• Electronics and Telecommunications Trends
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• v.6 no.3
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• pp.101-109
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• 1991

전화통화시, 송화측음에 따른 송화자의 음성레벨의 변화와 송화시 전화기를 통한 실내소음에 따른 음성레벨의 변화에 대해 알아보았다. 그 결과, 송화자는 항상 자신의 귀로 되돌아 오는 음성의 심리적 크기를 일정하게 유지하려는 모니터 기능을 보여, 측음의 크기와 송화시 실내소음의 크기에 따라 자신의 음성레벨을 제어하는 경향을 보였다.

• Proceedings of the KIEE Conference
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• 2007.07a
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• pp.1866-1867
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• 2007

본 논문에서는 전투기 조종석에서의 음성인식 기술 적용과 관련하여 전투기 음성인식 시스템의 개요, 역사, 구성 및 실제 사용되고 있는 음성명령어(Command Syntax)에 대하여 알아보고, 전투기에 적용되고 있는 음성인식 시스템의 발전 추세를 분석한다.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.39 no.1
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• pp.20-33
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• 2002

Usage of ATM AAL2 packets becomes dominant to increase transmission efficiency of voice traffic in the backbone network. In case of voice service that uses AAL2 mechanism, if resources of network are enough, connection of new call is accepted. However, due to packets generated by the new call, transmission delay of packets from old calls can increase sharply. To control this behavior, in this paper we present an AAL2 buffer management scheme that allocates a virtual buffer to each call and after calculating its propagation delay variation(PDV), decides to drop packets coming from each call according to the PDV value. We show that this packet dropping algorithm can effectively prevent abrupt QoS degradation of old calls. To do this, we analyze AAL2 packet composition process to find a critical factor in the process that influences the end-to-end delay behavior and model the process by K-policy M/D/1 queueing system and MIN(K, Tc)-policy M/D/1 queueing system. From the mathematical model, we derive the probability generating function of AAL2 packets in the buffer and mean waiting time of packets in the AAL2 buffer. Analytical results show that the AAL2 packet dropping algorithm can provide stable AAL2 packetization delay and ATM cell generation time even if the number of voice sources increases dramatically. Finally we compare the analytical result to simulation data obtained by using the COMNET Ⅲ package.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.41 no.1
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• pp.92-100
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• 2016

It is informed that the Voice over LTE(VoLTE) which serves voice and message on IP networks is better in terms of economies of scale than the legacy voice service on 2G/3G circuit-switched networks because of its technological and cost efficiency. In addition, services of voice and data are running on a single LTE network and as a result VoLTE has the more economies of scope. But, there is no study about how much technology-efficiency VoLTE has compared to circuit-based voice service and how much voice charge can be reduced as VoLTE grows up. This paper analyzes empirically cost-efficiency of VoLTE against circuit-based voice service and quantifies the reduction of voice charge as 2G/3G voice traffic shifts to VoLTE. The results describe the first is that the average cost of the total voice traffic rises shortly just after the investment of LTE network for providing VoLTE but it will soon have a capacity available to reduce the charge due to VoLTE's outstanding cost efficiency on the assumption that voice traffic is fixed, and the second is that the charge can be cut to 60% of the current rate in case of all the voice traffic moves to VoLTE. The latter proves partially the validation of data-focusing pricing plan. Our results are expected to become basic data for network operators' establishing pricing strategies and for policy makers' inducing price cutting.

• Journal of Advanced Navigation Technology
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• v.15 no.6
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• pp.946-952
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• 2011

In term of inaccuracy of information and increasing channel occupancy time, the use of voice communication in Air Traffic Control has many problems. In order to improve it, ICAO proposed digital communication and ADS-B system that is more effective for voice communication in ATC. For improvement of effectiveness to add additional parameter to designated ADS-B In-Out data group, many studies being performed. In this paper, we analysis voice communication for reduce the communication traffic in ATC and simulate to add aiming altitude parameter for comparative effect analysis of communication traffic between pilot and controller. The result of the analysis were successfully validated that reduction of communication traffic in ADS-B environments.

• Journal of Korea Multimedia Society
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• v.6 no.7
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• pp.1219-1230
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• 2003

We analyze an integrated voice/data CDMA system, where the whole channels are divided into voice prioritized channels and voice non-prioritized channels. For real-time voice service, a preemptivc priority is granted in the voice prioritized channels. And, for delay-tolerant data service, the employment of buffer is considered. On the other hand, the transmission permission probability in best-effort packet-data service is controlled by estimating the residual capacity available for users. We build a 2-dimensional markov chain about prioritized-voice and stream-data services and accomplish numerical analysis in combination with packet-data traffic based on residual capacity equation.

• The Journal of the Acoustical Society of Korea
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• v.16 no.1
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• pp.100-103
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• 1997

In this paper, we analyze the time delayed effect of speech feature. Here, the time delayed effect means that the current feature vector of speech is under the influence of the previous feature vectors. In this paper, we use a set of LPC driven cepstal coefficients and evaluate the time delayed effect of cepstrum with the performance of the speech recognition system. For the experiments, we used the speech database consisting of 22 words which uttered by 50 male speakers. The speech database uttered by 25 male speakers was used for training, and the other set was used for testing. The experimental results show that the time delayed effect is large in the lower orders of feature vector but small in the higher orders.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.15 no.9
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• pp.5763-5768
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• 2014

Maintaining a voice color is important when compounding both the normal voice because an emotion is not expressed with various emotional voices in a single synthesizer. When a synthesizer is developed using the recording data of too many expressed emotions, a voice color cannot be maintained and each synthetic speech is can be heard like the voice of different speakers. In this paper, the speech data was recorded and the change in the voice color was analyzed to develop an emotional HMM-based speech synthesizer. To realize a speech synthesizer, a voice was recorded, and a database was built. On the other hand, a recording process is very important, particularly when realizing an emotional speech synthesizer. Monitoring is needed because it is quite difficult to define emotion and maintain a particular level. In the realized synthesizer, a normal voice and three emotional voice (Happiness, Sadness, Anger) were used, and each emotional voice consists of two levels, High/Low. To analyze the voice color of the normal voice and emotional voice, the average spectrum, which was the measured accumulated spectrum of vowels, was used and the F1(first formant) calculated by the average spectrum was compared. The voice similarity of Low-level emotional data was higher than High-level emotional data, and the proposed method can be monitored by the change in voice similarity.

• Electronics and Telecommunications Trends
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• v.24 no.5
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• pp.10-19
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• 2009

본 논문에서는 통신과 방송서비스가 하나의 기기 또는 단말장치 안에서 결합되고 단말 내부에서는 디바이스의 통합에 따라 코덱의 개수를 최소화하기 위한 음성기반 오디오 융합코덱의 기술동향에 대해 기술한다. 하지만 기술적으로 완전히 태생이 다른 음성과 오디오 코덱을 진정한 의미에서 융합할 수 있는 기술적 모델과 기법은 아직 개발되지 않고 있다. 본 고에서는 이러한 시도의 일환으로 ITU-T SGl6을 중심으로 진행되고 있는 음성기반 코덱을 점진적 대역폭 확장 기술을 사용하여 광대역 음성, 슈퍼와이드 밴드 및 향후 오디오 대역까지 커버할 수 있는 임베디드 가변비트율 코덱기술을 중심으로 기술동향의 분석을 시도한다.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.65-68
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• 2002

현재 음성인식기는 다 채널의 음성입력방식을 사용하고 있는 추세이다. 이런 방법으로 음성인식기를 사용할 때에 자동적으로 음성을 검출하는 음성입력 방식은 발성자와 마이크간의 거리에 따라 Glottal Spectrum 성분이 변하는 특성을 가지고 있다. 이러한 Glottal Spectrum 성분은 a=R1/R0 (LPC 포락선의 기울기) 로 나타낼 수 있다. 본 논문에서는 발성자와 마이크 거리에 따른 Glottal Spectrum 성분을 비교 분석 하고자 한다.