• The Journal of the Acoustical Society of Korea
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• v.8 no.5
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• pp.70-77
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• 1989

In this paper, Korean language recognition using the phoneme is studied. The experiment is carried out by dividing 545 isolated words into phonemes. Using linear prediction coefficients the recognition rate of consonants, vowels, and end-consonants are $87.3(\%), 91.0(\%), 91.7(\%)$, respectively. Recognition rate of isolated words combined with the phonemes is $71.4(\%)$. Itakura-saito distortion measure is used to phoneme segmentation and phoneme recognition.

• The Journal of the Acoustical Society of Korea
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• v.24 no.7
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• pp.395-401
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• 2005

In this paper, the high-band codec for bandwidth scalable wideband speech codec is proposed. The wideband input speech signal is separated into low-band signal and high-band signal, and the low-band signal is encoded by the standard narrow-band speech codec and the high-band signal is encoded by the proposed codec. In the high-band codec. the signal is transformed into frequency domain by MLT on a subframe basis, and MLT coefficients are splitted into magnitude and sign for quantization. The magnitudes of MLT coefficients are arranged into several time-frequency bands and each band is quantized in 2D-DCT domain, where the low-band information is utilized for better performance. The sign of MLT coefficient is quantized based on a priority selection process with the weighting measurement. The objective and subjective performance of wideband speech codec including the proposed high-band codec is measured, and it is confirmed that the proposed codec has better performance than 32kbps G.722.1.

• The Journal of the Acoustical Society of Korea
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• v.21 no.5
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• pp.494-502
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• 2002

In this paper, the Low Rank Approximation (LRA) method to suppress the interference of signals from temporal fluctuations is applied. The reverberation signals and temporally fluctuating signals are separated from the measured data using the Ink. The Singular value decomposition (SVD) method is applied to extract the low rank and the temporally stable reverberation was extracted using the LRA. The reverberation suppression is performed on the LRA residual value obtained by removing the approximate reverberation signals. In overall, the method can be applied to the suppression of reververation in active sonar system as well as to the modeling of reverberation.

• Journal of Digital Convergence
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• v.11 no.7
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• pp.209-214
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• 2013

Improve the recognition performance of speech recognition systems as a method for recognizing human listening skills were incorporated into the system. In noisy environments by separating the speech signal and noise, select the desired speech signal. but In terms of practical performance of speech recognition systems are factors. According to recognized environmental changes due to noise speech detection is not accurate and learning model does not match. In this paper, to improve the speech recognition feature extraction using gamma tone and learning model using acoustic model was proposed. The proposed method the feature extraction using auditory scene analysis for human auditory perception was reflected In the process of learning models for recognition. For performance evaluation in noisy environments, -10dB, -5dB noise in the signal was performed to remove 3.12dB, 2.04dB SNR improvement in performance was confirmed.

• The Journal of the Acoustical Society of Korea
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• v.41 no.5
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• pp.525-533
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• 2022

We propose an effective method for suppressing both side and grating lobes by applying 2-dimensional Fourier Transform to the received channel data during the receive focusing process of an ultrasound imaging system. When the signal from the image point is focused, the channel signals have the same DC value across the channels. However, even after echoes from outside an imaging point are focused, they are manifested as having different spatial frequencies depending on their incident angles. Therefore, after the receive focusing delay time is applied, 2-D Fourier Transform is performed on the time-channel data to separate the DC component and other frequency components in the spectral domain, and the weighting value is defined using the ratio of the two values. The side lobe and grating lobe were suppressed by multiplying the ultrasound image by a weighting value. Ultrasound images with a frequency of 5 MHz were simulated in a 64-channel linear array. The grating lobe appearing in the ultrasound image was completely removed by applying the proposed method. In addition, the side lobe was reduced and the lateral resolution was greatly increased. Results of computer simulation on a human organ mimicking image show that the proposed method can aid in better lesion diagnosis by increasing the image contrast.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.257-262
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• 2004

본 논문에서는 초음파 웨이브렛 지연시간을 이용한 초고온 온도 측정법을 기초로 하여, 이 기술을 바탕으로 두 번째 단계인 용융물 온도에서 내구성을 갖는 초음파 센서(UTS : Ultrasonic Temperature Sense)를 설계하여 약 $2300^{\circ}C$까지 실험로 내부의 온도를 측정하고자 한다. 이때 UTS 설계의 중요 인수는 센서 봉 외부 표면과 시스(sheath) 내부 표면의 두 텅스텐 재료가 비접촉 상태로 요구된다. 만약 이들 두 재료가 고온의 상태에서 접촉되면 음향적 분로인 Shunting 현상이 발생한다. 이 현상을 물리적으로 억제하기 위한 센서 설계가 필요하게 되며, 이 센서 설계의 성공 여부의 첫째 요구 조건으로서 센서 내부의 구조적으로 음향 Shunting 현상을 억제하는 기술이 필요로 하게 된다. 이들 센서의 내부 구조에 상호 접촉을 피하기 위해서 작은 공간에 새롭게 구조적 분리가 가능한 텅스텐 재료인 Standoff를 제작하여 설치하였다. 그러나 본 실험에서는 제안된 Standoff적용한 출력 신호의 신호 대 잡음 비는 소량의 개선 가능성을 확인하였으나, 다양한 Standoff의 설계와 제작이 지속적으로 진행되어야 할 것이다.

• The Journal of the Acoustical Society of Korea
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• v.28 no.8
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• pp.697-705
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• 2009

An unsupervised (blind) method is proposed aiming at extracting rhythmic sources from commercial polyphonic music whose number of channels is limited to one. Commercial music signals are not usually provided with more than two channels while they often contain multiple instruments including singing voice. Therefore, instead of using conventional modeling of mixing environments or statistical characteristics, we should introduce other source-specific characteristics for separating or extracting sources in the under determined environments. In this paper, we concentrate on extracting rhythmic sources from the mixture with the other harmonic sources. An extension of nonnegative matrix factorization (NMF), which is called nonnegative matrix partial co-factorization (NMPCF), is used to analyze multiple relationships between spectral and temporal properties in the given input matrices. Moreover, temporal repeatability of the rhythmic sound sources is implicated as a common rhythmic property among segments of an input mixture signal. The proposed method shows acceptable, but not superior separation quality to referred prior knowledge-based drum source separation systems, but it has better applicability due to its blind manner in separation, for example, when there is no prior information or the target rhythmic source is irregular.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.157-160
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• 2000

본 논문에서는 멀티미디어 정보에서 원하는 정보를 추출하는 멀티미디어 인덱싱 중 오디오 인덱싱의 전처리 부격인 음성/음악 분류실험을 하였다. 오디오 인덱싱에 있어서 음성/음악 분류기는 원 오디오 신호에서 정보를 가진 음성 부분을 분리하는 역할을 한다. 실험에서는 음성/음악 분류에서 널리 쓰이는 멜캡스트럼(Mel Cepstrum), 정규화 로그 에너지(normalized log energy), 영교차(Zero-Crossings)를 특징 파라미터로 사용하였다[l, 2, 3]. 특징공간은 GMM(Gaussian Mixture Model)에 의해 모델링 되었고, 오디오 신호의 분류는 각각 3가지 분류항목(음성, 음악, 음성+음악)과 2가지 분류항목(음성, 음악)을 적용하였다. 실험결과 3가지 분류항목 적용시와 2가지 분류항목 적용시 모두 멜캡스트럼을 사용하였을 때 가장 좋은 결과를 보였다.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.41-44
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• 2001

본 논문에서는 음성합성 과정에서 음성유닛을 연결할 때 모음의 결합부분에서 포만트의 불일치로 일어나는 부자연스러운 합성음이 발생되는 문제점을 개선하기 위해서 앞에 오는 음성 유닛과 뒤에 오는 합성 유닛의 포만트 변경에 관한 방법을 제안한다. 요즘에 연구되는 코퍼스 방식에선 에너지와 피치와 음순지속시간 등을 기준으로 유닛을 선택한 후 연결하지만, 스펙트럼의 불일치가 이루어진다. 이런 스펙트럼의 불일치는 음질의 저하를 유도한다. 그래서 앞 음성유닛의 연결부분의 일정부분과 뒤 음성 유닛의 연결부분의 일정부분의 포만트를 천이시켜 일치시켜줌으로써 음질을 향상시켰다. 음성신호를 FFT한 후 magnitude와 phase를 분리한 후 앞 음성의 연결부분의 magnitude와 뒷 음성의 연결부분의 magnitude를 기준으로 linear interpolation한 값을 목표치로 이동하고 다시 합하여 원 신호를 복원하는 방식으로 포만트를 변경시켰다.

• The Journal of the Acoustical Society of Korea
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• v.10 no.5
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• pp.11-19
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• 1991

음성 신호의 합성기법들 중에서 파형코딩법은 음질이 우수하기 때문에 분석에 의한 합성법으로 많이 사용하고 있다. 그렇지만 음원과 성도의특성을 분리하지 않고 파형의 잉여분만을 제거한 후에 파 형자체를 저장하기 때문에 규칙에 의한 합성기법으로 사용하기에는 어려움이 많다. 본 논문은 파형코딩 법 중 선형 PCM 코딩법으로 저장된 음성파형에 대해 피치를 양분할 수 있는 주기반분법을 제안하여 파형자체의 음원을 분리하지 않고 피치 주기를 변경시킬 수 있는 새로운 피치 변경법을 제안하였다. 따 라서 음질이 우수한 파형코딩 합성법으로 규칙에 의한 합성을 수행할 수 있다.