• Proceedings of the Acoustical Society of Korea Conference
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• 1998.08a
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• pp.136-141
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• 1998

In order to enhance the performance of a CELP coder at low bit rates, it would be necessary to make the CELP excitation have the peaky pulse characteristic. In this paper we introduce an excitation signal with peaky pulse characteristic. It is obtained by using a two-tap pitch predictor. Samples of the signal have different gains according to their amplitudes by the predictor. In voiced sound the signal has the desirable peaky pulse characteristic, and its periodicity is well reproduced. Particularly, peaky pulses at voiced onset and a burst of plosive sound are clearly reconstructed.

• Proceedings of the Acoustical Society of Korea Conference
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• 1995.06a
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• pp.137-140
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• 1995

In the paper, CDMA's vocoder algorithm, QCELP, was analyzed. And, 16-bit programmable DSP core for QCELP was designed. When it is used two MACs in DSP, we can implement low-power DSP and estimate decrease of parameter computation speed. Also, we implemented in FIFO memory using register file to increase the access time of the data. This DSP was designed using logic synthesis tool, COMPASS, by top-down design methodology. Therefore, it is possible to cope with rapid change at mobile communication market.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.13 no.5
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• pp.2313-2318
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• 2012

With the widely applications of tactical radio networks, end-to-end secure speech communication in the heterogeneous network has become a very significant security issue. High-grade end-to-end speech security can be achieved using encryption algorithms at user ends. However, the use of encryption techniques results in a problem that encrypted speech data cannot be directly transmitted over heterogeneous tactical networks. That is, the decryption and re-encryption process must be fulfilled at the gateway between two different networks. In this paper, in order to solve this problem and to achieve optimal end-to-end speech security for heterogeneous tactical environments, we propose a novel mechanism for end-to-end secure speech transmission over ultra high frequency (UHF) and public switched telephone network (PSTN) and evaluate against the performance of conventional mechanism. Our proposed mechanism has advantages of no decryption and re-encryption at the gateway, no processing delay at the gateway, and good inter-operability over UHF and PSTN.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.89-92
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• 2004

본 논문의 목적은 이동통신 환경 하에서 음성 인식과 음성 부호화를 성능의 저하 없이 동시에 수행하기 위한 기법을 개발하는 것에 있다. 이를 구현하기 위해 통신상에서 전송되는 음성 특징 파라미터는 기존 음성 부호화기의 LPC 대신 음성 인식 파라미터인 MFCC를 사용하였다. 따라서 음성 인식 성능은 향상된다 하지만 음성 재생을 위해 MFCC를 LPC로 변환하는 과정에서 오차가 발생하여 전송되는 bit 수에 비해 만족할만한 음질을 얻을 수 없다. 따라서 이 오차를 보상하여야 하며 이를 위한 변수를 추가하여 음질을 개선시켰다. 그 결과 음질과 음성 인식에서 안정된 성능을 보이는 음성 부호화기를 개발하였다.

• Journal of the Korean Institute of Telematics and Electronics A
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• v.33A no.5
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• pp.172-179
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• 1996

In the paper, QCELP (qualcomm code excited linear predictive), CDMA (code division multiple access)'s vocoder algorithm, was analyzed. And then, a ptich parameter seaarch architecture for 16-bit programmable DSP(digital signal processor) for QCELP was designed. Because we speed up the parameter search through high speed DSP using two MACs, we can satisfy speech codec specifiction for the digital celluar. Also, we implemented in FIFO(first-in first-out) memory using register file to increase the access time of data. This DSP was designed using COMPASS, ASIC design tool, by top-down design methodology. Therefore, it is possible to cope with rapid change at mobile communication market.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.179-182
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• 2002

본 논문에서는 음성 압축기를 사용하는 디지털 이동통신 환경에서 한글 음성 인식기의 성능을 분석하기 위하여 다양한 표준 음성 압축기를 이용하여 음성 압축기의 구조, 전송률, 전송 채널의 에러율에 대한 성능을 측정하여 비교하였다. 동일한 구조의 음성 압축기에 대하여 전송률의 증가에 따라 음성 인식률이 증가하지만, 음성 압축기의 구조에 따라 동일 전송률에서도 많은 성능 차이가 발생하는 것을 확인하였다. 특히 IS-127 EVRC의 인식 성능이 매우 떨어지는 것을 알 수 있고, EVRC의 잡음 제거기와 가변 전송률에 의하여 음성 인식 성능이 저하되는 것을 확인하였다. 이를 통하여 청취 음질과 음성 인식 성능 사이의 상관 관계가 높지 않는 것을 알 수 있다. 모든 음성 압축기에 대하여 채널 에러율과 음성 인식기의 성능은 매우 밀접한 관계가 있음을 확인하였고, 평균적으로 채널 에러율 $1.0\%$에서 인식률이 $0.6\%$ 감소하고, 에러 $5.0\%$에서 인식률이 $1.8\%$ 감소한다.

• The Journal of the Acoustical Society of Korea
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• v.25 no.6
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• pp.282-290
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• 2006

This paper proposes a new Digital Reverberator that models Analog Helical Coil Spring Reverberator for guitar amplifiers. While the conventional digital reverberators are proposed to provide better sound field mainly based on room acoustics, no algorithm or analysis of digital reverberators those model Helical Coil Spring Reverberator was proposed. Considering the fact that approximately $70{\sim}80$ percent of guitar amplifiers are still with Helical Coil Spring Reverberator, research was performed based not on Room Acoustics but on Helical Coil Spring Reverberator itself as an effector. After performing simulations with proposed algorithm, it was confirmed that the Digital Reverberator by proposed algorithm provides perceptually equivalent response to the conventional Analog Helical Coil Spring Reverberators.

• The Journal of the Acoustical Society of Korea
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• v.20 no.3
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• pp.42-47
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• 2001

This paper improves the performance of VAD (Voice Activity Detector) in G.723.1 Annex A 6.3kbps/5.3kbps dual rate speech coder, which is developed for Internet Phone and videoconferencing. The VAD decision is based on a three-level energy threshold. We evaluates for processing time, speech quality, and bit rate. The processing time is reduced due to the accuracy of VAD decision on the silence period. On subjective quality test there is almost no difference compared with the G.723.1. In order to measure the bit rate we count the active speech frame (VAD=1) and we can reduce more bit rate as silence periods are shown.

• The Journal of the Acoustical Society of Korea
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• v.30 no.4
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• pp.190-196
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• 2011

Bandwidth extension is a technique to improve speech quality and intelligibility, extending from 300-3400 Hz narrowband speech to 50-7000 Hz wideband speech. This paper designs an artificial bandwidth extension (ABE) module embedded in the AMR (adaptive multi-rate) decoder, reducing LPC/LSP analysis and algorithm delay of the ABE module. We also introduce a fast search codebook mapping method for ABE, and design a low power BWE technique based on the AMR decoder. The proposed ABE method reduces the computational complexity and the algorithm delay, respectively, by 28 % and 20 msec, compared to the traditional DTE (decode then extend) method. We also introduce a weighted classified codebook mapping method for constructing the spectral envelope of the wideband speech signal.

• The Journal of the Acoustical Society of Korea
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• v.20 no.1
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• pp.43-49
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• 2001

EVRC (Enhanced Variable Rate Codec) has been adopted as a standard coder for the CDMA digital cellular system in North America and Korea, and known to provide good call quality at 8kbps. In this paper, fast implementation algorithms for EVRC encoder are proposed. The proposed algorithms are based on both efficient pitch detection scheme and fast fixed codebook search algorithm. In the codebook search, computational complexity is reduced down to 70％ of the original EVRC by limiting the number of pulse position combination and by using a truncated impulse response. The proposed algorithms enable us to implement the EVRC with much smaller computational works. Also, informal subjective tests confirmed that the difference in the speech quality between the original EVRC and the proposed method was indistinguishable.