• 제목/요약/키워드: speech rates

검색결과 271건 처리시간 0.02초

Temporal Variation Due to Tense vs. Lax Consonants in Korean

  • Yun, II-Sung
    • 음성과학
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    • 제11권3호
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    • pp.23-36
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    • 2004
  • Many languages show reverse durational variation between preceding vowel and following voiced/voiceless (lax/tense) consonants. This study investigated the likely effects of phoneme type (tense vs. lax) on the timing structure (duration of syllable, word, phrase and sentence) of Korean. Three rates of speech (fast, normal, slow) applied to stimuli with the target word /a-Ca/ where /C/ is one of /p, p', $p^h$/. The type (tense/lax) of /C/ caused marked inverse durational variations in the two syllables /a/ and /Ca/ and highly different durational ratios between them. Words with /p', $p^h$/ were significantly longer than that with /p/, which contrasts with many other languages where such pairs of words have a similar duration. The differentials between words remained up to the phrase and sentence level, but in general the higher linguistic units did not statistically differ within each level. Thus, the phrase is suggested as a compensatory unit of phoneme type effects in Korean. Different rates did not affect the general tendency. Distribution of time variations (from normal to fast and slow) to each syllable (/a/ and /Ca/) was also observed.

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음질향상 기법과 모델보상 방식을 결합한 강인한 음성인식 방식 (A Robust Speech Recognition Method Combining the Model Compensation Method with the Speech Enhancement Algorithm)

  • 김희근;정용주;배건성
    • 음성과학
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    • 제14권2호
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    • pp.115-126
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    • 2007
  • There have been many research efforts to improve the performance of the speech recognizer in noisy conditions. Among them, the model compensation method and the speech enhancement approach have been used widely. In this paper, we propose to combine the two different approaches to further enhance the recognition rates in the noisy speech recognition. For the speech enhancement, the minimum mean square error-short time spectral amplitude (MMSE-STSA) has been adopted and the parallel model combination (PMC) and Jacobian adaptation (JA) have been used as the model compensation approaches. From the experimental results, we could find that the hybrid approach that applies the model compensation methods to the enhanced speech produce better results than just using only one of the two approaches.

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음성명령에 의한 모바일로봇의 실시간 무선원격 제어 실현 (Real-Time Implementation of Wireless Remote Control of Mobile Robot Based-on Speech Recognition Command)

  • 심병균;한성현
    • 한국생산제조학회지
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    • 제20권2호
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    • pp.207-213
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    • 2011
  • In this paper, we present a study on the real-time implementation of mobile robot to which the interactive voice recognition technique is applied. The speech command utters the sentential connected word and asserted through the wireless remote control system. We implement an automatic distance speech command recognition system for voice-enabled services interactively. We construct a baseline automatic speech command recognition system, where acoustic models are trained from speech utterances spoken by a microphone. In order to improve the performance of the baseline automatic speech recognition system, the acoustic models are adapted to adjust the spectral characteristics of speech according to different microphones and the environmental mismatches between cross talking and distance speech. We illustrate the performance of the developed speech recognition system by experiments. As a result, it is illustrated that the average rates of proposed speech recognition system shows about 95% above.

Word-boundary and rate effects on upper and lower lip movements in the articulation of the bilabial stop /p/ in Korean

  • Son, Minjung
    • 말소리와 음성과학
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    • 제10권1호
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    • pp.23-31
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    • 2018
  • In this study, we examined how the upper and lower lips articulate to produce labial /p/. Using electromagnetic midsagittal articulography, we collected flesh-point tracking movement data from eight native speakers of Seoul Korean (five females and three males). Individual articulatory movements in /p/ were examined in terms of minimum vertical upper lip position, maximum vertical lower lip position, and corresponding vertical upper lip position aligned with maximum vertical lower lip position. Using linear mixed-effect models, we tested two factors (word boundary [across-word vs. within-word] and speech rate [comfortable vs. fast]) and their interaction, considering subjects as random effects. The results are summarized as follows. First, maximum lower lip position varied with different word boundaries and speech rates, but no interaction was detected. In particular, maximum lower lip position was lower (e.g., less constricted or more reduced) in fast rate condition and across-word boundary condition. Second, minimum lower lip position, as well as lower lip position, measured at the time of maximum lower lip position only varied with different word boundaries, showing that they were consistently lower in across-word condition. We provide further empirical evidence of lower lip movement sensitive to both different word boundaries (e.g., linguistic factor) and speech rates (e.g., paralinguistic factor); this supports the traditional idea that the lower lip is an actively moving articulator. The sensitivity of upper lip movement is also observed with different word boundaries; this counters the traditional idea that the upper lip is the target area, which presupposes immobility. Taken together, the lip aperture gesture is a good indicator that takes into account upper and lower lip vertical movements, compared to the traditional approach that distinguishes a movable articulator from target place. Respective of different speech rates, the results of the present study patterned with cross-linguistic lenition-related allophonic variation, which is known to be more sensitive to fast rate.

파킨슨 환자의 클리어 스피치 전후 음향학적 공기역학적 특성 (An aerodynamic and acoustic characteristics of Clear Speech in patients with Parkinson's disease)

  • 신희백;고도홍
    • 말소리와 음성과학
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    • 제9권3호
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    • pp.67-74
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    • 2017
  • An increase in speech intelligibility has been found in Clear Speech compared to conversational speech. Clear Speech is defined by decreased articulation rates and increased frequency and length of pauses. The objective of the present study was to investigate improvement in immediate speech intelligibility in 10 patients with Parkinson's disease (age range: 46 to 75 years) using Clear Speech. This experiment has been performed using the Phonatory Aerodynamic System 6600 after the participants read the first sentence of a Sanchaek passage and the "List for Adults 1" in the Sentence Recognition Test (SRT) using casual speech and Clear Speech. Acoustic and aerodynamic parameters that affect speech intelligibility were measured, including mean F0, F0 range, intensity, speaking rate, mean airflow rate, and respiratory rate. In the Sanchaek passage, use of Clear Speech resulted in significant differences in mean F0, F0 range, speaking rate, and respiratory rate, compared with the use of casual speech. In the SRT list, significant differences were seen in mean F0, F0 range, and speaking rate. Based on these findings, it is claimed that speech intelligibility can be affected by adjusting breathing and tone in Clear Speech. Future studies should identify the benefits of Clear Speech through auditory-perceptual studies and evaluate programs that use Clear Speech to increase intelligibility.

효율적인 음성신호의 전송을 위한 4배속 가변 변환율 ADPCM기법 및 DSP를 이용한 실시간 구현 (Variable Quad Rate ADPCM for Efficient Speech Transmission and Real Time Implementation on DSP)

  • 한경호
    • 조명전기설비학회논문지
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    • 제18권1호
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    • pp.129-136
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    • 2004
  • 본 논문에서는 ADPCM을 이용하여 보다 효율적인 음성전송을 위한 4배속 가변임계값을 갖는 음성 부호화법을 제안하였으며 TMS320C6711-DSP를 사용하여 제안된 알고리즘의 실시간 음성 처리 구현을 다루었다. 본 논문에서는 ADPCM 알고리즘을 개선하여 입력 신호 및 주변 환경에 따라 변환율을 16[kbps], 24[kbps], 32[kbps], 40[kbps]로 가변하는 음성부호화 방법을 제시하고 이를 DSP를 이용하여 Encoding과 Decoding을 실시간으로 구현하여 좋은 음질의 음성 신호를 보다 적은 비트 수로 전송하고자 하였다. 이를 위하여 영교차율(Zero Crossing Rate)을 이용하여 소음의 우세한 정도를 구분하여, 임계값을 가변 시키도록 하였으며 소음이 우세한 환경에서는 4가지 변환율 가운데 낮은 변환율을 사용하도록 하여 전송 비트 수를 줄이도록 하였으며, 소음이 열세한 환경에서는 높은 변환율을 사용하도록 하여 좋은 음질을 갖도록 하였다. 음성 대화의 많은 구간이 거의 묵음인 구간이 많으므로, 음질은 40[kbps] 수준의 음질을 유지하면서 이 보다 낮은 데이터 비트수를 유지할 수 있음을 시뮬레이션과 실험으로 보였다. 구현에 사용된 TMS320C6711-DSK board는 128[K]의 플래쉬 메모리를 가지고 있고 1333MIPS의 처리속도를 가지므로 제안된 알고리즘을 실시간 구현하기에 충분한 조건을 가지고 있다.

자유 대화에서의 한국어 원어민 화자와 한국어 고급 학습자들의 발화 속도 비교 (A Comparative Study on the Speech Rate of Advanced Korean(L2) Learners and Korean Native Speakers in Conversational Speech)

  • 홍민경
    • 한국어교육
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    • 제29권3호
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    • pp.345-363
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    • 2018
  • The purpose of this study is to compare the speech rate of advanced Korean(L2) learners and Korean native speakers in spontaneous utterances. Specifically, the current study investigated the difference of the two groups' speech pattern according to utterance length. Eight advanced Korean(L2) learners and eight Korean native speakers participated in this study. The data were collected by recording their conversation and physical measurements (speaking rate, articulatory rates, pause and several types of speech disfluency) were taken on extracted 120 utterances from 12 out of the 16 participants. The findings show that advanced Korean learners' speech pattern is similar to that of Koreans in the short-length utterance. However, in the long-length utterance, two groups show different speech patterns; while the articulatory rate of Korean native speakers increased in the long-length utterance, that of Korean learners decreased. This suggests that the frequency of speech disfluency factors might affect this result.

서울 코퍼스 20대 남성의 성대진동 개시시간 연구 (A study on the voice onset times of the Seoul Corpus males in their twenties)

  • 이유리;윤규철
    • 말소리와 음성과학
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    • 제8권4호
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    • pp.1-8
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    • 2016
  • The purpose of this work is to examine the voice onset times (VOTs) of the three types of plosives from the Seoul Corpus male speakers in their twenties. In addition, the factors known to affect VOTs were analyzed, including the place and manner of articulation, speakers, location in words, type of following vowels and speech rates calculated from the three consecutive words. Much of the findings agreed with those from earlier studies on Korean and other languages and new discoveries were made.

이동형 단말기를 위한 다채널 입력 기반 비정상성 잡음 제거기 (Multi-channel input-based non-stationary noise cenceller for mobile devices)

  • 정상배;이성독
    • 한국지능시스템학회논문지
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    • 제17권7호
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    • pp.945-951
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    • 2007
  • 잡음의 제거는 음성을 인터페이스로 하는 기기들에 필수적이라고 할 수 있다. 실질적으로 통화 품질이나 음성 인식률은 음성 입력부의 주변에서 들어오는 원치 않는 가산성 잡음에 의해서 크게 열화된다. 본 논문에서는 기본적으로 두 개의 마이크로폰을 이용한 잡음제거 방법을 제안한다. 마이크를 여러 개 사용했을 때의 장점은 방향 정보를 이용할 수 있다는 것인데 이는 사람 목소리, 음악 소리 등의 비정상성 잡음을 제거하는 데에 유용하다. 제안된 잡음제거 알고리즘은 위너필터에 기반 한다고 볼 수 있다. 위너필터에 의한 잡음제거를 위해서는 검출하고자 하는 음성과 제거하고자 하는 잡음의 주파수 응답이 동시에 추정 가능해야 한다. 이를 위해서 주파수 영역에서 스펙트럼 분류를 시행하여 위너필터 기반의 잡음제거에 필요한 정보를 얻는다. 제안된 알고리즘을 이용한 성능은 잘 알려진 프로스트 (Frost) 알고리즘 및 적응 모드 컨트롤러를 갖는 generalized sidelobe canceller (GSC)와 비교하였다. 성능의 지표로는 객관적 음질 평가의 방법 중에서 널리 쓰이고 있는 perceptual evaluation of speech quality (PESQ) 및 음성 인식률이 사용되었다.

VoIP 코더들의 프레임손실은닉 알고리즘 성능평가 (Performance Evaluation of Frame Erasure Concealment Algorithms in VoIP Coders)

  • 한승호;문광;한민수
    • 대한음성학회:학술대회논문집
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    • 대한음성학회 2004년도 춘계 학술대회 발표논문집
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    • pp.235-238
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    • 2004
  • Frame erasures cause speech quality degradation in wireless communication networks or packet networks. The degradation becomes worse when consecutive frame erasures occur. Speech coders have a frame erasure concealment(FEC) mechanism to compensate for frame erasures. It is meaningful to evaluate the performance of FEC mechanisms for frame erasures that occur in communications networks. In this paper, various frame erasures are designed. And the FEC algorithms of speech coders are evaluated and analyzed with the Perceptual Evaluation of Speech Quality(PESQ). It is found that the performances vary in accordance with frame erasure types, frame erasure rates, and utterance lengths.

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