• The Journal of the Acoustical Society of Korea
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• 제20권3E호
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• pp.18-24
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• 2001

Speaking rate represents how many phonemes in speech signal have in limited time. It is various and changeable depending on the speakers and the characters of each phoneme. The preprocessing to remove the effect of variety of speaking rate is necessary before recognizing the speech in the present speech recognition systems. So if it is possible to estimate the speaking rate in advance, the performance of speech recognition can be higher. However, the conventional speech vocoder decides the transmission rate for analyzing the fixed period no regardless of the variety rate of phoneme but if the speaking rate can be estimated in advance, it is very important information of speech to use in speech coding part as well. It increases the quality of sound in vocoder as well as applies the variable transmission rate. In this paper, we propose the method for presenting the speaking rate as parameter in speech vocoder. To estimate the speaking rate, the variety of phoneme is estimated and the Line Spectrum Pairs is used to estimate it. As a result of comparing the speaking rate performance with the proposed algorithm and passivity method worked by eye, error between two methods is 5.38% about fast utterance and 1.78% about slow utterance and the accuracy between two methods is 98% about slow utterance and 94% about fast utterances in 30 dB SNR and 10 dB SNR respectively.

• The Journal of the Acoustical Society of Korea
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• 제28권3E호
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• pp.99-108
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• 2009

In this paper, we suggest criteria for objective measurement of speech quality in mobile VoIP (Voice over Internet Protocol) services over wireless mobile internet such as mobile WiMAX networks. This is the case that voice communication service is available under other networks. When mobile VoIP service users in the mobile internet network based on packet call up PSTN and mobile network users, but there have not been relevant quality indexes and quality standards for evaluating speech quality of mobile VoIP. In addition, there are many factors influencing on the speech quality in packet network. Especially, if the degraded speech with packet loss transfers to the other network users through the handover, voice communication quality is significantly deteriorated by the transformation of speech codecs. In this paper, we eventually adopt the Gilbert-Elliot channel model to characterize packet network and assess the voice quality through the objective speech quality method of ITU-T P. 862. 1 MOS-LQO for the various call scenario from mobile VoIP service user to PSTN and mobile network users under various packet loss rates in the transmission channel environments. Our simulation results show that transformation of speech codecs results in the degraded speech quality for different transmission channel environments when mobile VoIP service users call up PSTN and mobile network users.

• 한국조명전기설비학회지:조명전기설비
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• 제7권3호
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• pp.49-56
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• 1993

본 논문에서는 지하전력구 환경하에서 통신신뢰도가 높고 확장성이 용이하며 복합전송(음성+데이터+영상신호)이 가능한 누설동축케이블을 전송 매체로 하여 지하전력구내 신호전송시스템을 설계 및 설치하였다.그리고 시험시스템에서 각종 전송특성시험을 실시하여 성능을 평가하였다. 음성신호의 경우, 전송손실 특성은 설계치보다 5∼10[dB]정도의 수신레벨이 향상되었으며 통화 품질평가시험에서 통화강도가 3(QSA 값), 명료도가 4(QRK 값)정도의 양호한 특성을 나타냈다. 또한 데이터 및 영상신호 전송특성에 대한 감시제어기능시험에서는 98[%]의 통신성공율을 확보할 수 있었다.그리고 전송로 및 시스템의 전송특성 측정결과에 의해서 중계기 없이 전송범위를 6[km]까지 확대할 수 있는 LCX 통신시스템을 전력구내 종합관리 시스템의 기본 구성방법으로 제시하였다.

• 대한전기학회:학술대회논문집
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• 대한전기학회 1999년도 추계학술대회 논문집 학회본부 B
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• pp.700-702
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• 1999

The protection of data which we want to keep secret from invalid users has become a main topic nowadays. This paper introduces a encryption scheme for protecting speech signals from eavesdropping. The proposed encryption scheme adopts a secure voice cryptographic algorithm based on the scrambling in frequency band. In order to improve the conventional speech signal encryption scheme, we have randomly permuted DCT coefficients of speech signal. Simulation results are included to show the performance of the proposed algorithm for secure transmission of speech signals.

• 한국소음진동공학회:학술대회논문집
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• 한국소음진동공학회 2008년도 춘계학술대회논문집
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• pp.551-556
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• 2008

Possibility of the remote eavesdropping through window glass is investigated using a laser sensor. Various thicknesses and types of glass windows are excited by maximum length sequency (MLS) signal and the vibration sound is detected by a laser doppler vibrometer. Intelligibility of the detected sound is evaluated using the speech transmission index (STI), which is based on the modulation transfer function (MTF). In order to identify the disturbing effect, different level of disturbing wave is generated by an outside speaker and a window shaker attached on the glass window. On the different thickness of glass windows, decrease effect of the speech intelligibility is analysed.

• 한국음향학회지
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• 제28권2호
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• pp.127-134
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• 2009

본 연구에서는 다양한 두께의 유리창을 대상으로 레이저 원격 도청 가능성을 검토한다. MLS 신호음을 사용하여 유리창을 진동시키고 레이저 도플러센서로 진동음을 탐지한다. 탐지된 진동음으로부터 음성정보의 인식 수준인 음성명료도를 객관적으로 평가한다. 평가에는 변조전송함수에 근거하여 결정되는 음성전송지수를 사용한다. 또한, 외부 스피커와 유리창 교란기로 각각 교란파를 발생시켜, 배경소음과 도청방지기의 음성명료도에 대한 교란효과를 평가한다. 다양한 두께의 유리창을 대상으로 레이저 원격 도청음의 음성인식 수준과 국산 도청 방지장치의 도청방지 성능을 평가하는 데에 본 연구의 목적이 있다.

• 한국화재소방학회논문지
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• 제33권1호
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• pp.105-112
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• 2019

본 연구에서는 T자형의 복도 연결 공간에서 비상 방송음이 명료하고 고르게 전달되는지를 건축음향 시뮬레이션을 이용하여 알아보았다. 복도 공간의 흡음성능 변화, 비상 방송용 확성기의 설치 위치와 간격을 변화시켜 보았으며 변화에 따른 음압 레벨 분포, 음성 전달 지수(STI, RASTI) 분포를 비교하였다. 시뮬레이션 결과 명료한 음성 전달을 위해서는 비상 방송용 확성기를 T자형 복도 연결부의 중심에서 약 10 m를 이격시켜 설치하는 것이 좋은 것으로 나타났다. NFSC의 25 m 설치 간격을 좁히는 경우 더욱 명료하고 충분한 음량을 갖는 비상 방송음이 고르게 전달될 수 있는 것으로 나타났다.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2000년도 추계종합학술대회 논문집(2)
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• pp.273-276
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• 2000

Recently, Turbo code has been considered for channel coding in IMT-2000(International Mobile Telecommunication-2000) system, because it offers better error correcting capability than the traditional convolution/viterbi coding . In this paper, a turbo code decoder for speech transmission in IMT-2000 system with frame size 192 bits, constrait length K=3, generator polynomials G(5,7) and code rate R=1/3 is designed using SOVA(Soft Output Viterbi Algorithm) and block interleaver

• 한국소음진동공학회:학술대회논문집
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• 한국소음진동공학회 2002년도 추계학술대회논문초록집
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• pp.374.1-374
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• 2002

The eventual purpose of this research is to make optimum standards for acoustic-environment by using not only physical characteristics but also subjective appraisals. First, basic Physical data were measured which were necessary to establish standards for acoustic environment in campus buildings, TSP has used to measure sound levels, reverberation times, clearness indexes, and speech-transmission-index. (omitted)

• 전자공학회논문지B
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• 제33B권2호
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• pp.88-94
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• 1996

In this paper, we present a half rate speech coder using trellis excitation. The coder combines code-excited linear prediction (CELP) system and trellis quantization method using the codebook expansion, and it produces higher speech quality than the typical CELP coder for the same transmission rate. A subjective comparison with 3~8 bit .$\mu$-law PCM indicates that the half rate coder provides speech quality between 5-bit and 6-bit $\mu$-law PCM .