• Proceedings of the IEEK Conference
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• 2002.06d
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• pp.187-190
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• 2002

본 논문에서는 지금까지 소개된 44.1KHz compact disc (CD)에서 48KHz digital audio tape (DAT)로의 표본화율 변환기법들에 대해서 가청 주파수 대역에서 100dB 이상의 dynamic range와 ±5x10­4dB 이하의 리플 크기를 유지할 수 있도록 각 기법들을 재설계하였으며, 메모리 요구량 및 계산량에 대해서 살펴보고자한다.

• Journal of Broadcast Engineering
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• v.11 no.1 s.30
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• pp.138-141
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• 2006

This paper evaluates the computational efficiency of sample-rate converters with rational factors in multi-stage structure in terms of memory requirement and multiplications per second. We describe resolution preserving and mutual prime conditions, and then present a method for designing the converter from which optimal rational-valued conversion factors for each stage can be yielded directly. As an example, we show an implementation of the 44.1-to-48KHz sample-rate converter in 2-stage structure.

• 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.

• Journal of Broadcast Engineering
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• v.9 no.4 s.25
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• pp.384-390
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• 2004

In this paper we compare the performance of 4 different algorithms for converting the sampling frequency of an audio from 44.1KHz to 48KHz. The algorithms considered here include the basic polyphase method. sine function based method. multi-stage method. and B-spline based method. For a fair comparison, the sampling rate converters using the 4 algorithms are redesigned under a high fidelity condition. Then, their H/W complexities are compared in terms of the computational complexity and the memory size. As a result, it is shown that the basic polyphase method and sine function based method outperform the other two in terms of the computational complexity, while the B-spline based method requires less memory than the others.

• 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.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.13 no.4
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• pp.179-191
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• 2013

The quincunx lattice is a non-separable sampling method in image processing. It treats the different directions more homogeneously and good frequency property than the separable two dimensional schemes. The high density discrete wavelet transformation is one that expands an N point signal to M transform coefficients with M > N. In two dimensions, this transform outperforms the standard discrete wavelet transformation in terms of shift-invariant. Although the transformation utilizes more wavelets, sampling rates are high costs. This paper proposed the high density discrete wavelet transform using quincunx sampling, which is a discrete wavelet transformation that combines the high density discrete transformation and non-separable processing method, each of which has its own characteristics and advantages. Proposed wavelet transformation can service good performance in image processing fields.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.38A no.12
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• pp.1030-1038
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• 2013

The proposed Alamouti coded OFDM effectively cancels Inter Carrier Interference (ICI) due to frequency offset between distributed antennas. The conventional Alamouti coded OFDM schemes to mitigate ICI utilize N-point Inverse Fast Fourier Transform/Fast Fourier Transform (IFFT/FFT) operations for OFDM modulation and demodulation processes with total N subcarriers. However, the performance degrades because ICI is also repeated in N periods due to the property of N-point IFFT/FFT operation. In order to avoid this problem, null data are used at the subcarriers with large ICI and thus, data rate decreases. The proposed scheme employs 2N-point IFFT/FFT instead of N-point IFFT/FFT in order to increase sampling rate. By increasing sampling rate, the amount of interference significantly decreases because the period of ICI also increases. The proposed scheme increases the data rate and improves the performance by reducing amount of ICI and the number of null-data. Furthermore, the gain of the performance and data rate of the proposed scheme is significant with higher modulation such as 16-Quadarature Amplitude Modulation (QAM) or 64-QAM.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.39A no.12
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• pp.746-755
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• 2014

Compressive sensing (CS) has drawn much interest as a new sampling technique that enables signals to be sampled at a much lower than the Nyquist rate. By noting that the block-based compressive sensing can still keep spatial correlation in measurement domain, in this paper, we propose a novel encoding technique for measurement data obtained in the block-based CS of natural image. We apply discrete wavelet transform (DWT) to decorrelate CS measurements and then assign a proper quantization scheme to those DWT coefficients. Thus, redundancy of CS measurements and bitrate of system are reduced remarkably. Experimental results show improvements in rate-distortion performance by the proposed method against two existing methods of scalar quantization (SQ) and differential pulse-code modulation (DPCM). In the best case, the proposed method gains up to 4 dB, 0.9 dB, and 2.5 dB compared with the Block-based CS-Smoothed Projected Landweber plus SQ, Block-based CS-Smoothed Projected Landweber plus DPCM, and Multihypothesis Block-based CS-Smoothed Projected Landweber plus DPCM, respectively.