• 인터넷정보학회논문지
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• 제19권6호
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• pp.1-7
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• 2018

본 논문에서는 합성 음성파형의 일그러짐을 제어하기 위하여 V/UV/S(Voiced / Unvoiced / Silence)의 스위칭을 사용하고, 피치구간마다 멀티펄스를 보정하며, 무성자음(Unvoiced)의 근사합성에 특정주파수를 이용하는 ACFBD-MPC(Amplitude Compensation Frequency Band Division - Multi Pulse Coding)와 LMS-MPC(Least Mean Square - Multi Pulse Coding)를 통합한 8kbps ACLMS-MPC(Amplitude Compensation and Least Mean Square - Multi Pulse Coding) 부호화 방식을 제안하고자 한다. 여러 방식을 통합하는데 있어서, 음성파형의 일그러짐을 줄이면서 유성음과 무성음의 비트율을 8kbps로 조정하는 것이 중요하다. 유성음과 무성음의 비트율을 8kbps로 조정함에 있어서, 개별피치를 이용하여 대표구간의 멀티펄스를 피치구간마다 복원함으로서 음성파형을 효율적으로 합성할 수 있다. 8kbps의 부호화 조건에서 ACLMS-MPC 방식을 구현하고 SNR를 평가한 결과, ACLMS-MPC의 SNR는 남자음성에서 15.0dB, 여자음성에서 14.3dB 임을 확인할 수 있었다. 따라서 ACLMS-MPC가 기존의 MPC, ACFBD-MPC, LMS-MPC에 비하여 남자음성에서 0.3dB~1.8dB, 여자음성에서 0.3dB~1.6dB 정도 개선된 것을 알 수 있었다. 이러한 방법들은 셀룰러폰이나 인터넷폰과 같이 낮은 비트율의 음원을 사용하여 음성신호를 부호화하는 방식에 활용할 수 있을 것으로 기대된다. 향후 멀티펄스 음원의 진폭과 위치를 동시에 보정하는 6.9kbps 음성부호화 방식의 음질평가를 수행하고자 한다.

• International journal of advanced smart convergence
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• 제6권4호
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• pp.19-25
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• 2017

After 40 years of age, the saliva glands are aged and the saliva is not made enough to cause xerostomia symptoms. Side effects such as hypertension medication or diuretics that the elderly take mainly can cause xerostomia syndrome. In addition, autoimmune diseases, diabetes, anemia, depression and other common diseases that cause xerostomia symptoms. If the saliva secretion is insufficient, tooth decay and gum disease are likely to occur, and the digestive ability of the saliva is also reduced due to the lack of amylase, which is a digestive element. Once the degenerated salivary gland is restored to its normal state, it is difficult to recover. In this paper, we give electrical stimulation to the masseter which is in contact with the large pituitary gland, and stimulate the salivary gland to the utmost by using speech recognition using words corresponding to oral gymnastics. Use the STM32F407VG to implement a system to relieve xerostomia.

• 한국통신학회논문지
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• 제10권4호
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• pp.159-167
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• 1985

男性話者가 發音한 韓國語 母音/아, 에, 이, 오, 우/의 聲門波를 직접 抽出하여 音聲에 따라 성문파가 각각 다르다는 것을 확인하였다. 具現한 5가지의 성문파로 母音을 다시 合成하여 聲門波形이 音聲合成에 미치는 영향을 波形的으로 비교하였다. 상문파의 모양, 개방시간과 폐쇄기간에 따라 合成音聲波形은 變化가 있었으며, 聲門波形이 合成音質向上의 중요 factor로 作用함을 알 수 있었다.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2003년도 신호처리소사이어티 추계학술대회 논문집
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• pp.45-48
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• 2003

In general, speech coding methods are classified into the following three categories: the waveform coding, the source coding and the hybrid coding. Fast speaking is possible to encode with a few information compared with slow speaking rate. In case of speaking rate, low frequency band is more important than high frequency band while listening. Speech vocoding technique is developing to way with low bit rate and complexity and high sound quality. the CELP type of vocoder support very good sound quality with low bit rate but these vocoders don't consider about the speaking rate. When we consider speaking rate and encode the frame depending on the speaking rate, the bit rate is able to reduce the bit rate than the conventional vocoder. We propose the technique to estimate the speaking rate and applied PSOLA technique in case of the frame of slow speaking rate. As a result of simulation bit rate can be reduced about 300 bps.

• 대한후두음성언어의학회지
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• 제19권2호
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• pp.101-106
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• 2008

The speech signal has been put into a form suitable for storage and analysis by computer, several different operation can be performed. Filtering, sampling and quantization are the basic operation in digiting a speech signal. The waveform can be displayed, measured and even edited, and spectra can be computed using methods such as the Fast Fourier Transform (FFT), Linear predictive Coding (LPC), Cepstrum and filtering. The digitized signal also can be used to generate spectrograms. The spectrograph provide major advantages to the study of speech. So, author introduces the basic techniques for the acoustic recording, digital signal processing and the principles of spectrum and spectrogram.

• 한국멀티미디어학회논문지
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• 제6권3호
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• pp.462-468
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• 2003

유성음원과 무성음원을 사용하는 음성부호화 방식에 있어서, 같은 프레임 안에 모음과 무성자음이 있는 경우에 음질저하 현상이 나타난다. 본 연구에서는 같은 프레임안에 유성음과 무성자음이 존재하지 않도록 FIR-STREAK 필터 와 zerocrossing rate을 이용한 개별피치 펄스를 사용하여 연속음성에서 무성자음을 포함한 천이구간(TSIUVC)을 탐색, 추출하는 방법을 제안한다. 또한 본 논문에서는 최송 자승법과 주파수 대역 분할을 이용한 TSIUVC 근사합성법을 제안하였다. 실험 결과, 0.547KHz 이하 2.813KHz 이상의 주파수 정보를 사용하여 TSIUVC 음성파형을 양호하게 근사합성할 수 있었으며, 최대 오차신호가 일그러짐이 적은 TSIUVC 근사합성 파형에 중요한 역할을 한다는 것을 알 수 있었다. 이 방법은 음성합성, 음성분석, 새로운 Voiced/Silence/TSIUVC의 음성부호화 방식에 활용할 수 있을 것으로 기대된다.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2009년도 정보 및 제어 심포지움 논문집
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• pp.37-39
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• 2009

In the research of speech recognition, locating the beginning and end of a speech utterance in a background of noise is of great importance. Since the background noise presenting to record will introduce disturbance while we just want to get the stationary parameters to represent the corresponding speech section, in particular, a major source of error in automatic recognition system of isolated words is the inaccurate detection of beginning and ending boundaries of test and reference templates, thus we must find potent method to remove the unnecessary regions of a speech signal. The conventional methods for speech endpoint detection are based on two simple time-domain measurements - short-time energy, and short-time zero-crossing rate, which couldn't guarantee the precise results if in the low signal-to-noise ratio environments. This paper proposes a novel approach that finds the Lyapunov exponent of time-domain waveform. This proposed method has no use for obtaining the frequency-domain parameters for endpoint detection process, e.g. Mel-Scale Features, which have been introduced in other paper. Comparing with the conventional methods based on short-time energy and short-time zero-crossing rate, the novel approach based on time-domain Lyapunov Exponents(LEs) is low complexity and suitable for Digital Isolated Word Recognition System.

• The Journal of the Acoustical Society of Korea
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• 제19권2E호
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• pp.8-13
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• 2000

In this paper, we present an adaptive formant tracking algorithm for speech using closed phase WRLS-VFF-VT method. The pitch synchronous closed phase methods is known to give more accurate estimates of the vocal tract parameters than the pitch asynchronous method. However the use of a pitch-synchronous closed phase analysis method has been limited due to difficulties associated with the task of accurately isolating the closed phase region in successive periods of speech. Therefore we have implemented the pitch synchronous closed phase WRLS-VFF-VT algorithm for speech analysis, especially for formant tracking. The proposed algorithm with the variable threshold(VT) can provide a superior performance in the boundary of phone and voiced/unvoiced sound. The proposed method is experimentally compared with the other method such as two channel CPC method by using synthetic waveform and real speech data. From the experimental results, we found that the block data processing techniques, such as the two-channel CPC, gave reasonable estimates of the formant/antiformant. However, the data windows used by these methods included the effects of the periodic excitation pulses, which affected the accuracy of the estimated formants. On the other hand the proposed WRLS-VFF-VT method, which eliminated the influence of the pulse excitation by using an input estimation as part of the algorithm, gave very accurate formant/bandwidth estimates and good spectral matching.

• 음성과학
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• 제9권4호
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• pp.237-252
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• 2002

MIT has been applied for nonfluent aphasic patients on the basis of lateralization of brain hemisphere. However, its applications for different languages have some inquiry for aphasic patients because of prosodic and rhythmic differences. The purpose of this study is to develop the Korean Melodic Intonation Therapy program using personal computer and its clinical effects for nonfluent aphasic patients. The algorithm was composed to voice analog signal, PCM, AMDF, Short-time autocorrelation function and center clipping. The main menu contains pitch, waveform, sound intensity and speech files on window. Aphasic patients' intonation patterns overlay on selected kMIT patterns. Three aphasic patients with or without kMIT training participated in this study. Four affirmative sentences and two interrogative sentences were uttered on CSL by stimulus of ST. VOT, VD, Hold and TD were measured on Spectrogram. In addition, articulation disorders and intonation patterns were evaluated objectively on spectrogram. The results indicated that nonfluent aphasic patients with kMIT training group showed some clinical effects of speech intelligibility based on VOT, TD values, articulation evaluation and prosodic pattern changes.

• 정보과학회 컴퓨팅의 실제 논문지
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• 제20권11호
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• pp.567-572
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• 2014

최근 네트워크와 컴퓨팅 기술의 발달로 정보기기가 소형화되고 이동성이 중요시되면서 간편하게 제어할 수 있는 음성 인식에 대한 수요가 증가하고 있다. 본 논문은 음성 인식 시스템의 일부로써 한국어 음소 중 모음 'ㅏ' 인식에 대한 연구 결과를 제시한다. 음소는 음성을 구성하고 있는 최소단위로서 음성을 인식하는데 매우 중요한 역할을 한다. 그러나 각각의 음소들을 정확하게 인식하려면 발음의 다양성 등으로 인해 많은 어려움이 존재한다. 본 논문에서는 한국어 음소 중 모음 'ㅏ'를 인식하기 위한 간단하고도 새로운 방식을 제안한다. 제안된 'ㅏ' 인식 휴리스틱은 파형 시퀀스의 공통 특징 추출을 기반으로 이루어졌으며, 이는 기존의 복잡한 방법에 비해 간단하면서도 실험 결과 90% 이상의 성공률로 'ㅏ'를 인식하는 것을 확인하였다.