• Speech Sciences
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• v.11 no.2
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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 Korean Institute of Communications and Information Sciences
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• v.21 no.2
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• pp.365-374
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• 1996

Finite wordlength effects of the FS-1016 CELP(Code Excite Linear Prediction) vocoder algorithm) is analyzed, and a block floating-Point implementation method is employed to improve the fixed-point performance. An efficient run-time integer wordlength estimation algorithm is developed, and the overall system performance. An efficient run-time integer wordlength estimation algorithm is developed, and the overall system performance is verified in real-time using a TMS320C50 emulation board. Autoscaler software that conducts simulation-based automatic scaling to provide a floating-point like programming environment is used for this application development.

• Proceedings of the IEEK Conference
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• 2001.06d
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• pp.119-122
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• 2001

본 논문에서는 음성압축 앨고리즘인 MPEG-4 CELP coder를 16 bit DSP 구현에 필요한 고정소수점 연산구조로 구현하였다. 기본 앨고리즘 중에 LSP 계수를 구하는 방법인 Chebyshev series method 대신 고정소수점 구현에 유리한 Real root method 앨고리즘을 사용하였다. 또한 cosine, log 둥 DSP 명령어가 지원하지 않는 수학 함수들은 미리 계산하여 테이블 적용기법을 사용하였고 고정 소수점 연산에 불리한 나눗셈 연산을 최대한 배제하였다. 고정 소수점 연산 구조로 변환한 후 부동 소수점 연산구조와의 비교를 통하여 오차를 최소화하도록 하였다 구현한 음성코더를 남, 여 각 5문장에 적용했을 때 부동 소수점 연산구조에 비교해 음질의 열화가 없음을 확인하였다.

• Proceedings of the IEEK Conference
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• 2006.06a
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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.

• Proceedings of the KAIS Fall Conference
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• 2010.05a
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• pp.257-260
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• 2010

In this paper, we propose the computation reduction methods of LSP(Line spectrum pairs) transformation that is mainly used in CELP vocoders. In order to decrease the computational time in real root method the characteristic of four proposed algorithms is as the following. First, scheme to reduce the LSP transformation time uses mel scale. Developed the second scheme is the control of searching order by the distribution characteristic of LSP parameters. Third, scheme to reduce the LSP transformation time uses voice characteristics. Developed the fourth scheme is the control of searching interval and order by the distribution characteristic of LSP parameters. As a result of searching time, computational amount, transformed LSP parameters, SNR, MOS test, waveform of synthesized speech, spectrogram analysis, searching time is reduced about 37.5%, 46.21%, 46.3%, 51.29% in average, computational amount is reduced about 44.76%, 49.44%, 47.03%, 57.40%. But the transformed LSP parameters of the proposed methods were the same as those of real root method.

• The Journal of the Acoustical Society of Korea
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• v.22 no.8
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• pp.717-726
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• 2003

This paper presents the methods that design the Wavelet-based pitch detector for 2,4 kbps Harmonic-CELP Coder, and that achieve the effective waveform interpolation by decision window shape of the transition region, Waveform interpolation coder operates by encoding one pitch-period-sized segment, a prototype segment, of speech for each frame, generate the smooth waveform interpolation between the prototype segments for voiced frame, But, harmonic synthesis of the prototype waveforms between previous frame and current frame occur not only waveform errors but also discontinuity at frame boundary on that case of pitch halving or doubling, In addtion, in transition region since waveform interpolation coder synthesizes the excitation waveform by using overlap-add with triangularity window, therefore, Harmonic-CELP fail to model the instantaneous increasing speech and synthesis waveform linearly increases, First of all, in order to detect the precise pitch period, we use the hybrid 1st pitch detector, and increse the precision by using 2nd ACF-pitch detector, Next, in order to modify excitation window, we detect the onset, offset of frame by GCI, As the result, pitch doubling is removed and pitch error rate is decreased 5.4% in comparison with ACF, and is decreased 2,66% in comparison with wavelet detector, MOS test improve 0.13 at transition region.

• Proceedings of the Acoustical Society of Korea Conference
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• 1994.06c
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• pp.205-209
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• 1994

This paper presents a variable rate speech coding method with phonetic segmentation, called for PSVXC. Multiple access techniques that require efficient encoding of speech to achieve capacity improvements are currently emerging in the cellular telephone system. The variable rate speech coder have the reduced average data rate required to transmit conversational speech. Each frame of active speech is classified into one of four phonetic classes. A distinct coding configuration and bit-rate is applied to each category. And also a split vector quantization is used to accurately quantize the LPC information using LSP parameters.

• The Journal of the Acoustical Society of Korea
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• v.12 no.2E
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• pp.33-37
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• 1993

부호여기선형예측 형태의 음성 부호화기에 있어서주된 단점은 요구되는 계산량이많다는 것이다. 본 논문에서는 음질의 저하없이 자기상관에 근거한 피치예측기에서 피치 찾는 시간을 줄이는 간단한 방법을 제안한다. 기본적인 발상은, 피치찾는 과정에서 나타나는 자기상관 함수의 특성에 근거하여, 음의 봉우리의 폭을 이전 양의 봉우리의 폭을 이용하여 추정하여 건너 뜀으로써 피치 찾는 범위를 자기상관함수의 양의 값쪽으로 제한하는 것이다. 이렇게 피치찾는 구간에 제약을 가함으로써 요구되는 계산량은 감소하게 된다. 제안된 방법을 적용한 결과, 평균 약 40%의 계산량 감소를 음질의 저하없이 얻을 수 있다.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.33B no.2
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• pp.88-94
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• 1996

In this paper, we present a half rate speech coder using trellis excitation. The coder combines code-excited linear prediction (CELP) system and trellis quantization method using the codebook expansion, and it produces higher speech quality than the typical CELP coder for the same transmission rate. A subjective comparison with 3~8 bit .$\mu$-law PCM indicates that the half rate coder provides speech quality between 5-bit and 6-bit $\mu$-law PCM .

• The Journal of the Acoustical Society of Korea
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• v.18 no.2E
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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.