• The Journal of the Acoustical Society of Korea
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• v.40 no.3
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• pp.247-253
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• 2021

Directional ultrasonic speakers that are used to hear sound only in a specific area have been continuously researched on various improvements in terms of sound quality and cost compared to general speakers. In this paper, we propose a DSP based hybrid-type ultrasonic speaker that can be heard at the same time as a general speaker in order to compensate for the sound in the low-band range, considering that it is difficult to hear the low-band sound below 500 Hz due to the sensor characteristics of the ultrasonic speaker. In the case of the system that is implemented by simply connecting a general speaker and an ultrasonic speaker, there are issues of high cost and difficulties of control as two amplifiers are used to playback ultrasonic and general sound sources. In addition, sound quality deteriorates due to the difference in playback time between ultrasonic and general sound sources. In order to improve issues of cost, control and sound quality, we developed hybrid-type ultrasonic speaker with a DSP based amplifier that can simultaneously playback by synchronizing the general sound source with the regenerated ultrasonic sound source, in addition to implement the existing CODEC functions such as Dynamic Range Control (DRC) and Equalizer (EQ).

• Speech Sciences
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• v.9 no.4
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• pp.253-263
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• 2002

In this paper, we propose new algorithm of speaker recognition which identifies the speaker using the information obtained by the intensive speech feature analysis such as pitch, intensity, duration, and formant, which are crucial parameters of individual voice, for candidates of high percentage of wrong recognition in the existing speaker recognition algorithm. For testing the power of discrimination of individual parameter, DTW (Dynamic Time Warping) is used. We newly set the range of threshold which affects the power of discrimination in speech verification such that the candidates in the new range of threshold are finally discriminated in the next stage of sound parameter analysis. In the speaker verification test by using voice DB which consists of secret words of 25 males and 25 females of 8 kHz 16 bit, the algorithm we propose shows about 1% of performance improvement to the existing algorithm.

• Speech Sciences
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• v.8 no.3
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• pp.93-104
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• 2001

The author analyzed acoustic parameters of ten Korean numbers produced by ten male students using Praat. Variations of f0, F1, F2 and F3 within and between speakers were examined by determining an average and standard deviation of the parameters of each number and by comparing the acoustic values with one another. Results showed that each subject produced the numbers within a certain range of variation across time. Thus, speaker identification can be more certain using dynamic information of the acoustic parameters within each vocalic segment. Also, percent difference of within-subjects' variation to that of between-subjects can be utilized to determine which sounds would be better stimuli for speaker identification. According to the criteria, the number '2' proved the best stimulus while the number '7' was the worst. Future studies will be necessary to explore robust methods of speaker identification under noisy conditions.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.14 no.2
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• pp.871-888
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• 2020

With the development of display engineering and technology, the screen and sound quality of information devices such as TVs are improving. The screen used LEDs via LCD and PDP and a large flat panel in the early CRT to create super-high resolution. The sound is improved by directly vibrating a thin and simple panel, such as an OLED. In our previous study, the exciter speaker was attached to the rear of the OLED panel to be used as the diaphragm of the speaker, and the sound quality was as good as that of the TV using the existing dynamic speaker. This method supplied the viewer with the direct sound coming from the panel, delivering clear sound, and the sound and image came from the same location, thus giving the viewer high immersion and maximizing the effect of information transfer. OLED exciter speakers, however, have a special directivity, which tends to slightly attenuate the tone at the very center of the screen. This study improves the sound quality by improving the structure of the exciter speaker and the radiated sound of the flat panel display. A 2-in-1 Exciter is made into a single core to improve the speaker's radiation pattern.

• The Journal of the Acoustical Society of Korea
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• v.11 no.2
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• pp.68-72
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• 1992

Recently, many researchers have studied the speaker recognition through the statistical processing method using Karhunen-Loeve Transform. However, the content of speaker's identity and the vocalization speed cause speaker recognition rate to be lowered. This parer studies the speaker recognition method using weighted orthogonal parameters which are weighted with eigen-values of speech so as to emphasize the speaker's identity, and optimal path which is made by DWP so as to normalize dynamic time feature of speech. To confirm this method, we compare the speaker recognition rate from this proposed method with that from the conventional statistical processing method. As a result, it is shown that this method is more excellent in speaker recognition rate than conventional method.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.7 no.7
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• pp.1539-1544
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• 2003

Recently, many researchers have studied the speaker recognition through the statistical processing method using Karhonen-Loeve Transform. However, the content of speaker's identity and the vocalization speed cause speaker recognition rate to be lowered. This parer studies the speaker recognition method using weighted parameters which are weighted with eigen-values of speech so as to emphasize the speaker's identity and optimal path which is made by DWP so as to normalize dynamic time feature of speech. To confirm this method, we compare the speaker recognition rate from this proposed method with that from the conventional statistical processing method. As a result, it is shown that this method is more excellent in speaker recognition rate than conventional method.

• The Transactions of the Korean Institute of Electrical Engineers
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• v.40 no.4
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• pp.351-359
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• 1991

This paper describes performances of speaker adaptation using a probabilistic spectral mapping matrix in hidden-Markov model(HMM) -based Korean isolated word recognition. Speaker adaptation based on probabilistic spectral mapping uses a well-trained prototype HMM's and is carried out by Viterbi, dynamic time warping, and forward-backward algorithms. Among these algorithms, the best performance is obtained by using the Viterbi approach together with codebook adaptation whose improvement for isolated word recognition accuracy is 42.6-68.8 %. Also, the selection of the initial values of the matrix and the normalization in computing the matrix affects the recognition accuracy.

• Language and Information
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• v.14 no.2
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• pp.103-129
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• 2010

In this paper, I show that anaphora across speakers shows both dynamic and static sides. To capture them all formally, I will adopt semantics based on the assumption that variables range over individual concepts that connect epistemic alternatives. As information increases, a variable can take a different range of possible individual concepts. This is captured by the notion of virtual individual (= vi), a set of individual concepts which are indistinguishable in an information state. The use of a pronoun involves two information states, one for the antecedent, which is always part of the common ground, and the other for the pronoun. Information increase changes vis for variables in the common ground. A pronoun can be used felicitously if there is a unique virtual individual in the information state for the antecedent which does not split in two or more distinctive virtual individuals in the information state for the pronoun. The felicity condition for cross-speaker anaphora can be satisfied in declaratives involving modality, interrogatives and imperatives in a rather less demanding way, because in these cases the utterance does not necessarily require non-trivial personal information for proper use of a pronoun.

• Speech Sciences
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• v.14 no.4
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• pp.145-156
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• 2007

The performance of speaker identification is severely degraded in noisy environments. Recently probability weighting method based on observation membership was proposed for overcoming the noise problem[1]. In the paper[1] the observation confidence was calculated from SNR with sigmoid function. However, estimating SNR needs additive calculation amount and estimated SNR is corrupted in dynamic noisy environments. In this paper we propose estimation methods of the observation confidence based on score-based reliabilities (SBR) of entropy and dispersion measures. Generally SBRs are obtained from speaker models' probabilities. The proposed methods are evaluated with ETRI speaker recognition DB. We compared the performances of the proposed methods with those in [1][8]. The experimental results show that the proposed methods can be successfully applied for the case where SNR is not available.

• The Journal of the Acoustical Society of Korea
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• v.22 no.5
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• pp.383-387
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• 2003

This paper deals with the method of speaker verification based on the conventional cohort of fixed size. In particular, a new cohort of variable size, which makes use of the distance between speaker models, is proposed: The density of neighboring speaker models within the fixed distance from each speaker is taken into account in the proposed method. The high density leads to the increase of cohort size, thus improving the speaker verification rate. On the other hand, the low density leads to its decrease, thus reducing the amount of computations. The simulation results show that the proposed method outperforms the conventional one, achieving a reduction in the EER.