• The Journal of the Acoustical Society of Korea
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• v.27 no.8
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• pp.427-434
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• 2008

In this paper, a transcoding algorithm for SMV and AMR speech coder is proposed. In the application requiring the interoperability of different networks, two speech coders must work together with the structure of cascaded connection, tandem. The tandem which is one of the simplest methods has several problems such as long delay, high complexity and the quality degradation due to twice complete encoding/decoding process. These problems can be solved by using transcoding algorithm. The proposed algorithm consists of LSP (Line Spectral Pair) conversion, pitch delay conversion, and fast fixed codebook search. The evaluation results show that the proposed algorithm achieves equivalent speech quality to that of tandem with reduced computational complexity and delay.

• The Journal of the Acoustical Society of Korea
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• v.20 no.1
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• pp.43-49
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• 2001

EVRC (Enhanced Variable Rate Codec) has been adopted as a standard coder for the CDMA digital cellular system in North America and Korea, and known to provide good call quality at 8kbps. In this paper, fast implementation algorithms for EVRC encoder are proposed. The proposed algorithms are based on both efficient pitch detection scheme and fast fixed codebook search algorithm. In the codebook search, computational complexity is reduced down to 70％ of the original EVRC by limiting the number of pulse position combination and by using a truncated impulse response. The proposed algorithms enable us to implement the EVRC with much smaller computational works. Also, informal subjective tests confirmed that the difference in the speech quality between the original EVRC and the proposed method was indistinguishable.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.5C
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• pp.548-554
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• 2003

Interoperability is ole the most important factors for a successful integration of the speech network. To accomplish communication between endpoints employing different speech coders, decoder and encoder of each endpoint coder should be placed in tandem. However, tandem coder often produces problems such as poor speech quality, high computational load, and additional transmission delay. In this paper, we propose an efficient transcoding algorithm that can provide interoperability to the networks employing ITU-T G.723.1［1］and TIA IS-127 EVRC［2］speech coders. The proposed transcoding algorithm is composed of four parts: LSP conversion, open-loop pitch conversion, fast adaptive codebook search, and fast fixed codebook search. Subjective and objective quality evaluation confirmed that the speech quality produced by the proposed transcoding algorithm was equivalent to, or better than the tandem coding, while it had shorter processing delay and less computational complexity, which is certified implementing on TMS320C62x.

• The Journal of the Acoustical Society of Korea
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• v.21 no.7
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• pp.648-655
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• 2002

This paper describes implementation of a multi-channel G.723.1 Annex A (G.723.1A) focused on code optimization using a high performance general purpose Digital Signal Processor (DSP), To implement a multi-channel G.723.1A functional complexities of the ITU-T G.723.1A fixed-point C-code are measures an analyzed. Then we sort and optimize C functions in complexity order. In parallel with optimization, we verify the bit-exactness of the optimized code using the ITU-T test vectors. Using only internal memory, the optimized code can perform full-duplex 17 channel processing. In addition, we further increase the number of available channels per DSP into 22 using fast codebook search algorithms, referred to as bit -compatible optimization.