• The Journal of the Acoustical Society of Korea
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• v.24 no.5
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• pp.248-254
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• 2005

In this Paper, the structure of a split bandwidth wideband speech coder and its highband coder for tone qualify elevation are Proposed. The lowband and highband by the split bandwidth method are encoded independently applying the G.729E and MLT (Modulated Lapped Transform) residual model. In the highband structure which is encoded by low bit rate of 4kbps, the MLT residual signals are distinguished to voice and unvoice signal . The voice signals are applied to MLT peak picking method by lowband pitch period. Because transformed MLT residual signals are represented by periodic signal that have periodic peak. The unvoice signals are applied to MLT which linear prediction spectral response is added and do vector quantization. Performance for proposed 15.8kbps wideband speech coder was verified through subjective listening test.

• The Journal of the Acoustical Society of Korea
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• v.23 no.6
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• pp.481-487
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• 2004

In this paper. the structure of bandwidth scalable wideband speech codec and its high-band codec are proposed. In the high-band codec. the signal is divided into frequency bands. and each band is quantized in DCT domain. The DCT coefficients are splitted into magnitude and sign, and each is quantized independently by a specialized method based on its characteristics. In addition. the quantized gain parameter in the low-band codec is utilized in the high-band codec for an enhanced performance. The bandwidth scalable wideband speech codec using G.729E for low-band and the proposed codec for high-band is developed, and it is confirmed that the proposed codec has better subjective performance than 24kbps G.722.1.

• The Journal of the Acoustical Society of Korea
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• v.27 no.6
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• pp.309-318
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• 2008

In this paper, we discuss a method for efficiently coding the high-band signal in the split-band coding approach where an input signal is divided into two bands and then each band may be encoded separately. Generally, and especially through the research on the artificial bandwidth extension (BWE), it is well known that there is a correlation between the two bands to some degree. Therefore, some coding gain could be achieved by utilizing the correlation. In the BWE-prediction-based coding approach, using a simple linear BWE function may not yield optimal results because the correlation has a non-linear characteristic. In this paper, we investigate the new coding scheme more in details. A few representative BWE functions including linear and non-linear ones are investigated and compared to find a suitable one for the coding purpose. In addition, it is also discussed whether there are some additional gains in combining the BWE coder with the predictive vector quantizer which exploits the temporal correlation.

• The Journal of the Acoustical Society of Korea
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• v.24 no.7
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• pp.395-401
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• 2005

In this paper, the high-band codec for bandwidth scalable wideband speech codec is proposed. The wideband input speech signal is separated into low-band signal and high-band signal, and the low-band signal is encoded by the standard narrow-band speech codec and the high-band signal is encoded by the proposed codec. In the high-band codec. the signal is transformed into frequency domain by MLT on a subframe basis, and MLT coefficients are splitted into magnitude and sign for quantization. The magnitudes of MLT coefficients are arranged into several time-frequency bands and each band is quantized in 2D-DCT domain, where the low-band information is utilized for better performance. The sign of MLT coefficient is quantized based on a priority selection process with the weighting measurement. The objective and subjective performance of wideband speech codec including the proposed high-band codec is measured, and it is confirmed that the proposed codec has better performance than 32kbps G.722.1.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.19 no.12
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• pp.2313-2327
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• 1994

As the communication environment evolves, there is an increasing need for multiresolution image coding. To meet this need, the entrophy constratined vector quantizer(ECVQ) for coding of image pyramids by spline wavelet transform is introduced in this paper. This paper proposes a new scheme for image compression taking into account psychovisual feature both in the space and frequency domains : this proposed method involves two steps. First we use spline wavelet transform in order to obtain a set of biorthogonal subclasses of images ; the original image is decomposed at different scale using a pyramidal algorithm architecture. The decomposition is along the vertical and horizontal directions and maintains constant the number of pixels required the image. Second, according to Shannon's rate distortion theory, the wavelet coefficients are vectored quantized using a multi-resolution ECVQ(entropy-constrained vector quantizer) codebook. The simulation results showed that the proposed method could achieve higher quality LENA image improved by about 2.0 dB than that of the ECVQ using other wavelet at 0.5 bpp and, by about 0.5 dB at 1.0 bpp, and reduce the block effect and the edge degradation.

• The Journal of the Acoustical Society of Korea
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• v.17 no.5
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• pp.57-62
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• 1998

본 논문에서는 입력신호의 시변특성에 따라 분석 필터의 대역을 가변 시키는 필터 뱅크의 구조를 제안한다. 제안된 필터뱅크는 일반적으로 32개의 균일한 대역으로 나누어 임 계대역의 표현을 적절히 표현하지 못하는 Polyphase 필터의 단점을 극복하면서 시스템 설 계에 높은 계산량을 요구하는 QMF-tree 필터의 단점을 보완한다. 본 연구에서는 분할 대역 은 4개에서 26개의 대역으로 가변하고, 웨이블릿 필터중 Daubechies필터를 사용하였다. 제 안된 구조의 부호화기는 128kbps에서 MPEG-a오디오와 비슷한 수준의 CD 음질을 유지하 며, 연산량 비교결과는 PolyPhase filter를 이용한 MPEG보다 부호화, 복호화 과정을 합쳐 다양한 전송률과 음원에서 평균 19%의 감소를 얻었다.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.45 no.2
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• pp.81-89
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• 2008

Wideband speech, characterized by a bandwidth of about 7 kHz (50-7000 Hz), provides a substantial quality improvement in terms of naturalness and intelligibility. Although higher data rates are required, it has extended its application to audio and video conferencing, high-quality multimedia communications in mobile links or packet-switched transmissions, and digital AM broadcasting. In this paper, we present a new bandwidth-scalable coder for wideband speech and audio signals. The proposed coder spits 8kHz signal bandwidth into two narrow bands, and different coding schemes are applied to each band. The lower-band signal is coded using the ITU-T G.729/G.729E coder, and the higher-band signal is compressed using a new algorithm based on the gammatone filter bank with an invertible auditory model. Due to the split-band architecture and completely independent coding schemes for each band, the output speech of the decoder can be selected to be a narrowband or wideband according to the channel condition. Subjective tests showed that, for wideband speech and audio signals, the proposed coder at 14.2/18 kbit/s produces superior quality to ITU-T 24 kbit/s G.722.1 with the shorter algorithmic delay.

• Journal of Broadcast Engineering
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• v.7 no.1
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• pp.37-44
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• 2002

In this paper. we propose a new video coding scheme with multi-resolution motion compensation and residual quantization. The main functional blocks of the proposed scheme include multi-level wavelet decomposition. motion estimation and motion compensation. raster scanning within each subband. formation of block trees and blocking partitioning. and adaptive arithmetic coding. Although the proposed ceding scheme is simple and requires low computational complexity, it produces bitstreams tilth good features. such as the embedded structure and SNR scalability. Experimental results demonstrate that the proposed coding scheme is quote competitive to and often outperforms other wavelet-based video coding schemes in the literature.

• Journal of Broadcast Engineering
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• v.8 no.2
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• pp.181-197
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• 2003

Notion Compensation(MC) technique using Sub-Band Coding with the hierarchical structure is efficient to estimate real motion. In the hierarchical pyramid method, low-band MC pyramid method is popular, where the upper layer estimate the glover motion and next lower layer estimate the local motion. The low-band MC pyramid scheme has two problems. First, because the quantization errors at lower layer are accumulated when using coding and quantizing, it is impossible to search the exact Motion Vector(MV) Second, because of the top-down search problem in the hierarchical structure, MV mismatch in upper layer causes serious MV in lower layer So. we propose new hierarchical MC pyramid method based on edge classification. In this Paper, we show that the performance of proposed Pass-band motion compensation pyramid technique is better than low-band motion compensation pyramid. Also, in the pyramid motion estimation, we propose initial MV estimation scheme based on the edge-pattern classification. As a result, we find that PSNR was increased.

• The Journal of the Acoustical Society of Korea
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• v.16 no.4
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• pp.64-70
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• 1997

This paper is aimed to propose a new wavelet audio signal coding scheme which reduces the complexity of well-known MPEG(Moving Picture Expert Group)-Audio. The filters of MPEG0audio apply subband technique on the 16-bits PCM audio to aquire bitstream of subband sample using dynamic bit allocation. If we use the wavelet coefficients instead of subband samples and 6 bands which is less than 32 bands of MPEG-audio, the complexity can be reduced. A new audio signal compression algorithm in this paper is based on wavelet transform and the proposed algorithm is compared with MPEG-audio. At the bitrate of 256kbps, the proposed algorithm maintains the CD(Compact-disc) quality. We were able to reduce the about 40% of complexity at encoder and about 70% at decoder.