• MALSORI
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• no.50
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• pp.99-110
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• 2004

The normal transmission characteristics of sound are hardly obtained due to the various noises and structural factors in a running car environment. It is due to the channel distortion of the original source sound recorded by microphones, and it seriously degrades the performance of the speech recognition in real driving environments. In this paper we analyze the degree of intelligibility under the various sound distortion environments by channels according to driving speed with respect to speech transmission index(STI) and compare the STI with rates of speech recognition. We examine the correlation between measures of intelligibility depending on sound pick-up patterns and performance in speech recognition. Thereby we consider the optimal location of a microphone in single channel environment. In experimentation we find that high correlation is obtained between STI and rates of speech recognition.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2006.11a
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• pp.150-155
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• 2006

Speech intelligibility is investigated on a conference room-window glass coupled system. Using MLS(Maximum Length Sequency) signal as a sound source, acceleration and velocity responses of the window glass are measured by accelerometer and laser doppler vibrometer. MTF(Modulation Transfer Function) is used to identify the speech transmission characteristics of the room and window system. STI(Speech Transmission Index) is calculated by using MTF and speech intelligibility of the room and the window glass is estimated. Speech intelligibilities by the acceleration signal and the velocity signal are compared and the possibility of the wiretapping is investigated. Finally, intelligibility of the conversation sound is examined by the subjective test.

• Speech Sciences
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• v.11 no.4
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• pp.67-73
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• 2004

Packet loss, jitter, and delay in the Internet are caused mainly by the shortage of network bandwidth. It is due to queuing and routing process in the intermediate nodes of the packet network. In the Internet whose bandwidth is changing very rapidly in time depending on the number of users and data traffic, controlling the peak transmission bit-rate of a VoIP. system depending on the channel condition could be very helpful for making use of the available network bandwidth. Adapting packet size to the channel condition can reduce packet loss to improve the speech quality. It has been shown in [1] that a VoIP system with an AMR speech codec provides better speech quality than VoIP systems with fixed rate speech codecs. With the adaptive codec mode assignment. algorithm proposed in [1], in this paper, we performed the voice transmission experiments using the wireless LAN through the real Internet environment. Experimental results are analyzed and discussed with our findings.

• Proceedings of the IEEK Conference
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• 2002.07c
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• pp.1796-1798
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• 2002

Speech transmission is common in many communications systems. In this paper, a technique to reduce the total bits required for expressing the speech data is proposed for the purpose of a packet transmission. A novel coding method is derived based on the concept of finding common information in sequential speech samples. Computer simulations demonstrate that the proposed scheme reduces the total bits re- quired in PCM approximately by half.

• The Journal of the Acoustical Society of Korea
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• v.13 no.5
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• pp.32-39
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• 1994

The speech transmission quality in telephone is quantified in terms of loudness rating, but this method has been validated only for the handset telephony. The transmission quality of loudspeaking telephony in any room must be evaluated not only with speech transmission but also with background noise, echo and reverberation since the effect of room acoustics is much stroger for loudspeaking telephoy. Therefore, it requires a better approach to specify the quality of loudspeaking telephony. By calcuating the speech transmission index (STI), a physical method for measuring the quality of speech transmission was proposed by Steeneken. In this paper, the application of a STI method for estimating articulation of loudspeaking telephony was discussed. And the STI measurement system with high speed calculation was also three rooms, having different reverberation times. The result show that the STI decreases as the reverberation time of rooms increases. It suggests that speech transmission index method can be useful evaluating articulation of a loudspeaking telephony including the sound field characteristics.

• Journal of Industrial Technology
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• v.27 no.A
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• pp.55-60
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• 2007

Possibility of the eavesdropping is investigated on a conference room-glass window coupled system. Speech intelligibility analysis is performed on the eavesdropping sound of the glass window. Using MLS(Maximum Length Sequency) signal as a sound source, acceleration and velocity responses of the glass window are measured by accelerometer and laser doppler vibrometer. MTF(Modulation Transfer Function) is used to identify the speech transmission characteristics of the room and window system. STI(Speech Transmission Index) is calculated by using MTF and speech intelligibility of the vibration sound is estimated. Speech intelligibilities by the acceleration signal and the velocity signal are compared.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.3C
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• pp.234-239
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• 2008

For the efficient coding and transmission, the speech signal can be classified into three distinctive classes: voiced, unvoiced, and transition classes. At low bit rate coding below 4 kbit/s, conventional sinusoidal transform coders synthesize speech of high quality for the purely voiced and unvoiced classes, whereas not for the transition class. The transition class including plosive sound and abrupt voiced-onset has the lack of periodicity, thus it is often classified and synthesized as the unvoiced class. In this paper, the efficient algorithm for the transition class detection is proposed, which demonstrates superior detection performance not only for clean speech but for noisy speech. For the detected transition frame, phase information is transmitted instead of magnitude information for speech synthesis. From the listening test, it was shown that the proposed algorithm produces better speech quality than the conventional one.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.18 no.2
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• pp.192-201
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• 1993

According to technical advances of telecommunication, communication network has changed to digital transmission from analog transmission network. In the long run, current network will be altered into ISDN which makes end-to-end digital communication. This transition of communication network brings about an important questions for networking plan, administration, and speech quality in order to achieve the effective and advanced telecommunication. Speech quality criterions and degradation factors of digital communication system differ from those of existing analog system because of other characteristics like single echo. It is, therefore, necessary to design new criterions and specifications for digital communication network. This Paper describes the relation between speech communication and speech transmission quality and describes the implementation of model system for quality evaluation of digital speech communication network. In addition, some applications of model system implemented are proposed.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.17 no.4 s.121
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• pp.363-369
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• 2007

The purpose of the study is to obtain acoustical information to prevent eavesdropping of the glass window. Speech intelligibility was investigated on the vibration sound detected from the glass window of a conference room. Objective test using speech transmission index(STI) was performed to estimate quantitatively the speech intelligibility. STI was determined based on tile modulation transfer function(MTF) of the room-glass window system. Using Maximum Length Sequency(MLS) signal as a sound source, impulse responses of the glass window and MTF were determined by signals from accelerometers and laser doppler vibrometer. Finally, speech intelligibility of the interior sound and window vibration were compared under different sound pressure levels and amplifier gains to confirm the effect of measurement condition on the speech intelligibility.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.17 no.9
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• pp.888-894
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• 2007

The speech sound is detected by the vibration measurement of the window glass. In this study, we investigate the effect of the disturbing waves by background noise and window shaker excitation on the speech intelligibility of the detected sound. Based upon Modulation Transfer Function(MTF), speech intelligibility of the sound is objectively estimated by Speech Transmission Index(STI) As the level of the disturbing wave varies, variation of the speech intelligibility is examined. Experimental result reveals how STI is influenced by the level and frequency characteristics of the disturbing wave. By using a customized window shaker for disturbing sound, we evaluate the efficiency and the frequency characteristics of the anti-eavesdropping system. The purpose of the study is to provide useful information to prevent the eavesdropping through the window glass.