• Proceedings of the IEEK Conference
• /
• 2000.06b
• /
• pp.250-253
• /
• 2000

In this paper, we Implement a new arithmetic computation for MPEG audio data to overcome the limitations of real number processing in the fixed-point arithmetics, such as: overheads in processing time and power consumption. We aims at efficiently dealing with real numbers by extending the fixed-point arithmetic manipulation for floating-point numbers in MPEG audio data, and implementing the DSP libraries to support the manipulation and computation of real numbers with the fixed-point resources.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.25 no.8B
• /
• pp.1460-1468
• /
• 2000

The goal of this research is the real-time implementation of MPEG-1 Layer III audio decoder using the fixed-point digital signal processor of TMS320C6201 The main job for this work is twofold: one is to convert floating-point operation in the decoder into fixed-point operation while maintaining the high resolution, and the other is to optimize the program to make it run in real-time with memory size as small as possible. We, especially, devote much time to the descaling module in the decoder for conversion of floating-point operation into fixed-point operation with high accuracy. The inverse modified cosine transform(IMDCT) and synthesis polyphase filter bank modules are optimized in order to reduce the amount of computation and memory size. After the optimization process, in this paper, the implemented decoder uses about 26% of maximum computation capacity of TMS320C6201. The program memory, data ROM, data RAM used in the decoder are about 6.77kwords, 3.13 kwords and 9.94 kwords, respectively. Comparing the PCM output of fixed-point computation with that of floating-point computation, we achieve the signal-to-noise ratio of more than 60 dB. A real-time operation is demonstrated on the PC using the sound I/O and host communication functions in the EVM board.

• ETRI Journal
• /
• v.19 no.3
• /
• pp.242-258
• /
• 1997

The Qualcomm code excited linear prediction (QCELP) speech coder was adopted to increase the capacity of the CDMA Mobile System (CMS). In this paper, we implemented the QCELP speech coding algorithm by using TMS320C50 fixed point DSP chip. Also the fixed point simulation was done with C language. The computation complexity of QCELP on TMS320C50 was 10k words and data memory was 4k words. In the normal call test on the CMS, where mobile to mobile call test was done in the bypass mode without double vocoding, mean opinion score for the speech quality was he Qualcomm code excited linear prediction (QCELP) speech quality was 3.11.

• Journal of Information Processing Systems
• /
• v.18 no.3
• /
• pp.293-301
• /
• 2022

Direction of arrival (DOA) estimation of space signals is a basic problem in array signal processing. DOA estimation based on the multiple signal classification (MUSIC) algorithm can theoretically overcome the Rayleigh limit and achieve super resolution. However, owing to its inadequate real-time performance and accuracy in practical engineering applications, its applications are limited. To address this problem, in this study, a DOA estimation algorithm with high parallelism and precision based on an analysis of the characteristics of complex matrix eigenvalue decomposition and the coordinate rotation digital computer (CORDIC) algorithm is proposed. For parallel and single precision, floating-point numbers are used to construct an orthogonal identity matrix. Thus, the efficiency and accuracy of the algorithm are guaranteed. Furthermore, the accuracy and computation of the fixed-point algorithm, double-precision floating-point algorithm, and proposed algorithm are compared. Without increasing complexity, the proposed algorithm can achieve remarkably higher accuracy and efficiency than the fixed-point algorithm and double-precision floating-point calculations, respectively.

• International Journal of Naval Architecture and Ocean Engineering
• /
• v.6 no.1
• /
• pp.175-185
• /
• 2014

This study presents the prediction of propagated wave profiles using the wave information at a fixed point. The fixed points can be fixed in either space or time. Wave information based on the linear wave theory can be expressed by Fredholm integral equation of the first kinds. The discretized matrix equation is usually an ill-conditioned system. Tikhonov regularization was applied to the ill-conditioned system to overcome instability of the system. The regularization parameter is calculated by using the L-curve method. The numerical results are compared with the experimental results. The analysis of the numerical computation shows that the Tikhonov regularization method is useful.

• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.22 no.1
• /
• pp.1-8
• /
• 2018

Recently, the convolutional neural network shows high performance in many computer vision tasks. However, convolutional neural networks require enormous amount of operation, so it is difficult to adopt them in the embedded environments. To solve this problem, many studies are performed on the ASIC or FPGA implementation, where an efficient representation method is required. The fixed-point representation is adequate for the ASIC or FPGA implementation but causes a performance degradation. This paper proposes a separate optimization of representations for the convolutional layers and the batch normalization layers. With the proposed method, the required bit width for the convolutional layers is reduced from 16 bits to 10 bits for the ResNet-50 neural network. Since the computation amount of the convolutional layers occupies the most of the entire computation, the bit width reduction in the convolutional layers enables the efficient implementation of the convolutional neural networks.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
• /
• v.2 no.2
• /
• pp.37-43
• /
• 2009

In this study, multi-channel digital hearing aid algorithm for low power system is proposed. First, MDCT(Modified Discrete Cosine Transform) method converts time domain of input speech signal into frequency domain of it. Output signal from MDCT makes a group about each channel, and then each channel signal adjusts a gain using LCF(Loudness Compensation Function) table depending on hearing loss of an auditory person. Finally, compensation signal is composed by TDAC and IMDCT. Its all of process make progress 16-bit fixed-point operation. We use fast-MDCT instead of MDCT for reducing system complexity and previously computed tables instead of log computation for estimating a gain. This algorithm evaluate through computer simulation.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.23 no.9A
• /
• pp.2365-2371
• /
• 1998

Discrete cosine transform (DCT) has wide applications in speech and image coding. In this paper, we propose a novel fast dCT scheme with the property of reduced multiplication stages and the smaller number of additions and multiplications. This exploits the symmetry property of the DCT kernel to decompose the N-point dCT to N/2 point, and can be generally applied recursively to $2^{m}$-point. The proposed algorithm has a structure that most of multiplications tend to be performed at final stage, and this reduces propagation of truncation error which could occur in the fixed-point computation. Also the minimization of the multiplication stages further decreases the error.

• International Journal of Fuzzy Logic and Intelligent Systems
• /
• v.5 no.2
• /
• pp.131-135
• /
• 2005

This paper proposes an efficient face recognition using independent component analysis(ICA) derived from the fixed point(FP) algorithm based on secant method. The secant method can exclude the complex computation of differential process from the FP based on Newton method. The proposed ICA has been applied to recognize the 20 Yale face images of $324\times324$ pixels. The experimental results show that the proposed ICA is superior to PCA not only in the restoration performance of basis images but also in the recognition performance of the trained images and the test images. Then negative angle as similarity measures has better recognition ratio than city-block and Euclidean.

• Journal of Advanced Marine Engineering and Technology
• /
• v.12 no.2
• /
• pp.21-34
• /
• 1988

Machinery enclosures are widely adopted to reduce the noise emission in various fields of application. Emitted noise, which is due to the vibration of enclosure's outer surface, is composed of two kinds of sound with different path of propagation. One is the "structure-borne sound", while the other is "air-borne sound". In order to get a most efficient machinery enclouser a prudent consideration upon the above structure-borne and air-borne sound is required, as the guiding principle of contermeasure for each noise is quite different. The controlling of input vibration and its isolation are major subjects for the structure-borne sound, and the specifications of absorbing members and damping panels are the major related matters for the air-borne sound. Hence, it seems very efficient to separate the total sounds into two categories with a great accuracy when one think of further reduction of noise from the existing enclosure, although its separating methods have not been made clear for many years. Author proposes an application method of experimental modal analysis to extract the structure-borne sound from the measured total radiation sound, as the air-borne sound is deduced by the vectorial difference between the measured total radiation sound and the calculated structure-borne sound. In order to calculate the correct structure-borne sound by the excitation at an arbitrary point on the enclosure structure, it is important to decide 1) how to estimate the enclosure's surface vibration velocity and 2) how to compute the radiation sound which is considered as the effect of vibration modes of enclosure surface. The former can be solved with total frequency response function calculated by the application of experimental modal analysis. The latter is to be solved by the author's new approaches for radiation sound computation by means of the Rayleigh's integral equation and the boundary-element method applied complex surface vibration velocity. As a first step, structure-borne sound by the excitation at an arbitry point on the rectangular plate with fixed edges, has been calculated to verified the reliability of the developed computation methods. The results of calculation show good agreements with those of the actual measurements.actual measurements.