• Journal of IKEEE
• /
• v.25 no.4
• /
• pp.697-702
• /
• 2021

Fast Fourier Transform (FFT) is a major signal processing block being widely used. For long-point FFT processing, usually more than 1024 points, its low-complexity implementation becomes very important while retaining high SQNR (Signal-to-Quantization Noise Ratio). In this paper, we present a low-complexity FFT algorithm with a simple dynamic scaling scheme. For the 2048-point pipelined FFT processing, we can reduce the number of general multipliers by half compared to the well-known radix-2 algorithm. Also, the table size for twiddle factors is reduced to 35% and 53% compared to the radix-2 and radix-2² algorithms respectively, while achieving SQNR of more than 55dB without increasing the internal wordlength progressively.

• Journal of the Institute of Electronics Engineers of Korea TC
• /
• v.43 no.12 s.354
• /
• pp.9-15
• /
• 2006

In this paper, signal processing techniques for noncoherent impulse-radio-based UWB (IR-UWB) communication system are proposed to provide system implementation of low power consumption and low complexity. The proposed system adopts a simple modulation technique of OOK (on-oft-keying) and noncoherent signal detection based on signal amplitude. In particular, a technique of a novel high speed digital sampler using a stable, lower reference clock is developed to detect nano-second pulses and recover digital signals from the pulses. Also, a 32 bits Turyn code for data frame synchronization and a convolution code as FEC are applied, respectively. To verify the proposed signal processing techniques for low power, low complexity noncoherent IR-UWB system, the proposed signal processing technique is implemented in FPGA and then a short-range communication system for wireless transmission of high quality MP3 data is designed and tested.

• Journal of Information Processing Systems
• /
• v.18 no.1
• /
• pp.97-114
• /
• 2022

Interest is increasing in electrocardiogram (ECG) signal analysis for embedded devices, creating the need to develop an algorithm suitable for a low-power, low-memory embedded device. Linear approximation of the ECG signal facilitates the detection of fiducial points by expressing the signal as a small number of vertices. However, dynamic programming, a global optimization method used for linear approximation, has the disadvantage of high complexity using memoization. In this paper, the calculation area and memory usage are improved using a linear approximated template. The proposed algorithm reduces the calculation area required for dynamic programming through local optimization around the vertices of the template. In addition, it minimizes the storage space required by expressing the time information using the error from the vertices of the template, which is more compact than the time difference between vertices. When the length of the signal is L, the number of vertices is N, and the margin tolerance is M, the spatial complexity improves from O(NL) to O(NM). In our experiment, the linear approximation processing time was 12.45 times faster, from 18.18 ms to 1.46 ms on average, for each beat. The quality distribution of the percentage root mean square difference confirms that the proposed algorithm is a stable approximation.

• Journal of the Institute of Convergence Signal Processing
• /
• v.19 no.2
• /
• pp.42-46
• /
• 2018

Adaptive signal processing is quite important in various signal and communication environments. In adaptive signal processing methods since the least mean square(LMS) algorithm is simple and robust, it is used everywhere. As the step is varied in the variable step(VS) LMS algorithm, the fast convergence speed and the small excess mean square error can be obtained. Various variable step LMS algorithms are researched for better performances. But in some of variable step LMS algorithms the computational complexity is quite large for better performances. The fixed step LMS algorithm with a low computational complexity merit and the variable step LMS algorithm with a fast convergence merit are combined in the proposed sporadic step algorithm. As the step is sporadically updated, the performances of the variable step LMS algorithm can be maintained in the low update rate using Fibonacci sequence. The performances of the proposed variable step LMS algorithm are proved in the adaptive equalizer.

• Transactions on Electrical and Electronic Materials
• /
• v.18 no.6
• /
• pp.320-322
• /
• 2017

In this paper, a high-performance, low-area gradient-magnitude calculator architecture is proposed, based on approximate image processing. To reduce the computational complexity of the gradient-magnitude calculation, vector properties, the symmetry axis, and common terms were applied in a hardware-resource-shared architec-ture. The proposed gradient-magnitude calculator was implemented using an Altera Cyclone IV FPGA (EP4CE115F29) and the Quartus II v.16 device software. It satisfied the output-data quality while reducing the logic elements by 23% and the embedded multipliers by 76%, compared with previous work.

• Journal of the Korean Institute of Telematics and Electronics S
• /
• v.35S no.4
• /
• pp.118-127
• /
• 1998

In this paper, we propose a voice personality transformation method which makes one person's voice sound like another person's voice. In order to transform the voice personality, vocal tract transfer function is used as a transformation parameter. Comparing with previous methods, the proposed method can obtain high-quality transformed speech with low computational complexity. Conversion between the vocal tract transfer functions is implemented by a linear mapping based on soft clustering. In this process, mean LPC cepstrum coefficients and mean removed LPC cepstrum modeled by the low dimensional vector are used as transformation parameters. To evaluate the performance of the proposed method, mapping rules are generated from 61 Korean words uttered by two male and one female speakers. These rules are then applied to 9 sentences uttered by the same persons, and objective evaluation and subjective listening tests for the transformed speech are performed.

• Proceedings of the Acoustical Society of Korea Conference
• /
• spring
• /
• pp.275-278
• /
• 2004

In TV sound system, loudspeakers are subject to severe size constraints. The small size of the transducer affects the low frequency signal performance of the system. Bass signal performance contributes significantly to the user perceived sound quality and a good bass signal reproduction is essential. Increasing the sound energy in the bass signal range is an unviable solution since the gain required are exceedingly high and signal distortion occurs because of the speaker overload. Recently methods are being proposed to invoke low frequency illusion using psychoacoustic phenomena of the missing fundamental. This paper proposes a simple and effective signal processing method to create bass signal illusion in TV speakers using the missing fundamental effect, at a complexity of 12 MIPS on Motorola 56371 audio DSP.

• Proceedings of the Korea Institute of Convergence Signal Processing
• /
• 2001.06a
• /
• pp.25-28
• /
• 2001

A new decision-feedback equalization technique for a filter bank-based orthogonal frequency division multiplexing (OFDM) data transmission system operating in a frequency selective multipath fading channel is presented in this paper. At the cost of relatively increased computational complexity in comparison to the conventional OFDM systems, the proposed system achieves a better performance in terms of bit error rates. The simulation results confirm the superiority and robustness of our method, particularly, in the low SNR channel environment.

• ETRI Journal
• /
• v.31 no.2
• /
• pp.240-242
• /
• 2009

A computationally efficient implementation of the progressive edge-growth algorithm is presented. This implementation uses an array of red-black (RB) trees to manage the layered structure of check nodes and adopts a new strategy to expand the Tanner graph. The complexity analysis and the simulation results show that the proposed approach reduces the computational effort effectively. In constructing a low-density parity check code with a length of $10^4$, the RB-tree-array-based implementation takes no more 10% of the time required by the original method.

• ETRI Journal
• /
• v.45 no.3
• /
• pp.492-504
• /
• 2023

Researchers have recently shown an increased interest in estimating the direction-of-arrival (DOA) of wideband noncircular sources, but existing studies have been restricted to subspace-based methods. An off-grid sparse recovery-based algorithm is proposed in this paper to improve the accuracy of existing algorithms in low signal-to-noise ratio situations. The covariance and pseudo covariance matrices can be jointly represented subject to block sparsity constraints by taking advantage of the joint sparsity between signal components and bias. Furthermore, the estimation problem is transformed into a single measurement vector problem utilizing the focused operation, resulting in a significant reduction in computational complexity. The proposed algorithm's error threshold and the Cramer-Rao bound for wideband noncircular DOA estimation are deduced in detail. The proposed algorithm's effectiveness and feasibility are demonstrated by simulation results.