• Proceedings of the Acoustical Society of Korea Conference
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• 1993.06a
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• pp.80-83
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• 1993

본 논문에서는 양질의 규칙합성을 얻기 위하여, 유성음에 대한 여기신호로 임펄스 스펙트럼과 노이즈 스펙트럼을 다중대역으로 혼합하여 생성한 여기신호를 규칙합성에 적용하는 방법을 제안한다. 이 방법에서는, 분석합성에서 각 프레임별로 요구되었던 혼합여기신호에 대한 정보량 문제를 해결하기 위해 유성음의 정상부분의 한 프레임에 대해 혼합여기신호를 구하여 규칙합성에 적용하였고, 정보량을 더욱 줄이는 방안으로, 켑스트럼 유클리디안 거리를 이용하여 유성음을 분류하여, 각 그룹에 대한 대표 여기신호를 규칙합성의 여기신호로 사용하였다. 제안된 방법으로 음성을 합성한 결과 양질의 합성음을 얻을 수 있음을 확인하였다.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.335-336
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• 2000

A psychoacoustic model based noise shaping method is proposed, where noise's presence with a host signal will not be perceptually noticeable. The derivation of imperceptible noise levels from the masking thresholds of the signal involves a deconvolution associated with the spreading function in the psychoacoustic model, which results in an ill-conditioned problem. In this paper, the problem is formulated as a constrained optimization, and it is demonstrated that the solution provides noise shaping where the noise excitation level conforms to the masking thresholds of the signal.

• Proceedings of the Acoustical Society of Korea Conference
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• 1998.06c
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• pp.155-158
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• 1998

IMBE(Improved Multi-Band Excitation) vocoders exhibit good performance at low data rates. The major drawback to IMBE coders is their large computational requirements. In this paper, thus, we propose a new pitch search method that preserves the quality of the IMBE vocoder with reduced complexity. The basic idea is to reduce computation complexity of the pitch searching by using the SAMDF. Applying the proposed method to the IMBE vocoder, we can get approximately 52.02% searching time reduction in the pitch search. There is no difference in voice quality between conventional IMBE and proposed IMBE.

• Proceedings of the Acoustical Society of Korea Conference
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• 1998.06e
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• pp.103-106
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• 1998

본 논문은 상용 FEM-BEM 프로그램을 사용하여 점-조화 가진(harmonic point excitation)에 의한 자유지지 경계 조건을 갖는 원통 셸(cylindrical shell)의 방사 효율(radiation efficiency) 실험의 결과와 비교하였다. 우선 충격 해머 실험(impact hammer test)을 통한 모드 시험(modal testing)으로 원통 셸의 공진 주파수(natural frequency)와 모드 형상(mode shape)의 특징을 살펴보고 다음으로 점-조화 가진에 의한 원통 셸의 방사 효율을 SYSNOISE와 ANSYS로 해석해 보았다. 동시에 음향 세기 실험을 통한 방사 효율을 측정하여 전산 해석의 결과와 실험의 결과를 비교해 보았다.

• Proceedings of the Acoustical Society of Korea Conference
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• 1995.06a
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• pp.141-145
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• 1995

Code excited linear prediction speech coders exhibit good performance at data rates as low as 4800 bps. The major drawback to CELP type coders is their large computational requirements. In this paper, we propose a new codebook search method that preserves the quality of the CELP vocoder with reduced complexity. The basic idea is to restrict the searching range of the random codebook by using a searching technique of the regular pulse excitation. Applying the proposed method to the CELP vocoder, we can get approximately 48% complexity reduction in the codebook search.

• Proceedings of the Acoustical Society of Korea Conference
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• 1995.06a
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• pp.164-167
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• 1995

RPE 음성부호화기에서 합성 필터로 인한 구동벡터 양자화잡음의 증폭효과를 분석하고 regular pulse 시퀀스의 양자화로 인한 성능감쇄를 줄이기 위해 pyramid vector 양자화방식을 도입하였다. 제안된 방식의 성능평가는 구동시퀀스 양자화를 위해 adaptive PCM을 이용하는 GSM 표준 RPE 방식과의 객관적 및 주관적 성능비교를 통해 수행하였다.T JDSMDQLRY 결과 제안된 방식은 대략 1dB의 SNR 및 segmental SNR 값 증가를 가져왔고, 또한 비공식 청취시험결과 명료도의 증가를 느낄 수 있었다.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.199-202
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• 2002

본 논문에서는 사람의 청각 모델을 기반으로 잡음에 의해 손상된 음성 신호로부터 잡음 신호의 마스킹 특성과 신호에너지의 지각(知覺)을 나타내는 임계대역(critical band)에서의 잡음 에너지에 대한 지각 패턴인 noise excitation pattern을 이용한 잡음 에너지 차감과 잡음 추정 오차에 의한 변형된 음성신호 내의 순음(tonal) 성분과 비순음(non-tonal)성분의 보정을 통해 효과적인 음성 품질의 개선을 위한 연구를 하였다.

• The Journal of the Acoustical Society of Korea
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• v.20 no.3
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• pp.34-41
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• 2001

In an IMBE (Improved Multi-band Excitation) speech coder, initial pitch estimation occupies most of the total computing time for the coder due to complex cost function and exhaustive search over candidate pitches. Future frames in initial pitch estimation cause inevitable time delay. Therefore, it is difficult to implement a real-time coder. Furthermore, unvoiced frames use the unnecessary pitch estimation as in the voiced frames. In this paper, each frame is determined voiced or unvoiced by Dyadic Wavelet Transform (DyWT) and, then, initial pitch estimation is performed only for voiced frame. Therefore different pitch estimation algorithms are employed between voiced and unvoiced frames incurring reduced time delay at transmitter and receiver. Simulation result show that the relative complexity of initial pitch estimation is reduced by 23％, and the processing time decreases down to 1/10 ∼ 1/1l of the IMBE coder while speech quality is almost maintained.

• The Journal of the Acoustical Society of Korea
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• v.22 no.6
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• pp.443-451
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• 2003

Since a codebook-based CELP coder models its excitation signal according to one of several bit rates pre-assigned to codebooks and synthesizes speech signal using codebooks, it can not support encoding of speech signal at an arbitrary bit rate in one encoder. The proposed variable bit rate speech coder encodes the excitation signal based on the bit rate assigned to a present frame of speech using one-dimensional SPIHT and wavelet transform. Also it does't need to model excitation signal (or codebook) to some types as CELP coder, and can encode excitation signal at various bit rates without exact pitch information according to user requirement. As a result, since the coder doesn't have a codebook structure, it has relatively low coder complexity and provides equal or better speech quality compared to G.729 and G.723.1 coder.

• The Journal of the Acoustical Society of Korea
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• v.31 no.7
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• pp.472-479
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• 2012

This paper is aimed to design 9.2 kbps low bits late transform excitation coder that target to voice and audio signal. To set up low bit rate, we used Band-selection in frequency domain and gain-shape quantization and AbS structure. To decrease lots of calculation from ABS structure, we used each band IDFT and synthesis. And we designed non-transfer band for performance by inserting comfort noise. We propose coder that has low bit rate and similar performance comparing with original 10.4 kbps AMR-WB+ TCX mode.