• Phonetics and Speech Sciences
• /
• v.4 no.4
• /
• pp.19-27
• /
• 2012

When two speakers read a text together, the produced speech has been shown to reduce a high degree of variability (e.g., pause duration and placement, and speech rate). This paper provides a quantitative analysis of speech rate variation exhibited in synchronous speech by examining the global and local patterns in two dialects of Mandarin Chinese (Taiwan and Shanghai). We analyzed the speech data in terms of mean speech rate and the reference of "Just Noticeable difference (JND)" within a subject and across subjects. Our findings show that speakers show lower and less variable speech rates when they read a text synchronously than when they read alone. This global pattern is observed consistently across speakers and dialects maintaining the unique local variation patterns of speech rate for each dialect. We conclude that paired speakers lower their speech rates and decrease the variability in order to ensure the synchrony of their speech.

• Speech Sciences
• /
• v.10 no.2
• /
• pp.85-95
• /
• 2003

In speech signal processing, it is very important to detect the pitch exactly in speech recognition, synthesis and analysis. If we exactly pitch detect in speech signal, in the analysis, we can use the pitch to obtain properly the vocal tract parameter. It can be used to easily change or to maintain the naturalness and intelligibility of quality in speech synthesis and to eliminate the personality for speaker-independence in speech recognition. In this paper, we proposed a new pitch detection algorithm. First, positive center clipping is process by using the incline of speech in order to emphasize pitch period with a glottal component of removed vocal tract characteristic in time domain. And rough formant envelope is computed through peak-fitting spectrum of original speech signal infrequence domain. Using the roughed formant envelope, obtain the smoothed formant envelope through calculate the linear interpolation. As well get the flattened harmonics waveform with the algebra difference between spectrum of original speech signal and smoothed formant envelope. Inverse fast fourier transform (IFFT) compute this flattened harmonics. After all, we obtain Residual signal which is removed vocal tract element. The performance was compared with LPC and Cepstrum, ACF. Owing to this algorithm, we have obtained the pitch information improved the accuracy of pitch detection and gross error rate is reduced in voice speech region and in transition region of changing the phoneme.

• The Journal of the Korea institute of electronic communication sciences
• /
• v.15 no.5
• /
• pp.807-812
• /
• 2020

This paper proposes a blind speech source separation algorithm using a microphone to separate only the target speech source signal in an environment in which various speech source signals are mixed. The proposed algorithm is a model of frequency domain representation based on independent component analysis method. Accordingly, for the purpose of verifying the validity of independent component analysis in the frequency domain for two speech sources, the proposed algorithm is executed by changing the type of speech sources to perform speech sources separation to verify the improvement effect. It was clarified from the experimental results by the waveform of this experiment that the two-channel speech source signals can be clearly separated compared to the original waveform. In addition, in this experiments, the proposed algorithm improves the speech source separation performance compared to the existing algorithms, from the experimental results using the target signal to interference energy ratio.

• Proceedings of the KSPS conference
• /
• 2003.10a
• /
• pp.29-32
• /
• 2003

This paper proposes an automatic pronunciation correction system which provides users with correction guidelines for each pronunciation error. For this purpose, we develop an HMM speech recognizer which automatically classifies pronunciation errors when Korean speaks foreign language. And, we collect speech database of native and nonnative speakers using phonetically balanced word lists. We perform analysis of mispronunciation types from the experiment of automatic mispronunciation detection using speech recognizer.

• Phonetics and Speech Sciences
• /
• v.9 no.4
• /
• pp.91-97
• /
• 2017

This study categorizes several voice characteristics by subjective listening assessment, and investigates correlation between voice characteristics and speech feature parameters. A model was developed to classify voice characteristics into the defined categories using SVM algorithm. To do this, we extracted various speech feature parameters from speech database for men in their 20s, and derived statistically significant parameters correlated with voice characteristics through ANOVA analysis. Then, these derived parameters were applied to the proposed SVM model. The experimental results showed that it is possible to obtain some speech feature parameters significantly correlated with the voice characteristics, and that the proposed model achieves the classification accuracies of 88.5% on average.

• Journal of the Korean Institute of Telematics and Electronics
• /
• v.27 no.4
• /
• pp.484-490
• /
• 1990

In speech coding scheme, because information bits have different error sensitivities over channel errors, the channel coder for combining with speech coding should be realized by the variable coding rate considering the bit importance of speech information bits. In realizing the 4 kbps channel coder for 12kbps speech, this paper have chosen the channel coding method by analyzing the hard-decision post-decoding error rate of RCPC(Rate Compatible Punctured Convolutional) codes and bit error sensitivity of 12 kbps speech. Under the coherent QPSK and Rayleigh fading channel, the performance analysis has showed that 10dB gain was obtained in speech SEGSNR by 4-level uneuqal error protection, which was compared with the caseof no channel coding at 7dB channel SNR.

• Phonetics and Speech Sciences
• /
• v.3 no.2
• /
• pp.89-96
• /
• 2011

In this paper, details of the design and implementation of a voice source simulator using Simulink and Matlab are discussed. This simulator is an implementation by model-based design concept. Voice sources can be analyzed and manipulated through various factors by choosing options from GUI input and selecting pre-defined blocks or user created ones. This kind of simulation tool can simplify the procedure of analyzing speech signals for various purposes such as voice quality analysis, pathological voice analysis, and speech coding. Also, basic analysis functions are supported to compare the original signal and the manipulated ones.

• The Journal of the Acoustical Society of Korea
• /
• v.17 no.4E
• /
• pp.41-47
• /
• 1998

Microphone is the first link in the speech recognition system. Depending on its type and mounting position, the microphone can significantly distort the spectrum and affect the performance of the speech recognition system. In this paper, characteristics of the speech signal for different microphones and microphone distances are investigated both in time and frequency domains. In the time domain analysis, the average signal-to-noise ration is measure ration is measured for the database we collected depending on the microphones and microphone distances. Mel-frequency spectral coefficients and mel-frequency cepstrum are computed to examine the spectral characteristics. Analysis results are discussed with our findings, and the result of recognition experiments is given.

• Phonetics and Speech Sciences
• /
• v.6 no.3
• /
• pp.57-62
• /
• 2014

The present study was performed to investigate and discuss the aerodynamic characteristics of whispered and normal speech during reading paragraph tasks. 39 normal females(18-23 yrs.) read 'Autumn' paragraph with whispered and normal phonation. Their readings were recorded and analyzed by 'Running Speech' in Phonatory Aerodynamic System(PAS) instrument. As results, during whispered speech, the total duration was longer and the numbers of inspiration were more frequently shown than normal speech. The Peak expiratory and inspiratory rate were higher in normal speech, but the expiratory and inspiratory volume were higher in whispered speech. By correlation analysis, both whispered and normal speech showed significantly high correlation between total duration and expiratory/inspiratory airflow duration; numbers of inspiration and inspiratory airflow duration; expiratory and inspiratory volume. These results show that whispered speech needs more respiratory effort but shows poorer aerodynamic efficacy during phonation than normal speech.

• The Transactions of the Korean Institute of Electrical Engineers D
• /
• v.50 no.12
• /
• pp.590-596
• /
• 2001

The pre-emphasis is an essential process for speech signal processing. Widely used two methods are the typical method using a fixed value near unity and te optimal method using the autocorrelation ratio of the signal. This study proposes a new pre-emphasis method using the short-term energy difference of speech signal, which can effectively compensate the glottal source characteristics and lip radiation characteristics. Using the proposed pre-emphasis, speech analysis, such as spectrum estimation, formant detection, is performed and the results are compared with those of the conventional two pre-emphasis methods. The speech analysis with 5 single vowels showed that the proposed method enhanced the spectral shapes and gave nearly constant formant frequencies and could escape the overlapping of adjacent two formants. comparison with FFT spectra had verified the above results and showed the accuracy of the proposed method. The computational complexity of the proposed method reduced to about 50% of the optimal method.