• Journal of the Korean Institute of Telematics and Electronics A
• /
• v.29A no.3
• /
• pp.96-105
• /
• 1992

The use of fixed-point digital signal processors, such as the TMS 320C25, requires scaling of data at each arithmetic step to prevent overflows while keeping the accuracy. A software which automatizes this process is developed for TMS 320C25. The programmers use a model of a hypothetical floating-point digital signal processor and a floating-point format for data representation. However, the program and data are automatically translated to a fixed-point version by this software. Thus, the execution speed is not sacrificed. A fixed-point variable has a unique binary-point location, which is dependent on the range of the variable. The range is estimated from the floating-point simulation. The number of shifts needed for arithmetic or data transfer step is determined by the binary-points of the variables associated with the operation. A fixed-point code generator is also developed by using the proposed automatic scaling software. This code generator produces floating-point assembly programs from the specifiations of FIR, IIR, and adaptive transversal filters, then floating-point programs are transformed to fixed-point versions by the automatic scaling software.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.22 no.3
• /
• pp.454-468
• /
• 1997

This paper presents a bitwidth optimization algorithm for efficient hardware sharing in digital signal processing system. The proposed algorithm determines the fixed-point representation for each signal through bitwidth optimization to generate the hardware requiring less area. To reduce the operator area, the algorithm partitions the abstract operations in the design description into several groups, such that the operations in the same group can share an operator. The partitioning result are fed to a high-level synthesis system to generate the pipelined fixed-point datapaths. The proposed algorithm has been implemented in SODAS-DSP an automatic synthesis system for fixed-point DSP hardware. Accepting the models of DSP algorithms in schematics, the system automatically generates the fixed-point datapath and controller satisfying the design constraints in area, speed, and SNR(Signal-to-Noise Ratio). Experimental results show that the efficiency of the proposed algorithm by generates the area-efficient DSP hardwares satisfying performance constraints.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
• /
• v.2 no.1
• /
• pp.37-42
• /
• 2009

The digital filter is essential element in digital signal processing area. It needs a high computational burden caused by multiplying and adding. The multiplier in digital filter is a dominant element, which occupies an wide area at the field of VLSI design, needs high power-consuming and also decides critical path that affects to filter performance. In this paper we proposed the simultaneous transform method which is represented 2's complementary representation to CSD and MSD representation to solve a complexity problem and to improve a computational speed. The performance of proposed method was implemented in VHDL and applied to an digital filters, was evaluated the decreasing of critical path delay.

• Proceedings of the KOSOMBE Conference
• /
• v.1994 no.12
• /
• pp.31-38
• /
• 1994

This paper proposed signal decomposition and multiresolution representation through wavelet transform using wavelet orthonormal basis. And it suggested most appropriate filter for scaling function in multiresoltion representation and compared two compression method, arithmetic coding and Huffman coding. Results are as follows 1. Daub18 coefficient is most appropriate in computing time, energy compaction, image quality. 2. In case of image browsing that should be small in size and good for recognition, it is reasonable to decompose to 3 scale using pyramidal algorithm. 3. For the case of progressive transmittion where requires most grateful image reconstruction from least number of sampls or reconstruction at any target rate, I embedded the data in order of significance after scaling to 5 step. 4. Medical images such as information loss is fatal have to be compressed by lossless method. As a result from compressing 5 scaled data through arithmetic coding and Huffman coding, I obtained that arithmetic coding is better than huffman coding in processing time and compression ratio. And in case of arithmetic coding I could compress to 38% to original image data.

• Proceedings of the KIEE Conference
• /
• 1999.07b
• /
• pp.991-993
• /
• 1999

This study proposed a new algorithm to assess autonomic function activity using Time-Frequency Representation(TFR). TFR is a way of describing the time-valiant energy of a signal. A discrete Wigner representation that is capable of filtering out any cross terms occuring in the Wigner-Ville Distribution(WVD) is used for time-variant energy distribution of heart rate variability(HRV) signals. And the marginal condition are evaluated to estimate power spectrum of HRV signals. The proposed algorithm showed that estimated power spectrum of HRV signals well describe the autonomic nerve system function and also showed the dynamics of autonomic nervous system response.

• Journal of KIISE:Software and Applications
• /
• v.31 no.10
• /
• pp.1412-1417
• /
• 2004

Sparse coding or independent component analysis (ICA) which is a holistic representation, was successfully applied to elucidate early auditor${\gamma}$ processing and to the task of sound classification. In contrast, parts-based representation is an alternative way o) understanding object recognition in brain. In this thesis we employ the non-negative matrix factorization (NMF) which learns parts-based representation in the task of sound classification. Methods of feature extraction from the spectro-temporal sounds using the NMF in the absence or presence of noise, are explained. Experimental results show that NMF-based features improve the performance of sound classification over ICA-based features.

• The Journal of the Korea institute of electronic communication sciences
• /
• v.15 no.5
• /
• pp.807-812
• /
• 2020

This paper proposes a blind speech source separation algorithm using a microphone to separate only the target speech source signal in an environment in which various speech source signals are mixed. The proposed algorithm is a model of frequency domain representation based on independent component analysis method. Accordingly, for the purpose of verifying the validity of independent component analysis in the frequency domain for two speech sources, the proposed algorithm is executed by changing the type of speech sources to perform speech sources separation to verify the improvement effect. It was clarified from the experimental results by the waveform of this experiment that the two-channel speech source signals can be clearly separated compared to the original waveform. In addition, in this experiments, the proposed algorithm improves the speech source separation performance compared to the existing algorithms, from the experimental results using the target signal to interference energy ratio.

• Journal of the Korean Society for Railway
• /
• v.18 no.1
• /
• pp.25-32
• /
• 2015

It has been subject to question whether a conventional traffic signal controller can support efficient operation at grade intersections in which vehicles and tram lines cross concurrently. This study proposes a list of functional requirements for a traffic signal controller to support effective operation of trams at such an intersection. The considered details includes the combinations of geometric conditions at which vehicle, bus, and tram lines are aligned, the types of exclusive tram signals, and the combinations of various signal phases, including exclusive tram phases, which are currently undefined in the nation. The results suggest that the controller should be capable of dealing with the NEMA standard overlap relays and a quadruple-ring phase representation scheme.

• Journal of the Institute of Electronics Engineers of Korea TC
• /
• v.43 no.2 s.344
• /
• pp.39-43
• /
• 2006

This paper deals with the AR Identification of unknown system using cumulant, which is the 3rd order statistics of output signal in the presence of the noise signal. The algorithms for identification of unknown system we applies to the AR identification method using the cumulant which is possible to the guarantees of global convergence and the representation of amplitude and phase information of system among with the method of parametric modeling. In the process of identification, we considered unknown system to the one of AR system. After the generation of input signal, it was being passed through the system then We use the its output signal that the noise is added. As a result of identification of AR system by changing the signal to noise ratio, we get the fairly good results compared to original system output values and confirmed that the pole was located in the unit circle of z transform.

• ETRI Journal
• /
• v.27 no.3
• /
• pp.330-333
• /
• 2005

The Inter-Channel Level Difference (ICLD) is a cue parameter to estimate spectral information in a binaural cue coding that has been recently in the spotlight as a multichannel audio signal compression technique. Even though the ICLD is an essential parameter, it is generally distorted by quantization. In this paper, a new modified ICLE representation method to minimize the quantization distortion is proposed by adopting a flexible determination of the reference channel and the unidirectional quantization. Our experimental result confirms that the proposed method improves the multichannel audio output quality even with the reduced bit-rate.