• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.7A
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• pp.520-528
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• 2003

This paper proposes a method of implementing synthetic stereo and an acoustic echo cancellation algorithm for multiple participant conference system. Synthetic stereo is generated by HRTF and two loudspeakers. A robust adaptive algorithm for synthetic stereo echo cancellation is proposed to reduce the weight misalignment due to near-end speech signals and ambient noises. The proposed adaptive algorithm is modified version of SMAP algorithm and the coefficients of adaptive filter is updated with cross correlation of input and estimation error signal normalized with sum of the autocorrelation of input signal and the power of the estimation error signal multiplied with projection order. This is more robust to projection order and ambient noise than conventional SMAP. Computer simulation show that the proposed algorithm effectively attenuates synthetic stereo acoustic echo.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.9B
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• pp.1785-1794
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• 1999

The affine projection algorithm has known t require less computational complexity than RLS but have much faster convergence than NLMS for speech-like input signals. But the affine projection algorithm is still much more computationally demanding than the LMS algorithm because it requires the matrix inversion. In this paper, we show that the affine projection algorithm can be realized with the Gram-Schmidt orthogonalizaion of input vectors. Using the derived relation, we propose an approximate but much more efficient implementation of the affine projection algorithm. Simulation results show that the proposed algorithm has the convergence speed that is comparable to the affine projection algorithm with only a slight extra calculation complexity beyond that of NLMS.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.39 no.2
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• pp.10-20
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• 2002

Various blind source separation (BSS) and independent component analysis (ICA) algorithms have been developed. However, comparison study for BSS/ICA algorithms has not been extensively carried out yet. The main objective of this paper is to compare various promising BSS/ICA algorithms in terms of several factors such as robustness to sensor noise, computational complexity, the conditioning of the mixing matrix, the number of sensors, and the number of training patterns. We propose several benchmarks which are useful for the evaluation of the algorithm. This comparison study will be useful for real-world applications, especially EEG/MEG analysis and separation of miked speech signals.

• The Journal of the Acoustical Society of Korea
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• v.32 no.2
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• pp.131-137
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• 2013

In this paper we propose an improved voice activity detection (VAD) algorithm using statistical models in the signal subspace domain. A uncorrelated signal subspace is generated using embedded prewhitening technique and the statistical characteristics of the noisy speech and noise are investigated in this domain. According to the characteristics of the signals in the signal subspace, a new statistical VAD method using GGD (Generalized Gaussian Distribution) is proposed. Experimental results show that the proposed GGD-based approach outperforms the Gaussian-based signal subspace method at 0-15 dB SNR simulation conditions.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.12 no.1
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• pp.195-200
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• 2012

We present a new algorithm which can improve the tracking performance of the double-talk robust algorithm. A detection method of the echo path change and a modification method for the update equation of the conventional adaptive filter are proposed. A duration of the high error signal to scale parameter ratio varies according to the call status and this property is used to detect the echo path change. The proposed update equation of the adaptive filter improves the tracking performance by prohibiting wrong selection of the error signal. Simulations using real speech signals and echo paths of the ITU-T G.168 standard confirmed that as compared to the conventional algorithm, the proposed algorithm improved the tracking performance by more than 4 dB.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.17 no.2
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• pp.225-232
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• 2017

Recently, the development of digital transmission technology of audio signals and the introduction of audio network equipment using digital transmission technology have been made. As a result, audio network technology and equipment are actively applied to the design and construction of audio systems. The meeting room is a place where a large number of participants exchange opinions and communicate with each other. In addition to using an electric acoustic device such as a microphone and a speaker, it improves the intelligibility of the conference room through an example using an audio network.

• Journal of Information Technology Applications and Management
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• v.21 no.3
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• pp.65-77
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• 2014

Ubiquitous learning has aroused great interest and is becoming a new way for foreign language education in today's society. However, how to increase the learners' initiative and their community cohesion is still an issue that deserves more profound research and studies. Emotional intelligence can help to detect the learner's emotional reactions online, and therefore stimulate his interest and the willingness to participate by adjusting teaching skills and creating fun experiences in learning. This is, actually the new concept of smart education. Based on the previous research, this paper concluded a neural mechanism model for analyzing the learners' emotional characteristics in ubiquitous environment, and discussed the intelligent monitoring and automatic recognition of emotions from the learners' speech signals as well as their behavior data by multi-agent system. Finally, a framework of emotional intelligence system was proposed concerning the smart foreign language education in ubiquitous learning.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.14 no.12
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• pp.2798-2804
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• 2010

The audio and image signal are corrupted by various noises in signal processing, many studies are being accomplished to restore those signals. In this paper, the algorithm is proposed to remove additive Gaussian noise and impulse noise at one dimension signal like an speech signal. The algorithm is composed to remove Gaussian noise after removing impulse noise. And the method using wavelet coefficient accumulation is used to remove the Gaussian noise, and the median filter based on element deviation is applied to remove the impulse noise. Also we compare existing methods using SNR(signal-to-noise ratio) as the standard of judgement of improvemental effect.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2013.10a
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• pp.442-444
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• 2013

In this paper, we may by gestures and voice, to know the effect of imitation on the life. Measuring how much about what is imitation unconsciously during conversation to collect data for measuring mimic such speak a short (1 sec) words. It can be used as a basis to predict the outcome of the interview further. Even used as a basis to predict future outcomes. We expect that our group and other researchers will continue to pursue this analysis with larger datasets and across different demographics.

• Journal of the Korean Institute of Telematics and Electronics
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• v.26 no.4
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• pp.151-163
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• 1989

A statistical approach for the segmentation of speed signals is described in this paper. The main idea of this algorithm is the use of three AR models. Two fixed models are identified at the stationary parts of the signal before and after the spectral change. Changes are detected when the distance between these two models is high. Another model is located between two fixed models and is used to estimate spectral change time. This segmentation algorithm has been tested with connected words and compared to classical methods. The results showed that it can provide more accurate locations of boundaries of segments and can reduce the amount of oversegmentation.