• Archives of Plastic Surgery
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• v.41 no.5
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• pp.472-479
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• 2014

Background Speech problems are a common clinical feature of the 22q11.2 deletion syndrome. The objectives of this study were to inventory the speech history and current self-reported speech rating of adolescents and young adults, and examine the possible variables influencing the current speech ratings, including cleft palate, surgery, speech and language therapy, intelligence quotient, and age at assessment. Methods In this cross-sectional cohort study, 50 adolescents and young adults with the 22q11.2 deletion syndrome (ages, 12-26 years, 67% female) filled out questionnaires. A neuropsychologist administered an age-appropriate intelligence quotient test. The demographics, histories, and intelligence of patients with normal speech (speech rating=1) were compared to those of patients with different speech (speech rating>1). Results Of the 50 patients, a minority (26%) had a cleft palate, nearly half (46%) underwent a pharyngoplasty, and all (100%) had speech and language therapy. Poorer speech ratings were correlated with more years of speech and language therapy (Spearman's correlation=0.418, P=0.004; 95% confidence interval, 0.145-0.632). Only 34% had normal speech ratings. The groups with normal and different speech were not significantly different with respect to the demographic variables; a history of cleft palate, surgery, or speech and language therapy; and the intelligence quotient. Conclusions All adolescents and young adults with the 22q11.2 deletion syndrome had undergone speech and language therapy, and nearly half of them underwent pharyngoplasty. Only 34% attained normal speech ratings. Those with poorer speech ratings had speech and language therapy for more years.

• MALSORI
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• no.62
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• pp.85-96
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• 2007

In this paper, we carried out recognition experiments for noisy speech having various levels of car noise and output of an audio system using the speech interface. The speech interface consists of three parts: pre-processing, acoustic echo canceller, post-processing. First, a high pass filter is employed as a pre-processing part to remove some engine noises. Then, an echo canceller implemented by using an FIR-type filter with an NLMS adaptive algorithm is used to remove the music or speech coming from the audio system in a car. As a last part, the MMSE-STSA based speech enhancement method is applied to the out of the echo canceller to remove the residual noise further. For recognition experiments, we generated test signals by adding music to the car noisy speech from Aurora 2 database. The HTK-based continuous HMM system is constructed for a recognition system. Experimental results show that the proposed speech interface is very promising for robust speech recognition in a noisy car environment.

• Journal of the Korean Institute of Telematics and Electronics S
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• v.34S no.5
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• pp.86-93
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• 1997

With recent development of speech recognition technology and multimedia computer systems, more potential applications of voice will become a reality. In this paper, we implement speech interface on the windows95 environment for practical use fo multimedia computers with voice. Speech interface is made up of three modules, that is, speech input and detection module, speech recognition module, and application module. The speech input and etection module handles th elow-level audio service of win32 API to input speech data on real time. The recognition module processes the incoming speech data, and then recognizes the spoken command. DTW pattern matching method is used for speech recognition. The application module executes the voice command properly on PC. Each module of the speech interface is designed and examined on windows95 environments. Implemented speech interface and experimental results are explained and discussed.

• Phonetics and Speech Sciences
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• v.6 no.1
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• pp.71-76
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• 2014

In this paper voice quality of normal speech is qualitatively classified by five components of breathy, creaky, rough, nasal, and thin/thick voice. To determine whether a correlation exists between a subjective measure of voice and an objective measure of voice, each voice is perceptually evaluated using the 1/2/3 scale by speech processing specialists and acoustically analyzed using speech analysis tools such as the Praat, MDVP, and VoiceSauce. The speech parameters include features related to speech source and vocal tract filter. Statistical analysis uses a two-independent-samples non-parametric test. Experimental results show that statistical analysis identified a significant correlation between the speech feature parameters and the components of voice quality.

• MALSORI
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• no.49
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• pp.63-75
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• 2004

Most existing telephone speech transmitted in current public networks is band-limited to 0.3-3.4 kHz. Compared with wideband speech(0-8 kHz), the narrowband speech lacks low-band (0-0.3 kHz) and high-band(3.4-8 kHz) components of sound. As a result, the speech is characterized by the reduced intelligibility and a muffled quality, and degraded speaker identification. Bandwidth extension is a technique to provide wideband speech quality, which means reconstruction of low-band and high-band components without any additional transmitted information. Our new approach considers to exploit harmonic synthesis method for reconstruction of low-band speech over the CELP coded speech. A spectral distortion measurement and listening test are introduced to assess the proposed method, and the improvement of synthesized speech quality was verified.

• Speech Sciences
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• v.7 no.1
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• pp.17-29
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• 2000

The STC(Sinusoidal Transform Coding) is a vocoding technique that uses a sinusoidal speech model to obtain high- quality speech at low data rate. It models and synthesizes the speech signal with fundamental frequency and its harmonic elements in frequency domain. To reduce the data rate, it is necessary to represent the sinusoidal amplitudes and phases with as small number of peaks as possible while maintaining the speech quality. As a basic research to develop a low-rate speech coding algorithm using the sinusoidal model, in this paper, we investigate the speech quality depending on the number of sinusoids. By varying the number of spectral peaks from 5 to 40 speech signals are reconstructed, and then their qualities are evaluated using spectral envelope distortion measure and MOS(Mean Opinion Score). Two approaches are used to obtain the spectral peaks: one is a conventional STFT (Short-Time Fourier Transform), and the other is a multiresolutional analysis method.

• Proceedings of the KSPS conference
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• 2006.05a
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• pp.101-104
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• 2006

The present study examined two acoustic characteristics(duration and intensity) of vowels produced by 4 cerebral palsic adults and 4 nondisabled adults in conversational and clear speech. In this study, clear speech means: (1) slow one's speech rate just a little, (2) articulate all phonemes accurately and increase vocal volume. Speech material included 10 bisyllabic real words in the frame sentences. Temporal-acoustic analysis showed that vowels produced by two speaker groups in clear speech(in this case, more accurate and louder speech) were significantly longer than vowels in conversational speech. In addition, intensity of vowels produced by cerebral palsic speakers in clear speech(in this case, more accurate and louder speech) was higher than in conversational speech.

• Speech Sciences
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• v.3
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• pp.167-176
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• 1998

Conventional time-scale modification methods have the problem that as the modification rate gets higher the time-scale modified speech signal becomes less intelligible, because they ignore the effect of articulation rate on speech characteristics. Results of research on speech perception show that the timing information of transient portions of a speech signal plays an important role in discriminating among different speech sounds. Inspired by this fact, we propose a novel scheme for modifying the time-scale of speech. In the proposed scheme, the timing information of the transient portions of speech is preserved, while the steady portions of speech are compressed or expanded somewhat excessively for maintaining overall time-scale change. In order to identify the transient and steady portions of a speech signal, we employ a simple method using LPC cepstral distance between neighboring frames. The result of the subjective preference test indicates that the proposed method produces performance superior to that of the conventional SOLA method, especially for very fast playback case.

• MALSORI
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• no.47
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• pp.109-120
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• 2003

This paper presents about the activities of speech database standardization in ETRI. Recently, with the support of government, ETRI and SiTEC have been gathering the large speech corpus for the domestic speech related companies. First, due to the lack of sharing the knowledge of speech database specification, the distributed speech database has a different format. Hence it seems to be needed to have the same format as soon as possible. ETRI and SiTEC are trying to find the better representation format of speech database. Second, we introduce a new description method of the annotation information of speech database. As one of the structured description method, XML based description will be applied to represent the metadata of the speech database. It will be continuously revised through the speech technology standard forum during this year.

• Speech Sciences
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• v.8 no.4
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• pp.45-58
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• 2001

Modulated components of a speech signal are frequently used for speech coding, speech recognition, and speech synthesis. Time-frequency representation (TFR) reveals some information about instantaneous frequency, instantaneous bandwidth and boundary of each component of the considering speech signal. In many cases, the extraction of AM-FM components corresponding to instantaneous frequencies is difficult since the Fourier spectra of the components with time-varying instantaneous frequency are overlapped each other in Fourier frequency domain. In this paper, an efficient method decomposing speech signal into AM-FM components is proposed. A variable bandwidth filter is developed for the decomposition of speech signals with time-varying instantaneous frequencies. The variable bandwidth filter can extract AM-FM components of a speech signal whose TFRs are not overlapped in timefrequency domain. Also, amplitude and instantaneous frequency of the decomposed components are estimated by using Hilbert transform.