• Speech Sciences
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• v.7 no.2
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• pp.225-234
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• 2000

This paper describes the fixed-point DSP implementation of a CELP(Code-excited linear prediction)-based speech coder. The effective realization methodologies to maximize the utilization of the DSP's architectural features, specifically parallel movement and pipelining are also presented together with the implementation results targeted for the ITU-T standard G.723.1 using Motorola DSP56362. The operation of the implemented speech coder is verified using the test vectors offered by the standard as well as using the peripheral interface circuits designed for the coder's real-time operation.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.21 no.5
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• pp.1096-1108
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• 1996

In this paper, we propose a temporal adaptive tranform 3D SBC coder with motion compensation, exploiting redundancy in the temporal domain. We propose a temporal adaptivity measure, by which the R-D optimal temporal transform can be chaosen. The base temporal subband frame is coded using H.261-like MC-DCT coder, while the higher temporal subband frames are coded using the 2D adaptive wavelet packet bases, considering the various energy distribution which results from the temporal variation. In encoding the subbands, we employ adaptive scanning methods, uniform step-size quantization with VLC, and coded/not-coded flag reduction technique using the quadtree structure. From the simulation results, the proposed adaptive 3D subband coder shows about 0.29~3.14 dB gain over the H.261 and the fixed 3D subband coder techniques.

• Proceedings of the KSME Conference
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• 2000.11a
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• pp.693-698
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• 2000

Aspects of classical information theory, such as rate distortion theory, investigate how to encode and decode information from an independently identically distributed source so that the asymptotic distortion rate between the source and its quantized representation is minimized. However, in most natural dynamics, the source state is highly corrupted by disturbances, and the measurement contains the noise. In recent coder-estimator sequence is developed for state estimation problem based on observations transmitted with finite communication capacity constraints. Unlike classical estimation problems where the observation is a continuous process corrupted by additive noises, the condition is that the observations must be coded and transmitted over a digital communication channel with finite capacity. However, coder-estimator sequence does not provide such a quantitative analysis as a variance for estimation error. In this paper, under the assumption that the estimation error is Gaussian distribution, a variance for coder-estimation sequence is proposed and its fitness is evaluated through simulations with a simple example.

• Proceedings of the IEEK Conference
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• 2001.06d
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• pp.115-118
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• 2001

This paper presents dispersed-pulse and random codebook for CELP coder. This coder operates on speech frames of 20ms and generates an excitation vector by convoluting dispersion vectors with signed pulses in an algebraic codevector. The improvement of pulse-based fixed codebook is performed at a low bit rate. A high performance fixed-codebook consists of a partial algebraic codebook and a random codebook in unvoiced and stationary noise regions. The proposed CELP coder is quantized with 4kb/s and is compared with G.729 (Bkb/s CS-ACELP). Subjective testing shows better quality than reference coders under some background noise conditions

• Proceedings of the Korea Information Processing Society Conference
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• 2002.11a
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• pp.167-170
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• 2002

본 고의 Multi-Media Caching System 은 이동 단말기의 보급 & 발달로 예상되는 Trans-coder 부하의 증가와 응답 시간 개선 하기 위한 것으로 기존의 Web 문서의 Caching System 과 비교 분석 하고, 빈번한 이동 단말기의 요구에 의해서 발생하는 않은 trans-coder 작업을 최소화 할 수 있는 XML 기반의 Caching System을 연구 하고자 한다. 즉, 기존의 Web Caching System의 장점과 XML의 재 사용 성, 확장 성, 플랫폼 독립성, 다양한 종류의 응용 프로그램과의 접목성 등의 장점을 살리고, Cache System Manager에 의해 최소환의 Trans-coder 동작이 일어나도록 한다. 따라서 Trans-coder 된 작업을 Caching System 에 보관 관리 하는 최적의 Caching System Manager 방법을 제안하고자 한다.

• Journal of information and communication convergence engineering
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• v.2 no.3
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• pp.139-144
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• 2004

In order to transmit various types of multimedia data (i.e. voice, video, and data) over a wireless channel, the coding and modulation scheme needs to be flexible and capable of providing a variable quality of service, data rates, and latency. In this paper, we study a mobile multimedia COMA network combined with the concatenated Reed-Solomon/Rate Compatible Punctured Convolution code (RS/RCPC). Also, this paper propose the combination of concatenated RS/RCPC coder and COMA RAKE receiver for multimedia COMA traffic which can be sent over wireless channels. From the results, using a frequency selective Rayleigh fading channel model, it is shown that concatenated RS/RCPC coder at the wireless physical layer can be effective in providing reliable wireless multimedia CDMA network. And the proposed scheme that combine concatenated RS/RCPC coder and CDMA RAKE receiver provides a significant gain in the BER performance over multi-user interference and multipath frequency selective fading channels.

• The Journal of the Acoustical Society of Korea
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• v.20 no.7
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• pp.63-68
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• 2001

This paper describes the details of ITU-T SGIS G.729A speech coder implementation using ARM9 Thumb/sup R/ processor core and various techniques used in the optimization process. ITU-T G.729 speech coder is the standard of the toll quality 8 kbit/s speech coding. The input to the speech encoder is assumed to be a 16 bits PCM signal at a sampling rate of 8000 samples per second. G.729A is reduced complexity version of the G.729 coder. This version is bit stream interoperable with the full version. The implemented coder requires 34.8 MIPS for the encoder and 8.1 MIPS for the decoder, 36.5 kBytes of program ROM and 6.3 kBytes of data RAM, respectively. The implemented coder is tested against the set of 9 test vectors provided by ITU-T for bit exact implementation.

• The Journal of the Acoustical Society of Korea
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• v.15 no.6
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• pp.59-65
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• 1996

This paper proposes a 4 kbps PSI-VSELP(Pitch Synchronous Innovation-Vector Sum Excited Linear Prediction) speech coder which produces speech equivalent to that of the conventional 4.8 kbps VSELP. Since the 'half-rate' is differently defined from country to country, there may be a need to reduce the bit rate of conventional half-rate coder. To minimize the degradation of speech quality caused by bit-rate reduction, it is desirable to perform bit-allocation based on the carefull consideration of the effect of various transmission parameters. This paper adopts this analytical approach for bit-allocation at 4 kbps. To improve the quality of the VSELP coder at 4 kbps, basis vectors which play the most important role in the performance, are optimized by an iterative closed-loop training process and the PSI technique is employed in the VSELP performance, are optimized by an iterative closed-loop training process and the PSI technique is employed in the VSELP coder. To demonstrate the performance of the proposed speech coder, we peformed experiments under the noiseless and error free conditions. From experimental results, even though the proposed 4 kbps PSI-VSELP coder showed lower scores in the objective measure, higher scores in subjective measure was obtained compared with those of the conventional 4.8 kbps VSELp.

• The Journal of the Acoustical Society of Korea
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• v.20 no.3
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• pp.34-41
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• 2001

In an IMBE (Improved Multi-band Excitation) speech coder, initial pitch estimation occupies most of the total computing time for the coder due to complex cost function and exhaustive search over candidate pitches. Future frames in initial pitch estimation cause inevitable time delay. Therefore, it is difficult to implement a real-time coder. Furthermore, unvoiced frames use the unnecessary pitch estimation as in the voiced frames. In this paper, each frame is determined voiced or unvoiced by Dyadic Wavelet Transform (DyWT) and, then, initial pitch estimation is performed only for voiced frame. Therefore different pitch estimation algorithms are employed between voiced and unvoiced frames incurring reduced time delay at transmitter and receiver. Simulation result show that the relative complexity of initial pitch estimation is reduced by 23％, and the processing time decreases down to 1/10 ∼ 1/1l of the IMBE coder while speech quality is almost maintained.

• 제어로봇시스템학회:학술대회논문집
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• 1989.10a
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• pp.984-986
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• 1989

101 coding scheme, one of sliding block coding techniques, provides practically attractive features in some compression applications for image sources such as facsimile. This paper presents a new simple model of 101 coder. The results show that the entropy of the output of the 101 coder can be reduced close to the rate distortion bound of a binary first-order markov source.