• ETRI Journal
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• v.27 no.5
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• pp.497-503
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• 2005

In this paper, we present a performance analysis for an MPEG-4 video codec based on the on-chip network communication architecture. The existing on-chip buses of system-on-a-chip (SoC) have some limitation on data traffic bandwidth since a large number of silicon IPs share the bus. An on-chip network is introduced to solve the problem of on-chip buses, in which the concept of a computer network is applied to the communication architecture of SoC. We compared the performance of the MPEG-4 video codec based on the on-chip network and Advanced Micro-controller Bus Architecture (AMBA) on-chip bus. Experimental results show that the performance of the MPEG-4 video codec based on the on-chip network is improved over 50% compared to the design based on a multi-layer AMBA bus.

• Proceedings of the IEEK Conference
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• 1998.10a
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• pp.143-146
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• 1998

A new stereoscopic codec. structure using MPEG-2 multiview profile is presented in this paper. In the suggested codec., the left image is coded with motion estimation in the base layerand the right image is coded with disparity estimation in the enhancement layer. Since it is possible to calculate rough motion of the right image sequence with disparity and motion of the left image sequence, motion compensation of the enhancement layer is performed without motion estimation. Since the proposed codec. does not perform motion estimation in the enhancement layer encoding, it is simple and reduces the encoding time. We compared the PSNR of encoded image with three different structured codec., and the experimental results show that suggested codec. has comparable with other codecs.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.33B no.4
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• pp.156-163
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• 1996

In this paper, we described the implementation and the sound quality assessment of a real-time stereo audio codec using TMS320C40 DSP (digital signal processing) chip for low bitrte and high quality audio. We implemented hardware and software in order to overcome a real-time processing problem of audio compression algorithm that can be produced by largely recursive computing and complexity of the process. We have studied five types of distortion that can be produced by perceptual coding and the codec was evaluated by eight test musics that are selected in SQAM (sound quality assessment material) 422-2-4-2 produced by EBU (european broadcast union). The subjective listening tests were carried out on the codec quality and preformance by double blind method in a listening room with eleven listeners. As a result, 5 grade-impairment scale was scored under minus one and the codec quality was evaluated to be perceptible, but not annoying.

• International Journal of Precision Engineering and Manufacturing
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• v.8 no.1
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• pp.38-45
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• 2007

We propose a preprocessing method for removing floating particles from underwater images based on an analysis of the image features. We compared baseline JPEG and wavelet codec methods to determine the method best suited for underwater images. The proposed preprocessing method enhanced the compression ratio and resolution, and provided an efficient means of compressing the images. The wavelet codec method yielded better compression ratios and image resolutions. The results suggest that the wavelet codec method linked with the proposed preprocess method provides an efficient codec processor and transmission system for underwater images that are used for searches and transmitted via ultrasonic communications.

• MALSORI
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• no.65
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• pp.143-152
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• 2008

This paper focuses on the real time implementation of an adaptive multi-rate (AMR) speech codec, that is a standard speech codec of IMT-2000, using the TMS320VC5510. The series of TMS320VC55x is a 16-bit fixed-point digital signal processor (DSP) having low power consumption for the use of mobile communications by Texas Instruments (TI) corporation. After we analyze the AMR algorithm and source code as well as the structure and I/O of 7MS320VC55x, we carry out optimizing the programs for real time implementation. The implemented AMR speech codec uses 55.2 kbyte for the program memory and 98.3 kbyte for the data memory, and it requires 709,878 clocks, i.e. about 3.5 ms, for processing a frame of 20 ms speech signal.

• Proceedings of the KOSOMBE Conference
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• v.1994 no.12
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• pp.92-95
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• 1994

As computer systems to make medical practice easy are widely used, a special hardware system processing medical data fast becomes more important. To meet the urgent demand for high speed image processing, especially image compression and decompression, we designed and implemented the medical image CODEC (COder/BECoder) based on MISD(Multiple Instruction Single Data stream) architecture to adopt parallelism in it. Considering not being a standart scheme of medical mage compression/decompress ion, the CODEC is designed programable and general. In this paper, we use JPEG (Joint Photographic Experts Group) algorithm to process images fast and evalutate it.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2007.06a
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• pp.510-513
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• 2007

가변길이 부호는 많은 이미지 및 영상 표준에서 폭넓게 사용되는 기법이다. 특히 국제 표준인 JVT와 중국 A/V 표준인 AVS는 엔트로피 코딩을 수행하기 위해 Exp-Golomb 코드에 기반한 UVLC(Universal Variable Length Code)를 채용하고 있다. 본 논문에서는 H.264/AVC의 엔트로피 코딩에서 사용되는 Exp-Golomb CODEC의 하드웨어 구현에 대해 연구하였다. 식의 간략화로 구현하기 어려운 log함수와 거듭제곱 연산을 하지 않으며, 첫 번째 1 검출기와 누산기 제어에 의한 배럴 쉬프터를 통하여 별도의 시간 지연 없이 부호화 및 복호화가 되도록 설계하였다. Xilinx ISE툴을 사용하여 합성하고, 보드 수준에서 PCI인터페이스를 사용하여 검증하였다. 본 논문에서 설계된 Exp-Glomb CODEC은 H.264/AVC 및 AVS와 같은 분야에서 응용이 가능할 것으로 예견된다.

• Speech Sciences
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• v.3
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• pp.177-186
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• 1998

This paper describes the procedure of implementing a real-time speech codec, G.723.1 which was developed by DSP Group and standardized by ITU-T, using fixed-point DSP, ADSP-2181. This codec has two bit rates associated with it, 5.3 and 6.3 kbit/s. We implemented only one bit rate, 6.3 kbit/s, of the two with fixed-point 32-bit precision. According to the result of the experiment, the amount of computational burden is about 55 MIPS and its quality is similar to the result of the PC simulation with floating-point arithmetic. In this paper, we proposed a method to use a fixed-point DSP and a procedure for developing a real-time speech codec using DSPs and finally developed a G.723.l speech codec for ADSP-2181.

• 한국HCI학회:학술대회논문집
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• 2008.02a
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• pp.230-235
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• 2008

This paper addresses issues associated with the real-time implementation of a wideband speech codec such as ITU-T G. 729. 1 on an ARM processor in order to provide an improved voice quality of a VoWiFi service. The real-time implementation features in optimizing the C-source code of G.729. 1 and replacing several parts of the codec algorithm with faster ones. The performance of the implementation is measured by the CPU time spent for G.729.1 on the ARM926EJ processor that is used for a VoWiFi phone. It is shown from the experiments that the G.729.1 codec works in real-time with better voice quality than G 729 codec that is conventionally used for VoIP or VoWiFi phones.

• The Journal of the Acoustical Society of Korea
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• v.16 no.1E
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• pp.41-48
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• 1997

The very purpose of subband codec is the attainment of data rate compression through the use of quantizer and optimum bit allocation for each decimated signal. Yet the question of floating-point quantization effects in subband codec has received scant attention. There has been no direct focus on the analysis of quantization errors, nor on design with quantization errors embedded explicitly in the criterion. This paper provides a rigorous theory for the modelling, analysis and optimum design of the general M-band subband codec in the presence of the floating-point quantization noise. The floating-point quantizers are embedded into the codec structure by its equivalent multiplicative noise model. We then decompose the analysis and synthesis subband filter banks of the codec into the polyphase form and construct an equivalent time-invariant structure to compute exact expression for the mean square quantization error in the reconstructed an equivalent time-invariant structure to compute exact expression for the mean square quantization error in the reconstructed output. The optimum design criteria of the subband codec is given to the design of the analysis/synthesis filter bank and the floating-point quantizer to minimize the output mean square error. Specific optimum design examples are developed with two types of filter of filter banks-orthonormal and biorthogonal filter bank, along with their perpormance analysis.