• 대한음성학회지:말소리
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• 제47호
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• pp.97-107
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• 2003

Speaker change detection is to detect automatically a point of time at which speaker was replaced. Since feature parameters used for speaker change detection depend not only on speaker characteristics but also on phonetic characteristics, spoken contents included in the feature parameters inevitably causes performance degradation of speaker change detection. In this paper, to alleviate this problem, a method to normalize phonetic variations in speech feature parameters is proposed for emphasizing changes due to speaker characteristics. Experimental results show that the proposed method improves the performance of speaker change detection.

• 음성과학
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• 제10권3호
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• pp.213-224
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• 2003

Spoken sound includes not only speaker's source but the characteristics of vocal tract and speech radiation. This paper is based on the theory of Park[1], who proposes the Phonation Type Index k; a variable that shows the characteristic of speaker's source excluding those of speaker's vocal tract and speech radiation. With Park's theory, we collect data by changing recording environments and expanding experimental data, and analyze the data collected to see whether or not the PTI k shows good discriminating power as a variable for speaker recognition. In the experiment, we repeatedly record 8 sentences ten times for each of 5 males in the environment of a recording room and an office, extract PTI k for each speaker, and measure the discriminating power for each speaker by using the value of PTI k. The result shows that PTI k has the excellent discriminating power of speakers. We also confirm that, even if the recording environment is changed, PTI k shows similar results.

• 한국소음진동공학회:학술대회논문집
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• 한국소음진동공학회 2004년도 추계학술대회논문집
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• pp.763-766
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• 2004

Recently, various acoustic artifacts that contains speaker have been produced such as cellular phone. Speaker consists of diaphragm generating sound and coil vibrating diaphragm. Generally, good speaker means that it has a wide frequency range, high output power rate to input power and flat sound pressure level in specified frequency range. Acoustic characteristic was estimated through the experiment and computer simulation, or sound power was controlled with acoustic sensitivity in a natural frequency range fer last decade. However, the flatness of sound pressure level has not been considered to enhance the sound quality of a speaker. Tn this study, a method for speaker design is proposed for a good acoustic characteristic, which is flatness of SPL(sound pressure level) and wideness between the first and second natural frequency. SYSNOISE is used fer acoustic analysis and ANSYS is used for harmonic response analysis and modal analysis. Optimization for acoustic characteristics of a speaker diaphragm is performed using ModelCenter. All analyses are done within a frequency domain. And we confirm that the experimental and computational simulations have similar trend.

• 대한전기학회논문지:시스템및제어부문D
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• 제51권2호
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• pp.69-77
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• 2002

Speaker verification can be classified in two categories, text-dependent speaker verification and text-independent speaker verification. In this paper, we discuss text-dependent speaker verification. Text-dependent speaker verification system determines whether the sound characteristics of the speaker are equal to those of the specific person or not. In this paper we obtain the speaker data using a sound card in various noisy conditions, apply a new Hidden LMS (Least Mean Square) adaptive algorithm to it, and extract LPC (Linear Predictive Coding)-cepstrum coefficients as feature vectors. Finally, we use a competitive learning neural network for speaker verification. The proposed hidden LMS adaptive filter using a neural network reduces noise and enhances features in various noisy conditions. We construct a separate neural network for each speaker, which makes it unnecessary to train the whole network for a new added speaker and makes the system expansion easy. We experimentally prove that the proposed method improves the speaker verification performance.

• 한국산학기술학회논문지
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• 제15권12호
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• pp.6993-6997
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• 2014

본 연구의 목적은 자성유체 스피커의 보이스 코일 방열 및 음질 저하 문제를 극복하기 위하여 무향실내에서 자성유체 스피커의 음 왜곡률, 음압레벨 및 주파수 특성과 같은 음질 성능 특성을 고찰하는 것이다. 이를 위하여 자성유체 스피커에서 자성유체 주입량 및 영구자석 자력을 변화시켜가면서 음압 성능을 측정하였다. 그리고 스피커의 음 왜곡률, 음압레벨 및 주파수 특성은 음향시스템 측정 프로그램인 Smarrt를 이용하여 측정하였다. 결과적으로, 자성유체의 주입량은 2.4 ml로 결정되었고, 자성유체를 주입할 경우 음 왜곡률 및 음압레벨은 향상되었다. 자성유체 스피커에서 영구자석의 자력을 8.06 mT에서 9.10 mT로 증가시킬 경우 주파수 특성 및 음압레벨은 더욱 향상되었다. 또한, 자성유체 스피커의 음 왜곡률은 일반 스피커에 비하여 약 0.01% 감소하였다.

• 한국소음진동공학회논문집
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• 제26권1호
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• pp.5-12
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• 2016

This paper describes the development process for vibration and acoustic characteristics of a balanced armature speaker. The design parameters were chosen in consideration of the influence of the bending stiffness of balanced armature which is the form of a cantilever structure in the speaker. For study of the performance of the speaker according to the design parameters, in the first step, we analyzed the characteristics of the velocity of the diaphragm to the electrical input. Next step, acoustic characteristics were analyzed by structural-acoustic coupled analysis. And the reliability of the analysis was verified by comparing the result of analysis with test results. Finally, we proposed a design method for implementing an enhanced balanced armature speakers through analysis method.

• 대한음성학회지:말소리
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• 제45호
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• pp.93-105
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• 2003

In this paper, we present a single framework based on GMM for speaker recognition. The proposed framework can simultaneously minimize environmental variations on mismatched conditions and adapt the bias free and speaker-dependent characteristics of claimant utterances to the background GMM to create a speaker model. We compare the closed-set speaker identification for conventional method and the proposed method both on TIMIT and NTIMIT. In the several sets of experiments we show the improved recognition rates on a simulated channel and a telephone channel condition by 7.2% and 27.4% respectively.

• 수산해양기술연구
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• 제30권1호
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• pp.25-32
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• 1994

An underwater speaker was made of a moving coil driver unite of usual speaker, acryl-boards, polyester resin, rubber and castor oil and it's frequency characteristics was measured in range of 250~600Hz in air water tank and sea. The results of measurements are follows: 1. Transmitting and receiving frequency of measurement frequency were similar in air, water tank and sea. 2. The input and output wave forms of a manufactured speaker which is not water-proof in air were similar to each other in 300~450Hz, but other frequencies showed distorted wave forms. 3. The input and output wave forms of an underwater speaker in water thank and sea were similar to each other in 250~600Hz. But output wave forms showed combination waves with very low frequency. 4. Transmitting and receiving frequency wave forms and resisting pressure of an underwater speaker at 80m in the depth of water were in good condition. Therefore it can be possible to use it as an underwater speaker.

• 한국통신학회논문지
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• 제34권3C호
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• pp.257-267
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• 2009

This paper presents a training method for neural networks and the employment of MSE (mean scare error) values as the basis of a decision regarding the identity claim of a speaker in a recurrent neural networks based speaker verification system. Recurrent neural networks (RNNs) are employed to capture temporally dynamic characteristics of speech signal. In the process of supervised learning for RNNs, target outputs are automatically generated and the generated target outputs are made to represent the temporal variation of input speech sounds. To increase the capability of discriminating between the true speaker and an impostor, a discriminative training method for RNNs is presented. This paper shows the use and the effectiveness of the MSE value, which is obtained from the Euclidean distance between the target outputs and the outputs of networks for test speech sounds of a speaker, as the basis of speaker verification. In terms of equal error rates, results of experiments, which have been performed using the Korean speech database, show that the proposed speaker verification system exhibits better performance than a conventional hidden Markov model based speaker verification system.

• 한국정보통신학회논문지
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• 제10권4호
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• pp.739-743
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• 2006

본 논문은 신경회로망을 이용한 화자적응 음성인식 제어시스템을 개발하였다. 화자종속시스템은 단일 화자의 음성만 등록하여 이용하므로 여러 화자의 음성을 인식하는 데는 문제가 있고, 화자독립시스템은 여러 화자를 인식한다. 본 연구 개발에서는 화자적응시스템을 구현하여 화자종속형의 단점을 보완하여 화자 독립과 화자 종속을 혼합하여 사용 할 수 있는 기능으로 화자 적용방법으로 구현하였고, 화자인증(Speaker Verification)도 가능하도록 프로그램 하였다.