• 제어로봇시스템학회:학술대회논문집
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• 2005.06a
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• pp.1435-1438
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• 2005

In hands-free telephone systems, the received speech signal is fed back to the microphone and constitutes the so-called echo. To cancel the effect of this time-varying echo path, it is necessary to device an adaptive filter between the receiving and the transmitting ends. For a typical FIR realization, the length of the fullband adaptive filter results in high computational complexity and low convergence rate. Consequently, subband adaptive filtering schemes have been proposed to improve the performance. In this work, we use deterministic approach to analyze the relationship between fullband and subband adaptive filtering structures. With block adaptive filtering structure as an intermediate stage, the analysis is divided into two parts. First, to avoid aliasing, it is found that the matrix of block adaptive filters is in the form of pseudocirculant, and the elements of this matrix are the polyphase components of the fullband adaptive filter. Second, to transmit the near-end voice signal faithfully, the analysis and the synthesis filter banks in the subband adaptive filtering structure must form a perfect reconstruction pair. Using polyphase representation, the relationship between the block and the subband adaptive filters is derived.

• MALSORI
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• no.67
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• pp.103-120
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• 2008

Basic components for multimodal interface, such as speech recognition, speech synthesis, gesture recognition, and multimodal fusion, have their own technological limitations. For example, the accuracy of speech recognition decreases for large vocabulary and in noisy environments. In spite of those technological limitations, there are lots of applications in which speech-oriented multimodal user interfaces are very helpful to users. However, in order to expand application areas for speech-oriented multimodal interfaces, we have to develop the interfaces focused on usability. In this paper, we introduce usability and user-centered design methodology in general. There has been much work for evaluating spoken dialogue systems. We give a summary for PARADISE (PARAdigm for Dialogue System Evaluation) and PROMISE (PROcedure for Multimodal Interactive System Evaluation) that are the generalized evaluation frameworks for voice and multimodal user interfaces. Then, we present usability components for speech-oriented multimodal user interfaces and usability testing guidelines that can be used in a user-centered multimodal interface design process.

• Annual Conference on Human and Language Technology
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• 2009.10a
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• pp.266-270
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• 2009

IT 관련 산업의 발전에 의한 저변의 확대로 장애우들의 IT 사용 수요가 늘고 있다. 본 논문에서는 IT분야에서 가장 기초적으로 활용되는 E-Mail을 시각 장애우가 활용 하는 데에 불편함이 없도록 하는 이메일 매니저 개발에 관련된 기법에 대해 논하고자 한다. TTS(Text-To Speech : 문자 텍스트를 음성으로 전환하여 들려줌)와 음성키보드(키보드 입력 시 입력한 문자를 음성으로 알려줌) 기능으로 시각 장애우가 이메일을 사용함에 있어 불편함을 느끼지 않도록 하였으며 본 시스템의 TTS 알고리즘은 국어 표준발음법을 참고로 하여 자바로 구현 하였다.

• Proceedings of the IEEK Conference
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• 2002.06d
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• pp.381-384
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• 2002

In speech signal processing, it Is very important to detect the pitch exactly in speech recognition, synthesis and analysis. but, it is very difficult to pitch detection from speech signal because of formant and transition amplitude affect. therefore, in this paper, we proposed a pitch detection using the spectrum flattening techniques. Spectrum flattening is to eliminate the formant and transition amplitude affect. In time domain, positive center clipping is process in order to emphasize pitch period with a glottal component of removed vocal tract characteristic. And rough formant envelope is computed through peak-fitting spectrum of original speech signal in frequency domain. As a results, well get the flattened harmonics waveform with the algebra difference between spectrum of original speech signal and smoothed formant envelope. After all, we obtain residual signal which is removed vocal tract element The performance was compared with LPC and Cepstrum, ACF 0wing to this algorithm, we have obtained the pitch information improved the accuracy of pitch detection and gross error rate is reduced in voice speech region and in transition region of changing the phoneme.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.29B no.1
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• pp.43-49
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• 1992

As a result of recent progress in speech synthesis techniques, the those new services using new techniques are going to introduce into the telephone communication system. In setting standards, voice quality is obviously an important criterion. It is very important to develope a quality evaluation method of synthesized speech for the diagnostic assessment of system algorithm, and fair comparison of assessment values. This paper has described several basic concepts and criterions for quality assessment (intelligibility) of synthesized speech by rule, and then a word selection method and the word list to be used in word intelligibility test were proposed. Finally, a test method for word intelligibility is described.

• Speech Sciences
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• v.10 no.1
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• pp.85-94
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• 2003

In this paper an algorithm to reduce the size of Text-to-Speech database is proposed. The algorithm is based on the characteristics of Korean phonemic units. From the initial database, a reduced phoneme unit set is induced by articulatory similarity of concatenating phonemes. Speech data is read by one female announcer for 1000 phonetically balanced sentences. All the recorded speech is then segmented by phoneticians. Total size of the original speech data is about 640 MB including laryngograph signal. To synthesize wave, RE-PSOLA (Residual-Excited Pitch Synchronous Overlap and Add Method) was used. The voice quality of synthesized speech was compared with original speech in terms of spectrographic informations and objective tests. The quality of the synthesized speech is not much degraded when the size of synthesis DB was reduced from 320 MB to 82 MB.

• Proceedings of the Korean Information Science Society Conference
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• 2000.10b
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• pp.278-280
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• 2000

인터넷의 중요성이 증가함에 따라, 웹 브라우저에 음성 인터페이스를 추가하는 연구와 개발이 이루어지고 있다. 그러나, 아직까지 기존의 모든 웹 문서가 HTML로 작성되어 있어, 효과적인 음성 인터페이스를 하기에는 많은 어려움이 있으며, 음성이 느린 출력 매체이므로 사용자가 빠르게 인지할 수 있는 방안이 연구되어야 한다. 본 논문에서는 사용자의 웹 액세스를 높이기 위하여, 웹 브라우저에 연결되는 웹 문서에서, 각각의 객체를 추출한 후, 사용자가 그 객체에 바로 액세스를 하거나 한국어 음성으로 그 정보를 알 수 있는 방법을 사용하여, 음성으로 제어할 수 있는 한국어 음성 웹 브라우저를 설계하고 구현하였다. 음성합성과 인식을 사용하여 브라우저를 제어하기 때문에, 노약자나 어린이 또는 시각장애인들이 쉽게 웹 서핑을 할 수 있도록 도와줄 수 있고, 또한 현재 사용되고 있는 웹 문서에서의 객체추출을 사용하기 때문에 특별히 문서의 변환이 필요 없는 장점이 있다.

• Annual Conference on Human and Language Technology
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• 2023.10a
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• pp.404-408
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• 2023

음성 사용자 인터페이스(Voice User Interface)에 대한 수요가 증가함에 따라 음성 합성(Speech Synthesis) 시스템에서 자연스러운 음성 발화를 모방하기 위해 적절한 위치에 휴지를 삽입하는 것이 주된 과업으로 자리잡았다. 대화의 연속성을 고려했을 때, 자연스러운 음성 기반 인터페이스를 구성하기 위해서는 대화의 맥락을 이해하고 적절한 위치에 휴지를 삽입하는 것이 필수적이다. 이에 따라 본 연구는 대화 맥락에 기반하여 적절한 위치에 휴지를 삽입하는 Long-Input Transformer 기반 휴지 예측 모델을 제안하고 한국어 대화 데이터셋에서 검증한 결과를 보인다.

• Journal of Digital Convergence
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• v.11 no.5
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• pp.285-290
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• 2013

In a MPC coding using excitation source of voiced and unvoiced, it would be a distortion of speech waveform. This is caused by normalization of synthesis speech waveform of voiced in the process of restoration the multi-pulses of representation section. To solve this problem, this paper present a method of position compensation(PC-MPC) in a multi-pulses each pitch interval in order to reduce distortion of speech waveform. I was confirmed that the method can be synthesized close to the original speech waveform. And I evaluate the MPC and PC-MPC using multi-pulses position compensation method. As a result, $SNR_{seg}$ of PC-MPC was improved 0.4dB for female voice and 0.5dB for male voice respectively. Compared to the MPC, $SNR_{seg}$ of PC-MPC has been improved that I was able to control the distortion of the speech waveform finally. And so, I expect to be able to this method for cellular phone and smart phone using excitation source of low bit rate.

• Journal of Digital Convergence
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• v.12 no.6
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• pp.271-276
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• 2014

In a speech coding system using excitation source of voiced and unvoiced, it would be involved a distortion of speech quality in case coexist with a voiced and unvoiced consonants in a frame. In this paper, I propose a method of 8kbps Multi-Pulse Speech Coding(FBD-MPC: Frequency Band Division MPC) by using TSIUVC(Transition Segment Including Unvoiced Consonant) searching, extraction and approximation-synthesis method in a frequency domain. I evaluate the 8kbps MPC and FBD-MPC. As a result, SNRseg of FBD-MPC was improved 0.5dB for female voice and 0.2dB for male voice respectively. Compared to the MPC, SNRseg of FBD-MPC has been improved that I was able to control the distortion of the speech waveform finally. And so, I expect to be able to this method for cellular phone and smart phone using excitation source of low bit rate.