• Phonetics and Speech Sciences
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• v.3 no.3
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• pp.173-179
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• 2011

The present study was performed to investigate the intelligibility of voice disorder patients by providing the various background noise levels. Four sets of 12-sentence-stimuli produced by 11 voice disorder patients were prepared, and 5 minute-news from radio broadcasting studio were used as a background noise. 30 listeners assigned intelligibility score of each sentence with visual analog scale. Each set of sentences was provided with 20dB, 10dB, 0dB noise (same intensity with stimuli), and, finally, with no noise. As results, as background noise level increased, intelligibility scores were lowered with statistical significance. Even though in the same severity, more loud background noise showed much lower scores than less loud noise. When 10dB noise was provided, intelligibility scores showed the biggest difference among the degree of severity.

• Journal of the Korean Institute of Telematics and Electronics T
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• v.35T no.1
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• pp.120-124
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• 1998

Tn the analysis of the audio signal characteristices and the codebook indices of LD-CELP, there are many cases of detecting codebook indices that are used previous. LD-CELP algorithm achives good quality of audio because it has analyzed a short term of audio signal. In spite of these advantage, the method has a drawback in which searching time of best codebook index inclose due to a complicated calculation for codebook index. This paper is proposed to decreasing the searching time of codebook index using a searching memory. As a simulation of the proposed method, searching time for codebook index is reduced 3.2%-11.7% as compared with LD-CELP.

• Phonetics and Speech Sciences
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• v.16 no.1
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• pp.57-65
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• 2024

Normal pressure hydrocephalus (NPH) is a condition wherein the cerebrospinal pressure in the brain is within the normal range, but the cerebrospinal fluid increases above the normal level, causing ventriculomegaly. In patients with NPH, the articulatory system exhibits reduced mobility and range, which may affect diadochokinesis (DDK) and speech intelligibility. In this study, we investigated the characteristics of DDK, including accurate-token DDK and all-token DDK including inaccurate tokens, in patients with NPH and healthy elderly adults (HE). We also examined the classification accuracy of DDK between the two groups. Finally, we investigated whether there was a correlation between speech intelligibility and DDKs in the NPH group. The results showed that NPH and HE groups differed significantly in both accurate-token DDK and all-token DDK, and their classification accuracy was relatively high. However, there was no correlation between speech intelligibility and DDK. The findings suggest that the DDK is a useful method for sensitively assessing speech motor performance in patients with NPH.

• The Journal of the Acoustical Society of Korea
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• v.32 no.4
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• pp.335-345
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• 2013

The present study investigates the effect of the inter-aural level difference(ILD) on the syllable articulation test in classrooms which can be occurred by the absorption of interior surfaces. In order to do this, the sound absorbing materials were installed in the classroom and sound pressure level(SPL) at each ear was measured using binaural recording systems. Also, syllable articulation tests were carried out at a classroom with and without sound absorption materials by 20 students who have normal hearing condition, in order to investigates the effect of the ILD on the speech intelligibility. As a result, it was found that the larger inter-aural level differences was occurred at the nearer positions to lateral walls after sound absorptions were applied to lateral walls in the classroom. At some places, the measured ILD was lager than JND of sound level (3dB). Also, it was shown that the correlation coefficient of inter-aural level difference with the score of syllable test has the significant result(-0.441). Thus, It is concluded that ILD can affect the subjective speech intelligibility in classrooms.

• The Journal of the Acoustical Society of Korea
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• v.27 no.7
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• pp.379-385
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• 2008

In this paper, we propose an effective speech reinforcement technique under the near-end and the far-end noise environments. In general, since the intelligibility of the far-end speech for the near-end listener is significantly reduced under near-end noise environments, we require a far-end speech reinforcement approach to avoid this phenomena. Specifically, based on the estimated background noise spectrum of the near-end, we reinforce the far-end speech spectrum by incorporating the more general cases under the near-end with background noise. Also, we propose the novel approach to reinforce the actual speech signal except for the noise signal in the far-end noisy speech signal. The performance of the proposed algorithm is evaluated by the CCR (Comparison Category Rating) test of the method for subjective determination of transmission quality in ITU-T P.800 under various noise environments and shows better performances compared with the conventional method.

• Proceedings of the Acoustical Society of Korea Conference
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• 1993.06a
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• pp.93-97
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• 1993

본 논문은 한국어 문어변환(TTS:Text-to-Speech) 시스템내에서의 음성합성시 음질 및 자연성 개선을 위한 연구 결과이다. 합성음 평가방법으로는 한국어 발음대사전에 수록된 빈도수 순위대로 추출한 음절(V형: 19개, CV형:80개, VC형:30개, CVC형: 100개, 총 229개)을 대상으로 규칙합성시킨 1음절어(합성음절수:229개)중 음절유형별로 15개씩 총 60개 음절을 20초간 3회 반복음의 녹음 테이프를 작성한 합성음에 대하여 사전지식이 없는 임의의 그룹을 선정하여 이해도, 명료도, 잡음감, 자연성 등 4 가지 항목에 대하여 오피니온 평가를 수행한 결과를 제시하였다.

• Proceedings of the Acoustical Society of Korea Conference
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• 1998.06e
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• pp.31-34
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• 1998

배경 잡음에 의해 저하된 음성을 복원하는 기술은 이미 오래 전부터 여러 가지 기법들이 연구되어왔다. 이들 기법 중, Spectral Subtraction 기법은 단일 채널에 의한 Speech Enhancement의 대표적인 방법이다. 그러나, 기존의 단일 채널 Speech Enhancement 기법의 중요한 단점은 Musical Noise라 불리는 잔존 Noise의 발생 및 목적신호가 왜곡된다는 것이다. 이 잔존 Noise에 의해 지금까지 연구 보고된 단일 채널 Speech Enhancement기법들은 거의 대부분 SNR은 향상되었지만 명료도의 향상이 곤란하였다고 보고되어왔다. 그러므로, 본 연구에서는 인간의 청각기구의 지각과정을 충실히 모방한 ROEX(Rounded Exponential) 청각 Filter를 이용하여 잔존 Noise인 Musical Noise를 억제시키는 기법을 제안하고자 한다.

• Proceedings of the Acoustical Society of Korea Conference
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• 1995.06a
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• pp.164-167
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• 1995

RPE 음성부호화기에서 합성 필터로 인한 구동벡터 양자화잡음의 증폭효과를 분석하고 regular pulse 시퀀스의 양자화로 인한 성능감쇄를 줄이기 위해 pyramid vector 양자화방식을 도입하였다. 제안된 방식의 성능평가는 구동시퀀스 양자화를 위해 adaptive PCM을 이용하는 GSM 표준 RPE 방식과의 객관적 및 주관적 성능비교를 통해 수행하였다.T JDSMDQLRY 결과 제안된 방식은 대략 1dB의 SNR 및 segmental SNR 값 증가를 가져왔고, 또한 비공식 청취시험결과 명료도의 증가를 느낄 수 있었다.

• The Journal of the Acoustical Society of Korea
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• v.13 no.5
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• pp.51-58
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• 1994

These times many kinds of man-machine Interfaces using speech signal, speech recognizers or speech synthesizers, are proposed and utilized in practice. Especially speech synthesis system is widely used in our life. But its assessment method is still in its first stage. In this paper we propose a method to generate multi-syllable nonsense wordset for the purpose of synthetic speech assessment and applies the wordset to one commercial text-to-speech system. Some results about the experiment is suggested and it is verified that the method to generate a nonsense wordset can be used to assess the intelligibility of the synthesizer in phoneme level or in phonemic environmental level.

• Journal of the Korean Institute of Telematics and Electronics S
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• v.35S no.6
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• pp.147-155
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• 1998

Conventional time-scale modification methods have the problem that as the modification rate gets higher the time-scale modified speech signal becomes less intelligible, because they ignore the effect of articulation rate on speech characteristics. In this paper, we propose a variable time-scale modification method based on the knowledge that the timing information of transient portions of a speech signal plays an important role in speech perception. After identifying steady protions only. The result of subjective preference test indicates that the proposed method produces performance superior to that of the conventional SOLA method.