• 대한후두음성언어의학회지
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• 제15권2호
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• pp.92-97
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• 2004

Background and Objectives : The purpose of this study was to examine objectively pre and post operative voice quality evaluation and intelligibility of alaryngeal voice using speech recognition program, HUVOIS. Materials and Methods : 2 laryngologists and 1 speech pathologist were evaluated 'G', 'R', 'B' in the GRBAS sclae and speech intelligibility using NTID rating scale from standard paragraph. And also acoustic estimates such as jitter, shimmer, HNR were obtained from Lx Speech Studio. Results : Speech recognition rate was not significantly different between pre and post operation for pathological vocie samples though voice quality(G, B) and acoustic values(Jitter, HNR) were significantly improved after post operation. In Alaryngeal voices, reed type electrolarynx 'Moksori' was the highest both speech intelligibility and speech recognition rate, whereas esophageal speech was the lowest. Coefficient correlation of speech intelligibility and speech recognition rate was found in alaryngeal voices, but not in pathological voices. Conclusion : Current study was not proved speech recognition program, HUVOIS during telephone program was not objective and efficient method for assisting subjective GRBAS scale.

• 대한음성학회지:말소리
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• 제52호
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• pp.133-144
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• 2004

The quality of narrowband speech can be enhanced by the bandwidth extension technology. This paper proposes a mixed excitation and an energy compensation method based on Gaussian Mixture Model (GMM). First, we employ the mixed excitation model having both periodic and aperiodic characteristics in frequency domain. We use a filter bank to extract the periodicity features from the filtered signals and model them based on GMM to estimate the mixed excitation. Second, we separate the acoustic space into the voiced and unvoiced parts of speech to compensate for the energy difference between narrowband speech and reconstructed highband, or lowband speech, more accurately. Objective and subjective evaluations show that the quality of wideband speech reconstructed by the proposed method is superior to that by the conventional bandwidth extension method.

• 전력전자학회:학술대회논문집
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• 전력전자학회 2002년도 전력전자학술대회 논문집
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• pp.727-729
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• 2002

In this paper, we proposed quad variable rates ADPCM coding method and its implementation on C6000 DSP, which is modified from the standard ADPCM of ITU G.726 for speech quality improvement considering the environmental noise Four coding rates, 16Kbps, 24Kbps, 32Kbps and 40Kbps are used for speech window samples and the rate decision threshold is decided by the environmental noise level. The object of the proposed method is to reduce the coding rate while retaining the speech quality and the speech quality is considerably close to 40Kbps single rate coder with the coding rate close to 16Kbps single rate coder under the environmental noise. The environmental noise level affects the coding rate and the noise level is calculated per every speech window samples. At high noise level, more samples are coded at higher rates to enhance the quality, but at low noise level, only the big speech signals are coded at higher rates and more speech samples are coded at lower coding rates to reduce the coding rates. The influence of the noise on tile speech signal is considerably high for small signals and the small signal has the higher ZCR (zero crossing rate). The method is simulated in PC and to be implemented on C6000 floating point DSP board in real time operations.

• ETRI Journal
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• 제43권1호
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• pp.82-94
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• 2021

In this work, six voiced/unvoiced speech classifiers based on the autocorrelation function (ACF), average magnitude difference function (AMDF), cepstrum, weighted ACF (WACF), zero crossing rate and energy of the signal (ZCR-E), and neural networks (NNs) have been simulated and implemented in real time using the TMS320C6713 DSP starter kit. These speech classifiers have been integrated into a linear-predictive-coding-based speech analysis-synthesis system and their performance has been compared in terms of the percentage of the voiced/unvoiced classification accuracy, speech quality, and computation time. The results of the percentage of the voiced/unvoiced classification accuracy and speech quality show that the NN-based speech classifier performs better than the ACF-, AMDF-, cepstrum-, WACF- and ZCR-E-based speech classifiers for both clean and noisy environments. The computation time results show that the AMDF-based speech classifier is computationally simple, and thus its computation time is less than that of other speech classifiers, while that of the NN-based speech classifier is greater compared with other classifiers.

• The Journal of the Acoustical Society of Korea
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• 제20권4E호
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• pp.31-38
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• 2001

A large-scale synthesis database for a unit selection based synthesis method usually retains redundant synthesis unit instances, which are useless to the synthetic speech quality. In this paper, to eliminate those instances from the synthesis database, we proposed a new pruning method called weighted vector quantization (WVQ). The WVQ reflects relative importance of each synthesis unit instance when clustering the similar instances using vector quantization (VQ) technique. The proposed method was compared with two conventional pruning methods through the objective and subjective evaluations of the synthetic speech quality: one to simply limit maximum number of instance, and the other based on normal VQ-based clustering. The proposed method showed the best performance under 50% reduction rates. Over 50% of reduction rates, the synthetic speech quality is not seriously but perceptibly degraded. Using the proposed method, the synthesis database can be efficiently reduced without serious degradation of the synthetic speech quality.

• 대한전자공학회논문지SP
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• 제42권4호
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• pp.103-110
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• 2005

음성 데이터베이스는 TTS 시스템에서 가장 중요한 요소 중의 하나이다. 특히, 내장형 TTS 시스템에서는 서버형 TTS 시스템에서보다 좀 더 작은 데이터베이스를 필요로 한다. 이러한 이유로, 음성합성 데이터의 압축과 통계적 축소과정의 비중은 내장형 TTS 시스템에서 아주 중요한 항목이라고 말할 수 있다. 그러나 이러한 압축과 통계적 축소과정은 합성음질의 저하를 유발시킨다. 본 논문에서는 고품질 내장형 TTS 시스템에서의 데이터 구축방법을 제안하며, MOS 테스트를 통한 합성음질을 검증한다.

• 한국정보통신학회논문지
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• 제20권3호
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• pp.464-470
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• 2016

본 논문에서는 음성의 특성 지표를 이용한 음성 인식용 데이터베이스 검증 시스템의 개발 내용을 소개하고 이 시스템의 핵심 기술인 음성 특성 지표 추출 알고리즘을 설명한다. 선행 연구에서는 본 시스템에 필요한 효과적인 음성 인식 성능 지표를 생성하기 위해 대표적인 음성 인식 성능 지표인 단어 오인식률(Word Error Rate, WER)과 상관도가 높은 여러 가지 음성 특성 지표들을 조합하여 새로운 성능 지표를 생성하였다. 생성된 음성 인식 성능 지표는 다양한 잡음 환경에서 각 음성 특성 지표를 단독으로 사용할 때보다 단어 오인식률과 높은 상관도를 나타내어 음성 인식 성능을 예측하는데 효과적임을 입증 하였다. 본 실험에서는 선행 연구에서 조합에 사용한 이차적인 음성 인식기에서 추출된 음향 모델 확률 값을 GMM(Gaussian Mixture Model) 음향 모델 확률 값으로 대체해 조합함으로써 시스템 구축 시 다른 음성 인식기에 대한 의존성을 감소시킨다.

• 한국음향학회지
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• 제32권3호
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• pp.214-226
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• 2013

음질 평가는 좋은 소리를 만들기 위해 필수적인 요소이며, 음향의 특성과 대상 시스템에 따라 다양한 방법이 사용되고 있다. 본 논문에서는 음질 평가법의 전반적인 방법론 및 전송 처리된 음향 신호의 품질 평가법에 대해 ITU-T, ITU-R, IEC, 그리고 ANSI 등의 권고안에 기술된 국제 표준을 중심으로 요약하고 분석하였다. 분야별로는 음성 명료도, 음성 음질, 그리고 오디오 음질 평가법을 다루었으며, 현재 사용되는 권고안의 기술적인 내용과 최신 연구 동향 및 향후 발전 방향 등에 대해 기술하였다.

• 한국음향학회지
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• 제37권6호
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• pp.512-517
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• 2018

음성 품질의 향상은 통신 분야의 주요 관심사이다. 본 논문에서는 VoIP(Voice-over-IP) 수신부에서의 배경잡음 및 패킷손실에 강인한 음질향상 방식을 제안한다. 제안된 방식에서는 하이브리드 마르코프 체인 기반 네트워크 지터추정, 추정된 지터를 이용한 적응적 플레이아웃 스케줄링, 그리고 진폭 및 위상 복원 기반의 음성 향상 방식 등을 결합하여 IP 네트워크를 통해 VoIP 수신부에 도착하는 음성신호의 품질을 향상시킨다. 실험결과는 제안된 방식이 송신부의 인코딩 전에 음성신호에 추가된 잡음을 제거하고 불안정한 네트워크 환경에서 양질의 음성을 제공하는 것을 확인할 수 있다.

• 말소리와 음성과학
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• 제10권2호
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• pp.77-84
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• 2018

The purpose of this study is to examine the acoustic characteristics of Parkinsonian speech, with respect to different utterance conditions, by employing acoustic/auditory-perceptual analysis. The subjects of the study were 15 patients (M=7, F=8) with Parkinson's disease who were asked to read out sentences under different utterance conditions (clear/casual). The sentences read out by each subject were recorded, and the recorded speech was subjected to cepstrum and spectrum analysis using Analysis of Dysphonia in Speech and Voice (ADSV). Additionally, auditory-perceptual evaluation of the recorded speech was conducted with respect to breathiness and loudness. Results indicate that in the case of clear speech, there was a statistically significant increase in the cepstral peak prominence (CPP), and a decrease in the L/H ratio SD (ratio of low to high frequency spectral energy SD) and CPP F0 SD values. In the auditory-perceptual evaluation, a decrease in breathiness and an increase in loudness were noted. Furthermore, CPP was found to be highly correlated to breathiness and loudness. This provides objective evidence of the immediate usefulness of clear speech intervention in improving the voice quality of Parkinsonian speech.