• IEIE Transactions on Smart Processing and Computing
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• 제2권4호
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• pp.197-202
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• 2013

Large gains in speech intelligibility can be obtained using the SNR-based binary mask approach. This approach retains the time-frequency (T-F) units of the mixture signal, where the target signal is stronger than the interference noise (masker) (e.g., SNR > 0 dB), and removes the T-F units, where the interfering noise is dominant. This paper introduces two alternative binary masks based on the distortion constraints to improve the speech intelligibility. The distortion constraints are induced by a gain function for estimating the short-time spectral amplitude. One binary mask is designed to retain the speech underestimated (T-F) units while removing the speech overestimated (T-F)units. The other binary mask is designed to retain the noise overestimated (T-F) units while removing noise underestimated (T-F) units. Listening tests with oracle binary masks were conducted to assess the potential of the two binary masks in improving the intelligibility. The results suggested that the two binary masks based on distortion constraints can provide large gains in intelligibility when applied to noise-corrupted speech.

• 방송공학회논문지
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• 제18권3호
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• pp.455-462
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• 2013

본 논문에서는 여러 개의 마이크를 이용하여 잡음을 제거하는 방법인 공간 필터로 전처리된 신호를 입력으로 하는 음성 왜곡 가중 다채널 위너 필터 (Spatially Preprocessed Speech Distortion Weighted Multi-channel Wiener Filter: SP-SDW-MWF)에 대해 소개하고, 가중치를 결정하는 방법을 제안한다. SP-SDW-MWF는 마이크로폰 어레이를 이용한 잡음 제거 알고리즘으로서 마이크로폰 불일치와 같은 오차에 강인한 것으로 알려져 있다. SP-SDW-MWF는 필터 계수를 최적화할 때 음성 왜곡과 잡음 제거에 대한 기준으로 나누어 가중치를 두고 있다. 이러한 가중치를 결정하기 위해, 본 논문에서는 전력 스펙트럼 밀도 오차를 평가 척도로 사용하여 마이크로폰으로부터 입력된 음성 신호와 잡음의 전력 스펙트럼 밀도의 비 (a priori SNR)를 이용하는 방법을 제안한다. 실험결과에서 나타난 바와 같이 a priori SNR에 따라 가변적인 가중치를 사용하는 것이 고정된 값을 가중치로 사용하는 것보다 향상된 성능을 보임을 알 수 있다.

• 음성과학
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• 제10권4호
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• pp.137-147
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• 2003

Speech synthesis can classify by synthesis way, that is waveform coding, source coding and mixture coding. Specially, waveform coding is suitable for high quality synthesis. However, it is not desirable by synthesis techniques of syllable or phoneme unit because it do not separate and handles excitation and formant part. Therefore, there is a need for pitch alteration method applied in synthesis by the rule in waveform coding. This study propose about pitch alteration method that use spectrum scaling after do to flatten spectra by subband linear approximation to minimize spectrum distortion. This paper show evaluation whether show excellency of some measure compared with LPC, Cepstrum, lifter function and method that propose. estimation method seeks distribution of each flattened signal and measured degree of flattened spectra Signal flattened is normalized, So that highest point amounts to zero, and distribution of signal ,whose average is zero, is calculated. this show result that measure the spectrum distortion rate to estimate performance of method that propose. The average spectrum distortion rate was kept below the average 2.12%, so the method that propose is superiors than existent method.

• 한국음향학회지
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• 제24권7호
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• pp.416-422
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• 2005

디지털 통신 시스템에서의 음성 인식은 음성 부호화기에 의한 음성 신호의 왜곡으로 인하여 성능이 크게 저하된다. 본 논문에서는 음성 부호화기에 의한 스펙트럼 왜곡을 분석하고 왜곡된 주파수 정보를 보상하는 전처리 과정을 통하여 음성 인식 성능을 향상시키는 방법을 제안한다. 현재 널리 사용되는 표준 음성 부호화기인 IS-127 EVRC, ITU G.729 CS-ACELP. IS-96 QCELP를 사용하여 부호화에 의한 왜곡을 분석하고, 모든 음성 부호화기에 공통으로 적용하여 왜곡을 보상할 수 있는 전처리 방법을 개발하였다. 본 논문에서 제안하는 왜곡 보상 방법을 세 종류의 음성부호화기에 각각 적용하였으며, 왜곡된 음성 신호에 대한 음성 인식률에 비하여 최대 $15.6\%$의 인식률 향상을 얻을 수 있었다.

• The Journal of the Acoustical Society of Korea
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• 제18권1E호
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• pp.32-36
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• 1999

In the area of the speech synthesis techniques, the waveform coding methods maintain the intelligibility and naturalness of synthetic speech. In order to apply the waveform coding techniques to synthesis by rule, we must be able to alter the pitches of synthetic speech. In this paper, we propose a new pitch altering method that compensates phase distortion of the cepstral pitch alteration method with time scaling method in the time domain. This method can remove some spectrum distortion which is occurred in conjunction point between the waveforms. For performance test the spectrum distortion rate was used as objective criterion and the MOS(Mean Opinion Score) was used as subjective criterion. As a result, the spectrum distortion and MOS are obtained by 0.66% and 3.9, respectively.

• 대한음성학회지:말소리
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• 제43호
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• pp.179-188
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• 2002

Improving the recognition performance of connected digit telephone speech still remains a problem to be solved. As a basic study for it, this paper analyzes the variation of feature parameters of Korean digit telephone speech according to channel distortion. As a feature parameter for analysis and recognition MFCC is used. To analyze the effect of telephone channel distortion depending on each call, MFCCs are first obtained from the connected digit telephone speech for each phoneme included in the Korean digit. Then CMN, RTCN, and RASTA are applied to the MFCC as channel compensation techniques. Using the feature parameters of MFCC, MFCC+CMN, MFCC+RTCN, and MFCC+RASTA, variances of phonemes are analyzed and recognition experiments are done for each case. Experimental results are discussed with our findings and discussions

• 음성과학
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• 제11권4호
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• pp.17-28
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• 2004

In this paper, we propose a statistical approach to improve the performance of spectral quantization of speech coders. The proposed techniques compensate for the distortion in a decoded line spectrum pairs (LSP) vector based on a statistical mapping function between a decoded LSP vector and its corresponding original LSP vector. We first develop two codebook-based probabilistic matching (CBPM) methods based on linear mapping functions according to different assumption of distribution of LSP vectors. In addition, we propose an iterative procedure for the two CBPMs. We apply the proposed techniques to a predictive vector quantizer used for the IS-641 speech coder. The experimental results show that the proposed techniques reduce average spectral distortion by around 0.064dB.

• 대한음성학회지:말소리
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• 제50호
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• pp.99-110
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• 2004

The normal transmission characteristics of sound are hardly obtained due to the various noises and structural factors in a running car environment. It is due to the channel distortion of the original source sound recorded by microphones, and it seriously degrades the performance of the speech recognition in real driving environments. In this paper we analyze the degree of intelligibility under the various sound distortion environments by channels according to driving speed with respect to speech transmission index(STI) and compare the STI with rates of speech recognition. We examine the correlation between measures of intelligibility depending on sound pick-up patterns and performance in speech recognition. Thereby we consider the optimal location of a microphone in single channel environment. In experimentation we find that high correlation is obtained between STI and rates of speech recognition.

• 대한음성학회:학술대회논문집
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• 대한음성학회 2005년도 추계 학술대회 발표논문집
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• pp.63-66
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• 2005

For the Large Corpus based TTS the consistency of the speech corpus is very important. It is because the inconsistency of the speech quality in the corpus may result in a distortion at the concatenation point. And because of this inconsistency, large corpus must be tuned repeatedly One of the reasons for the inconsistency of the speech corpus is the different glottal characteristics of the speech sentence in the corpus. In this paper, we adjusted the glottal characteristics of the speech in the corpus to prevent this distortion. And the experimental results are showed.

• 말소리와 음성과학
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• 제6권1호
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• pp.85-93
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• 2014

This paper investigated the effects of frequency band limitation on perceptual characteristics of Korean vowels. Monosyllabic speech (144 syllables of CV type, 56 syllables of VC type, 8 syllables of V type) produced by two announcers were low- and high-pass filtered with cutoff frequencies ranging from 300 to 5000 Hz. Six listeners with normal hearing performed perception tests by types of filter and cutoff frequencies. We reported phoneme recognition rates and types of perception error of band-limited Korean vowels to examine how frequency distortion in the process of speech transmission affect listener's perception.