• The Journal of the Acoustical Society of Korea
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• v.29 no.1
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• pp.41-47
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• 2010

The pitch alteration technique for voice conversion is classified in time domain, frequency domain and hybrid domain. The Hybrid domain method has a merit of clearness and natural-ness of pitch altered speech but has the major drawback of long processing time. In this paper, we proposed a new method that can reduce the processing time of pitch alteration in time-frequency hybrid domain. We omitted the bit-reversing process of FFT and IFFT in changing the processing domain. Therefore we can reduce the processing time by 86.26% to the conventional method with same quality.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.47 no.5
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• pp.100-108
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• 2010

An autocorrelation method is used in pitch estimation. Autocorrelation values in time and frequency domains, which have different characteristics, correspond to the pitch period and fundamental frequency, respectively. We utilize an integrated autocorrelation method in time and frequency domains. It can remove the errors of pitch doubling and having. In the time and frequency domains, pitch period and fundamental frequency have reciprocal relation to each other. Especially, fundamental frequency estimation ends up as an error because of the resolution of FFT. To reduce these artifacts, interpolation methods are applied in the integrated autocorrelation domain, which decreases pitch errors. Moreover, only for the pitch candidates found in a time domain, the corresponding frequency-domain autocorrelation values are calculated with reduced computational complexity. Using linear interpolation, we can decrease the required number of FFT coefficients by 8 times. Thus, compared to the conventional methods, computational complexity can be reduced by 9.5 times.

• The Journal of the Acoustical Society of Korea
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• v.20 no.3
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• pp.34-41
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• 2001

In an IMBE (Improved Multi-band Excitation) speech coder, initial pitch estimation occupies most of the total computing time for the coder due to complex cost function and exhaustive search over candidate pitches. Future frames in initial pitch estimation cause inevitable time delay. Therefore, it is difficult to implement a real-time coder. Furthermore, unvoiced frames use the unnecessary pitch estimation as in the voiced frames. In this paper, each frame is determined voiced or unvoiced by Dyadic Wavelet Transform (DyWT) and, then, initial pitch estimation is performed only for voiced frame. Therefore different pitch estimation algorithms are employed between voiced and unvoiced frames incurring reduced time delay at transmitter and receiver. Simulation result show that the relative complexity of initial pitch estimation is reduced by 23％, and the processing time decreases down to 1/10 ∼ 1/1l of the IMBE coder while speech quality is almost maintained.

• Proceedings of the Acoustical Society of Korea Conference
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• 1998.06c
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• pp.155-158
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• 1998

IMBE(Improved Multi-Band Excitation) vocoders exhibit good performance at low data rates. The major drawback to IMBE coders is their large computational requirements. In this paper, thus, we propose a new pitch search method that preserves the quality of the IMBE vocoder with reduced complexity. The basic idea is to reduce computation complexity of the pitch searching by using the SAMDF. Applying the proposed method to the IMBE vocoder, we can get approximately 52.02% searching time reduction in the pitch search. There is no difference in voice quality between conventional IMBE and proposed IMBE.

• Speech Sciences
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• v.12 no.1
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• pp.107-119
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• 2005

The current study considers that the distinctive acoustic properties and variations in the closed plateau are realized by four different pitch accents(/Hi*_ H*/ or /Hi*_$h*_f$ for an emphatic phrase and /hi*_ H*/ for a neutral phrase) in an intermediate phrase in the French intonational structure. Thus, an attempt is made to define the acoustic property of the CP in the ip according to the duration time and pitch range, while different combinations of the four pitch accents of the CP are used to explain the way a speaker will highlight. The duration time of the CP was measured at about 0.67 sec. for males and 0.75 sec. for females. The duration properties of the plateau in the CP were found to control the pitch range based on two different prominent pitch accents, which appeared in more than two APs. Therefore, the ip was identified as having a hierarchical level in the French intonational structure, along with the AP and IP. In addition, the CP in the ip was used as a specific location to explain the pragmatic meaning of the rhythm using the two acoustic factors and different combinations of the four pitch accents.

• Journal of the Institute of Convergence Signal Processing
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• v.7 no.4
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• pp.178-183
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• 2006

As a lot of researches on the speech signal processing are performed due to the recent rapid development of the information-communication technology. the pitch period is used as an important element to various speech signal application fields such as the speech recognition. speaker identification. speech analysis. or speech synthesis. A variety of algorithms for the time and the frequency domains related with such pitch period detection have been suggested. One of the pitch detection algorithms for the time domain. AMDF (average magnitude difference function) uses distance between two valley points as the calculated pitch period. However, it has a problem that the algorithm becomes complex in selecting the valley points for the pitch period detection. Therefore, in this paper we proposed the modified AMDF(M-AMDF) algorithm which recognizes the entire minimum valley points as the pitch period of the speech signal by using the rotation transform of AMDF. In addition, a threshold is set to the beginning portion of speech so that it can be used as the selection criteria for the pitch period. Moreover the proposed algorithm is compared with the conventional ones by means of the simulation, and presents better properties than others.

• The Journal of the Acoustical Society of Korea
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• v.27 no.1E
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• pp.11-16
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• 2008

In this paper, a pitch cycle waveform (PCW) is extracted, quantized, and interpolated in a time domain to synthesize high-quality speech at low bit rates. The pre-alignment technique is proposed for the accurate and efficient PCW extraction, which predicts the current PCW position from the previous PCW position assuming that pitch periods evolve slowly. Since the pitch periods are different frame by frame, the original PCW is converted into the fixed-dimension PCW using the dimension-conversion method, and subsequently quantized by code-excited linear predictive (CELP) coding. The excitation signal for the linear predictive coding (LPC) synthesis filter is generated using the time-domain interpolation and interlink of the quantized PCW's. The coder operates at 4.2 kbit/s and 3.2 kbit/s depending on the pitch period. Informal listening test demonstrates the effectiveness of the proposed coding scheme.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.147-150
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• 2002

This paper describes procedures to implement a real-time pitch analyzer using Matlab. Matlab is a multi-purpose signal-processing tool. Using this tool real-time analysis tool is implemented. To make it real-time we used data acquisition toolbox which comes with Matlab. Autocorrelation method was used as a basic algorithm. The resulting pitch informations are displayed in two different forms, i.e. instantaneous pitch plot and pitch track. V/UV decision is performed using zero crossing rate and energy Informations based on 500 utterances.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• v.9 no.2
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• pp.1019-1022
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• 2005

As recent information communication technology is rapidly developed, a lot of researches related to speech signal processing have been processed. So pitch period is applied as important factor to many application fields such as speech recognition, speaker identification, speech analysis and synthesis. Therefore, many algorithms related to pitch detection have been proposed in time domain and frequency domain and AMDF(average magnitude difference function) which is one of pitch detection algorithms in time domain chooses time interval from valley to valley as pitch period. But, in selection of valley point to detect pitch period, complexity of the algorithm is increased. So in this paper we proposed pitch detection algorithm using rotation transform of AMDF, that taking the global minimum valley point as pitch period and established a threshold about the phoneme in beginning portion, to exclude pitch period selection. and compared existing methods with proposed method through simulation.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.16 no.6 s.111
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• pp.627-633
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• 2006

The tire is considered as one of the important noise sources having an influence on vehicle's performance. The Pattern noise of a tire is the transmission sound of airborne noise. On smooth asphalt road, Pattern noise is amplified with the velocity. In recent, the study on the reduction of Pattern noise is energetically processed. Pattern noise is strongly related with pitch sequence. To reduce the pattern noise, tire's designer has to randomize the sequence of pitch. The FFT is a traditional method to evaluate the level of the randomization of the pitch sequence, but gives no information on time-varying, instantaneous frequency. In the study, we found that Time-Frequency transform is a useful method to non-stationary signal such as tire noise.