• The Journal of the Acoustical Society of Korea
• /
• v.17 no.2
• /
• pp.43-49
• /
• 1998

본 논문에서는 고성능 CELP 음성 압축기를 위한 "Boaseline 코드벡터"와 "Implied 코드벡터"로 구성되는 새로운 구조의 코드북을 제안한다. Implied 코드벡터는 피치 주기 이 전의 합성음으로부터 구하여지며 여기(勵起)신호의 피치 구조를 강화하여 합성음의 음질을 향상시킨다. Implied 코드벡터는 전달되지 않고 인코더 및 디코더에서 각각 합성음을 이용 하여 독립적으로 구하여진다. 또한 펄스와 랜덤 성분을 모두 가지는 복합 여기방식을 이용 하여 음질을 더욱 향상시킨다. 제안된 코드북 구조를 이용하여 10msec프레임을 가지는 8kbps CELP 음성 압축기를 설계하여 하나의 DSP칩에 실시간 구현 하였고, 이것의 성능을 SNRseg와 MOS로 측정하였다. 평균 SNRseg는 12.14dB로 CS-ACELP의 SNRseg보다 6dB 높고, 조용한 환경에서의 MOS는 3.80으로 G.729 CS-ACELP의 MOS보다 0.02 높다.

• Proceedings of the Korean Information Science Society Conference
• /
• 1999.10c
• /
• pp.271-273
• /
• 1999

본 논문에서는 CELP 부호화기의 계산량을 줄이면서도 고음질의 음성을 합성할 수 있는 코드북 구조를 제안한다. 제안한 코드북 구조는 불규칙 코드북과 희박 중첩형 코드북 두 개의 코드북의 합으로 여기 신호를 표현한다. codebook I에서 잔류신호와 오차가 적은 여기신호열을 구한 후, 이 여기신호열에 codebook II의 여기신호열을 합하여 최적의 여기신호열을 구한다. 또한 이로 인한 전송비트수의 증가를 막기위해 홀수 프레임에서는 두 개 코드북의 index를, 짝수 프레임에서는 codebook I의 여기신호열은 그대로 사용하고 codebook II에서만 검색하여 전송하는 방법을 사용하였다. 이러한 2중 코드북 구조는 두 개의 여신호열의 합으로 표현되고 각각의 서로 다른 코드북 이득을 사용하기 때문에 정확한 이득을 표현할 수 있어 기존의 개선 알고리듬보다 더 나은 음질을 제공할 수 있다. 검색시간이 빠르고, 본 코드북 구조를 갖는 4.8kbps CELP형 부호화기를 설계하여 컴퓨터 모의 실험한 결과, 같은 전송률을 갖는 DoD CELP 부호화기보다 segSNR가 0.53dB 더 높게 나타났다.

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

• Journal of Advanced Navigation Technology
• /
• v.4 no.1
• /
• pp.67-77
• /
• 2000

This paper proposes a reduction scheme for codebook search time in the adaptive codebook using wavelet transformed coefficients. In a CELP coder, pitch estimation with a combined open loop and closed loop search in adaptive codebook needs a lengthy search. More precisely, the pitch search using autocorrelation function over all possible ranges has been shown inefficient compared to the consuming time. In this paper, we propose a new adaptive codebook search algorithm which ensures the same position for the pitch with maximum wavelet coefficient over various scaling factors in Dyadic wavelet transform. A new adaptive codebook search algorithm reduces 25% conventional search time with almost the same quality of speech.

• Proceedings of the Acoustical Society of Korea Conference
• /
• spring
• /
• pp.51-54
• /
• 2000

This paper describes the fixed-point DSP implementation of a CELP(code-excited linear prediction)-based speech coder. The effective realization methodologies to maximize the utilization of the DSP's architectural features, specifically Parallel movement and pipelining are also presented together with the implementation results targeted for the ITU-T standard G.723.1 using Motorola DSP56309. The operation of the implemented speech coder is verified using the test vectors offered by the standard as well as using the peripheral interface circuits designed for the coder's real-time operation.

• The Journal of the Acoustical Society of Korea
• /
• v.22 no.3E
• /
• pp.101-109
• /
• 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.

• MALSORI
• /
• no.49
• /
• pp.63-75
• /
• 2004

Most existing telephone speech transmitted in current public networks is band-limited to 0.3-3.4 kHz. Compared with wideband speech(0-8 kHz), the narrowband speech lacks low-band (0-0.3 kHz) and high-band(3.4-8 kHz) components of sound. As a result, the speech is characterized by the reduced intelligibility and a muffled quality, and degraded speaker identification. Bandwidth extension is a technique to provide wideband speech quality, which means reconstruction of low-band and high-band components without any additional transmitted information. Our new approach considers to exploit harmonic synthesis method for reconstruction of low-band speech over the CELP coded speech. A spectral distortion measurement and listening test are introduced to assess the proposed method, and the improvement of synthesized speech quality was verified.

• The Journal of the Acoustical Society of Korea
• /
• v.25 no.6
• /
• pp.282-290
• /
• 2006

This paper proposes a new Digital Reverberator that models Analog Helical Coil Spring Reverberator for guitar amplifiers. While the conventional digital reverberators are proposed to provide better sound field mainly based on room acoustics, no algorithm or analysis of digital reverberators those model Helical Coil Spring Reverberator was proposed. Considering the fact that approximately $70{\sim}80$ percent of guitar amplifiers are still with Helical Coil Spring Reverberator, research was performed based not on Room Acoustics but on Helical Coil Spring Reverberator itself as an effector. After performing simulations with proposed algorithm, it was confirmed that the Digital Reverberator by proposed algorithm provides perceptually equivalent response to the conventional Analog Helical Coil Spring Reverberators.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.33 no.9C
• /
• pp.691-696
• /
• 2008

A new vocoder of Code Excited Linear Predictive (CELP) based on Adaptive Multi Rate (AMR) 7.4kbit/s mode is proposed in this paper. The proposed vocoder achieves a better compression rate in an environment of Speaker Dependent Coding System (SDSC) and is efficiently used for systems, such as OGM(Outgoing message) and TTS(Text To Speech), which needs only one person's speech. In order to enhance the compression rate of a coder, a new Line Spectral Pairs(LSP) code-book is employed by using Centroid Neural Network (CNN) algorithm. In comparison with original(traditional) AMR 7.4 Kbit/s coder, the new coder shows 27% higher compression rate while preserving synthesized speech quality in terms of Mean Opinion Score(MOS).

• Journal of the Institute of Electronics Engineers of Korea SP
• /
• v.42 no.2 s.302
• /
• pp.131-142
• /
• 2005

The low rate speech coders under 4 kbit/s are based on sinusoidal transform coding (STC) or multiband excitation (MBE). Since the harmonic coders are not efficient to reconstruct the transient segments of speech signals such as onsets, offsets, non-periodic signals, etc, the coders do not provide a natural speech quality. This paper proposes method of a efficient transient model :d a multi-mode low rate coder at 2.4 kbit/s that uses harmonic model for the voiced speech, stochastic model for the unvoiced speech and a model using aperiodic pulse location tracking (APPT) for the transient segments, respectively. The APPT utilizes the harmonic model. The proposed method uses different models depending on the characteristics of LPC residual signals. In addition, it can combine synthesized excitation in CELP coding at time domain with that in harmonic coding at frequency domain efficiently. The proposed coder shows a better speech quality than 2.4 kbit/s version of the mixed excitation linear prediction (MELP) coder that is a U.S. Federal Standard for speech coder.