• KSII Transactions on Internet and Information Systems (TIIS)
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• 제13권2호
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• pp.962-977
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• 2019

The human voice is a convenient method of information transfer between different objects such as between men, men and machine, between machines. The development of information and communication technology, the voice has been able to transfer farther than before. The way to communicate, it is to convert the voice to another form, transmit it, and then reconvert it back to sound. In such a communication process, a vocoder is a method of converting and re-converting a voice and sound. The CELP (Code-Excited Linear Prediction) type vocoder, one of the voice codecs, is adapted as a standard codec since it provides high quality sound even though its transmission speed is relatively low. The EVRC (Enhanced Variable Rate CODEC) and QCELP (Qualcomm Code-Excited Linear Prediction), variable bit rate vocoders, are used for mobile phones in 3G environment. For the real-time implementation of a vocoder, the reduction of sound quality is a typical problem. To improve the sound quality, that is important to know the size and shape of noise. In the existing sound quality improvement method, the voice activated is detected or used, or statistical methods are used by the large mount of data. However, there is a disadvantage in that no noise can be detected, when there is a continuous signal or when a change in noise is large.This paper focused on finding a better way to decrease the reduction of sound quality in lower bit transmission environments. Based on simulation results, this study proposed a preprocessor application that estimates the SNR (Signal to Noise Ratio) using the spectral SNR estimation method. The SNR estimation method adopted the IMBE (Improved Multi-Band Excitation) instead of using the SNR, which is a continuous speech signal. Finally, this application improves the quality of the vocoder by enhancing sound quality adaptively.

• 한국음향학회:학술대회논문집
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• 한국음향학회 1993년도 학술논문발표회 논문집 제12권 1호
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• pp.280-287
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• 1993

Code Excited Linear Prediction(CELP) speech coders exhibit good performance at data rates as low as 4800bps. The major drawback to CELP type paper, a comparative performance study of three pitch searching algorithms for the CELP vocoder was conducted. For each of the algorithms, a standard pitch searching algorithm was used by the sequential pitch searching algorithm that was implimented in the QCELP vocoder. The algorithms used in this study were 1) using the skip table(TABLE), 2) using the symmetrical property of the autocorrelation(SYMMT), and 3) using the preprocessing autocorrelation(PREPC). Performance scores are presented for each of the three pitch searching algorithms based on computation speed and on pitch prediction error.

• 한국음향학회:학술대회논문집
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• 한국음향학회 1994년도 제11회 음성통신 및 신호처리 워크샵 논문집 (SCAS 11권 1호)
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• pp.188-194
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• 1994

Code excited linear prediction speech coders exhibit good performance at data rates as low as 4800bps. But the major drawback to CELP type coders is their large computational requirements. Therefore, in this paper a comparative performance study of three pitch searching algorithms for the CELP vocoder was conducted. For each of the algorithms, a standard pitch searching algorithm was used by the full pitch searching algorithm that was implimented in the QCELP vocoder. The algorithms used in this study is to reduce the pitch searching time 1) using the skip table, 2) using the symmetrical property of the autocorrelation , and 3) using the preprocessing autocorrelation, 4) using the positive autocorrelation, 5) using the preliminary pitch. Performance scores are presented for each of the five pitch searching algorithms based on computation speed and on pitch prediction error.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2001년도 제14회 신호처리 합동 학술대회 논문집
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• pp.525-528
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• 2001

본 논문은 음성 및 음악을 위한 새로운 4kbps 다중 모드 하모닉 변환 여기 부호화 방법을 제안한다. 제안된 부호화방법은 음성/음악 분류기에 의해 분류된 신호를 각각 하모닉-잡음 여기모델과 MLT 여기모델로 부호화한다. 하모닉-잡음 여기모델에서는 전이구간과 유/무성음 혼합신호의 모델링오차 개선을 위해 MP(Matching Pursuit)방법과 혼합된 잡음스펙트럴을 표현하기 위한 캡스트럽 LPC 잡음 모델, 빠른 정현파 합성법을 제안한다. 음악에서는 비트할당 효율을 높이기위한 LP 적응 피크 분석을 적용한 MLT(Modulated Lapped Transform) 부호화 방법을 제안한다. 제안된 방법을 적용한 4kbps 음성부호화 방법은 전이구간에서의 향상된 모델링 구조를 보여주었으며, 주관적음질 평가 8kbps QCELP 보다 MOS 0.2 정도 향상된 결과를 얻었다.

• 대한음성학회지:말소리
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• 제46호
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• pp.65-76
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• 2003

In this paper, we employ the adaptive comb filtering for effective noise reduction in mobile communication environment. Adaptive comb filtering is a well-known method for noise reduction, but requires correct pitch period and must be applied just in voiced speech frames. To satisfy these requirements we use two kinds of information extracted from speech packets, one of which is the pitch period information measured precisely by a speech coder and the other is the frame rate information related to a decision on speech or silence frame. Experiments on speech recognition system confirm the efficiency of this method. Feature parameters employing this method give superior performance in noise environment to those extracted directly from output speech.

• 한국음향학회:학술대회논문집
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• 한국음향학회 1998년도 학술발표대회 논문집 제17권 2호
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• pp.373-376
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• 1998

통신 수단의 발달로 휴대단말기의 사용이 증가하고 있으며, 이와 함께 휴대단말기에서의 음성인식에 대한 수요도 증가하고 있다. 휴대단말기의 경우 저 전송율을 가지는 음성 부호화기를 사용하게 되며, 이러한 저전송율의 음성 부호화기에서의 음성인식을 수행할 경우 인식 성능이 저하되는 현상을 보이게 된다. 본 논문에서는 이러한 문제를 해결하기 위하여 LSP 파라메터 기반의 거리척도에 관하여 비교 검토하였으며, 적은 훈련 데이터에서 사용 가능한 화자 종속 음성인식 방법으로 Dynamic Time Warping(DTW)과 변형된 Hidden Markov Model(HMM)에 관하여 검토하였다. QCELP 음성 부호화기에서 인식 어휘 당 2번의 훈련 데이터만을 이용한 화자종속 인식방법을 사용한 결과 95% 이상의 인식 성능을 얻을 수 있었다.

• The Journal of the Acoustical Society of Korea
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• 제22권3E호
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• pp.101-109
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• 2003

A multi-mode harmonic transform coding (MMHTC) for speech and music signals is proposed. Its structure is organized as a linear prediction model with an input of harmonic and transform-based excitation. The proposed coder also utilizes harmonic prediction and an improved quantizer of excitation signal. To efficiently quantize the excitation of music signals, the modulated lapped transform(MLT) is introduced. In other words, the coder combines both the time domain (linear prediction) and the frequency domain technique to achieve the best perceptual quality. The proposed coder showed better speech quality than that of the 8 kbps QCELP coder at a bit-rate of 4 kbps.

• 한국음향학회:학술대회논문집
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• 한국음향학회 2001년도 추계학술발표대회 논문집 제20권 2호
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• pp.85-88
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• 2001

Line spectrum pair(LSP) 계수는 양자화 오류에 강하고. 선형 릴간에 효율적이며, 필터의 안정성 판정이 용이하므로 LPC를 대신하여 음성 부호화에 널리 사용되고 있다. 일반적으로 LSP 계수간에는 일정한 상관관계가 나타나고, 이 특성을 이용하면 LSP 계수의 부호량을 줄일 수 있는 가능성이 있나. 본 논문에서는 LSP 계수를 압축하기 위해 principal component analysis(PCA)를 사용한 방법을 제안한다. 제안된 방법에서는 LSP 계수를 Karhunen-Loeve(KL) 변환해 에너지가 집중되는 고유치(eigenvalue)와 고유벡터(eigenvector)를 찾고 값을 양자화 한다. 성능 평가를 위해 2.4kbps MELP(mixed excitation linear prediction)와 8kbps QCELP(qualcumn code excited linear prediction) 음성 부호화기를 사용해 결과 값을 비교했고, 압축률이 증가하는 것을 확인했다.

• 전자공학회논문지B
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• 제33B권10호
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• pp.72-79
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• 1996

We developed a speech coder that can transfer data as well as speech for CDMA digital cellular system. We describe the design method of the speech coder that uses QCELP algorithm for speech coding. The speech coder is implemented on a single fixed-point DSP chip (TMS320C50). the coder has the complexity such as 4K words in RAM, 10K words in ROM, and 33 MIPS in execution time. The developed speech coder is fully tested and successfully working on the CDMA base station system.

• 한국통신학회논문지
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• 제21권10호
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• pp.2634-2644
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• 1996

Two different low rate speech coders using one of four types of lattice vector quantizers(LVQ's) with fairly low complexity were investigated for an application to mobile communications. More specifically, two-stage vector quantizer-lattic vector quantizer(VQ-LVQ) systems and vector differenctial pulse code modulation(VDPCM)systems with lattice vector quantizers simulated to encode the line spectrum frequencies of various sentences at the rate 22 to 39 bits per 20 msec frame. The simulation results showed that the VDPCM system with the lattice VQ can save up to 10 bits/fram compared to the quantization scheme used in QCELP system. For the VQ-LVQ system, the spherical quasi-uniform LVQ below 36 bits/frame outperformed the other 3 types of LVQ's and the pyramidal quasi-uniform LVQ at 37 bits/frame outperformed the other 3 types of LVQ's with the spectral distortion 0.97.