, recognition rate improved as 3.9%.

• Proceedings of the IEEK Conference
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• 2000.06d
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• pp.98-101
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• 2000

On CELP type Vocoders G.723.1 6.3kbps/5.3kbps Dual Rate Speech Codec, which is developed for Internet Phone and videoconferencing, uses VAD(Voice Activity Detection)/CNG (Comfort Noise Generator) in order to reduce the bit rate in a silence period. In order to reduce the bit rate effectively in this paper, we first set the boundary condition of the energy threshold to prevent the consumption of unnecessary processing time, and use three decision rules to detect an active frame by energy, pitch gain and LSP distance. To evaluate the performance of the proposed algorithm we use silence-inserted speech data with 0, 5, 10, 20dB of SNR. As a result when SNR is over 5dB, the bit rate is reduced up to about 40% without speech degradation and the processing time is additionally decreased.

• Proceedings of the IEEK Conference
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• 2004.06a
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• pp.11-14
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• 2004

Digital Video Broadcasting - Satellite (DVB-S) [1] (EN 300 421(bibliography)) was introduced as a standard in 1994. However, by combing with higher order modulation, promise more powerful alternatives to the DVB-S / DVB-DSNG coding and modulation schemes. Variable rate coding and modulation (VCM) may employed to provide different levels of error protection to different service components. Adaptive coding and modulation (ACM) provides more exact channel protection and dynamic link adaptation to propagation conditions, targeting each individual receiving terminal. By these reasons, DVB-S2 introduced. This paper derives exact symbol error rate(SER) of 16-Amplitude Phase Shift Keying(APSK) modulation by using Craig's formula. 16-APSK modulation is used in DVB-S2. The difference between Union Bound and Craig's formula is 1.26dB in low SNR and 0.1dB in high SNR.

• The Journal of the Acoustical Society of Korea
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• v.14 no.1
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• pp.5-12
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• 1995

In the case where the set of training vectors constitute clusters, the codevectors of the codebook which is used to compression for speech and images in the vector quantization are regarded as the central vectors of the clusters constituted by given training vectors. In this work, we consider the distribution of Euclidean distance obtaining in the process of searching for the minimum distance between vectors, and propose the method searching for the proper number of and the central vectors of clusters. And then, the proposed method shows more than the about 4[dB] SNR than the LBG algorithm and the competitive learning algorithm

• Journal of information and communication convergence engineering
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• v.10 no.2
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• pp.162-167
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• 2012

This paper proposes a speech processing system based on a model of the human auditory system and a noise reduction neural network with fast Fourier transform (FFT) amplitude and phase spectrums for noise reduction under background noise environments. The proposed system reduces noise signals by using the proposed neural network based on FFT amplitude spectrums and phase spectrums, then implements auditory processing frame by frame after detecting voiced and transitional sections for each frame. The results of the proposed system are compared with the results of a conventional spectral subtraction method and minimum mean-square error log-spectral amplitude estimator at different noise levels. The effectiveness of the proposed system is experimentally confirmed based on measuring the signal-to-noise ratio (SNR). In this experiment, the maximal improvement in the output SNR values with the proposed method is approximately 11.5 dB better for car noise, and 11.0 dB better for street noise, when compared with a conventional spectral subtraction method.

• Journal of the Korean Institute of Telematics and Electronics
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• v.21 no.6
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• pp.105-114
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• 1984

A fast pseudo-noise (PN) sequence acquisition algorithm for the direct-sequence (DS) spread spectrum system is proposed. The basic concept of the algorithm has been adopted from that of the classical sliding correlator. Mathematical modeling, analysis and computer simulation of the proposed system have been done. The results of analysis and computer simulation show that the acquisition system yields a significant performance improvement over the sliding correlator. Its acquisition time takes only 45 ms when signal-to-noise ratio(SNR) is -18dB. The algorithm developed has been implemented in hardware and its experimental result is also given.