• 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.

• Journal of the Korea Society of Computer and Information
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• v.15 no.7
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• pp.91-98
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• 2010

Large corpus-based concatenating Text-to-Speech (TTS) systems can generate natural synthetic speech without additional signal processing. Because the improvements in the natualness, personality, speaking style, emotions of synthetic speech need the increase of the size of speech DB, it is necessary to prune the redundant speech segments in a large speech segment DB. In this paper, we propose a new method to construct a segmental speech DB for the Korean TTS system based on a clustering algorithm to downsize the segmental speech DB. For the performance test, the synthetic speech was generated using the Korean TTS system which consists of the language processing module, prosody processing module, segment selection module, speech concatenation module, and segmental speech DB. And MOS test was executed with the a set of synthetic speech generated with 4 different segmental speech DBs. We constructed 4 different segmental speech DB by combining CM1(or CM2) tree clustering method and full DB (or reduced DB). Experimental results show that the proposed method can reduce the size of speech DB by 23% and get high MOS in the perception test. Therefore the proposed method can be applied to make a small sized TTS.

• The Journal of the Acoustical Society of Korea
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• v.10 no.5
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• pp.11-19
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• 1991

음성 신호의 합성기법들 중에서 파형코딩법은 음질이 우수하기 때문에 분석에 의한 합성법으로 많이 사용하고 있다. 그렇지만 음원과 성도의특성을 분리하지 않고 파형의 잉여분만을 제거한 후에 파 형자체를 저장하기 때문에 규칙에 의한 합성기법으로 사용하기에는 어려움이 많다. 본 논문은 파형코딩 법 중 선형 PCM 코딩법으로 저장된 음성파형에 대해 피치를 양분할 수 있는 주기반분법을 제안하여 파형자체의 음원을 분리하지 않고 피치 주기를 변경시킬 수 있는 새로운 피치 변경법을 제안하였다. 따 라서 음질이 우수한 파형코딩 합성법으로 규칙에 의한 합성을 수행할 수 있다.

• The Transactions of the Korea Information Processing Society
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• v.4 no.1
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• pp.149-160
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• 1997

This paper is concerned with the problem of speaker-adaptive speech synthes is method using a mapped codebook designed by fuzzy mapping on FLVQ (Fuzzy Learning Vector Quantization). The FLVQ is used to design both input and reference speaker's codebook. This algorithm is incorporated fuzzy membership function into the LVQ(learning vector quantization) networks. Unlike the LVQ algorithm, this algorithm minimizes the network output errors which are the differences of clas s membership target and actual membership values, and results to minimize the distances between training patterns and competing neurons. Speaker Adaptation in speech synthesis is performed as follow;input speaker's codebook is mapped a reference speaker's codebook in fuzzy concepts. The Fuzzy VQ mapping replaces a codevector preserving its fuzzy membership function. The codevector correspondence histogram is obtained by accumulating the vector correspondence along the DTW optimal path. We use the Fuzzy VQ mapping to design a mapped codebook. The mapped codebook is defined as a linear combination of reference speaker's vectors using each fuzzy histogram as a weighting function with membership values. In adaptive-speech synthesis stage, input speech is fuzzy vector-quantized by the mapped codcbook, and then FCM arithmetic is used to synthesize speech adapted to input speaker. The speaker adaption experiments are carried out using speech of males in their thirties as input speaker's speech, and a female in her twenties as reference speaker's speech. Speeches used in experiments are sentences /anyoung hasim nika/ and /good morning/. As a results of experiments, we obtained a synthesized speech adapted to input speaker.

• The Journal of the Acoustical Society of Korea
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• v.38 no.5
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• pp.614-620
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• 2019

In this paper, we propose a method to effectively determine the representative style embedding of each emotion class to improve the global style token-based end-to-end speech synthesis system. The emotion expressiveness of conventional approach was limited because it utilized only one style representative per each emotion. We overcome the problem by extracting multiple number of representatives per each emotion using a k-means clustering algorithm. Through the results of listening tests, it is proved that the proposed method clearly express each emotion while distinguishing one emotion from others.

• MALSORI
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• no.43
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• pp.89-99
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• 2002

Recently, researches into ways of improving large vocabulary continuous speech recognition and speech synthesis are being carried out intensively as the field of speech information technology is progressing rapidly. In the field of speech recognition, developments of stochastic methods such as HMM require large amount of speech data for training, and also in the field of speech synthesis, recent practices show that synthesis of better quality can be produced by selecting and connecting only the variable size of speech data from the large amount of speech data. In this paper we design and discuss linguistic contents for speech copora for speech recognition and synthesis to be shared in common.

• Proceedings of the KSLP Conference
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• 2003.11a
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• pp.236-236
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• 2003

음성합성을 위한 운율연구는 실험음성학과 음성공학 분야에서 활발히 이루어져 왔고, 가시적인 성과도 거두어 왔다. 최근 운율의 자연성을 향상시키기 위한 노력은 corpus-based unit selection에서와 같이, 대단위 음성자료에 의존하게 되면서 급격한 자연성의 향상을 가져온 것이 사실이지만, 여전히 새로 주어진 자료에 대해서 부자연스러운 운율을 극복해야 한다는 과제를 안고 있다. 이 논문은 길이, 억양, 에너지 등의 운율요소 가운데, 우선적으로 분절음의 길이에 초점을 두고, 대규모 음성자료를 바탕으로 한 자료기반형 길이예측모델을 지양하고 언어학적 지식과 언어자질을 이용한 지식기반형 길이예측모델을 제시하고자 한다. 모델링을 위한 방법으로는, 기본적으로 자료기반형 모델링 기법이지만 언어자질을 이용한 길이예측이 가능한 Classification and Regression Tree (CART) 모델링과 전통적인 지식기반형 모델링인 sequential rule systems을 확대시킨 Sums-of-Products 모델링을 사용한다. (중략)

• 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.

• The Journal of the Acoustical Society of Korea
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• v.20 no.3
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• pp.15-23
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• 2001

This paper presents a voice conversion technique that modifies the utterance of a source speaker as if it were spoken by a target speaker. Feature parameter conversion methods to perform the transformation of vocal tract and prosodic characteristics between the source and target speakers are described. The transformation of vocal tract characteristics is achieved by modifying the LPC cepstral coefficients using Linear Multivariate Regression (LMR). Prosodic transformation is done by changing the average pitch period between speakers, and it is applied to the residual signal using the LP-PSOLA scheme. Experimental results show that transformed speech by LMR and LP-PSOLA synthesis method contains much characteristics of the target speaker.

• Proceedings of the KSPS conference
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• 2000.03a
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• pp.1-16
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• 2000