• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2003.05a
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• pp.786-790
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• 2003

The purpose of this research is to observe that the acoustic characters of classroom have some difference by several conditions. TSP has used to measure impulse response and such physical indexes as RT(Reverberation Time), D$\sub$50/, and STI(Speech-Transmission-Index) are computed by it. we investigate difference under some conditions such as when students were present at each classroom and when was not so, and when professor used a microphone and unused it. In this study, we found that reverberation time when people take a seat is lower than was not so. we wish to help one who work for construction industry, as they build a kind of classroom

• MALSORI
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• no.44
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• pp.73-82
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• 2002

Channel distortion degrades the performance of speech recognizer in telephone environment. It mainly results from the bandwidth limitation and variation of transmission channel. Variation of channel characteristics is usually represented as baseline shift in the cepstrum domain. Thus undesirable effect of the channel variation can be removed by subtracting the mean from the cepstrum. In this paper, to improve the recognition performance of Korea connected digit telephone speech, channel compensation methods such as CMN (Cepstral Mean Normalization), RTCN (Real Time Cepatral Normalization), MCMN (Modified CMN) and MRTCN (Modified RTCN) are applied to the static MFCC. Both MCMN and MRTCN are obtained from the CMN and RTCN, respectively, using variance normalization in the cepstrum domain. Using HTK v3.1 system, recognition experiments are performed for Korean connected digit telephone speech database released by SITEC (Speech Information Technology & Industry Promotion Center). Experiments have shown that MRTCN gives the best result with recognition rate of 90.11% for connected digit. This corresponds to the performance improvement over MFCC alone by 1.72%, i.e, error reduction rate of 14.82%.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.5 no.2
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• pp.279-287
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• 2001

In this paper, we discuss the scale of subjective speech quality measurement over VoIP(Voice over IP) network which is a component of broadband networks. Objective parameters of multimedia services like PSNR or jitter can easily measured and defined, but these factors are not easily meet the user's perceptual recognition. We suggest the speech quality measurement scale through the subjective measurement for end-to-end speech quality composed of sender-side quality, transmission quality, receiver-side quality, which provide the degree of correctness of representation of speaker, the degree of impairment caused by various factors, the degree of recognition of processed speech, respectively. Also, we examined the proposed method and verify it's availability.

• The Journal of the Acoustical Society of Korea
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• v.29 no.5
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• pp.347-353
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• 2010

In VoIP system, the speech quality is seriously degraded due to packet loss, and the degree of degradation by each packet loss depends on the characteristics of the corresponding packet. Therefore, it is possible to improve the speech quality of VoIP by selectively controlling the packet to be lost during transmission based on the expected degradation by the loss of each packet. In this paper, a new scheme to improve speech quality of DiffServ-based VoIP by assigning priority to each packet is proposed, and a method to determine the priority of each packet is developed. The performance of proposed method was measured in packet loss environment based on Gilbert model, and it was verified both objectively and subjectively that the speech quality is improved by the proposed method.

• MALSORI
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• no.43
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• pp.45-55
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• 2002

High reliability of automatic speaker recognition regardless of voice conditions is necessary for forensic application. Audio recordings in real cases are not consistent in voice conditions, such as duration, time interval of recording, given text or conversational speech, transmission channel, etc. In this study the variation of verification error rate of ASR system with the voice conditions was investigated. As a result in order to decrease both false rejection rate and false acception rate, the various voices should be used for training and the duration of train voices should be longer than the test voices.

• The Journal of the Acoustical Society of Korea
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• v.42 no.2
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• pp.133-142
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• 2023

The office environment is changing according to work types, Information Technology (IT) advancements, and the Coronavirus disease (COVID)-19 situation. In order for office space users to perform their tasks comfortably and efficiently, it is necessary to secure individual privacy as well as easy communication among members. In Korea, the demand for improving the acoustic performance of office spaces is also increasing, but the related performance criteria and guidelines have not been established. In this study, standardization of office space acoustic performance measurement and evaluation methods and European countries' acoustic performance criteria were compared and reviewed. It is proposed to comprehensively review international standardization trends and acoustic performance standards in each country and to establish and utilize criteria for evaluating the acoustic performance and satisfaction of office spaces in Korea through our survey. Considering the international standardization direction and compatibility with communication and Public Address (PA) systems, it is appropriate to establish criteria using the speech transmission index or Speech Transmission Index (STI) application index. This criterion will be highly utilizable and compatible. In addition, since the office furniture industry is interested in improving the acoustic performance of office space, it is necessary to establish a labelling system for speech level reduction of office furniture.

• 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.

• The Journal of the Acoustical Society of Korea
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• v.27 no.7
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• pp.379-385
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• 2008

In this paper, we propose an effective speech reinforcement technique under the near-end and the far-end noise environments. In general, since the intelligibility of the far-end speech for the near-end listener is significantly reduced under near-end noise environments, we require a far-end speech reinforcement approach to avoid this phenomena. Specifically, based on the estimated background noise spectrum of the near-end, we reinforce the far-end speech spectrum by incorporating the more general cases under the near-end with background noise. Also, we propose the novel approach to reinforce the actual speech signal except for the noise signal in the far-end noisy speech signal. The performance of the proposed algorithm is evaluated by the CCR (Comparison Category Rating) test of the method for subjective determination of transmission quality in ITU-T P.800 under various noise environments and shows better performances compared with the conventional method.

• MALSORI
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• no.43
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• pp.57-72
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• 2002

The technology of speech recognition has a wide field of application. The range of such technology is spreading into mobile computing having the large amount of movement for communication equipments at the present time. Particularly, recognition in internet environment is rapidly moving into mobile environment. Because of these environments, users want the faster speed of data transmission and the lighter portable equipment for data access. That is PDA(Personal Digital Assistant). Therefore, we designed a triphone-based text independent vocabulary recognizer for the implementation of speech control in this paper. The text independent vocabulary recognizer is based on the state .joint algorithm with decision trees

• Journal of the Korean Institute of Telematics and Electronics
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• v.25 no.4
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• pp.430-436
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• 1988

LSF(Line Spectral Frequency) derived from LPC has known as a very useful transmission parameter of speech signals, for it has a good linear interpolation characteristics and a low spectrum distortion at low bit rates coding. This paper presents that it is possible to extract directly the formant frequencies of speech signals from LSF parameter without application of FFT algorithm by comparing the distribution of LSF parameter with the frequency distribution of analysis filter. This paper suggests the advanced algorithm that results in improving the speed of convergence at analytic solution method. Also, for the flexibility of parameters, the process that transforms from LSF to LPC is presented.