• Journal of Industrial Technology
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• v.27 no.A
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• pp.211-218
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• 2007

As demands for interaction of humans and robots are increasing, robots are expected to be equipped with intelligibility which humans have. Especially, for natural communication, hearing capabilities are so essential that speech recognition technology for robot is getting more important. In this paper, we implement a speech recognizer suitable for robot applications. One of the major problem in robot speech recognition is poor speech quality captured when a speaker talks distant from the microphone a robot is mounted with. To cope with this problem, we used wireless transmission of commands recognized by the speech recognizer implemented using TMS320VC5510 DSK. In addition, as for implementation, since TMS320VC5510 DSP is a fixed-point device, we represent efficient realization of HMM algorithm using fixed-point arithmetic.

• Proceedings of the KSPS conference
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• 2006.11a
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• pp.19-23
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• 2006

This paper is for the comparison of prosodic phrasing in Korean spontaneous speech and read speech. For this comparison, The subjects read the transcriptions from their own spontaneous speech. The number of IP in spontaneous speech is more than in read speech, while The number of AP has no difference between them. A accentual phrase in spontaneous speech has less syllable than read speech.

• The Magazine of the IEIE
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• v.24 no.9
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• pp.91-98
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• 1997

Text-to-Speech(WS) technology has been attracting a lot of interest among speech engineers because of its own benefits. Namely, the possible application areas of talking computers, emergency alarming systems in speech, speech output devices for speech-impaired, and so on. Hence, many researchers have made significant progresses in the speech synthesis techniques in the sense of their own languages and as a result, the quality of current speech synthesizers are believed to be acceptable to normal users. These are partly why the MPEG group had decided to include the WS technology as one of its MPEG-4 functionalities. ETRI has made major contributions to the current MPEG-4 775 appearing in various MPEG-4 documents with relatively minor contributions from AT&T and NW. Main MPEG-4 functionalities presently available are; 1) use of original prosody for synthesized speech output, 2) trick mode functions for general users without breaking synthesized speech prosody, 3) interoperability with Facial Animation(FA) tools, and 4) dubbing a moving/anlmated picture with lip-shape pattern informations.

• Annual Conference on Human and Language Technology
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• 2003.10d
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• pp.279-283
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• 2003

ETRI가 참여하고 있는 자동 통역 관련 국제 컨소시엄인 C-STAR에서는 여행자 영역의 대규모 다국어 병렬 말뭉치를 공동으로 구축하였고, 현재 각 기관에서는 이를 이용한 대화체 음성 언어 번역 시스템을 개발 중이다. ETRI에서는 핵심어 처리, 통계정보를 이용하는 구 단위 자동 설정, 설정된 구의 자동대응 및 재배치 등을 특징으로 하는 구 기반 직접 번역 방식(PDMT: Phrase-based Direct Machine Translation)의 번역 방법론을 제안하고 관련 연구를 진행하고 있다. 본 논문에서는 ETRI 대화체 음성 언어 번역 시스템의 구성에 대해 알아보고 PDMT 번역 방법론의 등장 배경과 그 구체적인 번역 방법 및 특징에 대해 자세히 논의하기로 한다.

• Phonetics and Speech Sciences
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• v.3 no.4
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• pp.85-91
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• 2011

Sasang constitution medicine utilizes voice characteristics to diagnose a person's constitution. To classify Sasang constitutional groups using speech information technology, this study aims at establishing the relationship between Sasang constitutional groups and their corresponding voice characteristics by investigating various speech feature variables. The speech variables include features related to speech source and vocal tract filter. Experimental results show that statistically significant correlation between voice characteristics and some speech feature variables is observed.

• Proceedings of the IEEK Conference
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• 2007.07a
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• pp.303-304
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• 2007

In this paper, we implement an automatic distance speech recognition system for voiced-enabled services. We first construct a baseline automatic speech recognition (ASR) system, where acoustic models are trained from speech utterances spoken by using a cross-talking microphone. In order to improve the performance of the baseline ASR using distance speech, the acoustic models are adapted to adjust the spectral characteristics of speech according to different microphones and the environmental mismatches between cross-talking and distance speech. Next we develop a voice activity detection algorithm for distance speech. We compare the performance of the base-line system and the developed ASR system on a task of PBW (Phonetically Balanced Word) 452. As a result it is shown that the developed ASR system provides the average word error rate (WER) reduction of 30.6 % compared to the baseline ASR system.

• The Journal of the Korea institute of electronic communication sciences
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• v.16 no.3
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• pp.465-470
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• 2021

A robust speech recognition technology is required that does not degrade the performance of speech recognition and the quality of the speech when speech recognition is performed in an actual environment of the speech mixed with noise. With the development of such speech recognition technology, it is necessary to develop an application that achieves stable and high speech recognition rate even in a noisy environment similar to the human speech spectrum. Therefore, this paper proposes a speech enhancement algorithm that processes a noise suppression based on the MMSA-STSA estimation algorithm, which is a short-time spectral amplitude method based on the error of the least mean square. This algorithm is an effective nonlinear speech enhancement algorithm based on a single channel input and has high noise suppression performance. Moreover this algorithm is a technique that reduces the amount of distortion of the speech based on the statistical model of the speech. In this experiment, in order to verify the effectiveness of the MMSA-STSA estimation algorithm, the effectiveness of the proposed algorithm is verified by comparing the input speech waveform and the output speech waveform.

• The Journal of the Acoustical Society of Korea
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• v.15 no.4
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• pp.24-31
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• 1996

In this paper, we present a speech translation system for hotel reservation, KT_STS(Korea Telecom Speech Translation System). KT-STS is a speech-to-speech translation system which translates a spoken utterance in Korean into one in Japanese. The system has been designed around the task of hotel reservation(dialogues between a Korean customer and a hotel reservation de나 in Japan). It consists of a Korean speech recognition system, a Korean-to-Japanese machine translation system and a korean speech synthesis system. The Korean speech recognition system is an HMM(Hidden Markov model)-based speaker-independent, continuous speech recognizer which can recognize about 300 word vocabularies. Bigram language model is used as a forward language model and dependency grammar is used for a backward language model. For machine translation, we use dependency grammar and direct transfer method. And Korean speech synthesizer uses the demiphones as a synthesis unit and the method of periodic waveform analysis and reallocation. KT-STS runs in nearly real time on the SPARC20 workstation with one TMS320C30 DSP board. We have achieved the word recognition rate of 94. 68% and the sentence recognition rate of 82.42% after the speech recognition tests. On Korean-to-Japanese translation tests, we achieved translation success rate of 100%. We had an international joint experiment in which our system was connected with another system developed by KDD in Japan using the leased line.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.41 no.10
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• pp.1183-1185
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• 2016

Low bit-rate speech coder in digital speech communications synthesizes speech using vocal tract model parameters. In this case, the spectra of the synthesized speech can be much distorted since the allocated bits for the parameters are considerably limited, which results in the degradation of speech intelligibility. In this paper, we propose a speech intelligibility improvement method using stochastic spectral equalizer. This method stochastically obtains the weight vector of each speech coder using spectral ratios between original and synthesized speech, then applies this weight vector to synthesized speech. From the experiments of objective speech intelligibility tests, we found that the performance of the proposed method is better than that of the conventional method.

• Proceedings of the KSPS conference
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• 2006.05a
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• pp.95-98
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• 2006

A prompter software is used, behind the camera, to scroll the script for a TV narrator. So far it has been manually operated by an assistant, who scrolls the caption following narrator's speech. Automating this procedure using a speech recognition technology has been investigated in this project. The developed auto-scrolling software was tested in offline and online, which shows performance good enough to replace an existing prompter software. This paper describes the whole development process and concerns to be cared.