• Proceedings of the Acoustical Society of Korea Conference
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• 1994.06c
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• pp.210-213
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• 1994

In this paper, we present a combination of trellis coded vector quantization and code-excited linear prediction coding, termed trellis excitation coding, for an efficient 4.8 kbps speech coding system. A training sequence-based algorithm is developed for designing an otimized codebook subject to the TEC structure. Also, we discuss the trellis symbol release rules that avoid excessive encoding delay. Finally, simulation results for the TEC coder are given at bit rate of 4.8 kbps.

• The Journal of the Acoustical Society of Korea
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• v.16 no.3E
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• pp.44-49
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• 1997

In this paper, we propose an adaptive encoding method of fixed codebook in CELP coders and implement an adaptive fixed code exited linear prediction(AF-CELP) speech coder. AF-CELP exploits the fact that the fixed codebook contribution to speech signal is also periodic like the adaptive codebook (or pitch filter) contribution. By modeling the fixed code book with the pitch lag and the gain from the adaptive codebook, AF-CELP can be implemented at low bit rates as well as low complexity. Listening tests show that a 6.4 kbit/s AF-CELP has a comparable quality to the 8 kbit/s CS-ACELP in background noise conditions.

• The Journal of the Acoustical Society of Korea
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• v.21 no.6
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• pp.531-542
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• 2002

Novel tandemless transcoding method for AMR and EVRC speech coders is proposed in this paper. In contrast to conventional tandem method, the parameters which is used commonly in speech coder where CELP algorithm is adapted are directly transcoded. The proposed algorithm is composed of LSP transcoding, pitch delay transcoding, gains transcoding and fixed codebook vector transcoding Evaluation results show that the novel algorithm achieves better speech quality than tandem method and reduce computational complexity and delay.

• Speech Sciences
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• v.11 no.4
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• pp.89-97
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• 2004

In this paper, an improved network echo canceller is proposed. The proposed echo canceller is based on the LTJ(lattice transversal joint) adaptive filter which uses informations from the speech decoder. In the proposed implementation method of the network echo canceller, the filer coefficients of the transversal filter part in the LTJ adaptive filter is updated every other sample instead of every sample. So its complexity can be lower than that of the transversal filter. And the echo cancellation rate can be improved by residual echo cancellation using the lattice predictor whose order is less than 10. Computational complexity of the proposed echo canceller is lower than that of the transversal filter but the convergence speed is faster than that of the transversal filter. The performance improvement of the proposed echo canceller was verified by the experiments using the real speech signal and speech coder.

• Speech Sciences
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• v.11 no.4
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• pp.17-28
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• 2004

In this paper, we propose a statistical approach to improve the performance of spectral quantization of speech coders. The proposed techniques compensate for the distortion in a decoded line spectrum pairs (LSP) vector based on a statistical mapping function between a decoded LSP vector and its corresponding original LSP vector. We first develop two codebook-based probabilistic matching (CBPM) methods based on linear mapping functions according to different assumption of distribution of LSP vectors. In addition, we propose an iterative procedure for the two CBPMs. We apply the proposed techniques to a predictive vector quantizer used for the IS-641 speech coder. The experimental results show that the proposed techniques reduce average spectral distortion by around 0.064dB.

• Journal of the Korean Institute of Telematics and Electronics
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• v.19 no.6
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• pp.84-90
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• 1982

In this paper we exploit the use of the intermittent property of speech to reduce the transmission rate or to increase signal-to-quantization noise ratio (SQNR) in coding speech by hybrid companding data modulation (HCDM). In this scheme we detect silence in speech by a speech/silence discriminator. HCDM coding is done only for speech portion. For silence that is detected in evert block of 5 ms, only the information indicating that the Since the HCDM coder transmits bina교 signal synchronously at a fixed rate, the use of a buffer and its efficient control is essential. By using the HCDM with silence detection in coding speech, we could improve SONR by as much as 6 dB over the conventional HCDM or reduce the transmission rate by one third of the HCDM rate.

• Proceedings of the Acoustical Society of Korea Conference
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• 1994.06c
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• pp.267-270
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• 1994

정보 및 통신의 필요성이 증대되면서 음성 부호화 방법에 관한 연구는 꾸준히 진행되어왔다. 특히, 이동통신에 대한 수요가 증가함에 따라 선진국에서는 기본 표준안을 완성하고, 채널 용량을 확대하기 위한 half-rate 표준화 작업이 한창 진행되고 있다. 본 논문에서는 표준화되거나 표준안으로의 가능성이 높은 음성 부호화 알고리즘들에 대해 서술한다. 또한 이로부터 향후 진행방향에 대해 언급하고자 한다.

• Proceedings of the Korean Institute of Communication Sciences Conference
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• 2004.07a
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• pp.211-211
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• 2004

• The Journal of the Acoustical Society of Korea
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• v.20 no.2
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• pp.3-9
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• 2001

In this paper, a random number generator for simulation of speech-waveform coders was designed. A random number generator having a desired probability density function and a desired power spectral density is discussed and experimental results are presented. The technique is based on Sondhi algorithm which consists of a linear filter and a memoryless nonlinearity. Several methods of obtaining memoryless nonlinearities for some typical continuous distributions are discussed. Sondhi algorithm is analyzed in the time domain using the diagonal expansion of the bivariate Gaussian probability density function. It is shown that the Sondhi algorithm gives satisfactory results when the memoryless nonlinearity is given in an antisymmetric form as in uniform, Cauchy, binary and gamma distribution. It is shown that the Sondhi algorithm does not perform well when the corresponding memoryless nonlinearity cannot be obtained analytically as in Student-t and F distributions, and when the memoryless nonlinearity can not be expressed in an antisymmetric form as in chi-squared and lognormal distributions.

• The Journal of the Acoustical Society of Korea
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• v.18 no.6
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• pp.38-43
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• 1999

The EVRC vocoder is a toll quality coder, but it shows significant degradation or the quality in weak RF environment. In this paper, the speech quality degradation phenomenon of the EVRC is analyzed, and two methods are proposed as the solution - the pitch prediction and the gradual increase. The preference tests for various Rf environment are performed for speech quality assessments and both the methods show better performance.