• KSII Transactions on Internet and Information Systems (TIIS)
• /
• v.9 no.12
• /
• pp.4934-4949
• /
• 2015

The unstable characteristic values of communication signals along with the varying SNR (Signal Noise Ratio) environment make it difficult to identify the modulations of signals. Most of relevant literature revolves around signal recognition under stable SNR, and not applicable for signal recognition at varying SNR. To solve the problem, this research developed a novel communication signal recognition algorithm based on Holder coefficient and cloud theory. In this algorithm, the two-dimensional (2D) Holder coefficient characteristics of communication signals were firstly calculated, and then according to the distribution characteristics of Holder coefficient under varying SNR environment, the digital characteristics of cloud model such as expectation, entropy, and hyper entropy are calculated to constitute the three-dimensional (3D) digital cloud characteristics of Holder coefficient value, which aims to improve the recognition rate of the communication signals. Compared with traditional algorithms, the developed algorithm can describe the signals' features more accurately under varying SNR environment. The results from the numerical simulation show that the developed 3D feature extraction algorithm based on Holder coefficient cloud features performs better anti-noise ability, and the classifier based on interval gray relation theory can achieve a recognition rate up to 84.0%, even when the SNR varies from -17dB to -12dB.

• Journal of IKEEE
• /
• v.25 no.2
• /
• pp.234-242
• /
• 2021

In case implementation of microwave photonic (MWP) systems for phased array radars (PARs), noise and time delay error should be minimized to obtain accurate beam direction. Time delay error in MWP systems is generated from signal noise and timing jitter. In this paper, noise and timing jitter in MWP systems for PAR is researched, also according to the amplification of an erbium-doped fiber amplifier, noise and timing jitter variation is verified by an experiment. Timing jitter is decreased and SNR is increased if we amplify the signal by using an erbium-doped fiber amplifier, up to the amplification rate of signal and noise is similar.

• Proceedings of the Korea Institute of Convergence Signal Processing
• /
• 2003.06a
• /
• pp.94-97
• /
• 2003

In this Paper, propose for MLP(multilayer perception) neural network that uses optimization recognition training scheme for the wavelet transform and the numeral image add to noise, and apply this system in Numeral Recognition. As important part of original image information preserves maximum using the wavelet transform, node number of neural network and the loaming convergence time did size of input vector so that decrease. Apply in training vector, examine about change of the recognition rate as optimization recognition training scheme raises noise of data gradually. We used original image and original image added 0, 10, 20, 30, 40, 50㏈ noise (or the increase of numeral recognition rate. In case of test image added 30∼50㏈, numeral recognition rate between the original image and image added noise for training Is a little But, in case of test image added 0∼20㏈ noise, the image added 0, 10, 20, 30, 40 , 50㏈ noise is used training. Then numeral recognition rate improved 9 percent.

• Journal of the Korean Society for Precision Engineering
• /
• v.23 no.5 s.182
• /
• pp.68-76
• /
• 2006

The ability to recognize human emotion is one of the hallmarks of human-robot interaction. Hence this paper describes the realization of emotion recognition. For emotion recognition from voice, we propose a new feature called frequency range of meaningful signal. With this feature, we reached average recognition rate of 76% in speaker-dependent. From the experimental results, we confirm the usefulness of the proposed feature. We also define the noise environment and conduct the noise-environment test. In contrast to other features, the proposed feature is robust in a noise-environment.

• The Journal of the Korea Contents Association
• /
• v.7 no.12
• /
• pp.254-261
• /
• 2007

In this paper, we propose a blind equalization method for advanced terrestrial digital multimedia broadcasting (AT-DMB) systems which use hierarchical modulation. The AT-DMB system adopts hierarchical 16-ary quadrature amplitude modulation (16-QAM) to ensure backward-compatibly with the differential quadrature phase shift keying (DQPSK) signal of the legacy terrestrial digital multimedia broadcasting (T-DMB) systems and to support higher transmission rate. Due to the hierarchical modulation, the conventional T-DMB signal and the additional signal have different error rate at same signal to noise ratio (SNR). By weighting the decided symbols differently according to the reliability of the symbols, i.e., high priority symbol with low error rate and low priority symbol with high error rate, we can improve the channel estimation accuracy. In this paper, we analyze SNR loss by hierarchical modulation and confirm it through simulations. Moreover, through simulations, we verify that the proposed weighting method improve BER compared to the no-weighting method.

• The Journal of the Acoustical Society of Korea
• /
• v.33 no.5
• /
• pp.335-340
• /
• 2014

In this paper, we propose a new sign subband adaptive filter to improve the convergence rate of the conventional sign subband adaptive filter which has been proposed to deal with colored input signal under the environment with impulsive noise. The existing sign subband adaptive filter does not increase the convergence speed by increasing the number of subband because each subband input signal is normalized by $l_2-norm$ of all of the subband input signals. We devised a new sign subband adaptive filter that normalizes each subband input signal with $l_2-norm$ of each subband input signal and increases the convergence rate by increasing the number of subband. We carried out a performance comparison of the proposed algorithm with the existing sign subband adaptive filter using a system identification model. It is shown that the proposed algorithm has faster convergence rate than the existing sign subband adaptive filter.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
• /
• 2002.11b
• /
• pp.116-121
• /
• 2002

Axial fans are widely used in heavy machines due to their ability to produce high flow rate for cooling of engines. At the same time, the noise generated by these fans causes one of the most serious problems. This work is concerned with the low noise technique of discrete frequency noise. To calculate the unsteady resultant force over the fan blade in an unsymmetric engine room. Time-Marching Free-Wake Method is used. From the calculations of unsteady force on fan blades, noise signal of an engine cooling fan is calculated by using an acoustic similarity law. Noise optimization is obtained from Neural Network which is constructed based on the calculated flow rate and noise spectrum.

• Journal of Sensor Science and Technology
• /
• v.18 no.6
• /
• pp.423-431
• /
• 2009

Standard uroflowmetry measures the urine weight using single load cell to evaluate the urinary flow rate. Impact noise should be introduced due to gravity when the urine stream falls down into the container upon the load cell. The present study placed three load cells on the three vertices of a regular triangle and the three signals were ensemble averaged to enhance the signal-to-noise ratio(SNR) regardless of how the urination was made. Simulated urination experiment was performed with three different urine collection methods. In all three methods, SNR of the averaged signal was much higher than each load cell signals. With no urine collection device, the present signal averaging technique resulted in SNR values higher by 10~15 dB than when dual funnels or upper funnel were used to guide the urine stream. Therefore, it was demonstrated that the three point measurement followed by with ensemble averaging could enable accurate uroflowmetric test without any specially made urine collection devices.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
• /
• 2009.10a
• /
• pp.723-726
• /
• 2009

In this research, We simulated performances of error reduction algorithm for the speech signal based on the microphone array-based beamforming method in speech recognition system and analyzed its performance. Also, we processed speech signal adopted from microphone array and maximum signal to noise ratio for each channel, and then compared them with signal to noise ratio of speech signal. Speech recognition rate is improved from 54.2% to 61.4% in case 1 and is improved from 41.2% to 50.5% in case 2 of the lower signal to noise ratio. Therefore the average reduction rates are showed 15.7% in case 1.

• Journal of the Korean Institute of Telematics and Electronics
• /
• v.23 no.5
• /
• pp.597-604
• /
• 1986

This paper describes various mismatch phenomena in differential pulse code modulation (DPCM) coding, such as the mismatch effects of probability density functin(pdf), signal variance, and correlation. At a high transmission rate(i.e., above 32 kbits/s), the performance of DPCM can be improved by matching the pdf shape between the input signal and the quantizer. However, the same gain cannot be obtained at a lower transmission rate. Also, it is shown that the gamma quantizer is realtively robust to the variation of pdf shaper and signal variance. Moreover, as the transmission rate increases, the performance of DPCM for the input signal with large variance is worse than that of DPCM for the signal with small variance due to the increase of overload noise. According to our simuladiton results, the mismatch effects of pdf shape and variance appear to yield more degradatin than that of correlation in a DPCM system.