• The Journal of the Acoustical Society of Korea
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• v.21 no.1
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• pp.5-10
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• 2002

This paper proposes new 16 kbps wideband speech codec with bandwidth of 7 kHz. The proposed codec decomposes the input speech signal into low-band and high-band signals using QMF (Quadrature Mirror Filter), then AMR (Adaptive Multi Rate) speech codec processes the low-band signal and new transform-domain codec based on G.722.1 wideband cosec compresses the high-band signal. The proposed codec allocates different number of bits to each band in an adaptive way according to the property of input signal, which provides better performance than the codec with the fixed bit allocation scheme. In addition, the proposed cosec processes high-band signal using wavelet transform for better performance. The performance of proposed codec is measured in a subjective method. and the simulations with various speech data show that the proposed coders has better performance than G.722 48 kbps SB-ADPCM.

• Broadcasting and Media Magazine
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• v.19 no.1
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• pp.48-55
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• 2014

One of the recent noticeable evolutions in mobile communication systems is that wideband-codec is deployed rapidly in VoLTE (Voice over Long Term Evolution) service or HD voice. This paper is concerned with next generation HD voice or VoLTE service that is coined to describe high quality communication with super-wideband voice codec. 3GPP EVS (Enhanced Voice Service) Codec is being standardized to develop the super-wideband voice codec. This paper deals with the codec design constraints, performance requirements, the status of standardization, and finally perspective on VoLTE service in future.

• The Journal of the Acoustical Society of Korea
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• v.22 no.4
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• pp.308-314
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• 2003

Wideband speech communications with 7㎑ bandwidth can provide high-quality speech services that are almost impossible with current narrow-band speech communications with 3.4 ㎑ bandwidth, and AMR wideband codec was recently developed for these services. The performance of AMR wideband codec is excellent due to its wideband information and partially to ACELP structure, but it requires high computational complexity especially in codebook search. In this paper, to solve this problem, an efficient codebook search method for AMR wideband codec is proposed. The proposed method first determines the coarse initial codevector, then improves the performance of codevector by replacing a poor pulse in codevector with better one iteratively. Simulations show that AMR wideband codec with proposed codebook search method has higher performance with much less computational cost than conventional AMR wideband codec.

• The Journal of the Acoustical Society of Korea
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• v.20 no.7
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• pp.76-82
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• 2001

In this paper, a novel quantization method for wideband signal codec with 7 kHz bandwidth is proposed. In the transform-based wideband signal codecs, the signal is transformed to frequency domain and the spectral coefficients in each frequency band are quantized based on human perceptual model, followed by Huffman coding. However, the property of each band varies with frequency, and the codec has poor performance when all bands are quantized with the same method. Therefore, a selective quantization method is proposed, which analyzes the band property and selects the quantization domain between frequency domain and time domain based on the quantization efficiency. It is confirmed that the proposed method has better performance than the quantizer of G722.1 codec.

• Journal of IKEEE
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• v.2 no.2 s.3
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• pp.233-238
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• 1998

In this paper, the implementation of dual rate ADPCM using G.726 16Kbps and 40Kbps speech codec algorithm is handled. For small signals, the low rate 16Kbps coding algorithm shows almost the same SNR as the high rate 40Kbps coding algorithm , while the high rate 40Kbps coding algorithm shows the higher SNR than the low rate 16Kbps coding algorithm fur large signal. To obtain the good trade-off between the data rate and synthesized speech quality, we applied low rate 16Kbps for the small signal and high rate 40Kbps for the large signal. Various threshold values determining the rate are applied for good trade-off between data rate and speech quality. The simulation result shows the good speech quality at a low rate comparing with 16Kbps & 40Kbps.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.11 no.8
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• pp.3076-3082
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• 2010

The techniques and the products which use various video compression codec are come out from army or civil field. In existing high-end PC environment, process of the video compression codec does not become a problem, but in embedded system environments which limited system resources, because the system load due to the high-resolution images compressed by high-density, issues of performance and utilization are highlighted. This paper proposes the DirectShow Filter interfaces which are a hardware method in order to solve the problem existing software algorithms for image compression performance and peripheral interfaces.

• Proceedings of the Korea Information Processing Society Conference
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• 2010.11a
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• pp.1586-1588
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• 2010

디지털 비디오 압축기술은 대용량의 영상을 화질 손실을 최소화하면서 압축률을 높이는 데에 그 목적이 있다. 모바일 디바이스에서 인코딩/디코딩시에는 프로세싱 오버헤드를 최소화 시켜야 할 필요성이 있다. 또한 특정 동영상을 압축할 시 불필요한 영상을 줄여 압축률을 높이고 타겟 오브젝트에 집중할 수 있도록 하기 위해서 Hierarchical Temporal Memory 를 사용해 오브젝트를 인식하고 타겟 오브젝트만 선택 압축하는 기술을 제안하고자 한다.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.40 no.4
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• pp.214-221
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• 2003

This paper shows an implementation of video codec (coder/decoder) on an embedded system. The video codec supports both H.261 and H.263 standards. For efficient real-time processing, the video codec is partitioned into a software module and a hardware module. Both modules are codesigned on an embedded system. The software module is processed on a real-time operating system and a RISC processor. It cooperates with the hardware module to compress and decompress images in real time. AMBA (Advanced Microcontroller Bus Architecture) AHB (Advanced High-performance Bus) is used as the system bus. The hardware module works both as AHB masters and as AHB slaves. The encoder part of the hardware module operates in a pipelines mode to compress images in real time. The video codec compresses 15 CIF frames and simultaneously decompresses 15 CIF frames in a second according to H.261 or H.263 standard at 33 MHz frequency.

• Broadcasting and Media Magazine
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• v.12 no.4
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• pp.28-39
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• 2007

다양한 디지털 기술의 발전으로 인하여 방송형태의 이동 멀티미디어 서비스가 다국적으로 제안되고, 국내에서는 이동 멀티미디어 방송 (DMB: Digital Multimedia Broadcasting)을 통하여 야외나 이동시에도 시청이 가능한 방송서비스가 활발해지고 있다. 휴대 및 이동수신 방송 환경에서 비디온 오디오 및 데이터를 포함한 멀티미디어 방송 서비스를 효율적으로 제공하기 위해서는 다양한 장소에서 수신 영상에 대한 품질 확보가 필수적이다. 본 논문에서는 현재 이동 멀티미디어 방송이 비디오 압축방식으로 채택하고 있는 H.264/AVC 압축 파라미터의 성능 연구에 대하여 기술한다. 현재 국내의 위성/지상파 DMB의 경우 비디오의 압축 방법으로 H.264/AVC baseline 1.3의 표준규격을 사용한다. 이러한 비디오 코덱(codec) 이용하여 비디오 영상을 압축할 경우 관련 파라미터(parameter) 조절이 가능한데, 비디오를 압축할 경우 관련 파라미터들을 어떻게 정하느냐에 따라 서로 다른 수신환경에서 압축 효율 및 재생된 비디오의 화질에 많은 영향을 미친다. 따라서 수신 환경에 가장 적합한 비디오 화질을 얻기 위해서는 관련 파라미터 설정이 매우 중요하다. 본 논문에서는 다양한 압축 파라미터들 중 화질에 많은 영향을 미치는 항목을 선정하여, 해당 파라미터의 변화가 재생된 비디오 화질에 미치는 영향을 객관적 평가척도인 PSNR, Bit-rate, 수행시간 등을 이용하여 분석하였다. 또한, 실험 결과를 바탕으로 이동 멀티미디어 방송 환경에서의 H.264 인코더의 적정 압축 파라미터 및 인코더의 성능 개선 방안을 제안한다.

• Proceedings of the IEEK Conference
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• 2001.06d
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• pp.49-52
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• 2001

This paper presents a FPGA Implementation of wavelet-based CODEC, which can compress 2-dimensional image. For real-time processing, a scheduling method of input image data is proposed and a new structure of MAC(multiplier-accumulator) is proposed for wavelet transforms. Also this study proposes global pipelining structure of wavelet CODEC and efficient buffering method at interfaces between each module with different clock frequency.