• Journal of the Korean Institute of Telematics and Electronics A
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• v.33A no.7
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• pp.55-63
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• 1996

In this paper, a multiple-access algorithm for PCN is proposed. The proposed algorithm provides th integrated service of information soruces and can be operated stably in high load state. Given bandwidth is efficiently used for it in the microcell environment. And system performance can be improved through the statistical-multiplexing technque. In order to process the speech signal usually requiring real-tiem processing, we adopt a random access of AlOHA type for th ebasic protocol sturcture and assume the form of ALOHA-reservation. We have analyzed the performance of the proposed algorithm through system throughput and packet delay in the microcell environment.

• Proceedings of the IEEK Conference
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• 2000.07b
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• pp.691-694
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• 2000

This paper is addressed to Internet telephony based service implementation. It describes an implementation method which uses ARS as a gateway which combines Internet and traditional public switched telephone network for Internet information service using telephony and fax, is different to traditional Internet telephony which provide enhanced speech quality and low cost functionality. This method allows telephony and/or fax user to get Internet information without additional Internet bill, Internet infrastructure and low connection quality from low signal bandwidth connected him. Implemented system is useful to a special kind information service such as climate information of Korea etc and simpler than WAP based service as for wireless mobile telephony user. We implement job opportunity information and advertisement service supported by Home page of Choong Book Small & Medium Business Administration and e-mail service supported by Korean Society for Rehabilitation of persons with Disabilities to demonstrate the system ability. As a result of test implementation, this service system works good fur blind persons and graduated persons without job, is expected to apply for special Internet information provider via Voice and Fax.

• The Journal of the Acoustical Society of Korea
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• v.15 no.1
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• pp.46-53
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• 1996

Adaptive LMS algorithm has been used in many application areas due to its low complexity. In this paper input signal is transformed into the subbands with arbitrary bandwidth. In each subbands the dynamic range can be reduced, so that the independent filtering in each subbands has faster convergence rate than the full band system. The DCT transform domain LMS adaptive filtering has the whitening effect of input signal at each bands. This leads the convergence rate to very high speed owing to the decrease of eigen value spread Finally, the filtered signals in each subbands are synthesized for the output signal to have full frequency components. In this procedure wavelet filter bank guarantees the perfect reconstruction of signal without any interspectra interference. In simulation for the case of speech signal added additive white gaussian noise, the suggested algorithm shows better performance than that of conventional NLMS algorithm at high SNR.

• The Journal of the Acoustical Society of Korea
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• v.25 no.1
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• pp.21-29
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• 2006

In this Paper a split bandwidth wideband speech coder and its highband coding method are Proposed. The coder uses a split-band approach. where the wideband input speech signal is split into two equal frequency bands from 0-4kHz and 4-8kHz. The lowband and the highband are coded respectively by the 11.8kb/s G.729 Annex E and the proposed coding method. After the LPC analysis, the highband is divided by two modes according to the properties of signals. In stationary mode. the highband signals are compressed by the mixture excitation model; CELP algorithm and W (Matching Pursuit) algorithm. The others are coded by the only CELP algorithm. We compare the performance of the new wideband speech coder with that of G.722 48kbps SB-ADPCM and G.722.2 12.85kbps in a subjective method. The simulation results show that the Performance of the proposed wideband speech coder has better than that of 48kbps G.722 and no better than that of 12.85kbps G.722.2.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.12 no.12
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• pp.5588-5593
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• 2011

In this paper, measure of change the speech analysis parameter by stimulating ears blood points corresponding to cardiac. To do this, we collected voice of before and after a stimulation corresponding points to ears to select normal heart having 10 subjects. We analyzed changes before and after corresponding points to ear in cardiac to apply Jitter, the second zFormant Frequency Bandwidths related to heart of elements of voice analysis. As a result of us experiment, we were able to analyze correlation of voice with cardiac according to corresponding points to ears decreased values of Jitter, the Second Formant Frequency Bandwidths of 90% of subjects. Finally, the effectiveness of proposed method is demonstrated by several experiments.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.8C
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• pp.575-582
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• 2008

In this paper we described the process and the results of real-time implementation of G.729.1 wideband speech codec which is standardized in SG15 of ITU-T. To apply the codec on ARM926EJ-S(R) processor core. we transformed some parts of the codec C program including basic operations and arithmetic functions into assembly language to operate the codec in real-time. G.729.1 is the standard wideband speech codec of ITU-T having variable bit rates of $8{\sim}32kbps$ and inputs quantized 16 bits PCM signal per sample at the rate of 8kHz or 16kHz sampling. This codec is interoperable with the G.729 and G.729A and the bandwidth extended wideband($50{\sim}7,000Hz$) version of existing narrowband($300{\sim}3,400Hz$) codec to enhance voice quality. The implemented G.729.1 wideband speech codec has the complexity of 31.2 MCPS for encoder and 22.8 MCPS for decoder and the execution time of the codec takes 11.5ms total on the target with 6.75ms and 4.76ms respectively. Also this codec was tested bit by bit exactly against all set of test vectors provided by ITU-T and passed all the test vectors. Besides the codec operated well on the Internet phone in real-time.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.8B
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• pp.830-838
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• 2009

Oriental medicine lacks diagnosis data in fixed quantity possible to express visually to patients by depending on clinician's intuition than Western medicine that continues to development by various diagnosis devices. For that, this paper intends to examine relation between heart and voice signal regarded as center organ and source of life and mind in order to implement objectification through the visualization of oriental diagnosis method above all. According to because the heart is related to the tongue among five organs, by thinking with sounds, we would design the way of identifying existence of heart diseases focused on the fact that lingual sound pronunciation of heart patient is inexact. For this, we achieved a comparison, analysis of statistical bandwidth and morphological modeling of the second formants frequency about a lingual sound for their voice constituted subject group of heart diseases and normal people. Finally, we analyzed interrelationship to the result of experiment by designed method.