• Phonetics and Speech Sciences
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• v.6 no.4
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• pp.171-179
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• 2014

Purposes: Speech acceptability means the subjective judgement of listeners regarding the naturalness and normality of the speech. The purpose of this study was to determine the predicting variables for speech acceptabilities of children with cochlear implants. Methods: Twenty seven children with CI participated. They had profound pre-lingual hearing loss without any additional disabilities. The mean of chronological ages was 8;9, and mean of age of implantation was 2;11. Speech samples of reading and spontaneous speech were recorded separately. Twenty college students who were not familiar to the speech of deaf children evaluated the speech acceptabilities using visual analog scale. 1 segmental (articulation) and 6 suprasegmental features (pitch, loudness, quality, resonance, intonation, and speaking rate) of speech were perceptually evaluated by 3 SLPs. Correlation and multiple regression analysis were performed to evaluate the predicting variables. Results: The means of speech acceptability for reading and spontaneous speech were 73.47 and 71.96, respectively. Speech acceptability of reading was predicated by the severity of intonation and articulation. Speech acceptability of spontaneous speech was predicated by the severity of intonation and loudness. Discussion and conclusion: Severity of intonation was the most effective variable to predict the speech acceptabilities of both reading and spontaneous speech. A further study would be necessary to generalize the result and to apply this result to intervention in clinical settings.

• Journal of rehabilitation welfare engineering & assistive technology
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• v.3 no.1
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• pp.27-34
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• 2009

Hearing impaired person's hearing loss has personally various shape, so existing symmetrical auditory filter of frequency band method wasn't properly simulated the hearing impaired person's various hearing loss shape. The shapes of auditory filter are asymmetrical different with each center frequency and each input level. Hearing impaired person which has hearing loss was differently changed with that of normal hearing people and it has different value for speech of quality through auditory filter. In this study, the asymmetrical auditory filter was simulated and then some tests to estimate the filter's performance objectively were performed. The experiment as simulated auditory filter's performance evaluation method used perceptual evaluation of speech quality (PESQ) and log likelihood ratio (LLR) for speech through auditory filter. In the test, processed speech was evaluated objective speech quality and distortion using PESQ and LLR value. When hearing loss processed, PESQ and LLR value have big difference between symmetrical and asymmetrical auditory filter. It means that the difference of the shape auditory filter may affect to speech quality. Especially, when hearing loss existed, auditory filter changing according to asymmetrical shape for each center frequency affected to perceive speech quality of the hearing impaired.

• Journal of Korea Multimedia Society
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• v.7 no.7
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• pp.896-904
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• 2004

In this paper, We present an efficient variable rate speech coder for spectral efficiency and packet processing technique for packet loss compensation of a voice codec with 10msec frame in VoIP service. Through disconnecting the users from the spectral resource during silence interval of about 60% period, a variable rate voice coder based on a voice activity detection(VAD) can increase spectral gain by two times. The performance of the method was analyzed by variation of detected voice activity factor and degraded speech frame ratio under various background noise level, and compared those of G.729B of ITU-T 8kbps standard speech codec. A method to compensate lost packets utilized addition of recovery data to a main stream and error concealment scheme for speech quality enhancement, the performance is verified by reconstructed speech quality. The proposed scheme can achieve spectral gain by two times or enhance speech quality by 3dB through reserved bandwidth of VAD. Therefore, the proposed method can enhance a spectral efficiency or speech quality of VoIP.

• The Journal of the Korea Contents Association
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• v.5 no.3
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• pp.167-173
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• 2005

In a speech 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 unvoiced consonants in a frame. So, I propose TSIUVC(Transition Segment Including Unvoiced Consonant) extraction method by using pitch pulses and Zero Crossing Rate in order to unexistent with a voiced and unvoiced consonants in a frame. And this paper present a TSIUVC approximate-synthesis method by using frequency band division. As a result, this method obtains a high quality approximation-synthesis waveform within TSIUVC by using frequency information of 0.547kHz below and 2.813kHz above. And the TSIUVC extraction rate was $91\%$ for female voice and $96.2\%$ for male voice respectively This method has the capability of being applied to a new speech coding of Voiced/Silence/TSIUVC, speech analysis, and speech synthesis.

• Phonetics and Speech Sciences
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• v.5 no.4
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• pp.33-41
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• 2013

This study investigated the perceptual and cepstral/spectral characteristics of phonation and their relationships in dysarthria in connected speech. Twenty-two participants were divided into two groups; the eleven dysarthric speakers were paired with matching age and gender healthy control participants. A perceptual evaluation was performed by three speech pathologists using the GRBAS scale to measure the cepstrual/spectral characteristics of phonation between the two groups' connected speech. Correlations showed dysarthric speakers scored significantly worse (with a higher rating) with severities in G (overall dysphonia grade), B (breathiness), and S (strain), while the smoothed prominence of the cepstral peak (CPPs) was significantly lower. The CPPs were significantly correlated with the perceptual ratings, including G, B, and S. The utility of CPPs is supported by its high relationship with perceptually rated dysphonia severity in dysarthric speakers. The receiver operating characteristic (ROC) analysis showed that the threshold of 5.08 dB for the CPPs achieved a good classification for dysarthria, with 63.6% sensitivity and the perfect specificity (100%). Those results indicate the CPPs reliably distinguished between healthy controls and dysarthric speakers. However, the CPP frequency (CPP F0) and low-high spectral ratio (L/H ratio) were not significantly different between the two groups.

• Journal of Internet Computing and Services
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• v.7 no.4
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• pp.57-66
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• 2006

In this paper, I propose a new method of Multi-Pulse Speech Coding(FBD-MPC: Frequency Band Division MPC) by using TSIUVC(Transition Segment Including UnVoiced Consonant) searching, extraction and approximation-synthesis method in a frequency domain. As, a result. the extraction rates of TSIUVC are 84.8%(plosive), 94.9%(fricative) and 92.3%(affricative) in female voice, 88%(plosive), 94.9%(fricative) and 92.3%(affricative) in male voice respectively. Also, I obtain a high quality approximation-synthesis waveforms within TSIUVC by using frequency information of 0.547kHz below and 2.813kHz above. I evaluate MPC by using switching information of voiced/unvoiced and FBD-MPC by using switching information of voiced/Silence/TSIUVC. As, a result, I knew that synthesis speech of FBD-MPC was better in speech quality than synthesis speech of the MPC.

• Journal of the Institute of Electronics and Information Engineers
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• v.52 no.7
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• pp.119-124
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• 2015

Speech enhancement has been required as a preprocessor for a noise robust speech recognition system. Codebook-based Speech Enhancement (CBSE) is highly robust in nonstationary noise environments compared with conventional noise estimation algorithms. However, its performance is severely degraded for the codevector combinations that have lower correlation with the input signal since CBSE depends on the trained codebook information. To overcome this problem, only the reliable codevector combinations are selected to be used to remove the codevector combinations that have lower correlation with input signal. The proposed method produces the improved performance compared to the conventional CBSE in terms of Log-Spectral Distortion (LSD) and Perceptual Evaluation of Speech Quality (PESQ).

• Speech Sciences
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• v.9 no.3
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• pp.77-86
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• 2002

The purpose of this study was to investigate the effects of Sea Dong-Il's technique on voice quality in patients with vocal nodules and phonasthenia (vocal fatigue). Ten patients (4 nodules and 6 vocal fatigue) participated in the study. Each subject was assessed acoustically (Fo, Jitter, Shimmer, NNE) in the first and last session. Dr. Speech (version 3.4, Tiger-DRS) was used to compare acoustic parameters of pre-and post-treatment. Sea Dong-Il's technique consisted of breathing exercise, relaxation exercise, and phonation exercise. The results were as follows: First, Sea Dong- Il's technique tended to be effective on improving voice quality in patients with phonasthenia and vocal nodules. Second, the nature of improvements were as follows: there was a significant difference between pre-and post-treatment in shimmer (p < .01) and NNE (p < .001), while there was no significant difference between pre-and post-treatment in Fo and Jitter. Finally, given the fact that the number of subjects was only 10, the jitter might have shown a significant difference if more subjects participated in the experiment.

• Proceedings of the KSPS conference
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• 2007.05a
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• pp.307-310
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• 2007

In this paper, a packet loss concealment (PLC) algorithm for CELP-type speech coders is proposed to improve the quality of decoded speech under a burst packet loss condition. The proposed algorithm is based on the recovery of voiced excitation using an estimate of the voicing probability and the generation of random excitation by permutating the previously decoded excitation. The voicing probability is estimated from the correlation using the previous correctly decoded excitation and pitch. The proposed algorithm is implemented as a PLC algorithm for G.729 and its performance is compared with PLC employed in G.729 by means of perceptual evaluation of speech quality (PESQ) and an A-B preference test under the random and burst packet losses with rates of 3% and 5%. It is shown that the proposed algorithm provides better speech quality than the PLC of G.729, especially under burst pack losses.

• MALSORI
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• no.49
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• pp.95-105
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• 2004

An automatic pronunciation correction system provides learners with correction guidelines for each mispronunciation. For this purpose we develope a speech recognizer which automatically classifies pronunciation errors when Koreans speak a foreign language. In order to develope the methods for automatic assessment of pronunciation quality, we propose a language model based score as a machine score in the speech recognizer. Experimental results show that the language model based score had higher correlation with human scores than that obtained using the conventional log-likelihood based score.