• Title/Summary/Keyword: Code-Excited Linear Prediction(CELP)

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Voice Activity Detection Algorithm base on Radial Basis Function Networks with Dual Threshold (Radial Basis Function Networks를 이용한 이중 임계값 방식의 음성구간 검출기)

  • Kim Hong lk;Park Sung Kwon
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.29 no.12C
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    • pp.1660-1668
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    • 2004
  • This paper proposes a Voice Activity Detection (VAD) algorithm based on Radial Basis Function (RBF) network using dual threshold. The k-means clustering and Least Mean Square (LMS) algorithm are used to upade the RBF network to the underlying speech condition. The inputs for RBF are the three parameters in a Code Exited Linear Prediction (CELP) coder, which works stably under various background noise levels. Dual hangover threshold applies in BRF-VAD for reducing error, because threshold value has trade off effect in VAD decision. The experimental result show that the proposed VAD algorithm achieves better performance than G.729 Annex B at any noise level.

A New Fast Pitch Search Algorithm using Line Spectrum Frequency in the CELP Vocoder (CELP보코더에서 Line Spectrum Frequency를 이용한 고속 피치검색)

  • Bae, Myung-Jin;Sohn, Sang-Mok;Yoo, Hah-Young;Byun, Kyung-Jin
    • The Journal of the Acoustical Society of Korea
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    • v.15 no.2
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    • pp.90-94
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    • 1996
  • Code Excited Linear Prediction(CELP) vocoder exhibits good performance at data rates below 8 kbps. The major drawback of CELP type coders is a large amount of computation. In this paper, we propose a new pitch searching method that preserves the quality of the CELP vocoder reducing computational complexity. The basic idea is that grasps preliminary pitches using the first formant of speech signal and performs pitch search only about the preliminary pitches. As applying the proposed method to the CELP vocoder, we can reduce complexity by 64% in the pitch search.

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Signal Enhancement of a Variable Rate Vocoder with a Hybrid domain SNR Estimator

  • Park, Hyung Woo
    • KSII Transactions on Internet and Information Systems (TIIS)
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    • v.13 no.2
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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.

Perceptual Quality Improvement of KLT based Entropy-Constrained Quantizer using a SAW Filter (SAW 필터를 이용한 KLT 기반 Entropy-Constrained Quantizer 성능 향상)

  • Lim, Dong-Seok;Kim, Moo Young
    • Proceedings of the Korean Society of Broadcast Engineers Conference
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    • 2013.06a
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    • pp.1-2
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    • 2013
  • KLT-AECQ 는 지각적인 성능 향상을 위하여 formant weighting 필터를 사용한다.Code Excited Linear Prediction(CELP) 코더는 사람의 음성신호를 압축하는 대표적인 방식이다. CELP 의 Rate-Distortion 성능을 향상 시키기 위해서 Karhunen-Loeve-Transform (KLT) 기반의 Classified Vector Quantization (KLT-CVQ) 방식이 제안되었으며, 이는 KLT 기반의 Adaptive Entropy-Constrained Quantization (KLT-AECQ) 방식으로 확장되었다. 기존의 KLT-AECQ 에서는 지각적인 성능 향상을 위하여 formant weighting 필터를 사용한다. 본 논문에서는 이 필터 대신에 Spectral Amplitude Warping (SAW) 필터를 적용함으로써, KLT-AECQ 코더의 지각적인 성능을 향상하였다.

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Real-time implementation of the G.723.1 voice coder using DSP56362 (DSP56362를 이용한 G.723.1 음성코덱의 실시간 구현)

  • Lee, Jae-Sik;Son, Yong-Ki;Chang, Tae-Gyu;Min, Byoung-Ki
    • Speech Sciences
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    • v.7 no.2
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    • pp.225-234
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    • 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 DSP56362. 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.

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Reduction in Computational Complexity of KLT-CVQ using UTV Decomposition (UTV 분해를 이용한 KLT-CVQ 코더의 계산량 개선)

  • Ju, Hyunho;Kim, Moo Young
    • Proceedings of the Korean Society of Broadcast Engineers Conference
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    • 2012.07a
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    • pp.176-177
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    • 2012
  • 사람의 음성을 압축하는 방법으로 Code Excited Linear Prediction (CELP) 코더가 주로 사용되어 왔다. CELP 코더의 수신단에서는 양자화 된 여기신호를 LPC 필터로 합성하여 신호를 복원한다. LPC 합성필터의 영향으로 양자화 된 여기신호의 보로노이 셀 모양이 변형되는 문제점이 있기 때문에 이런 문제점을 해결하기 위해서 Karhunen-Loeve-Transform based Classify vector Quantization (KLT-CVQ) 코더가 제안되었다. 기존 KLT-CVQ 코더는 KLT 변환과 class 선택을 위해서 Eigen Value Decomposition (EVD)을 이용해서 eigen vector와 eigen value를 계산한다. 본 논문에서는 EVD 대신에 UTV Decomposition (UTVD)을 이용하여 KLT-CVQ의 계산량 문제점을 개선하는 방법을 제안한다.

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On a Reduction of Pitch Searching Time by Separating the Speech Components in the CELP Vocoder (성분분리에 의한 CELP 보코더의 피치 검색시간 단축에 관한 연구)

  • Hyeon, Jin-Il;Byeon, Gyeong-Jin;Han, Gi-Cheon;Kim, Jong-Jae;Yu, Ha-Yeong;Kim, Jae-Seok;Kim, Dae-Sik;Bae, Myeong-Jin
    • The Journal of the Acoustical Society of Korea
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    • v.14 no.1E
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    • pp.22-29
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    • 1995
  • Code excited Linear Prediction(CELP) vocoder exhibits good performance at data rates below 4.8 kbps. The major drawback of CELP type coders is their large amount of computation. In this paper, we propose a new pitch searching method that preseves the quality of the CELP vodocer reducing computational complexity. The basic idea is that pregrasps preliminary pitches about signal and performs pitch search only about the preliminary pitches. Applying the proposed method to the CELP vocoder, we can reduce complexity about 90% in th pitch search.

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A Study on an Improvement of the Performance by Spectrum Analysis with Variable Window in CELP Vocoder (CELP 부호화기에서 가변 윈도우 스펙트럼 분석에 의한 성능 향상에 관한 연구)

  • Min So-Yeon;Kim Eun-Hwan;Bae Myung-Jin
    • Journal of the Korea Society of Computer and Information
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    • v.10 no.6 s.38
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    • pp.233-238
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    • 2005
  • In general CELP(Code Excited Linear Prediction) type vocoders provide good speech qualify around 4.8kbps. Among them, G.723.1 developed for Internet Phone and video-conferencing includes two vocoders, 5.3kbps ACELP(Algebraic-CELP) and 6.3kbps MP-MLQ(Multi-Pulse Maximum Likelihood Quantization) In order to improve the speech qualify in CELP vocoder, in this paper. we proposed a new spectrum analysis algorithm with variable window In CELP vocoder, the spectrum of the synthesised speech signal is distorted because the fixed size windows is used for spectrum analysis. So we have measured the spectral leakage and in order to minimize the spectral leakage have adjusted the window size. Applying this method G.723.1 ACELP, we can got SD(Spectral Distortion) reduction 0.084(dB), residual energy reduction 6.3$\%$ and MOS(Mean Opinion Score) improvement 0.1.

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Real-time implementation of the 2.4kbps EHSX Speech Coder Using a $TMS320C6701^TM$ DSPCore ($TMS320C6701^TM$을 이용한 2.4kbps EHSX 음성 부호화기의 실시간 구현)

  • 양용호;이인성;권오주
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.29 no.7C
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    • pp.962-970
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    • 2004
  • This paper presents an efficient implementation of the 2.4 kbps EHSX(Enhanced Harmonic Stochastic Excitation) speech coder on a TMS320C6701$^{TM}$ floating-point digital signal processor. The EHSX speech codec is based on a harmonic and CELP(Code Excited Linear Prediction) modeling of the excitation signal respectively according to the frame characteristic such as a voiced speech and an unvoiced speech. In this paper, we represent the optimization methods to reduce the complexity for real-time implementation. The complexity in the filtering of a CELP algorithm that is the main part for the EHSX algorithm complexity can be reduced by converting program using floating-point variable to program using fixed-point variable. We also present the efficient optimization methods including the code allocation considering a DSP architecture and the low complexity algorithm of harmonic/pitch search in encoder part. Finally, we obtained the subjective quality of MOS 3.28 from speech quality test using the PESQ(perceptual evaluation of speech quality), ITU-T Recommendation P.862 and could get a goal of realtime operation of the EHSX codec.c.

A Very Low-Bit-Rate Analysis-by-Synthesis Speech Coder Using Zinc Function Excitation (Zinc 함수 여기신호를 이용한 분석-합성 구조의 초 저속 음성 부호화기)

  • Seo Sang-Won;Kim Jong-Hak;Lee Chang-Hwan;Jeong Gyu-Hyeok;Lee In-Sung
    • The Journal of the Acoustical Society of Korea
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    • v.25 no.6
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    • pp.282-290
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    • 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.