• The Journal of the Acoustical Society of Korea
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• v.11 no.4
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• pp.14-21
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• 1992

The waveform coding is concerned with simply preserving the wave shape of speech signal through a redundancy reduction process. In area of the speech synthesis, the waveform codings with high quality are mainly used to the synthesis by analysis. However, because the parameters of this coding are not classified as either excitation parameters and vocal tract parameters, it is difficult to applying the waveform coding to the synthesis by rule. In this paper, we proposed a new pitch alternation method that can change the pitch periods in the waveform coding by using the cepstrum analysis. Thus, it is possible that the waveform coding is carried out the synthesis by rule in speech processing.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.81-84
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• 2004

패킷망을 통한 음성 통신이 발달됨에 따라 패킷 스위칭 채널 환경에서 계층적 구조를 가지는 광대역 음성 부호화기의 개발에 대한 요구가 늘어나고 있다. 본 논문에서는 이러한 차세대 패킷 전화용 광대역 음성 부호화기의 상위 대역에 대해서 효율적인 양자화 방법을 제안한다. 먼저 전체 프레임을 다수의 짧은 부프레임으로 구분하고, 각각의 부프레임에 MLT(Modulated Lapped Transform)변환을 적용하여 주파수 영역으로 변환하여 2차원 구조의 데이터 행렬을 생성한다. 이러한 2차원 구조의 데이터를 크기와 부호로 분리하고, 크기는 2차원 DCT를 사용하여 시간과 주파수 영역에서의 신호 압축을 동시에 얻을 수 있게 하였다. 이와 같은 새로운 구조를 활용하여 기존의 방법보다 Energy Compaction 효과를 높이고 양자화 성능을 향상시킬 수 있었다. 또한 Core Layer의 부호화된 파라미터를 상위 대역의 양자화에 이용함으로써 그 성능을 향상시킬 수 있는 방법을 제안한다.

• 한국정보통신설비학회:학술대회논문집
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• 2003.08a
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• pp.234-239
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• 2003

디지털방송, DAB 등과 같은 디지털 오디오 방송 서비스를 위한 디지털 시스템을 설계하기 위해서는 오디오 음질을 평가하기 위한 방법이 필수적이다. 기존의 방식은 인간의 귀를 이용한 주관적 방식을 이용함으로서 많은 시간과 비용을 들이게 되며, 음질평가를 하는 사람의 주관적 의견에 많이 좌우하게 된다. 그러나 최근 ITV-R에서는 오디오 음질의 객관적 평가를 위한 BS.1387(PEAQ)를 제안함으로 많은 시간과 비용을 절감하고 신뢰할 수 있는 결과를 얻게 되었다. PEAQ는 인간의 귀에서의 신호의 처리과정과 인식과정을 심리음향모델과 인식모델로 분리하여 구성함으로써 주관적 평가의 SDG(Subjective Difference Grade)에 대응하는 ODG(Objective Difference Grade)를 구하게 된다. 본 논문에서는 이러한 PEAQ의 심리음향 모델과 인식 모델을 원리와 과정을 평가 분석하였다.

• Proceedings of the Acoustical Society of Korea Conference
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• 1998.08a
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• pp.309-314
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• 1998

언어학적 단위인 형태소의 특성을 유지하면서 음성인식 과정에 적합한 분리 기준의 새로운 디코딩 단위인 의사형태소를 정의하였다. 이러한 필요성을 확인하기 위해 새로이 정의된 37개의 품사 태그를 갖는 의사 형태소를 표제어 단위로 삼아 발음사전 생성과 형태소 해석에 초점을 두고 한국어 연속음성 인식 시스템을 구성하였다. 각 음성신호 구간에 해당되는 의사 형태소가 인식되면 언어모델을 사용하여 구성된 의사 형태소 단위의 상위 5개 문장을 기반으로 시작 시점과 끝 시점, 그리고 확률 값을 가진 의사 형태소 격자를 생성하고, 음성 사전으로부터 태그 정보를 격자에 추가하였다. Tree-trellis 탐색 알고리즘 기반에 의사 형태소 접속정보를 사용하여 음성언어 형태소 해석을 수행하였다. 본 논문에서 제안한 의사 형태소를 문장의디코딩 단위로 사용하였을 경우, 사전의 크기면에서 어절 기반의 사전 entry 수를 현저히 줄일 수 있었으며, 문장 인식률면에서 문자기반 형태소 단위보다 약 20% 이상의 인식률 향상을 얻을 수있었다. 뿐만 아니라 형태소 해석을 수행하기 위해 별도의 분석과정 없이 입력값으로 사용되며, 전반적으로 문자을 구성하는 디코딩 수를 안정화 시킬 수 있었다. 이 결과값은 상위레벨 언어처리를 위한 입력？으로 사용될 뿐만 아니라, 언어 정보를 이용한 후처리 과정을 거쳐 더 나은 인식률 향상을 꾀할 수 있다.

• The Journal of the Acoustical Society of Korea
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• v.23 no.5
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• pp.420-427
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• 2004

This paper presents real time implementation of 4kbps EHSX (Enhanced Harmonic Stochastic Excitation) speech coder that combines the harmonic vector excitation coding with time-separated transition coding. The harmonic vector excitation coding uses the harmonic excitation coding for voiced frames and used the vector excitation coding with the structure of analysis-by-synthesis for unvoiced frames, respectively. For transition frames mixed with voiced and unvoiced signal, we use the time-separated transition coding. In this paper. we present the optimization methods of implementation speech coder on the EMS320C6701/sup (R)/ DSP. To reduce the complex for real-time implementation. we perform the optimization method in algorithm by replacing the complex sinusoidal synthesis method with IFFT. and we apply fully pipelines hand assembly coding after converting it from floating source to fixed source. To generate a more efficient code. we also make use or the available EMS320C6701/sup (R)/ resources such as Fastest67x library and memory organization.

• The Journal of the Acoustical Society of Korea
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• v.14 no.5
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• pp.89-93
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• 1995

To reduce the redundancy within samples that resulted from uniform sampling method, nonuniform sampling or nonredundant-sample coding methods can be considered. However, it is well known that when conventional nonuniform sampling methods are applied directly to speech signal, the required amount of data is comparable to or mure than that by uniform sampling method like PCM. To overcome this problem, a new nonuniform sampling method is proposed, in which nonuniform sampling is applied to the low-pass filtered speech signal and higher band is compensated by 8 colored Gaussian random noise with various noise levels. By this method, speech signal waveform can be encoded by 1.8 times larger compression ratio than the conventional nonuniform sampling method.

• The Journal of the Acoustical Society of Korea
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• v.20 no.5
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• pp.83-91
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• 2001

This paper proposes the Focused beamforming to estimate the location of target residing near to the observation platform in the underwater environment. The Focused beamforming technique provides the location of target by the coherent summation of a series of incident spherical waveforms considering distinct propagation delay times at the sensor array. But due to the movement of the observation platform and the variation of the underwater environment, the shape of the sensor array is no longer to be linear but it becomes distorted as the platform moves. Thus the Focused beamforming should be peformed regarding to the geometric shape variation at each time. To estimate the target location, the artificial image plane comprised of cells is constructed, and the delays are calculated from each cell where the target could be proximity to sensors for the coherent summation. After the coherent combining, the beam pattern can be obtained through the Focused beamforming on the image plane. Futhermore to compensate the variation of the shape of the sensor array, the paper utilizes the Nth-order polynomial approximation to estimate the shape of the sensor array obeying the water pulley modeling. Simulation results show the performance of the Focused beamforming for different frequency bands of the radiated signal.

• The Journal of the Acoustical Society of Korea
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• v.21 no.3
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• pp.245-255
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• 2002

This thesis deals a method to deliver more realistic sound by cancelling the cross-talk which is inherent to the 5.1 channel speaker system. The acoustical model for cross-talk cancellation is the free field model. This model minimizes distortion of sound. I used the bark scale sound quality compensation which based on psycho-acoustic. For the surround channels, band-limited sound quality compensation is performed in the frequency domain. I also performed the sound quality assessment test on the traditional 2 channel stereo and 5.1 channel system. This test is performed in the test chamber which satisfies the ITU-R specifications. I uses the IACC (Inter-Aural Cross-Correlation) to determine the preferences of the amateur and the golden ear experts to asses the trans-aural filter. According to the result from the proposed method, I got more the 38 dB separation rates with the Dolby standard speaker array. The results on the diffusion by the subjective test with the experts shows 0.4 point increased then before.

• Proceedings of the KIEE Conference
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• 2006.10c
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• pp.416-418
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• 2006

In this paper, a new acoustic echo cancellation approach based on the DUET algorithm and scaling factor estimation is proposed to solve the scaling ambiguity in case of blind separation based acoustic echo cancellation in a noisy environment. In hands-free full-duplex communication system. acoustic noises picked up by the microphone are mixed with echo signal. For this reason, the echo cancellation system may provide poor performance. For that purpose, a degenerate unmixing estimation technique, adjusted in the time-frequency domain, is employed to separate undesired echo signals and noises. Also, since scaling and permutation ambiguities have not been solved in the blind source separation algorithm, kurtosis for the desired signal selection and a scaling factor estimation algorithm are utilized in this rarer for the separation of an echo signal. Simulation results demonstrate that the proposed approach yields better echo cancellation and noise reduction performances, compared with conventional methods.

• The Journal of the Acoustical Society of Korea
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• v.20 no.5
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• pp.110-115
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• 2001

In this paper, we propose a beamformer adequate for the nested away that is generally used for multiple frequency band signal processing. The nonisotropic beam pattern of channel in this array causes two problems: the bearing-estimate error of mainlobe and the difference between design and output in sidelobe level. By separating the time delay among channel signals and the time delay among sensor signals in channel, we can remove the effects of the nonisotropic beam pattern of channel in the beamformer output. Through this process, a method to correct simultaneously these problems is proposed.