• Journal of the Korea Institute of Information and Communication Engineering
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• v.18 no.7
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• pp.1740-1748
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• 2014

This paper focuses on listening test and acoustic analysis of patients' speech with velopharyngeal insufficiency (VPI) and normal speakers' simulation speech. In this research, a set consisting of 50-words, vowels and single syllables is determined for speech database construction. A web-based listening evaluation system is developed for a convenient/automated evaluation procedure. The analysis results show the trend of incorrect recognition for VPI speech and the one for simulation speech are similar. Such similarity is also confirmed by comparing the formant locations of vowel and spectrum of consonant sounds. These results show that the simulation method for VPI speech is effective at generating the speech signals similar to actual VPI patient's speech. It is expected that the simulation speech data can be effectively employed for our future work such as acoustic model adaptation.

• Journal of Biomedical Engineering Research
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• v.28 no.2
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• pp.205-211
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• 2007

A novel envelope extraction algorithm for speech processor of cochlear implants, called adaptation algorithm, was developed which is based on a adaptation effect of the inner hair cell(IHC)/auditory nerve(AN) synapse. We achieved acoustic simulation and hearing experiments with 12 normal hearing persons to compare this adaptation algorithm with existent standard envelope extraction method. The results shows that speech processing strategy using adaptation algorithm showed significant improvements in speech recognition rate under most channel/noise condition, compared to conventional strategy We verified that the proposed adaptation algorithm may yield better speech perception under considerable amount of noise, compared to the conventional speech processing strategy.

• Speech Sciences
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• v.11 no.4
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• pp.53-66
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• 2004

The nonlinear speech enhancement method with interactive parallel-extended Kalman filter is applied to speech contaminated by additive white noise. To represent the nonlinear and nonstationary nature of speech. we assume that speech is the output of a nonlinear prediction HMM (NPHMM) combining both neural network and HMM. The NPHMM is a nonlinear autoregressive process whose time-varying parameters are controlled by a hidden Markov chain. The simulation results shows that the proposed method offers better performance gains relative to the previous results [6] with slightly increased complexity.

• Proceedings of the Korean Institute of Communication Sciences Conference
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• 1984.04a
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• pp.18-21
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• 1984

In this paper, the tree coding using the (M, L) multi-path search algorithm has teen investigated. A hybrid adaptation scheme which employs a block adaptation as well as a sequential dadptation is described for application in quantization and compression of speech signals. Simulation results with the gybrid adaptation scheme indicate that a relatively good speech quality can be obtained at rate about 8Kbps. All necessary parameters such as MlL and filter-order were found from simulation and these parameters turned out to be a good compromise between the complexity and overall performance.

• Proceedings of the KIEE Conference
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• 1999.11c
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• pp.700-702
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• 1999

The protection of data which we want to keep secret from invalid users has become a main topic nowadays. This paper introduces a encryption scheme for protecting speech signals from eavesdropping. The proposed encryption scheme adopts a secure voice cryptographic algorithm based on the scrambling in frequency band. In order to improve the conventional speech signal encryption scheme, we have randomly permuted DCT coefficients of speech signal. Simulation results are included to show the performance of the proposed algorithm for secure transmission of speech signals.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.21 no.5
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• pp.1172-1179
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• 1996

General adaptive noise canceller(ANC) suffers from the misadjustment of adaptive filter weights, because of the gradient-estimate noise at steady state. In this paper, an adaptive noise cancellation algorithm with speech detector which is distinguishing speech from silence and adaptation-transient region is proposed. The speech detector uses property of adaptive prediction-error filter which can filter the highly correlated speech. To detect speech region, estimation error which is the output of the adaptive filter is applied to the adaptive prediction-error filter. When speech signal apears at the input of the adaptive prediction-error filter. The ratio of input and output energy of adaptive prediction-error filter becomes relatively lower. The ratio becomes large when the white noise appears at the input. So the region of speech is detected by the ratio. Sign algorithm is applied at speech region to prevent the weights from perturbing by output speech of ANC. As results of computer simulation, the proposed algorithm improves segmental SNR and SNR up to about 4 dBand 11 dB, respectively.

• Proceedings of the KSPS conference
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• 2005.11a
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• pp.121-124
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• 2005

In this paper, we propose a sound localization algorithm for two simultaneous speakers. Because speech is wide-band signal, there are many frequency sub-bands in that two speech sounds are mixed. However, in some sub-bands, one speech sound is more dominant than other sounds. In such sub-bands, dominant speech sounds are little interfered by other speech or noise. In speech sounds, overtones of fundamental frequency have large amplitude, and that are called 'Harmonic structure of speech'. Sub-bands inharmonic structure are more likely dominant. Therefore, the proposed localization algorithm is based on harmonic structure of each speakers. At first, sub-bands that belong to harmonic structure of each speech signal are selected. And then, two speakers are localized using selected sub-bands. The result of simulation shows that localization using selected sub-bands are more efficient and precise than localization methods using all sub-bands.

• The Transactions of The Korean Institute of Electrical Engineers
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• v.62 no.12
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• pp.1750-1754
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• 2013

A variable bit rate speech coder is presented. The coder is based on the observation that a speech signal can be viewed as a combination of piecewise linear signals in a short time period. The encoder detects the sample points where the slope of the signal changes, which are called the inflection points in this paper. The coder transmits the location and value for the detected inflection sample, but only the location information for the noninflection samples. In the decoder, the noninflection samples are estimated with interpolation of the received information. Several factors affecting the performance of the coder have been tested through simulation. Simulation results show that the linear interpolation produces 1 ~ 5 dB improvement over the cubic spline interpolation. And the -law companding does not provide any benefit when it is applied before the inflection detection. With low threshold values in the inflection point detection, the coder shows better MOS and more than 16 dB improvement in SNR compared to the continuously variable slope delta modulation (CVSDM).

• MALSORI
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• no.51
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• pp.137-149
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• 2004

A weather forecast service with speech recognition is described. This service allows users to get the weather information of all the cities by saying the city names with just one phone call, which was not provided in the previous weather forecast service. Speech recognition is implemented in the intelligent peripheral (IP) of the advanced intelligent network (AIN). The AIN is a telephone network architecture that separates service logic from switching equipment, allowing new services to be added without having to redesign switches to support new services. Experiments in speech recognition show that the recognition accuracy is 90.06% for the general users' speech database. For the laboratory members' speech database, the accuracies are 95.04% and 93.81%, respectively in simulation and in the test on the developed system.

• The Journal of the Acoustical Society of Korea
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• v.23 no.8
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• pp.597-601
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• 2004

This paper introduces a pre-rejection technique for wireless channel distorted speech with application to automatic speech recognition (ASR) Based on analysis of distorted speech signals over a wireless communication channel. we propose a method to reject the channel distorted speech with a small computational load. From a number of simulation results. we can discover that tile pre-rejection algorithm enhances the robustness of speech recognition operation.