• The Journal of the Acoustical Society of Korea
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• v.24 no.7
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• pp.402-409
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• 2005

In this paper. we propose a coding method using a matching pursuit algorithm in a strongly periodic highband signal. Also. we propose an efficient quantizer for the estimated parameters : spectral magnitude and phase. Based on the error concealment principle and sinusoidal model. the MP algorithm requires the high-precision pitch period estimation. To estimate more accurate pitch period. the refined pitch obtained from lowband speech is used. which increases the efficiency of bit allocation. The spectral magnitude parameters are quantized by the method which is combined with MDCT (Modified Discrete Cosine Transform) and multi-stage structure. The spectral phase quantizer uses the $2{\pi}$ modular characteristic of phases and the weighted function by spectral magnitudes. To evaluate the efficiency of the proposed method. we applied it to analysis-by-synthesis system. Furthermore we suggest the possibillity of scalable wideband speech codecs based on band-split structure.

• Phonetics and Speech Sciences
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• v.1 no.4
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• pp.97-103
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• 2009

This paper describes a new algorithm for encoding spectral envelope in the time domain alias cancellation (TDAC) part of G.729.1. The spectral envelope and modified discrete cosine transform (MDCT) coefficients of the weighted code-excited linear predictive (CELP) coding error in lower-band and the higher-band input signal are encoded in the TDAC part. In order to reduce allocation bits for spectral envelope coding, a new algorithm using sub-band correlation between adjacent frames is proposed. In addition, to improve the quality of decoded signals, two bit allocation strategies using reduced bits from the proposed algorithm are proposed. The performance of the proposed algorithm is evaluated in terms of objective quality and bit reduction rates. Experimental results show that the proposed algorithm increases the quality of sounds significantly.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2021.06a
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• pp.32-35
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• 2021

본 논문에서는 시간축 정보량을 감축하여 오디오 부호화 효율을 개선하기 위한 기술을 제안한다. 시간축 정보량 감축 방법은 종전의 오디오 코덱에서도 활용되었던 대표적인 기술로 TNS(temporal noise shaping) 기술이 있다. 그러나 TNS 기술은 오디오 신호의 천이구간에서 선별적으로 유효하게 동작하며 그 효율성도 간헐적으로 나타나는데 이는 MDCT(modified discrete cosine transform)에서 예측 과정을 수행하는 구조적인 문제를 갖고 있기 때문이다. 본 논문에서는 종전의 TNS 기술의 취약점을 보완한 ITES(intensive temporal envelope shaping) 기술을 제안하였다. 제안 기술은 TNS 보다 유효한 오디오 시간영역 정보량을 예측하고 감축하였으며, 개선된 음질을 나타냄을 주관적 평가를 수행하여 검증하였다.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.49 no.4
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• pp.16-22
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• 2012

In this paper, the forensic marking algorithm is proposed using psychoacoustic model II and MDCT for high-quality audio. The proposed forensic marking method, that inserts the user fingerprinting code of the audio content into the selected sub-band, in which audio signal energy is lower than the spectrum masking level. In the range of the one frame which has 2,048 samples for FFT of original audio signal, the audio forensic marking is processed in 3 sub-bands. According to the average attack of the fingerprinting codes, one frame's SNR is measured on 100% trace ratio of the collusion codes. When the lower strength 0.1 of the inserted fingerprinting code, SNR is 38.44dB. And in case, the added strength 0.5 of white gaussian noise, SNR is 19.09dB. As a result, it confirms that the proposed audio forensic marking algorithm is maintained the marking robustness of the fingerprinting code and the audio high-quality.

• Journal of Biomedical Engineering Research
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• v.27 no.5
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• pp.282-290
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• 2006

We have developed a multi band digital hearing aid algorithm emphasizing feedback cancellation and a hardware module to evaluate the performance of our algorithm. The hearing aids should be able to compensate for individual hearing loss characteristics of hearing impaired person. Thus hearing aids need the function of multi-bands amplification and the capabilities of feedback cancellation that can remove howling caused by acoustic feedback. In this paper, we proposed a digital hearing aid algorithm which has multi-bands compensation using modified discrete cosine transform (MDCT) and can efficiently remove acoustic feedbacks. Moreover, we have developed digital hearing aid hardware module, which can evaluate hearing aid algorithms in real time operation. The developed algorithm and hardware module were verified through computer simulation and clinical tests. Through operational experiments, good performances in real time operation environment and an efficient howling cancellation were also observed. The developed hardware module can operate in stable condition and it is expected to become a good hardware platform for developing hearing aid algorithms.