• Journal of Biomedical Engineering Research
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• v.32 no.3
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• pp.198-206
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• 2011

Hearing devices like cochlear implant, vibrant soundbridge, etc. try to offer better sound for people. In hearing devices, several beamformers including conventional directional microphone are applicable to noise reduction. Each beamformer has different directional response and it could change sound intelligibility or quality for listeners. Therefore, we investigated the performance of three beamformers, which are first and second order directional microphone, and broadband beamformer(BBF) with a computer simulation assuming hearing device microphone configuration. We also calculated objective measurements which have been used to evaluate speech enhancement algorithms. In the simulation, a single speech and a single babble noisewere propagated from the front and $135^{\circ}$ azimuth degrees respectively. Microphones were configured in an end-fire array and the spacing was varied in comparison. With 3 cm spacing, BBF had about 3 dB higher enhanced SNR than that of directional microphones. However, enhancement of segmental SNR and frequency weighted segmental SNR were similar between the first order directional microphone and broadband beamformer. In addition when steady state noise was used, broadband beamformer showed the increased performance and had the highest enhanced SNR, and segmental SNR.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.40 no.5
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• pp.924-930
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• 2015

In this paper, we propose a novel approach of low computational complexity to improve the speech quality of the small acoustic equipment in noisy environment. The conventional gain control algorithm suppresses the noise of input signal, and then the part of wide dynamic range compression (WDRC) amplifies the undesired signal. The proposed algorithm controls the gain of hearing aids according to speech present probability by using the output of a voice activity detection (VAD). The performance of the proposed scheme is evaluated under various noise conditions by using objective measurement and yields superior results compared with the conventional algorithm.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.5C
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• pp.478-486
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• 2007

Double-talk detection errors can result in either large residual echo or distorting the near-end talker's input speech. Thus accurate double-talk detection is an important problem in the acoustic echo canceller to improve the speech quality. In the double-talk detection algorithm using a cross-correlation coefficient, double-talk detection errors can occur in the initial convergence period of an adaptive filter or in noisy environment since the cross-correlation coefficient becomes large in such situations. In this paper, we propose a new double-talk detection algorithm based on the cross-correlation method using a normalized error signal power to reduce the double-talk detection errors. The experimental results have shown the performance improvement of an acoustic echo canceller as well as the noise-robustness of the proposed double-talk detector.

• Speech Sciences
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• v.1
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• pp.201-211
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• 1997

Acoustically a naturally-spoken vowel is composed of five formants. However, the acoustic quality of a vowel is known to be mostly determined by $F_1\;and\;F_2$. The main purpose of this study was to examine how synthesized vowels with $F_1\;and\;F_2$ are perceived by Korean native speakers. In addion, we are interested in finding whether the synthesized vowels are perceived differently by standard Korean speakers and Kyungnam regional dialect speakers. In the experiment 9 Seoul standard Korean speakers and 9 Kyungnam dialect speakers heard 536 vowels synthesized in vowel space with $F_1\;by\;F_2$ and categorized them into one of 10 Korean vowels. The resultant vowel map showed that each Korean vowel occupies an unique area in the two-dimensional vowel space of $F_1\;by\;F_2$, and confirmed that $F_1\;and\;F_2$ play important roles in the perception of vowels. The results also showed that the Seoul speakers and the Kyungnam speakers perceive the synthesized vowels differently. For example, /e/ versus /$\varepsilon$/ contrast, /y/, and /$\phi$/ are perceived differently by the Seoul speakers, whereas they were perceptually confused by the Kyungnam speakers. These results might be due to the different vowel systems of the standard Korean and the Kyungnam regional dialect. While the latter uses a six-vowel system which has no /e/ vs /$/ contrast, /v/ vs /i/ contrast, /y/, and /$\phi$/, the former recognizes these as different vowels. This result suggests that the vowel system of differing dialect restricts the perception of the Korean vowels. Unexpectedly /i/ does not occupy any area in the vowel apace. This result suggests that /i/ cannot be synthesized without $F_3$.

• Phonetics and Speech Sciences
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• v.2 no.4
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• pp.165-171
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• 2010

Vocal process granuloma can occur commonly by laryngopharyngeal reflux (LPR), vocal abuse or misuse. It has been reported that voice therapy is employed with medication therapy for the patients who has vocal process granuloma, however research about effect of voice therapy can be hardly founded. For that matter, the primary aim of this study was to evaluate the effect of therapeutic method we implement. Thirty one patients who has been diagnosed with vocal process granuloma from January, 2007 to June, 2009 participated in this study. 19 patients among them are provided voice therapy and medication, 12 patients take only medication. Voice therapy is implemented ranging from 5 to 19 sessions (mean: 8.6 sessions). We provided explanation about problem each patient has, voice rest, SKMVTT$^{(R)}$, abdominal breathing, and relaxation in session. All subjects were examined by videostroboscopy, perceptual assessment, acoustic and aerodynamic measures. Consequantly, the greater part of the patients (78.9%) who is treated by voice therapy and medication are confirmed disappearance or decrease of granuloma, it shows better results compared with the group provided only medication (66.7%). Especially, the period of drug administration is 3.7 months in the group runs parallel with voice therapy, the period of other group is 7.8 months. The results of acoustic and aerodynamic measures after treating indicates there are significant decrease in Jitter, Shimmer, and NHR, and increase in MPT, Psub (p<.05). However, there is no large difference statistically even though voice quality has improved since the therapy. In conclusion, it is verified that the voice therapy to the vocal process granuloma patients taking medication is effectual method, we recommend combining voice therapy with medication when treatment is needed for the vocal process granuloma patients.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.11C
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• pp.1573-1580
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• 2004

In this paper, an adaptation mode controller for adaptive microphone array system is proposed for high-quality speech acquisition in real environments. To ensure proper adaptation of the adaptive array algorithm, the proposed adaptation mode controller uses not only temporal information, but also spatial information. The proposed adaptation mode controller is constructed with two processing stages: an initialization stage and a running stage. In the initialization stage, a sound source localization technique is adopted, and a signal correlation characteristic is used in the running stage. For the adaptive may algorithm, a generalized sidelobe canceller with an adaptive blocking matrix is used. The proposed adaptation mode controller can be used even when the adaptive blocking matrix is not adapted, and is much stable than the power ratio method. The proposed algorithm is evaluated in real environment, and simulation results show 13dB SINR improvement with the speaker sitting 2m distance from the may.

• The Journal of Internal Korean Medicine
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• v.42 no.3
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• pp.323-339
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• 2021

Objectives: This study investigates the effect of electro-acupuncture on aphasia after stroke. Methods: A search of OASIS, NDSL, PubMed, Cochrane, and CNKI was executed between 4 January 2021 and 4 February 2021, with no limitation on publication year. Extraction and selection from the studies were made by 3 authors. The quality of the studies was evaluated using Cochrane's risk of bias (RoB) tool. Results: 10 studies met the selection criteria. As the treatment site for electro-acupuncture, GV20 (Baihui) was used the most. In all studies, the region located on the head was used for treatment without distinguishing between acupoints and areas of scalp acupuncture, and the stimulation was organized into 3 conditions: speed, intensity, and time. The outcome indicators used before and after treatment focused on the evaluation of language function and the degree of aphasia. The results showed that using electro-acupuncture with speech rehabilitation therapy for aphasia after stroke was more effective than using speech rehabilitation therapy alone. Conclusions: In this review, electro-acupuncture for aphasia after stroke was found to have a significant effect compared to the previous treatment alone. However, because of limitations, information was not reliable enough. Additional research is needed to produce more objective evidence.

• The Journal of the Korea institute of electronic communication sciences
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• v.17 no.3
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• pp.409-416
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• 2022

This paper relates to the blind noise separation method of time-delayed convolutive mixed signals. Since the mixed model of acoustic signals in a closed space is multi-channel, a convolutive blind signal separation method is applied and time-delayed data samples of the two microphone input signals is used. For signal separation, the mixing coefficient is calculated using an inverse model rather than directly calculating the separation coefficient, and the coefficient update is performed by repeated calculations based on secondary statistical properties to estimate the speech signal. Many simulations were performed to verify the performance of the proposed blind signal separation. As a result of the simulation, noise separation using this method operates safely regardless of convolutive mixing, and PESQ is improved by 0.3 points compared to the general adaptive FIR filter structure.

• The Journal of the Acoustical Society of Korea
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• v.29 no.1
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• pp.69-75
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• 2010

In this paper, the pitch is detected after the elimination of formant ingredients by flattening the spectrum in frequency domain. In order to remove impact of formant and transition frequency in the signal spectrum, formant envelop is made by linear interpolation with any points each sub-band and the spectrum of speech signal is compensated by the reverse of the envelop interpolated linearly after we divide frequency band into several segment based on LSP and detect the points. The experimental result showed the proposed method appeared an outstanding performance in compared with LPC, Cepstrum, Lifter methods. The method reduced the gross error rate 1.30% than the LPC method which appeared a good performance except the proposed method. Also, the proposed method showed low error rate in noise environment.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.18 no.12
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• pp.1841-1854
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• 1993

In this paper, the packet voice protocol for the transmission of voice signal onto ethernet is implemented in a personal computer (PC). The packet voice protocol used is a modified one from CCITT G.764 packetized voice protocol. The hardware system to facilitate the voice communication onto ethernet is divided into telephone interface, speech processing, PC interface and controllers. The software structure of the protocol is designed according to the OSI seven layer architecture and is divided into three routines : ethernet device driver, telephone interface, and processing routine of the packet voice protocol. Experiments through ethernet with telephone interface show that this packet voice communication achieves satisfactory quality when the network traffic is light.