• International Journal of Fuzzy Logic and Intelligent Systems
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• 제15권4호
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• pp.300-304
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• 2015

A low bit rate speech coder based on the non-uniform sampling technique is proposed. The non-uniform sampling technique is based on the detection of inflection points (IP). A speech block is processed by the IP detector, and the detected IP pattern is compared with entries of the IP database. The address of the closest member of the database is transmitted with the energy of the speech block. In the receiver, the decoder reconstructs the speech block using the received address and the energy information of the block. As results, the coder shows fixed data rate contrary to the existing speech coders based on the non-uniform sampling. Through computer simulation, the usefulness of the proposed technique is shown. The SNR performance of the proposed method is approximately 5.27 dB with the data rate of 1.5 kbps.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2006년도 하계종합학술대회
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• pp.349-350
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• 2006

This paper presents very low bit rate speech coder, ZFE-CELP(ZINC Function Excitation-Code Excited Linear Prediction). The ZFE-CELP speech codec is based on a ZINC function and CELP modeling of the excitation signal respectively according to the frame characteristic such as a voiced speech and an unvoiced speech. And this paper suggest strategies to improve the speech quality of the very low bit rate speech coder.

• 한국멀티미디어학회논문지
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• 제10권12호
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• pp.1551-1558
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• 2007

The standard of evaluating function of speech coder for mobile telecommunication can be shown in channel capacity, noise immunity, encryption, complexity and encoding delay largely. This study is an algorithm to reduce complexity applying to CDMA(Code Division Multiple Access) mobile telecommunication system, which has a benefit of keeping the existing advantage of telecommunication quality and low transmission rate. This paper has an objective to reduce the computing complexity by controlling the frequency band nonuniform during the changing process of LSP(Line Spectrum Pairs) parameters from LPC(Line Predictive Coding) coefficients used for EVRC(Enhanced Variable-Rate Coder, IS-127) speech coders. Its experimental result showed that when comparing the speech coder applied by the proposed algorithm with the existing EVRC speech coder, it's decreased by 45% at average. Also, the values of LSP parameters, Synthetic speech signal and Spectrogram test result were obtained same as the existing method.

• 전기학회논문지
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• 제62권12호
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• pp.1750-1754
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• 2013

A variable bit rate speech coder is presented. The coder is based on the observation that a speech signal can be viewed as a combination of piecewise linear signals in a short time period. The encoder detects the sample points where the slope of the signal changes, which are called the inflection points in this paper. The coder transmits the location and value for the detected inflection sample, but only the location information for the noninflection samples. In the decoder, the noninflection samples are estimated with interpolation of the received information. Several factors affecting the performance of the coder have been tested through simulation. Simulation results show that the linear interpolation produces 1 ~ 5 dB improvement over the cubic spline interpolation. And the -law companding does not provide any benefit when it is applied before the inflection detection. With low threshold values in the inflection point detection, the coder shows better MOS and more than 16 dB improvement in SNR compared to the continuously variable slope delta modulation (CVSDM).

• 한국음향학회지
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• 제21권6호
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• pp.510-521
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• 2002

다중대역 여기부호화 (MBE: multi-band excitation) 음성 부호화기는 고조파 대역별로 유/무성음 판단을 수행함으로써 한 프레임 내에서 유성음과 무성음이 혼합되는 경우를 잘 모델링할 수 있다. 하지만 같은 주파수 대역에서는 유성음 성분과 무성음 성분이 공존할 수 없다. 또한 유/무성음 판단 과정에서 경험에 의한 임계치와의 비교 과정이 필요하므로 원음 스펙트럼과 합성음 스펙트럼간의 오류가 큰 경우가 발생하는 단점이 있다. 본 논문에서는 모든 주파수 영역에서 유성음 성분과 무성음 성분이 혼합되는 것을 허용하는 혼합다중대역 여기 부호화 (MMBE: mixed multi-band excitation) 음성 모델을 제안하고, 모델 파라미터인 주파수 영역 혼합함수를 임계치와의 비교없이 효과적으로 추정할 수 있는 방법을 제시한다. 제안한 음성 모델을 적용한 2.6 kbps 음성 부호화기를 구현해 본 결과, 2.9 kbps의 전송률을 갖는 MBE음성 부호화기에 비해서 낮은 전송률에서도 더 우수한 합성음 음질을 가지는 것으로 나타났다.

• 대한음성학회지:말소리
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• 제54호
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• pp.27-43
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• 2005

Existing standard speech coders can provide speech communication of high quality while they degrade the performance of speech recognition systems that use the reconstructed speech by the coders. The main cause of the degradation is that the spectral envelope parameters in speech coding are optimized to speech quality rather than to the performance of speech recognition. For example, mel-frequency cepstral coefficient (MFCC) is generally known to provide better speech recognition performance than linear prediction coefficient (LPC) that is a typical parameter set in speech coding. In this paper, we propose a speech coder using MFCC instead of LPC to improve the performance of a server-based speech recognition system in network environments. However, the main drawback of using MFCC is to develop the efficient MFCC quantization with a low-bit rate. First, we explore the interframe correlation of MFCCs, which results in the predictive quantization of MFCC. Second, a safety-net scheme is proposed to make the MFCC-based speech coder robust to channel error. As a result, we propose a 8.7 kbps MFCC-based CELP coder. It is shown from a PESQ test that the proposed speech coder has a comparable speech quality to 8 kbps G.729 while it is shown that the performance of speech recognition using the proposed speech coder is better than that using G.729.

• 한국음향학회지
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• 제22권6호
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• pp.443-451
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• 2003

코드북 기반의 CELP 부호기는 코드북에 미리 할당된 부호화 비트율에 따라서 여기 신호를 모델링한 후 코드북을 이용하여 음성신호를 합성한다. 따라서 임의의 다양한 비트율을 하나의 부호기에서 지원하지 못하는 단점이 있다. 본 논문에서 제안하는 가변 비트율 부호기는 웨이블렛 변환 (wavelet transform과 1차원 SPIHr (one dimensional SPIHT)를 이용하여 현재 프레임에 할당되는 비트수에 따라서 여기신호를 부호화한다. 또한 CELP 부호기의 경우처럼 특정한 몇 가지 형태로 여기신호(또는 코드북)를 모델링할 필요가 없고, 정확한 피치정보가 없어도 여기신호를 사용자의 요구에 따라 다양한 비트율로 부호화할 수 있다. 그 결과 코드북이 존재하지 않기 때문에 부호기의 복잡도가 낮으며, CELP 기반의 G.729와 G.723.1 부호기와의 음질 비교 결과 동등하거나 나은 결과를 보여준다.

• 한국통신학회논문지
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• 제41권10호
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• pp.1183-1185
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• 2016

디지털 음성통신에서의 저 비트율 음성부호화기는 음성발성모델의 파라미터를 사용하여 음성을 합성한다. 이 경우, 파라미터에 할당된 비트가 매우 한정적이기 때문에 합성된 음성의 스펙트럼이 크게 왜곡될 수 있으며, 이는 명료도 저하의 요인이 된다. 본 논문에서는 통계적 스펙트럼 이퀄라이저를 이용한 명료도 향상 기법을 제안한다. 본 기법은 각각의 음성부호화기별로 원음과 합성음의 스펙트럼 비율을 이용하여 통계적으로 가중치 벡터를 구하며, 이를 합성 음성에 적용한다. 객관적인 음성명료도 평가 실험을 통해, 제안한 기법이 기존의 방법보다 성능이 우수함을 확인하였다.

• 음성과학
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• 제11권2호
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• pp.259-269
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• 2004

In this paper, we propose a new excitation enhancement technique to improve the speech quality of low bit-rate code-excited linear prediction (CELP) coders. The proposed technique is based on a harmonic model and it is employed only in the decoding process of speech coders without any additional bits. We develop the procedure of harmonic model parameter estimation and harmonic generation, and apply this technique to a current state-of-the-art low bit rate speech coder, ITU-T G.729 Annex D. Also, its performance is measured by using the ITU-T P.862 PESQ score and compared to those of the phase dispersion filter and the long-term postfilter applied to the decoded excitation. It is shown that the proposed excitation enhancement technique can improve the quality of decoded speech and provide better quality for male speech than other techniques.

• The Journal of the Acoustical Society of Korea
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• 제18권2E호
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• pp.24-30
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• 1999

In this paper, we propose a weighting dependent mixed source model (WD-MSM) coder that is an improved version of a CELP-based mixed source model (C-MSM) coder. The coder classifies speech segments into three types : voiced, unvoiced and mixed. The excitation for a voiced frame is an adaptive source, and the excitation for an unvoiced frame is a stochastic source. The coder has a modified mixed source for a mixed frame. We apply different weighting functions for three classes. Simulation results show that the proposed coder at 4 kbits/s yields very good performance both subjectively and objectively.