• 말소리와 음성과학
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• 제4권3호
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• pp.25-30
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• 2012

This study investigated whether native Korean listeners could use visual cues for the identification of the English consonants /b/ and /v/. Both auditory and audiovisual tokens of word minimal pairs in which the target phonemes were located in word-initial or word-medial position were used. Participants were instructed to decide which consonant they heard in $2{\times}2$ conditions: cue (audio-only, audiovisual) and location (word-initial, word-medial). Mean identification scores were significantly higher for audiovisual than audio-only condition and for word-initial than word-medial condition. Also, according to signal detection theory, sensitivity, d', and response bias, c were calculated based on both hit rates and false alarm rates. The measures showed that the higher identification rate in the audiovisual condition was related with an increase in sensitivity. There were no significant differences in response bias measures across conditions. This result suggests that native Korean speakers can use visual cues while identifying confusing non-native phonemic contrasts. Visual cues can enhance non-native speech perception.

• 말소리와 음성과학
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• 제6권1호
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• pp.95-101
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• 2014

This paper investigated the effects of frequency band limitation on perceptual characteristics of Korean consonants. 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 test by types of filter and cutoff frequencies. We reported phoneme recognition rates and types of perception error of band-limited Korean consonants to examine how frequency distortion in the process of speech transmission affect listener's perception. The results showed that recognition rates varied with the following factors: position in a syllable, manner of articulation, place of articulation, and phonation types. Consonants in the final position were stronger to the frequency band limitation than those in the initial position. Fricatives and Affricates are stronger than stops. Fortis consonants were less stronger than their lenis or aspirated counterparts. Types of perception error also varied depending on such factors as consonant's place of articulation: In case of bilabial stops, they were perceived as alveolar stops with while in cases of alveolar and velar stops, there were changes in phonation types without any change in the place of articulation.

• 전자공학회논문지 IE
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• 제43권2호
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• pp.22-29
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• 2006

본 논문에서는 모바일 환경에 따른 속삭임의 사용이 증가하는 데 따른 속삭임 인식을 위하여 음성인식에 많이 사용되고 있는 특징벡터들을 은닉 마코프 모델을 이용, 정상어 모델, 속삭임 모델, 정상어, 속삭임 통합 모델들에 인식 시험하고 결과를 분석하여 가장 적합한 인식 시스템을 찾으려고 하였다. 인식 시험을 통하여 속삭임의 인식은 정상어 모델로 인식하는 시스템은 낮은 인식률로 실용성이 없으며 속삭임 모델을 별도로 사용하는 것이 85%이상의 가장 높은 인식률을 보였다. 또한 '정상어+속삭임' 모델도 인식률은 조금 벌어지나 가능성을 확인할 수 있었다. 특징벡터로는 속삭임 모델을 사용하는 경우 MFCC 혹은 PLCC를 사용하는 것이 거의 유사하게 높은 인식률을 얻을 수 있었으나 '정상어+속삭임' 모델을 사용하는 경우 PLCC를 특징벡터로 사용하는 것이 속삭임 인식에서 가장 좋은 결과를 보였다.

• ETRI Journal
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• 제36권5호
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• pp.772-782
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• 2014

Two-microphone binary mask speech enhancement (2mBMSE) has been of particular interest in recent literature and has shown promising results. Current 2mBMSE systems rely on spatial cues of speech and noise sources. Although these cues are helpful for directional noise sources, they lose their efficiency in diffuse noise fields. We propose a new system that is effective in both directional and diffuse noise conditions. The system exploits two features. The first determines whether a given time-frequency (T-F) unit of the input spectrum is dominated by a diffuse or directional source. A diffuse signal is certainly a noise signal, but a directional signal could correspond to a noise or speech source. The second feature discriminates between T-F units dominated by speech or directional noise signals. Speech enhancement is performed using a binary mask, calculated based on the proposed features. In both directional and diffuse noise fields, the proposed system segregates speech T-F units with hit rates above 85%. It outperforms previous solutions in terms of signal-to-noise ratio and perceptual evaluation of speech quality improvement, especially in diffuse noise conditions.

• 말소리와 음성과학
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• 제9권1호
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• pp.27-39
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• 2017

This paper deals with how speech rate in Korean is affected by the sociolinguistic factors such as region, gender and generation. Speech rate was quantified as articulation rate (excluding physical pauses) and speaking rate (including physical pauses), both expressed as the number of syllables per second (sps). Other acoustic measures such as pause frequency and duration were also examined. Four hundred twelve subjects were chosen from Korean Standard Speech Database considering their age, gender and region. The result shows that generation has a significant effect on both speaking rate and articulation rate. Younger speakers produce their speech with significantly faster speaking rate and articulation rate than older speakers. Mean duration of total pause interval and the total number of pause of older speakers are also significantly different to those of younger speakers. Gender has a significant effect only on articulation rate, which means male speakers' speech rate is characterized by faster articulation rate, longer and more frequent pauses. Finally, region has no effect both on speaking and articulation rates.

• 한국정보통신학회:학술대회논문집
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• 한국해양정보통신학회 2009년도 추계학술대회
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• pp.723-726
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• 2009

본 연구에서는, 음성인식 시스템에서의 마이크 어레이 기반으로 한 beamforming 방법을 기반으로 음성신호에 대한 에러감소 알고리듬의 성능평가를 위한 시뮬레이션 하였으며 그 성능을 분석하였다. 또한, 마이크 어레이로 부터 취득한 음성신호로 부터 각 채널에 대한 최대 신호대잡음비 구하고 음성신호별로 신호대잡음비를 비교 검토하였다. 음성 인식률은 경우1에서는 54.2%에서 61.4%로, 경우2에서는 더 낮은 신호대잡음비로 41.2%에서 50.5%로 각각 개선됨을 알 수 있었다. 따라서 평균 에러감소율은 경우1에서 15.7%를 보였다.

• 한국음향학회지
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• 제18권3호
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• pp.10-16
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• 1999

본 연구에서는 소량의 음성 데이터만으로 적응화가 가능한 MAPE(최대사후확률추정)을 이용한 연속음성 인식시스템 개발에 대해 연구하였다. 음절단위 모델을 구축한 후 적응화 하고자 하는 화자의 데이터를 연결학습법과 Viterbi 알고리즘으로 음절단위의 추출을 자동화 한 후 MAPE로 적응화하였다. 자동차 제어문에 대해 화자 적응화한 경우의 인식률(O(n)DP인 경우)은 77.18%로 적응화 전의 결과보다 약 6%향상되었다.

• 한국정보처리학회논문지
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• 제7권4호
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• pp.1299-1304
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• 2000

In a speed coding system using excitation source of voiced and unvoiced, it would be involved a distortion of speech quality in case coexist with a voiced and an unvoiced consonants in a frame. So, I propose TSIUVC(Transition Segment Including UnVoiced Consonant) searching, extraction ad approximation-synthesis method in order to uncoexistent with a voiced and unvoiced consonants in a frame. This method based on a zerocrossing rate and pitch detector using FIR-STREAK Digital Filter. As a result, the extraction rates of TSIUVC are 84.8% (plosive), 94.9%(fricative), 92.3%(affricative) in female voice, and 88%(plosive), 94.9%(fricative), 92.3%(affricative) in male voice respectively, Also, I obain a high quality approximation-synthesis waveforms within TSIUVC by using frequency information of 0.547kHz below and 2.813kHz above. This method has the capability of being applied to speech coding of low bit rate, speech analysis and speech synthesis.

• 음성과학
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• 제2권
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• pp.57-66
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• 1997

This paper studies the effects of the method, CMS(Cepstral Mean Subtraction), (which compensates for some of the speech distortion. caused by telephone channels), on the performance of the text-dependent speaker identification system. This system is based on the VQ(Vector Quantization) and HMM(Hidden Markov Model) method and chooses the LPC-Cepstrum and Mel-Cepstrum as the feature vectors extracted from the speech data transmitted through telephone channels. Accordingly, we can compare the correct recognition rates of the speaker identification system between the use of LPC-Cepstrum and Mel-Cepstrum. Finally, from the experiment results table, it is found that the Mel-Cepstrum parameter is proven to be superior to the LPC-Cepstrum and that recognition performance improves by about 10% when compensating for telephone channel using the CMS.

• 말소리와 음성과학
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• 제4권2호
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• pp.121-126
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• 2012

This study deals with how stuttering changes in its frequency in a situation where adult participants who stutter are exposed to one type of background noise, that is, multi-talker babble noise. Eight American English-speaking adults who stutter participated in this study. Each of the subjects read aloud sentences under each of three speaking conditions (i.e., typical solo reading (TSR), typical choral reading (TCR), and multi-talker babble noise reading (BNR)). Speech fluency was computed based on a percentage of syllables stuttered (%SS) and speaking rate was also assessed to examine if there was significant change in rates as a measure of vocal change under each of the speaking conditions. The study found that participants read more fluently both during BNR and during TCR than during TSR. The study also found that participants did not show significant changes in speaking rate across the three speaking conditions. Some discussion was provided in relation to the effect of multi-talker babble noise on the frequency of stuttering and its further speculation.