• Proceedings of the IEEK Conference
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• 2000.11d
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• pp.199-202
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• 2000

Speaking Rate has variety depends on the situation and habit of speakers. It has been many studied about speaking rate In speaker recognition. The study of speaking rate in speech recognition is one of considerable matter when It is recognized the speakers and it is measured by many speech data base and complicate estimation for accuracy. In this paper, conventional vocoder process the speech signal when encoding and transmitting without regard to speaking rate so in order to apply the speaking rate for vocoder It should be considered the simpler algorithm and less computation amount than the conventional method of speaking rate used In speech recognition. We proposed the speaking rate algorithm which is used the simple parameter with Line Spectrum Pair (LSP). The proposed peaking rate method is measured by the information of LSP in speech. We measured the variety rate of phenomenon about utterances which have different velocity, respectively. As a result, It has distinct variation rate of phenomenon between utterances uttered fast and slow and the rate is 42.8% higher in case of uttered fast than in case of uttered slow.

• ETRI Journal
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• v.38 no.6
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• pp.1064-1073
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• 2016

An adaptive speech streaming method to improve the perceived speech quality of a software-based multipoint control unit (SW-based MCU) over IP networks is proposed. First, the proposed method predicts whether the speech packet to be transmitted is lost. To this end, the proposed method learns the pattern of packet losses in the IP network, and then predicts the loss of the packet to be transmitted over that IP network. The proposed method classifies the speech signal into different classes of silence, unvoiced, speech onset, or voiced frame. Based on the results of packet loss prediction and speech classification, the proposed method determines the proper amount and bitrate of redundant speech data (RSD) that are sent with primary speech data (PSD) in order to assist the speech decoder to restore the speech signals of lost packets. Specifically, when a packet is predicted to be lost, the amount and bitrate of the RSD must be increased through a reduction in the bitrate of the PSD. The effectiveness of the proposed method for learning the packet loss pattern and assigning a different speech coding rate is then demonstrated using a support vector machine and adaptive multirate-narrowband, respectively. The results show that as compared with conventional methods that restore lost speech signals, the proposed method remarkably improves the perceived speech quality of an SW-based MCU under various packet loss conditions in an IP network.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.47 no.1
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• pp.175-182
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• 2010

In this paper, we propose a robust speech reinforcement technique to enhance the intelligibility of the degraded speech signal under the ambient noise environments based on soft decision scheme incorporating a speech absence probability (SAP) with speech reinforcement gains. Since the ambient noise significantly decreases the intelligibility of the speech signal, the speech reinforcement approach to amplify the estimated clean speech signal from the background noise environments for improving the intelligibility and clarity of the corrupted speech signal was proposed. In order to estimate the robust reinforcement gain rather than the conventional speech reinforcement method between speech active periods and nonspeech periods or transient intervals, we propose the speech reinforcement algorithm based on soft decision applying the SAP to the estimation of speech reinforcement gains. The performances of the proposed algorithm are evaluated by the Comparison Category Rating (CCR) of the measurement for subjective determination of transmission quality in ITU-T P.800 under various ambient noise environments and show better performances compared with the conventional method.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.25 no.7
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• pp.462-469
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• 2015

This study was carried out to examine characteristics of various acoustic indicators evaluating speech intelligibility such as reverberation time(T₃₀), D₅₀, C₅₀ and speech transmission index(STI) in Korean elementary school classrooms. Results showed that mean T₃₀ at middle frequencies(500 Hz to 2000 Hz) measured in 9 classrooms was about 0.75 s, which exceeds a regulation specified on American National Standards(ANSI); 0.60 s. Mean D₅₀, C₅₀ and STI were 60 % to 66 %, +2 dB to +3 dB, and 0.65, respectively. The maximum difference in D₅₀ and C₅₀ according to different receiver points in a classroom was 13 % and 2.5 dB, while the maximum difference in T₃₀ was 0.03 s. Whereas STI measured in classrooms has relatively low correlation with other indicators, correlation between D₅₀ and C₅₀ was high, R²=.9964. In addition, T₃₀ and C₅₀ were fitted well as logarithmic regression curve with R²=.9610. It was +3.73 dB in C₅₀ and 68 % in D₅₀ which are the value corresponding to 0.60 s in T₃₀ on this curve.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.22 no.12
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• pp.1604-1610
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• 2018

The speech recognition service is used as an ancillary means of communication by converting and visualizing the speaker's voice into text to the hearing impaired persons. However, in open environments such as classrooms and conference rooms it is difficult to provide speech recognition service to many hearing impaired persons. For this, a method is needed to efficiently provide it according to the surrounding environment. In this paper, we propose a system that recognizes the speaker's voice and transmits the converted text to many hearing impaired persons as messages. The proposed system uses the MQTT protocol to deliver messages to many users at the same time. The end-to-end delay was measured to confirm the service delay of the proposed system according to the QoS level setting of the MQTT protocol. As a result of the measurement, the delay between the most reliable Qos level 2 and 0 is 111ms, confirming that it does not have a great influence on conversation recognition.

• Journal of Advanced Navigation Technology
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• v.11 no.1
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• pp.87-92
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• 2007

The conventional CELP(code excited linear prediction) type vocoder has no V/UV(voiced/unvoiced) classifier. So, the unvoiced speech is processed like the voiced speech. In this paper, to reduce the bit rate, we propose a new V/UV decision algorithm minimized error rate and preprocessing computation. This V/UV classifier use the LSP(line spectrum pair) parameter which is acquired spectrum analysis process in CELP vocoders. Applying this method to the 5.3kbps ACELP(algebraic code excited linear prediction) in the G.723.1, we can get the transmission bits rate reduction of 6% approximately without degradation of speech quality.

• Journal of the Korean Institute of Educational Facilities
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• v.11 no.1
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• pp.5-14
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• 2004

Since 1990s in Korea, elementary school classrooms have been designed toward open education system in pursuit of variety of educational purpose. Also, the architectural designs of schools have been acomplished for individual school not based on the standard design code. The present paper aims to investigate the acoustic environment of existing classrooms and to compare the sound insulation capacity between the ordinary classrooms and the newly built classrooms for open education. The current acoustical situation of elementary classrooms was analyzed using field measurements and questionnaire survey. In order to this, Three elementary schools were selected which were built in 1978, 1996 and 2000 respectively. Room acoustical parameters including Reverberation time(RT), Definition(D50), Speech Intelligibility(RASTI), Transmission loss(TL) and STC were measured in a classroom in each elementary school classroom. Each measurement was undertaken with the windows and doors being open or closed. As the result, it was found that the transmission loss between rooms in open classrooms is, $5{\sim}6dB$ in average, inferior than the ordinary classrooms. The RASTI of 0.70 was measured in newly built classrooms which is better than old classrooms(0.70) and open classrooms(0.73). This was shown as same in the speech definition measurements. This results from the condition of sealing and airtightness of classrooms and floor materials. The results denote that open classrooms have poor acoustic condition in sound insulation and speech intelligibility.

• The Journal of the Acoustical Society of Korea
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• v.38 no.4
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• pp.396-405
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• 2019

The speech privacy of closed office rooms located in a university campus was measured and assessed in terms of SPC (Speech Privacy Class) values. The measurements of two quantities, the LD (Level Difference) between a source and a receiving room, and the background noise level ($L_b$) at the receiving room were carried out in 5 rooms located in 3 different buildings in the university campus. Each of the 5 rooms was adjacent to both offices and corridors through walls. The TL (Transmission Loss) between the source and the receiver room was also measured to compare the difference of two standard methods, ASTM E2836-10 and KS F 2809. The present results show that the speech privacy of the 5 office rooms is not met the requirement for a minimum SPC values of 70. A minimum LD value of 41 dB between the source and the receiver room should be achieved for having a SPC value of 70 when the mean measured value of $L_b$ at the receiving room is 29.2 dB. That is, the TL(avg) value averaged over the octave bands from 160 Hz to 5000 Hz between the source and the receiver room should be or greater than 40 dB. The most important architectural factor influencing the LD value is the presence of openings, such as doors, and windows, on the adjacent walls between the source and receiving room. Therefore, if the opening of the adjacent wall is replaced by an opening with high sound insulation, the appropriate SPC value of the research and office rooms can be achieved.

• Phonetics and Speech Sciences
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• v.6 no.1
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• pp.85-93
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• 2014

This paper investigated the effects of frequency band limitation on perceptual characteristics of Korean vowels. Monosyllabic speech (144 syllables of CV type, 56 syllables of VC type, 8 syllables of V type) produced by two announcers were low- and high-pass filtered with cutoff frequencies ranging from 300 to 5000 Hz. Six listeners with normal hearing performed perception tests by types of filter and cutoff frequencies. We reported phoneme recognition rates and types of perception error of band-limited Korean vowels to examine how frequency distortion in the process of speech transmission affect listener's perception.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2002.11b
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• pp.743-748
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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. 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. In addition to physical characteristics, questionnaires were given to university students to given subjective appraisals. For instance, questions about volume or clearness of lectures. The relevancy between physical characteristics and subjective appraisals was studied.