• Proceedings of the Korean Institute of Intelligent Systems Conference
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• 1993.06a
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• pp.1321-1325
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• 1993

This paper investigates the use of Fuzzy vector quantizer(FVQ) in speech synthesis. To compress speech data, we employ K-means algorithm to design codebook and then FVQ technique is used to analysize input speech vectors based on the codebook in an analysis part. In FVQ synthesis part, analysis data vectors generated in FVQ analysis is used to synthesize the speech. We have fined that synthesized speech quality depends on Fuzziness values in FVQ, and the optimum fuzziness values maximized synthesized speech SQNR are related with variance values of input speech vectors. This approach is tested on a sentence, and we compare synthesized speech by a convensional VQ with synthesized speech by a FVQ with optimum Fuzziness values.

• Journal of Broadcast Engineering
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• v.18 no.6
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• pp.919-926
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• 2013

The acoustic signal from a distance sound source in an enclosed space often produces reverberant sound that varies depending on room impulse response. The estimation of the level of reverberation or the quality of the observed signal is important because it provides valuable information on the condition of system operating environment. It is also useful for designing a dereverberation system. This paper proposes a speech quality estimation method based on the harmonicity of received signal, a unique characteristic of voiced speech. At first, we show that the harmonic signal modeling to a reverberant signal is reasonable. Then, the ratio between the harmonically modeled signal and the estimated non-harmonic signal is used as a measure of standard room acoustical parameter, which is related to speech clarity. Experimental results show that the proposed method successfully estimates speech quality when the reverberation time varies from 0.2s to 1.0s. Finally, we confirm the superiority of the proposed method in both background noise and reverberant environments.

• The Journal of the Acoustical Society of Korea
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• v.29 no.3E
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• pp.131-139
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• 2010

This paper proposes a novel spectral hole substitution technique for low bit-rate audio coding. The spectral holes frequently occurring in relatively weak energy bands due to zero bit quantization result in severe quality degradation, especially for harmonic signals such as speech vowels. The enhanced aacPlus (EAAC) audio codec artificially adjusts the minimum signal-to-mask ratio (SMR) to reduce the number of spectral holes, but it still produces noisy sound. The proposed method selectively predicts the spectral shapes of hole bands using either intra-band correlation, i.e. harmonically related coefficients nearby or inter-band correlation, i.e. previous frames. For the bands that have low prediction gain, only the energy term is quantized and spectral shapes are replaced by pseudo random values in the decoding stage. To minimize perceptual distortion caused by spectral mismatching, the criterion of the just noticeable level difference (JNLD) and spectral similarity between original and predicted shapes are adopted for quantizing the energy term. Simulation results show that the proposed method implemented into the EAAC baseline coder significantly improves speech quality at low bit-rates while keeping equivalent quality for mixed and music contents.

• Speech Sciences
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• v.13 no.1
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• pp.95-105
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• 2006

The primary goal of therapeutic intervention in dysarthric speakers is to increase the speech intelligibility. Decision of critical features to increase the intelligibility is very important in speech therapy. The purpose of this study is to know the effects of pitch increasing training (PIT) on speech of a subject with Parkinson's disease (PD). The PIT program is focused on increasing pitch while a vowel is sustained with the same loudness. The loudness level is somewhat higher than that of the habitual loudness. A 67-year-old female with PD participated in the study. Speech therapy was conducted for 4 sessions (200 minutes) for one week. Before and after the treatment, acoustic, perceptual and speech naturalness evaluation was peformed for data analysis. Speech and voice satisfaction index (SVSI) was obtained after the treatment. Results showed Improvements in voice quality and speech naturalness. In addition, the patient's satisfaction ratings (SVSI) indicated a positive relationship between improved speech production and their (the patient and care-givers) satisfaction.

• Journal of Electrical Engineering and Technology
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• v.11 no.6
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• pp.1793-1801
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• 2016

Speech enhancement is a challenging problem due to the diversity of noise sources and their effects in different applications. The goal of speech enhancement is to improve the quality and intelligibility of speech by reducing noise. Many research works in speech enhancement have been accomplished in English and other European Languages. There has been limited or no such works or efforts in the past in the context of Tamil speech enhancement in the literature. The aim of the proposed method is to reduce the background noise present in the Tamil speech signal by using wavelets. New modified thresholding function is introduced. The proposed method is evaluated on several speakers and under various noise conditions including White Gaussian noise, Babble noise and Car noise. The Signal to Noise Ratio (SNR), Mean Square Error (MSE) and Mean Opinion Score (MOS) results show that the proposed thresholding function improves the speech enhancement compared to the conventional hard and soft thresholding methods.

• Speech Sciences
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• v.13 no.1
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• pp.109-116
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• 2006

In this paper, our study focuses on obtaining the differences of emotional speech in three different vocal tract sections. The vocal tract area was computed from the area function of the emotional speech. The total vocal tract was divided into 3 sections (vocal fold section, middle section and lip section) to acquire the differences in each vocal tract section of emotional speech. The experiment data include 6 emotional speeches from 3 males and 3 females. The 6 emotions consist of neutral, happiness, anger, sadness, fear and boredom. The measured difference is computed by the ratio through comparing each emotional speech with the normal speech. The experimental results present that there is not a remarkable difference at lip section, but the fear and sadness have a great change at the vocal fold part.

• 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.

• The Transactions of The Korean Institute of Electrical Engineers
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• v.59 no.4
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• pp.831-840
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• 2010

In this paper, we propose a hearing impairment simulator considering reduced frequency selectivity and asymmetrical auditory filter of the hearing impaired, and we verified the reduced frequency selectivity and asymmetrical auditory filter affected in speech perception through experiments. The reduced frequency selectivity has made embodied by spectral smearing using LPC(linear prediction coding). The shapes of auditory filter are asymmetrical different with each center frequency. Hearing impaired person which has hearing loss was differently changed with that of normal hearing people and it has different value for speech of quality through auditory filter. The experiments confirmed subjective test and objective test. The subjective experiments are composed of 4 kinds of tests: pure tone test, SRT(speech reception threshold) test, and WRS(word recognition score) test without spectral smearing, and WRS test with spectral smearing. The experiment of the hearing impairment simulator was performed from 9 subjects who have normal ears. The amount of spectral smearing was controlled by LPC order. The asymmetrical auditory filter of proposed hearing impairment simulator was simulated and then some tests to estimate the filter's performance objectively were performed. The objective experiment as simulated auditory filter's performance evaluation method used PESQ(perceptual evaluation of speech quality) and LLR(log likelihood ratio) for speech through auditory filter. The processed speech was evaluated objective speech quality and distortion using PESQ and LLR value. When hearing loss processed, PESQ and LLR value have big difference according to asymmetrical auditory filter in hearing impairment simulator.

• 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.38 no.4
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• pp.406-413
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• 2019

This paper proposes a speech enhancement method through the process of combining the gain function with spectrum subtraction method in the two microphone array with close spacing. A speech enhancement method that uses a gain function estimated by the SNR (Signal-to Noise Ratio) based on the multi frequency band coherence function causes the performance degradation in high correlation between input noises of two channels. A new speech enhancement method is proposed where the weighted gain function is used by combining the gain function from the spectral subtraction. The performance evaluation of the proposed method was shown by comparison with PESQ (Perceptual Evaluation of Speech Quality) value which is an objective quality evaluation test provided by the ITU-T (International Telecommunications Union Telecommunication). In the PESQ tests, the maximum 0.217 of PESQ value is improved in the various background noise environments.