• Journal of the Korean Society of Laryngology, Phoniatrics and Logopedics
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• v.9 no.2
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• pp.142-146
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• 1998

Background : It has been established that the fundamental frequency(Fo) of the vowels varies systemically as a function of vowel height. Specifically, high vowels have a higher Fo than low vowels. Two major explanations or hypotheses dominate contemporary accounts of fired to explain the mechanisms underlying intrinsic variation in vowel Fo, source-tract coupling hypothesis and tongue-pull hypothesis. Objectives : Total laryngectomy surgery necessiates removal of all structures between the hyoid bone and the tracheal rings. Therefore, the assumption that no direct interconnection exists between the tongue and pharyngoesophageal segment that would mediate systematic variation in vowel Fo appears quite reasonable. If tongue-pull hypothesis is correct, systemic differences in Fo between high versus low vowels produced by esophageal speakers would not Or expected. We analyzed the Fo in the vowels of esophageal voice. Materials and method : The subjects were 11 cases of laryngectomee patients with fluent esophageal voice. The five essential vowels were recorded and analyzed with computer speech analysis system(Computerized Speech Lab). The Fo was measured using acoustic waveform, automatically and manually, and narrow band spectral analysis. Results : The results of this study reveal that intrinsic variation in vowel Fo is clearly evident in esophageal speech. By analysis using acoustic waveform automatically, the signals were too irregular to measure the Fo precisely. So the data from automatic analysis of acoustic waveform is not logical. But the Fo by measuring with manually calculated acoustic waveform or narrowband spectral analysis resulted in acceptable results. These results were interpreted to support neither the source-tract coupling nor the tongue-pull hypotheses and led us to offer an alternative explanation to account for intrinsic variation of Fo.

• Proceedings of the Acoustical Society of Korea Conference
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• 1998.06c
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• pp.159-162
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• 1998

Speech synthesis coding is classified into three categories: waveform coding, source coding and hybrid coding. To obtain the synthetic speech with high quality, the synthesis by waveform coding is desired. However, it is difficult to apply waveform coding to synthesis by syllable or phoneme unit, because it does not divide the speech into excitation and formant component. Thus it is required to alter the excitation in waveform coding for applying waveform coding to synthesis by rule. In this paper we propose a new pitch alteration method that minimizes the spectrum distortion by using the behavior of cepstrum. This method splits the spectrum of speech signal into excitation spectrum and formant spectrum and transforms the excitation spectrum into cepstrum domain. The pitch of excitation cepstrum is altered by zero insertion or zero deletion and the pitch altered spectrum is reconstructed in spectrum domain. As a result of performance test, the average spectrum distortion was below 2.29%.

• Speech Sciences
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• v.12 no.1
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• pp.27-35
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• 2005

Differential instrumentation for the diagnoses of individuals with Cleft palate has been used to objectively measure speech problems. The Cepstrum Method was used to study the vocal tract transfer function. The vocal tract transfer function and the source spectrum should be considered in the evaluation of nasal resonance. The aim of this study was to collect quantitative data on the acoustic Instrumentation used for evaluating hypernasality. Normal subjects (9 male, 21 female; 37 male children, 20 female children) and individuals with VPI (13 male, 8 female; 16 male children, 9 female) participated in this study. The vowel /i/ was selected to gauge the severances of hypernasality Spectral and Cepstral studies using CSL was used to identify the acoustic characteristics. Cepstrum analysis shows significant differences in quefrency and amplitude. The quefrency of normal groups was shorter than that of the VPI groups, while the amplitude of normal groups was lower than that of the VPI groups. This may have significance in the evaluation 'of nasal resonance.

• MALSORI
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• no.45
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• pp.63-77
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• 2003

The purpose of voice color conversion is to change the speaker identity perceived from the speech signal. In this paper, we propose a new voice color conversion algorithm through the formant shifting and the spectrum-tilt modification in the frequency domain. The basic idea of this technique is to convert the positions of source formants into those of target speaker's formants through interpolation and decimation and to modify the spectrum-tilt by utilizing the information of both speakers' spectrum envelops. The LPC spectrum is adopted to evaluate the position of formant and the information of spectrum-tilt. Our algorithm enables us to convert the speaker identity rather successfully while maintaining good speech quality, since it modifies speech waveforms directly in the frequency domain.

• Journal of the Korea Computer Industry Society
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• v.5 no.2
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• pp.269-274
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• 2004

Noisy speech recognition is one of most important problems in speech recognition. In this paper, a method which efficiently removes the mixed noise with speech, is proposed. The proposed method is based on the ICA to separate the mixed noise. ICA(Independent component analysis) is a signal processing technique, whose goal is to express a set of random variables as linear combinations of components that are statistically as independent from each other as possible.

• 전자공학회논문지 IE
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• v.49 no.2
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• pp.9-17
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• 2012

One of the problems on the application of the speech recognition system in the real world is the degradation of the performance by acoustical distortions. The most important source of acoustical distortion is the additive noise. This paper describes a spectral compensation technique based on a spectral peak enhancement scheme followed by an efficient noise subtraction scheme for noise robust speech recognition. The proposed methods emphasize the formant structure and compensate the spectral tilt of the speech spectrum while maintaining broad-bandwidth spectral components. The recognition experiments was conducted using noisy speech corrupted by white Gaussian noise, car noise, babble noise or subway noise. The new technique reduced the average error rate slightly under high SNR(Signal to Noise Ratio) environment, and significantly reduced the average error rate by 1/2 under low SNR(10 dB) environment when compared with the case of without spectral compensations.

• Phonetics and Speech Sciences
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• v.6 no.4
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• pp.91-99
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• 2014

Speaking styles tend to have an influence on spectral characteristics of produced speech. There are not many studies on the spectral characteristics of speech because of complicated processing of too much spectral data. The purpose of this study was to examine spectral characteristics and formant bandwidths of English vowels produced by nine American males with different speaking styles: clear or conversational styles; high- or low-pitched voices. Praat was used to collect pitch-corrected long-term averaged spectra and bandwidths of the first two formants of eleven vowels in the speaking styles. Results showed that the spectral characteristics of the vowels varied systematically according to the speaking styles. The clear speech showed higher spectral energy of the vowels than that of the conversational speech while the high-pitched voice did the same over the low-pitched voice. In addition, front and back vowel groups showed different spectral characteristics. Secondly, there was no statistically significant difference between B1 and B2 in the speaking styles. B1 was generally lower than B2 when reflecting the source spectrum and radiation effect. However, there was a statistically significant difference in B2 between the front and back vowel groups. The author concluded that spectral characteristics reflect speaking styles systematically while bandwidths measured at a few formant frequency points do not reveal style differences properly. Further studies would be desirable to examine how people would evaluate different sets of synthetic vowels with spectral characteristics or with bandwidths modified.

• The KIPS Transactions:PartB
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• v.9B no.2
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• pp.223-230
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• 2002

In a speech coding system using excitation source of voiced and unvoiced, it would be involves a distortion of speech waveform in case coexist with a voiced and an unvoiced consonants in a frame. This paper present a new method of TSIUVC (Transition Segment Including Unvoiced Consonant) approximate-synthesis by using Least Mean Square. The TSIUVC extraction is based on a zero crossing rate and IPP (Individual Pitch Pulses) extraction algorithm using residual signal of FIR-STREAK Digital Filter. As a result, This method obtain a high Quality approximation-synthesis waveform by using Least Mean Square. The important thing is that the frequency signals in a maximum error signal can be made with low distortion approximation-synthesis waveform. This method has the capability of being applied to a new speech coding of Voiced/Silence/TSIUVC, speech analysis and speech synthesis.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.7 no.6
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• pp.1175-1179
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• 2006

In a speech coding system using excitation source of voiced and unvoiced, it would be a distortion of speech waveform in case of exist a voiced and an unvoiced consonants in a frame. In this paper, I propose a new method of TSIUVC(Transition Segment Including Unvoiced Consonant) approximate-synthesis by using Least Mean Square. As a result, a method by using Least Mean Square was obtained a high quality approximation-synthesis waveform . The important thing is that the frequency signals in a maximum error signal can be made with low distortion approximation-synthesis waveform. This method has the capability of being applied to a new speech coding of Voiced/Silence/TSIUVC, speech analysis and synthesis.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.17 no.7
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• pp.49-53
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• 2016

Recently, the use of signal compression methods to improve the efficiency of wireless networks have increased. In particular, the MPC system was used in the pitch extraction method and the excitation source of voiced and unvoiced to reduce the bit rate. In general, the MPC system using an excitation source of voiced and unvoiced would result in a distortion of the synthesis speech waveform in the case of voiced and unvoiced consonants in a frame. This is caused by normalization of the synthesis speech waveform in the process of restoring the multi-pulses of the representation segment. This paper presents an ACFBD-MPC (Amplitude Compensation Frequency Band Division-Multi Pulse Coding) using amplitude compensation in a multi-pulses each pitch interval and specific frequency to reduce the distortion of the synthesis speech waveform. The experiments were performed with 16 sentences of male and female voices. The voice signal was A/D converted to 10kHz 12bit. In addition, the ACFBD-MPC system was realized and the SNR of the ACFBD-MPC estimated in the coding condition of 8kbps. As a result, the SNR of ACFBD-MPC was 13.6dB for the female voice and 14.2dB for the male voice. The ACFBD-MPC improved the male and female voice by 1 dB and 0.9 dB, respectively, compared to the traditional MPC. This method is expected to be used for cellular telephones and smartphones using the excitation source with a low bit rate.