• IEMEK Journal of Embedded Systems and Applications
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• v.1 no.2
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• pp.31-36
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• 2006

As two dimensional bar code can collect data and information quickly, it is used and recognized as a useful tool for the many industrial application field. But the information capacity of two dimensional bar code is still limited. Recently, the two dimensional AD bar code (analog-digital code) that can increase its application range and overcome capacity limitation is developed. In this paper, we implement an effective system which can transform text information into voice using two dimensional AD bar code and TTS(Text To Speech). It can be transmitted to blind people by capturing the AD bar code on the papers or the books.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.30B no.5
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• pp.43-51
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• 1993

A new Hangul code called Truecode is developed for accomodating to the future computing environments of graphical user interface and multimedia as well as for corresponding with the invention principle of Hangul. Truecode is not a forced two-byte code of syllable unit, as completion-type of combination-type, currently used, but a one byte code of phoneme unit, which can represent initial consonant, vowel, and final consonant each. It is quite different from three-byte code of syllable unit and also does not require the fill code used for three-byte code. We expect great contribution to the Hangul culture from Truecode's some important following features. It can express all the Korean characters we may imagine and does not cause any problem in communication. As well as we may use direct connection font, we can assign ont-to-one correspondence between Truecode and a keyboard with three sets. Truecode has a good advantage in developing application softwares of Hangul and it can nicely be applied to the fields of speech recognition and artificial intelligence using natural language.

• The Journal of the Acoustical Society of Korea
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• v.21 no.4
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• pp.394-400
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• 2002

In digital speech communications, the quality of service can be increased by speech coding scheme that is adaptive to the error rate of voice packet transmission. However, current communication protocol in cellular and internet communications does not provide the function that transmits the channel error information. To solute this problem, in this paper, new method for real-time transmission of channel error information is proposed, where channel error information is embedded in voice packet. The proposed method utilizes the pulse positions of codevector in ACELP speech codec, which results in little degradation in speech quality and low false alarm rate. The simulations with various speech data show that the proposed method meets the requirement in speech quality, detection rate, and false alarm rate.

• Journal of Advanced Navigation Technology
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• v.11 no.1
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• pp.87-92
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• 2007

The conventional CELP(code excited linear prediction) type vocoder has no V/UV(voiced/unvoiced) classifier. So, the unvoiced speech is processed like the voiced speech. In this paper, to reduce the bit rate, we propose a new V/UV decision algorithm minimized error rate and preprocessing computation. This V/UV classifier use the LSP(line spectrum pair) parameter which is acquired spectrum analysis process in CELP vocoders. Applying this method to the 5.3kbps ACELP(algebraic code excited linear prediction) in the G.723.1, we can get the transmission bits rate reduction of 6% approximately without degradation of speech quality.

• Journal of the Korean Institute of Telematics and Electronics A
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• v.33A no.5
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• pp.172-179
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• 1996

In the paper, QCELP (qualcomm code excited linear predictive), CDMA (code division multiple access)'s vocoder algorithm, was analyzed. And then, a ptich parameter seaarch architecture for 16-bit programmable DSP(digital signal processor) for QCELP was designed. Because we speed up the parameter search through high speed DSP using two MACs, we can satisfy speech codec specifiction for the digital celluar. Also, we implemented in FIFO(first-in first-out) memory using register file to increase the access time of data. This DSP was designed using COMPASS, ASIC design tool, by top-down design methodology. Therefore, it is possible to cope with rapid change at mobile communication market.

• 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.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.52-55
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• 1999

다중 전송율(16, 24, 32, 40kbps)을 제공하는 G.726 부호화기는 ADPCM (Adaptive Differential Pulse Code Modulation) 부호화법을 사용한다. 본논문에서는 G.726 ADPCM 알고리즘을 C&S Technology에서 개발한 음성 신호 처리를 위한 범용 DSP인 CSD17C00A 칩을 이용하여 실시간 응용이 가능하도록 구현하였다. G.726에 대한 양방향 평가는 Codec Loopback test을 통해 수행되었으며, W-T에서 제공한 테스트 절차에 따라 평가되었다. 본 논문에서 구현된 G.726 부호화기는 평균 11 MIPS의 계산 속도를 갖고, 프로그램 메모리 크기는 2.8K Words이고, 데이터 메모리 크기는 550 Words 를 필요로 하였다.

• Proceedings of the Acoustical Society of Korea Conference
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• 1994.06c
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• pp.210-213
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• 1994

In this paper, we present a combination of trellis coded vector quantization and code-excited linear prediction coding, termed trellis excitation coding, for an efficient 4.8 kbps speech coding system. A training sequence-based algorithm is developed for designing an otimized codebook subject to the TEC structure. Also, we discuss the trellis symbol release rules that avoid excessive encoding delay. Finally, simulation results for the TEC coder are given at bit rate of 4.8 kbps.

• Proceedings of the IEEK Conference
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• 2003.07e
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• pp.2240-2243
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• 2003

In this paper. we present a new speech recognition system using DSP chip. DSP chip used TMS320c6711 of TI. We designed hardware system including acoustic model, word list and code book in flash memory. The word candidates are recognized based on CV, VCCV, and VC units HMM. This system can be applied to various electric ＆ electronic devices: home automation, robotics etc.

• Phonetics and Speech Sciences
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• v.1 no.3
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• pp.103-108
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• 2009

This paper proposes a method for building a transliteration dictionary, which is based on pronouncing information extracted from two kinds of existing dictionaries. Also, it proposes a method for transforming the pronouncing information into Korean translitered words. To express the pronouncing information, we define Phoman code system. In order to avoid phonetic estimation process of English words which is the most important problem, the proposed method uses the pronouncing information extracted from the existing dictionaries. Therefore, unlike previous approaches, the proposed method does not need any incomplete phonetic estimation process so that it can produce accurate transliteration results. The proposed method has been fully implemented.