• Electronics and Telecommunications Trends
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• v.8 no.3
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• pp.95-107
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• 1993

단말장치의 송출전력기준은, 단말장치 이용자가 음성통화 이외에 사용하는 신호에 대해서 송출되는 레벨의 크기를 크게 하여 교환기의 과부하 및 다른 회선으로의 누화 등을 야기시켜 타 이용자에게 피해를 끼칠수 있어 그 레벨의 허용범위를 규제한 것으로, 단말장치 인증(형식승인)시 적용되는 기술기준 항목중의 하나로 활용되고 있다. 본 고에서는 이러한 단말장치 송출전력에 대해 적용하고 있는 각국의 기준치와 그에 따른 측정방법등에 대한 동향을 조사분석하였다.

• The Journal of the Acoustical Society of Korea
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• v.22 no.3
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• pp.242-249
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• 2003

When special voice control application services (VCS) such as voice-controlled web browsing or voice-controlled stock transactions are introduced in cellular systems, a channel quality better than that for ordinary voice communications service (OVS) is necessary in order to keep a suitable grade of VCS. To avoid ai. congestion, calls are normally admitted if there exists a channel-processing resource not occupied by other calls in the base as well as the interference level at the receiver is not higher than a predefined threshold. The threshold is usually 10㏈ noise-rise over the background noise level for voice communications service. When the base admits VCS attempts in exactly the same manner as it handles OVS calls. the same fraction of those will be not successful in taking the channel and then blocked. If the same noise-rise threshold is used as 10 ㏈, however, the admitted VCS calls might suffer from bad channel qualify and finally be dropped. From the user's point of view, the forced termination of ongoing calls is significantly undesirable than blocking new call attempts. When using a lower noise-rise threshold for VCS. on the other hand, the blocking probability of VCS gets higher than that of OVS. In this paper, a call admission policy that gives a priority to VCS is considered in order to reduce the blocking probability and keep an adequate channel quality.

• Fire Science and Engineering
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• v.25 no.2
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• pp.101-106
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• 2011

In case of an emergency such as a fire on a building and there is a need to evacuate the occupant in that building, it is important to have the guidance information effectively delivered to the evacuating occupants to guide them toward a safe direction using audio sensual media. And, it is also very important to prevent the evacuating occupants getting lost or falling astray, away from the direction toward safety. The purpose of this study, in this respect, is to examine the possible application of the precedent sound effect, with which the evacuating occupants may get a sense of the direction where the announcement comes from. With such an effect, an experiment was conducted to measure the extent to which people can hear the preceding and the following sound in terms of the acoustic pressure level changes and delay time changes, with a view to make the optimal evacuation-guidance announcement or sound. The optimal evacuation guidance sound (announcement) per each of the experimental indoors environments were as follows; 1) Regarding the optimal condition for the evacuation guidance announcement sound in the space of a lecture room, the direction of the advanced sound is positively recognized when the follow-up sound has the delaying time of 10 ms~50 ms in comparison with the advanced sound or when there is no difference between the acoustic pressures of the advanced and follow-up sounds or the acoustic pressure of the advanced sound is higher than that of the follow-up sound. 2) Regarding the optimal evacuation guidance announcement sound in the space of a hall, the advanced sound is positively recognized when the follow-up sound has the delaying time of 20 ms~60 ms in comparison with the advanced sound. 3) Regarding the optimal evacuation guidance announcement sound in the space of a gymnasium, the advanced sound is positively recognized when the follow-up sound has the delaying time of 10 ms~40 ms in comparison with the advanced sound or when the sound pressure of the advanced sound has a higher level than or the same level as that of the follow-up sound.

• Proceedings of the KAIS Fall Conference
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• 2010.05a
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• pp.241-244
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• 2010

ITU-T에서 인터넷 폰과 화상회의에 사용하기 위하여 개발된 G.723.1 음성 부호화기는 잡음 구간에서의 전송률을 낮추기 위한 방법으로 VAD(Voice Activity Detector)와 CNG(Comfort Noise Generator)를 사용하고 있다. 여기서 VAD는 최종적으로 현재 프레임의 에너지 레벨을 비교하여 음성의 활동 유무를 판정하고 있다. 하지만 G.723.1 VAD에서는 보다 안정적인 판정을 위해 음성 활동 구간 사이에 삽입되어 있는 묵음 구간에 대해서는 거의 대부분 음성이 활동하는 영역으로 판정을 하고 있다. 본 논문에서는 묵음 구간에 대해 보다 정확한 판정을 통하여 기존의 방법에 비해 전송율을 더욱 감소시킬 수 있는 방법을 제안한다. 실험에서는 묵음구간을 길게 조절한 문장을 사용하여 측정한 결과 약 50% 정도의 전송율을 감소시킬 수 있었으며, MOS 테스트 결과, 음질의 열하는 발생하지 않았다.

• Proceedings of the KAIS Fall Conference
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• 2007.11a
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• pp.194-197
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• 2007

.ITU-T 국제 표준화 기구에서 인터넷 폰과 화상회의를 목적으로 개발된 G.723.1 음성 부호화기는 잡음구간에서의 전송률을 낮추기 위한 방법으로 VAD(Voice Activity Detector)와 CNG(Comfort Noise Generator)를 사용하고 있다. 이중 VAD는 최종적으로 현재 프레임의 에너지 레벨을 비교하여 음성의 활동 유무를 판정하고 있다. 하지만 G.723.1 VAD에서는 보다 안정적인 판정을 위해 음성 활동 구간 사이에 삽입되어 있는 묵음 구간에 대해서는 거의 대부분 음성이 활동하는 영역으로 판정을 하고 있다. 따라서 본 논문에서는 묵음 구간에 대해 보다 정확한 판정을 통하여 기존의 방법에 비해 전송율을 더욱 감소시킬 수 있는 방법을 제안한다. 실험에서는 묵음구간을 길게 조절한 문장을 사용하여 측정한 결과, 약 47% 정도의 전송율을 감소시킬 수 있었으며, MOS test 결과, 음질의 열하는 거의 발생하지 않았다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.5B
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• pp.448-456
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• 2003

In this parer, an ABR (Available Bit Rate) congestion control algorithm for voice transmission in ATM networks was proposed. To deal with the network congestion problem, not only the buffer level of a switch but also the variation of the buffer level were considered. Also, to resolve the unfairness among sources where the bit transfer rates vary, a loading factor that is used to adjust the bit rate was introduced. To show the superiority of this paper over others, simulation was done with a network of 7 voice sources and 4 switches, which was represented by Petri net model. ExSpect was used for simulation. The simulation results showed that there was improvement in network utilization and that unfairness among sources were resolved a lot.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.14 no.2
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• pp.134-139
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• 2021

Currently, the autonomous vehicle market is commercializing a level 3 autonomous vehicle, but it still requires the attention of the driver. After the level 3 autonomous driving, the most notable aspect of level 4 autonomous vehicles is vehicle stability. This is because, unlike Level 3, autonomous vehicles after level 4 must perform autonomous driving, including the driver's carelessness. Therefore, in this paper, we propose the Emergency-notification and Driver-response Confirmation System(EDCS) for an autonomousvehicle safety that notifies the driver of an emergency situation and recognizes the driver's reaction in a situation where the driver is careless. The EDCS uses the emergency situation delivery module to make the emergency situation to text and transmits it to the driver by voice, and the driver response confirmation module recognizes the driver's reaction to the emergency situation and gives the driver permission Decide whether to pass. As a result of the experiment, the HMM of the emergency delivery module learned speech at 25% faster than RNN and 42.86% faster than LSTM. The Tacotron2 of the driver's response confirmation module converted text to speech about 20ms faster than deep voice and 50ms faster than deep mind. Therefore, the emergency notification and driver response confirmation system can efficiently learn the neural network model and check the driver's response in real time.

• Proceedings of the Acoustical Society of Korea Conference
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• 1998.06c
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• pp.167-170
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• 1998

음성신호의 특징적인 성분인 피치를 검출하는 알고리즘 중 실시산 구현이 손쉬운 3단계를 클리핑 알고리즘을 PC상에서의 처리를 위하여 구현하였다. 이 알고리즘을 통하여 검출되는 피치의 안정성 및 정확성을 높이기 위해서 적용된 창함수, LPF, 클리핑 자기상관값계산, 비선형 감쇄, 등의 전처리 필터링과, 배수피치 검출 및 정정, 메디언 필터링을 사용하여 피치를 검출하였다. 또한 이 알고리즘을 이용하여 DSP의 도움을 얻지 않고 PC상에서 음성을 분석하여 스펙트로그램, 파형, 에너지, 피치 등을 출력하는 프로그램인 Visual Analysis Tool for sounds(VAT)의 출력화면을 통하여 피치검출을 나타내었다.

• Proceedings of the Acoustical Society of Korea Conference
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• 1998.06e
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• pp.343-346
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• 1998

잡음환경에서의 음성인식 성능향상을 위하여 본 논문에서는 스펙트럼 차감법 이후에 남아 있는 잔여 잡음으로 인한 mismatch를 극복하는 수단으로 기존의 스펙트럼 차감법에서의 flooring factor를 사용하는 대신에 target 잡음레벨을 이용하여 잡음 마스킹을 적용하는 스펙트럼 차감법과 잡음 마스킹의 hybrid 방식을 사용한다. 이 방법은 낮은 SNR에서 개선되지 않는 기존의 잡음 마스킹이 가지는 약점을 극복하고 동시에 스펙트럼 차감버에서의 잔여 잡음 문제를 완화시킬 수 있었다. 특히 시간/주파수 영역 smoothing을 적용함으로써 스펙트럼 차감법과 잡음 마스킹의 hybrid 방식의 적용 이후에도 여전히 남아 있는 일부 잡음을 추가적으로 감소시켰으며, 더욱 향상된 인식성능을 얻을 수 있었다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.21 no.2
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• pp.326-332
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• 1996

In this paper, a method to design the sending loudness rating(SLR) is proposed and the desirable transmission characteristics are considered in order to specify the transmission quality, based on the loudness ratings, for the digital telephone system that is a terminal for digital speech communication. To specify the desirable SLR for digital telephone system, the subjective test defining the preferred range of inout level for CODEC was performed. From the test results, it was identified that the optimal input level for CODEC is -15dB and the range not to cause the quantization noise and the distortion of CODEC fall within -12dB and -18dB.