• The Journal of the Acoustical Society of Korea
• /
• v.28 no.4
• /
• pp.383-391
• /
• 2009

In speech signal processing, speech signal corrupted by noise should be enhanced to improve quality. Usually noise estimation methods need flexibility for variable environment. Noise profile is renewed on silence region to avoid effects of speech properties. So we have to preprocess finding voice region before noise estimation. However, if received signal does not have silence region, we cannot apply that method. In this paper, we proposed SNR estimation method for continuous speech signal. The waveform which is stationary region of voiced speech is very correlated by pitch period. So we can estimate the SNR by correlation of near waveform after dividing a frame for each pitch. For unvoiced speech signal, vocal track characteristic is reflected by noise, so we can estimate SNR by using spectral distance between spectrum of received signal and estimated vocal track. Lastly, energy of speech signal is mostly distributed on voiced region, so we can estimate SNR by the ratio of voiced region energy to unvoiced.

• Proceedings of the IEEK Conference
• /
• 1998.10a
• /
• pp.1247-1250
• /
• 1998

In speech synthesis, LF model is widely used for excitation signal for voice source coding system. But LF model does not represent the harmonic frequencies of excitation signal. We propose an effective method which use sinusoidal functions for representing the harmonics of voice source signal. The proposed method could achieve more exact voice source waveform and better synthesized speech quality than LF model.

• ETRI Journal
• /
• v.27 no.4
• /
• pp.453-456
• /
• 2005

The ultra-wideband (UWB) signal radiation process in an antenna is different from that of a narrowband signal. In this paper, we study the degradation of the desired signal component according to the antenna structure and location of a receiver in a bipolar time-hopping UWB system. And we propose a receiver structure with an adaptive template waveform generator to compensate for the degradation caused by a realistic TX-RX antenna system.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
• /
• v.22 no.11
• /
• pp.1070-1077
• /
• 2011

This paper introduces a radar signal processing technique for intelligent rear view monitoring of an automobile. The linear frequency modulation-frequency shift keying(LFM-FSK) waveform, which is the combination of frequency modulation continuous wave(FMCW) and frequency shift keying(FSK) waveform, is employed to simultaneously estimate the range, relative aspect angle, and velocity of an automobile. Hence, it can be applied to monitor the rear view of an automobile. FMCW waveform has high range resolution capability, but it produces ghost targets under a multiple target environment. In contrast, FSK waveform can provide high velocity resolution and avoids the problem of ghost targets. However, it fails to identify multiple targets along the radar's line of sight. With LFM-FSK waveform, we can estimate the ranges and velocities of multiple targets with very high resolution, which avoids the ghost target problem of an FMCW waveform. Simulation result shows that LFM-FSK wavefrom is suitable for use in the lane change assistance system for an automobile.

• The Journal of the Acoustical Society of Korea
• /
• v.14 no.4
• /
• pp.89-95
• /
• 1995

The waveform coding technique has concerned with simply preserving the waveform shape of speech signal through a redundancy reduction process. In the case of speech synthesis, the waveform coding with high sound quality is mainly used to the synthesis by analysis. However, since the parameters of this coding are not classified into either excitation or vocal tract parameters, it is difficult to applying the waveform coding to the synthesis by rule. In order to apply the waveform coding to the synthesis by rule, the pitch alteration technique is required in prosody control. In this paper, we propose a new pitch alteration method that can change the pitch period in waveform coding by scaling the time-axis and compensating the spectrum. This is relevant to the time-frequency domain method were the phase components of the waveform is preserved with a little spectrum distortion of 2.5 % and less for 50% pitch change.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
• /
• v.25 no.9
• /
• pp.930-937
• /
• 2014

Inverse synthetic aperture radar (ISAR) images can be generated by radar which radiates the electromagnetic wave to a target and receives signal reflected from the target. ISAR images can be widely used to target detection and recognition. This paper proposed a method of generation of high resolution ISAR images by synthesizing frequency spectrums of each stepped chirp waveform in one burst and sub-sampling in frequency domain. This process is performed over entire bursts during coherent processing interval. Conventional ISAR image generation method using stepped frequency waveform has a severe problem of short unambiguous range, loading to ghost phenomenon. However, this problem can be resolved by the proposed method. In simulations, we generate high resolution ISAR image of the moving target which is Boeing-737 aircraft model composed of several ideal point scatterers.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.36 no.12B
• /
• pp.1728-1735
• /
• 2011

In this paper, satellite data link waveform is proposed which is based on the waveform of Link-16 but LDPC code is studied instead of CCSK code in order to be optimized to satellite nonlinear channel environment and transmission characteristics. And the DSM (Demux, Selective CH switch, and Mux) transponder structure is suggested which can remove all of the jamming signal out of the transmission signal band and convert uplink hopping frequency to desired ones of downlink. The results of BER and anti-jamming performance analysis shows that the required Eb/No and processing gain in the worst case partial band jammer of the proposed waveform are 2.5dB and 52dB respectively and the anti-jamming capacity improvement of DSM transponder is maximum 2dB.

• Journal of the Korean Institute of Telematics and Electronics
• /
• v.12 no.2
• /
• pp.43-46
• /
• 1975

In a chopper amplifier the input signal of the AC amplifier is a train of square-waves. In the rase of square-wave the operatiom of AC amplifier is much different from that of ordinary sinusoidal wave. In this paper for the purpose of contributing to the design of chopper amplifier destortions of waveforms in the amplifier were investigated. When the time constant of each stage is equal the waveform in each stave apppears as square wave whose top is exponentially decaying. And when each stave has different time constant the waveform in n-th stage is composed of n-square waves whose tops are exponentially decaying.

• Journal of the Korea Academia-Industrial cooperation Society
• /
• v.12 no.9
• /
• pp.4119-4124
• /
• 2011

In a MPC coding using excitation source of voiced and unvoiced, it would be a distortion of speech waveform in case of increasing or decreasing of speech signal amplitude in a frame. This is caused by normalization of synthesis speech signal in the process of restoration the multi-pulses of representation section. To solve this problem, this paper present a method of amplitude compensation(AC-MPC) in a multi-pulses each pitch interval in order to reduce distortion of speech waveform. I was confirmed that the method can be synthesized close to the original speech waveform. And I evaluate the MPC and AC-MPC using amplitude compensation method. As a result, SNRseg of AC-MPC was improved 0.7dB for female voice and 0.7dB for male voice respectively. Compared to the MPC, SNRseg of AC-MPC has been improved that I was able to control the distortion of the speech waveform finally. And so, I expect to be able to this method for cellular phone and smart phone using excitation source of low bit rate.

• Proceedings of the KOSOMBE Conference
• /
• v.1995 no.11
• /
• pp.265-268
• /
• 1995

Signal averaging technique to improve signal-to-noise ratio has widely been used in various fields, especially in electrophysiology. Estimation of the EP(evoked potential) signal using the conventional averaging method fails to correctly reconstruct the original signal under EEG(electroencephalogram) noise especial]y when the latency times of the evoked potential are not identical. Therefore, a technique based on the bispectrum averaging was proposed for recovering signal waveform from a set o noisy signals with variable signal dalay. In this paper an improved bispectrum estimation technique of the RP signal is proposed using a confidence thresholding of the EP signal in frequency domain in which energy distribution of the EP signal is usually not uniform. The suggested technique is coupled with the conventional bispectrum estimation technique such as least square method and recursive method. Some results with simulated data and real EP signal are shown.