• Journal of the Korean Institute of Intelligent Systems
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• v.14 no.6
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• pp.759-763
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• 2004

In this paper, real-time voice change method using pitch change technique is proposed to change one's voice to the other voice. For this purpose, sampling rate change method using DFT (Discrete Fourier Transform) method and time scale modification method using SOLA (Synchronized Overlap and Add) method is combined to change pitch. In order to evaluate the performance of the proposed method, voice transformation experiments were conducted. Experimental results showed that original speech signal is changed to the other speech signal in which original speaker's identity is difficult to find. The system is implemented using TI TMS320C6711DSK board to verify the system runs in real time.

• The Journal of the Acoustical Society of Korea
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• v.19 no.3
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• pp.11-19
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• 2000

This research deals with the implementation procedures of a chorus processing DSP for karaoke system. It is necessary to compress the chorus data to store as many choruses as we can. We apply MPEG-1 audio algorithm to compress the chorus data. And the chorus system must be accompanied with the karaoke that can change the key and the tempo. So the chorus DSP must be able to change the key and tempo of the chorus data. We apply SOLA (Synchronized Overlap and Add) to do it. We designed the chorus DSP that can compress the chorus, change the key and tempo. And we verified the chorus DSP logic using FPGA. The used FPGA are two FLEX10K100s made by ALTERA. Finally we make the ASIC chip of chorus DSP and verify its operation.

• The Journal of the Acoustical Society of Korea
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• v.7 no.4
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• pp.31-53
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• 1988

In this work we study extensively the structures and performance characteristics of the block least mean-square (BLMS) adaptive digital filters (ADF's) that can be realized efficiently using the fast Fourier transform (FFT). The weights of a BLMS ADF realized using the FFT can be adjusted either in the time domain or in the frequency domain, leading to the time-domain BLMS(TBLMS) algorithm or the frequency-domain BLMS (FBLMS) algorithm, respectively. In Part Ⅰof the paper, we first present new results on the overlap-add realization and the number-theoretic transform realization of the FBLMS ADF's. Then, we study how we can incorporate the concept of different frequency-weighting on the error signals and the self-orthogonalization of weight adjustment in the FBLMS ADF's , and also in the TBLMS ADF's. As a result, we show that the TBLMS ADF can also be made to have the same fast convergence speed as that of the self-orthogonalizing FBLMS ADF. Next, based on the properties of the sectioning operations in weight adjustment, we discuss unconstrained FBLMS algorithms that can reduce two FFT operations both for the overlap-save and overlap-add realizations. Finally, we investigate by computer simulation the effects of different parameter values and different algorithms on the convergence behaviors of the FBLMS and TBLMS ADF's. In Part Ⅱ of the paper, we will analyze the convergence characteristics of the TBLMS and FBLMS ADF's.

• Proceedings of the Acoustical Society of Korea Conference
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• 1998.08a
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• pp.339-343
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• 1998

Sinusoidal model 은 음성신호처리의 넓은 분야에 적용되고 있는 방법으로 고음질의 합성음을 생성해 낼 수 있고, 조작이 용이하다는 장점을 가지고 있다. 본 논문에서는 Analysis-by-synthesis/Overlap-add Sinusoidal model 이라는 방법을 이용하여 시간축 변환과 dam성 변환을 수행하였다. 특히 본 논문에서는 음질향상을 위하여 시간축 변환시에는 정적인 구간과 변화하는 구간을 구별하여 서로 다른 시간축 변환비를 이용하였고, 기존의 LPC 방법에 비해 스펙트럼 포락선을 보다 잘 추정하는 Improved Cepstrum을 이용하여 음정변환에 적용하였다. 또 서로 다른 문맥에서 얻어진 음성단위들을 결합할 때 생기는 위상차이를 극복하기 위하여, 기본주파수 성분이 일치하도록 시간축을 이동하여 합성하였다. 실험결과 본 논문에서 적용한 방법들을 통해 기존 방식에 비해 개선된 음질을 얻을 수 있었다.

• The Journal of the Acoustical Society of Korea
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• v.33 no.4
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• pp.262-267
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• 2014

Recently, long pulses are transmitted for target detection in active sonar application. Matched filtering implemented by simple convolution algorithm, requires massive computational power for long replica. The computational loads are reduced significantly by implementing the convolution in the frequency domain with overlap add method, but the performance degrades for specified input/output system delay which constrains the size of FFT function. For performance improvement, the replica could be partitioned into uniform blocks (FDL) by re-using IFFT operations, or variable blocks of increasing length (MC) by using the largest possible blocks to calculate the convolution. In this paper, by combining the strong points of the two methods, we propose a new filter partition structure that allows for further optimization of the previous two methods.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.4 no.2
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• pp.96-102
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• 2011

In this study, we performed algorithm development of automatic symptom degree for patient with hallux valgus one of the representative foot disease of morden. And this study proposes an efficient automated technique that is different from the original analog diagnosis for treatment and surgery of hallux valgus using digital image process. And we used X-Ray images of both a normal and a patient with hallux valgus in the procedure. First, we marked the standard angle on the X-Ray image of normal through Overlap & Add technique. Then we created a standard image through thinning filter and roberts filter(edge detection algorithm). Second, we used sobel filter of edge detection algorithm on the X-Ray image of patient. Moreover, we went another overlap & add technique procedure with both normal and patient image that we made. With the output, we projected the display detection image onto the screen. Finally, with the display detection image, we could measure and project the diagnosis angle of hallux valgus. And this confirms that this method is much more practical and applicable for another orthopedics disease than the prior one.

• Proceedings of the IEEK Conference
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• 1999.11a
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• pp.676-679
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• 1999

In this paper, we presents a method to control prosody of the synthetic speech using sampling rate conversion technique. In prosody control, the conventional methods perform overlap and add. So the synthetic speech has a distortion and the voice quality is not satisfied. Using sampling rate conversion technique, we can get high Qualify of the synthetic speech. Also we can control various talking speeds according to speaker's patterns.

• Proceedings of the IEEK Conference
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• 1999.11a
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• pp.1127-1130
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• 1999

In this paper, using one method of Additive Synthesis, Analysis-by-synthesis/Overlap-Add (ABS/OLA) method, analysis and synthesis of musical tones is processed. But peak detection of frequency domain is processed by proposed method considering the view of acoustics. It is that that harmonics frequency is times of main frequency. Using this fact, peak detection of frequency domain is useful for detection of tonal component identified musical note. It is possible to realize high-quality lour bit rate audio.

• The Journal of the Acoustical Society of Korea
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• v.7 no.4
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• pp.54-76
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• 1988

In Part Ⅰ of the paper, we have developed various block least mean-square (BLMS) adaptive digital filters (ADF's) based on a unified matrix treatment. In Part Ⅱ we analyze the convergence behaviors of the self-orthogonalizing frequency-domain BLMS (FBLMS) ADF and the unconstrained FBLMS (UFBLMS) ADF both for the overlap-save and overlap-add sectioning methods. We first show that, unlike the FBLMS ADF with a constant convergence factor, the convergence behavior of the self-orthogonalizing FBLMS ADF is governed by the same autocorrelation matrix as that of the UFBLMS ADF. We then show that the optimum solution of the UFBLMS ADF is the same as that of the constrained FBLMS ADF when the filter length is sufficiently long. The mean of the weight vector of the UFBLMS ADF is also shown to converge to the optimum Wiener weight vector under a proper condition. However, the steady-state mean-squared error(MSE) of the UFBLMS ADF turns out to be slightly worse than that of the constrained algorithm if the same convergence constant is used in both cases. On the other hand, when the filter length is not sufficiently long, while the constrained FBLMS ADF yields poor performance, the performance of the UFBLMS ADF can be improved to some extent by utilizing its extended filter-length capability. As for the self-orthogonalizing FBLMS ADF, we study how we can approximate the autocorrelation matrix by a diagonal matrix in the frequency domain. We also analyze the steady-state MSE's of the self-orthogonalizing FBLMS ADF's with and without the constant. Finally, we present various simulation results to verify our analytical results.

• Journal of KIISE:Information Networking
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• v.28 no.4
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• pp.540-549
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• 2001

Internet telephony service is very cheap and very easy to introduce the value-added service than the POTS, but is difficult to the QoS of telephone service. The existing Internet typically offers 'best effort' services only, which do not make any commitment about delay, packet loss and jitter. This paper compensates the low quality of the speech for packet loss or delay using SOLA algorithm in Internet phone. SOLA algorithm is a popular technique for Time Scale Modification of speech and audio signal. In the proposed algorithm, the receiver expands the received packet under resonable threshold, and hence compensates the QoS of speech. From the simulation, this algorithm can conceals packet loss considerably, and is also improved the quality of the speech.