• The Journal of Korean Institute of Communications and Information Sciences
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• v.16 no.12
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• pp.1382-1397
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• 1991

The QM-Coder based on the recommandation of the JBIG committee has been studied for applying it to G3 FAX. In the point of view of implementation. the QM-Coder is modified to have following properties : 1) initialized at each start of the new line for protecting the error propagation, 2) pixel context for error estimation, fill bit insertion to meet minimum scan line time of 10 msec. and it is found that the modified QM-Coder is useful for encoding half tone images for low end G3 FAX machine, For higher compression ration than QM-Coder in encoding of the image binarized by ordered orther thchniques, a rearranging process before applying the QM-Coder is used and the resultant compression ratic was increased 1.13∼1.31 times than that of the conventional QM-Coder.

• The Journal of the Acoustical Society of Korea
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• v.21 no.5
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• pp.462-470
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• 2002

Wideband speech, characterized by a bandwidth of 50∼7000 ㎐, sounds more natural and intelligible, and is less tiring to listen to when compared to narrowband speech characterized by a bandwidth of 300∼3400 ㎐. Wideband speech coders, however, have not been as successful as the narrowband speech coders because of their higher bit rate. In this paper, we propose a new wideband speech coder which combines the European standard of a narrowband speech coder, i.e., GSM-EFR, and a transform coder using the discrete wavelet transform. The proposed wideband speech coder operates as follows input speech is first split into two subbands with equal bandwidth and the two subband signals are coded and decoded by each subband coder. A GSM-EFR is adopted as a lower subband coder and a subband coder with wavelet transformed speech is designed for a upper subband coder. The total bit rate of the proposed coder is 18.9kbps (12.2 kbps for lower band coder and 6.7 kbps for upper band coder), and informal listening test results have shown that the proposed coder has comparable speech quality to that of G.722 with 56 kbps.

• The Journal of the Acoustical Society of Korea
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• v.20 no.5
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• pp.35-47
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• 2001

This paper describes an efficient harmonic-CELP speech coder by taking advantages of harmonic and CELP coders into account. According to frame voicing decision, the proposed harmonic-CELP coder adopts the RP-VSELP coder as a fast CELP in case of an unvoiced frame, or an improved harmonic coder in case of a voiced frame. The proposed coder has main features as follows: simple pitch detection, fast harmonic estimation, variable dimension harmonic vector quantization, perceptual weighting reflecting frequency resolution, fast harmonic synthesis, naturalness control using band voicing, and multi-mode. These features make the proposed coder require very low complexity, compared with HVXC coder To demonstrate the performance of the proposed coder, a 2.4 kbps coder has been implemented and compared with reference coders. From results of informal listening tests, the proposed coder showed good quality while requiring low delay and complexity.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.1C
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• pp.35-44
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• 2004

In this paper, we design an efficient VLSI architecture of entropy coding algorithm in JPEG2000. Entropy coder is a context-based binary arithmetic encoder, and composed of a Context Extractor(CE) and an Arithmetic Coder(AC). We speed-up CE by skipping no-operation bits in coding passes, and AC is to be performed based on MQ coder. Because of using Qe value associated with each allowed context and probability estimation, MQ coder is a multiplication free coder that reduces computation loads and makes simple the structure of arithmetic coder. We have developed and synthesized the VHDL models of CE and AC pairs using Xilinx FPGA technology. The proposed architecture operates up to 30MHz.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.195-198
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• 2002

This paper presents an improved MELP Coder using Fourier magnitude compensation method chosen the new 2.4 kbit/s U.S. federal Standard. Although the MELP is quite good, it has some distortion for low-pitch male speakers. An improved MELP coder includes a post processing for the fourier magnitude model that allows the MELP to reconstruct the lower frequency spectrum more accurately and improve the speech quality. In this new compensation algorithm, the harmonic magnitudes in the low frequencies are adaptively modified by removing the effect of the two filters. Also, the bit rate of the improved MELP coder is the same as that of the Federal Standard MELP coder. formal quality tests show that the improved MELP coder was preferred over the Federal Standard MELP coder by $80.8\%$.

• The Journal of the Acoustical Society of Korea
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• v.18 no.2E
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• pp.24-30
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• 1999

In this paper, we propose a weighting dependent mixed source model (WD-MSM) coder that is an improved version of a CELP-based mixed source model (C-MSM) coder. The coder classifies speech segments into three types : voiced, unvoiced and mixed. The excitation for a voiced frame is an adaptive source, and the excitation for an unvoiced frame is a stochastic source. The coder has a modified mixed source for a mixed frame. We apply different weighting functions for three classes. Simulation results show that the proposed coder at 4 kbits/s yields very good performance both subjectively and objectively.

• Journal of Advanced Navigation Technology
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• v.19 no.6
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• pp.614-622
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• 2015

Binary MQ arithmetic coding is widely used recently as a basic entropy coder in multimedia coding system. MQ coder esteems high in compression efficiency to be used in JBIG2 and JPEG2000. The importance of arithmetic coding is increasing after it is adopted as an unique entropy coder in HEVC standard. In the binary MQ coder, arithmetic approximation without multiplication is used in the process of recursive subdivision of range interval. Because of the MPS/LPS exchange activity happened in MQ coder, output byte tends to increase. This paper proposes an enhanced binary MQ arithmetic coder to make use of a lookup table for AQe using quantization skill in order to reduce the deficiency. Experimental results show that about 4% improvement of compression in case of JBIG2 for bi-level image compression standard. And also, about 1% improvement of compression ratio is obtained in case of lossless JPEG2000 coding. For the lossy JPEG2000 coding, about 1% improvement of PSNR at the same compression ratio. Additionally, computational complexity is not increasing.

• The Journal of the Korea Contents Association
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• v.10 no.9
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• pp.88-96
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• 2010

We propose the efficient channel coding method combined with a 2.4kbps speech coder. The code rate of a channel coder is given by 1/2 and 1/2 rate convolutional coder is obtained from the punctured convolutional coder with rate of 1/3. The punctured convolutional coder is used for a variable rate allocation. The puncturing method according to the importance of the output data of the source encoder is applied for the convolutional coder. The importance of output data is analyzed by evaluating the bit error sensitivity of speech parameter bits. The performance of proposed coder is analyzed and simulated in Rayleigh fading channel and AWGN channel. The experimental results with 2.4kbps EHSX coder show that the variable rate channel coding method is superior to non-variable channel coding method from the subjective speech quality.

• Journal of information and communication convergence engineering
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• v.5 no.2
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• pp.177-180
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• 2007

We compared CS-ACELP with QCELP speech coder in CDMA cellular under channel error environment and experimented performance with its measured value under channel error environment. Also, we specified the effective coding scheme to overcome. CS-ACELP speech coder using a LSP vector quantizer shows transparent speech quality from the results that SD is 0.92dB and outlier frames over 2dB is 2.9% in the BER 0.10% condition. CS-ACELP speech coder which is utilizing MA predictor shows better results on SVR and SEGSNR than QCELP speech coder(IS-96) adopting DPCM type predictor when bit error occurs from BER 0.01% to 0.50%.

• Journal of Korea Multimedia Society
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• v.10 no.12
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• pp.1551-1558
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• 2007

The standard of evaluating function of speech coder for mobile telecommunication can be shown in channel capacity, noise immunity, encryption, complexity and encoding delay largely. This study is an algorithm to reduce complexity applying to CDMA(Code Division Multiple Access) mobile telecommunication system, which has a benefit of keeping the existing advantage of telecommunication quality and low transmission rate. This paper has an objective to reduce the computing complexity by controlling the frequency band nonuniform during the changing process of LSP(Line Spectrum Pairs) parameters from LPC(Line Predictive Coding) coefficients used for EVRC(Enhanced Variable-Rate Coder, IS-127) speech coders. Its experimental result showed that when comparing the speech coder applied by the proposed algorithm with the existing EVRC speech coder, it's decreased by 45% at average. Also, the values of LSP parameters, Synthetic speech signal and Spectrogram test result were obtained same as the existing method.