• The Journal of the Acoustical Society of Korea
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• v.13 no.5
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• pp.77-82
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• 1994

In speech signal processing, it is very important to detect the pitch exactly. Although various methods for detecting the pitch of speech signals have been developed, it is difficult to exactly extract the pitch for wide range of speakers and various utterances. Thus we propose a new pitch detection algorithm which takes advantage of the G-peak extraction. It is a method to detect the pitch period of the voiced signals by finding MZCI (maximum zero-crossing interval) of the G-peak which is defined as cut-off bandwidth rate of LPF (low pass filter). This algorithm performs robustly with a gross error rate of 3.63% even in 0 dB SNR environement. The gross error rate for clean speech is only 0.18%. Also it is able to process all courses with high speed.

• The Journal of the Acoustical Society of Korea
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• v.30 no.7
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• pp.390-395
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• 2011

This paper covers the studies on common voice parameters through the voice analysis of female main news anchors on weekday evening by the station, and differences of relative voices and sounds among stations. To examine voice characteristics, 6 voice parameters were analyzed and it showed anchors of each station had distinctive characteristics of voices and phonations over all fields except the speech rate, and there were also differences in sound systems. As major analysis parameters, basic pitch, tone of the 1st formant and pitch ratio, level of closeness by pitch bandwidth, type of sentence closing through average pitch position within pitch bandwidth, average speech rate, and acoustic tone analysis by energy distribution by frequency band were used. Analyzed values and results could be referred to and utilized in the criteria of phonation characteristics for domestic female news anchors.

• Proceedings of the Korean Institute of Communication Sciences Conference
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• 1994.06a
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• pp.139-142
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• 1994

• Proceedings of the Korean Society of Precision Engineering Conference
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• 2011.06a
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• pp.1025-1026
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• 2011

• The Journal of the Acoustical Society of Korea
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• v.30 no.4
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• pp.190-196
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• 2011

Bandwidth extension is a technique to improve speech quality and intelligibility, extending from 300-3400 Hz narrowband speech to 50-7000 Hz wideband speech. This paper designs an artificial bandwidth extension (ABE) module embedded in the AMR (adaptive multi-rate) decoder, reducing LPC/LSP analysis and algorithm delay of the ABE module. We also introduce a fast search codebook mapping method for ABE, and design a low power BWE technique based on the AMR decoder. The proposed ABE method reduces the computational complexity and the algorithm delay, respectively, by 28 % and 20 msec, compared to the traditional DTE (decode then extend) method. We also introduce a weighted classified codebook mapping method for constructing the spectral envelope of the wideband speech signal.

• Proceedings of the IEEK Conference
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• 1998.06a
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• pp.615-618
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• 1998

음성부호화의 주요목적은 대역 제한된 전송 대역폭에 전송을 하기위한 음성압축, 명료성과 자연성을 유지하는 고음질 음성합성, 그리고 처리 속도등의 요소에 따라 달라진다. 일반적으로 음성 부호화 방법은 파형 부호화범, 신호원 부화화법, 그리고 혼성 부호화법으로 나누어질 수 있다. 이러한 방법으로 전송되어진 음성은 다시 합성을 하게되는데, 이때 고음질을 유지할 수 있는 PSOLA법을 사용하였다. 본 논문에서 제안한 방법으로 전송되어진 음성은 다시 합성을 하게되는데, 이때 고음질에 유지 할 수 있는 PSOLA법을 사용하였다. 본 논문에서 제안한 방법은 기존의 PSOLA 합성법에서 사용되어지는 hanning 윈도우가 음성이 갖는 golttal wave shape의 특성에 적합하지 않다는 것을 이용하여 기존의 hanning 윈도우가 아닌 비대칭성을 가진 새로운 형태의 비대칭 윈도우(asymmetry window)를 제안하였다. 비대칭 윈도우의 형태는 위도우를 중심으로 왼쪽편은 기울기가 심하고, 오른쪽은 기울기가 완만하여 음성의 기울기에 적합한 웨이팅을 갖는 형태이다. 제안한 비대칭 윈도우를 사용하여 PSOLA 합성을 하였을 경우 SNR 2~3dB 정도 향상되었음을 알 수 있다.

• Proceedings of the Acoustical Society of Korea Conference
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• 1995.06a
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• pp.221-224
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• 1995

In speech signal processing, it is necessary to detect exactly the pitch. The algorithms of pitch extraction with have been proposed until now are difficult to detect pitches over wide range speech signals. In this paper, thus, we proposed a new pitch detection algorithm that used a low pass filter with variable bandwidth. It is the method that preprosses to find the first formant of speech signals by the FFT at each frame and detects the pitches for signals LPFed with the cut off frequency according to the first formant. Applying the method, we obtained the pitch contours, improving the accuracy of pitch detection in some noise environments.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.14 no.6
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• pp.699-712
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• 1989

This paper describes a method of Korean speech synhes is using format VOCODER. The parameters of speech synthes is are a follows, 1) format F1, F2, and F3 by spectrum moment method and F4, F5 using the length of vocal tract. 2) pitch frequencies obtained by optimu, Comb method using AMDF. 3) short time average energy and short time mean amplitude. 4) The decision method of bandwidth reportd by Fant. 5) voicde/unvoiced discrimination using zerocrossing. 6) excitation wave reported by Rosenberg. 7) gaussian white noise. Synthesis results are in fairly good agreement with original speech.

• Journal of Wind Energy
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• v.2 no.1
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• pp.74-81
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• 2011

This paper deals with the application of LQ control to the power curve tracking control of wind turbine. However, two more additional tasks are required to apply the LQR theory to wind turbine control. One is the tracking problem instead of regulation, because the wind turbine is controlled as variable speed and variable pitch. The other is LQ integral control., because the rotor speed should be tightly controlled without any steady state error. Starting from the analysis of wind characteristics, design requirement of a wind turbine control system is defined. A design procedure of LQ tracking with integral control is introduced. The performance of LQ tracking system is analyzed and evaluated by numeric simulation.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.7C
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• pp.962-970
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• 2004

This paper presents an efficient implementation of the 2.4 kbps EHSX(Enhanced Harmonic Stochastic Excitation) speech coder on a TMS320C6701$^{TM}$ floating-point digital signal processor. The EHSX speech codec is based on a harmonic and CELP(Code Excited Linear Prediction) modeling of the excitation signal respectively according to the frame characteristic such as a voiced speech and an unvoiced speech. In this paper, we represent the optimization methods to reduce the complexity for real-time implementation. The complexity in the filtering of a CELP algorithm that is the main part for the EHSX algorithm complexity can be reduced by converting program using floating-point variable to program using fixed-point variable. We also present the efficient optimization methods including the code allocation considering a DSP architecture and the low complexity algorithm of harmonic/pitch search in encoder part. Finally, we obtained the subjective quality of MOS 3.28 from speech quality test using the PESQ(perceptual evaluation of speech quality), ITU-T Recommendation P.862 and could get a goal of realtime operation of the EHSX codec.c.