• MALSORI
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• no.60
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• pp.181-190
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• 2006

This paper describes a speech enhancement scheme that leads to significant improvements in recognition performance when used in the ASR front-end. The proposed approach is based on adaptive comb filtering and an MMSE-related parameter estimator. While adaptive comb filtering reduces noise components remarkably, it is rarely effective in reducing non-stationary noises. Furthermore, due to the uniformly distributed frequency response of the comb-filter, it can cause serious distortion to clean speech signals. This paper proposes an improved comb-filter that adjusts its spectral magnitude to the original speech, based on the speech absence probability and the gain modification function. In addition, we introduce the modified comb filtering-based speech enhancement scheme for ASR in mobile environments. Evaluation experiments carried out using the Aurora 2 database demonstrate that the proposed method outperforms conventional adaptive comb filtering techniques in both clean and noisy environments.

• Proceedings of the KSPS conference
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• 2003.05a
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• pp.207-210
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• 2003

In this paper, the combinations of speech enhancement techniques are experimented. Specifically, the spectral subtraction, KLT based comb-filtering, and their combinations are applied to the Aurora2 database. The results show that recognition accuracy is improved when KLT based comb-filtering is applied after spectral subtraction.

• MALSORI
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• no.46
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• pp.65-76
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• 2003

In this paper, we employ the adaptive comb filtering for effective noise reduction in mobile communication environment. Adaptive comb filtering is a well-known method for noise reduction, but requires correct pitch period and must be applied just in voiced speech frames. To satisfy these requirements we use two kinds of information extracted from speech packets, one of which is the pitch period information measured precisely by a speech coder and the other is the frame rate information related to a decision on speech or silence frame. Experiments on speech recognition system confirm the efficiency of this method. Feature parameters employing this method give superior performance in noise environment to those extracted directly from output speech.

• Phonetics and Speech Sciences
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• v.2 no.1
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• pp.77-85
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• 2010

This paper proposes an efficient feature vector processing technique to guard the Speech Emotion Recognition (SER) system against a variety of noises. In the proposed approach, emotional feature vectors are extracted from speech processed by comb filtering. Then, these extracts are used in a robust model construction based on feature vector classification. We modify conventional comb filtering by using speech presence probability to minimize drawbacks due to incorrect pitch estimation under background noise conditions. The modified comb filtering can correctly enhance the harmonics, which is an important factor used in SER. Feature vector classification technique categorizes feature vectors into either discriminative vectors or non-discriminative vectors based on a log-likelihood criterion. This method can successfully select the discriminative vectors while preserving correct emotional characteristics. Thus, robust emotion models can be constructed by only using such discriminative vectors. On SER experiment using an emotional speech corpus contaminated by various noises, our approach exhibited superior performance to the baseline system.

• Proceedings of the KSPS conference
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• 2003.05a
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• pp.203-206
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• 2003

In this paper, we employ the adaptive comb filtering for effective noise reduction in mobile communication environment. Adaptive comb filtering is a well- known method for noise reduction, but requires the correct pitch period and must be applied just in voiced speech frames. To satisfy these requirements we use two kinds of information extracted from speech packets, one of which is the pitch period information measured precisely by a speech coder and the other is the frame rate information related to a decision on speech or silence frame. Experiments on speech recognition system confirm the efficiency of this method. Feature parameters employing this method give superior performance in noise environment to those extracted directly from output speech.

• Journal of Satellite, Information and Communications
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• v.10 no.2
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• pp.86-89
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• 2015

Formula acoustic characteristics of the room with a double layer, are compared through simulation and actual measurement. The rear area of the under- balcony speakers will cause a delay difference between the main speaker. In the mid / bass parts do not generate sufficient pressure is lowered and comb-Filtering phenomenon occurs significantly. The lower right area of the under- balcony speakers and speaker distance is the sound pressure of the under- balcony speakers to around 2 ~ 3m bigger than the main speakers and the sound image matches the pulpit is broken. Also, under area is more than 5 ~ 6m from the balcony outside speakers and causes differ by more than 10dB lower than the under- balcony speakers depending on the local laws of Translator wins Well, the main speaker at mid / high frequency sounds do not enter the sound pressure variations will drop by a significant. Appropriate arrangement and the output of the speaker according to the position under the balcony, and output of the main speakers are requested to minimize this problem sound. The proper sound design direction for the under- balcony speakers must be presented in order to improve the lower balcony area more pleasant acoustic environment.

• The Proceeding of the Korean Institute of Electromagnetic Engineering and Science
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• v.10 no.3
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• pp.2-17
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• 1999

Interpolated second order polynomials(ISOP's) are proposed to design efficient cascaded integrator-comb(CIC)-based decimation filters for a programmable downconverter. It is shown that some simple ISOP's can effectively reduce the passband droop caused by CIC filtering with little degradation in aliasing attenuation. In addition, ISOP's are shown to be useful for simplifying halfband filters that usually follow CIC filtering. As a result, a modified half band filter(MHBF) is introduced which is simpler than conventional halfband filters. The proposed decimation filter for a programmable downconverter is a cascade of a CIC filter, an ISOP, MHBF's and a programmable finite impulse response(FIR) filter. A procedure for designing the decimation filter is developed. In particular, an optimization technique that simultaneously designs the decimation filter is developed. In particular, an optimization technique that simultaneously designs the ISOP and programmable FIR filters is presented. Design examples demonstrate that the proposed method leads to more efficient programmable downconverters than existing ones.

• Journal of Satellite, Information and Communications
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• v.5 no.1
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• pp.32-37
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• 2010

The algorithm of multiple channel signal processing requires the flexibility of variable frequency band, efficient allocation of transmission power, and flexible frequency band reallocation to satisfy various service types which requires different transmission rates and frequency band. There are three methods including per-channel approach, multiple tree approach, and block approach performing frequency band reallocation method by channelization and dechannelization in the multiple-channel signal. This paper proposes an improved per-channel approach for converting the frequency band of multiple carrier signals efficiently. The proposed algorithm performs decimation and interpolation using CIC(cascaded integrator comb filter), half-band filter, and FIR filter. In addition, it performs filtering of each sub-channel, and reallocates channel band through FIR low-pass filter in the multiple-channel signal. The computer simulation result shows that the perfect reconstruction of output signal and the flexible frequency band reallocation is performed efficiently by the proposed algorithm.

• Korean Journal of Optics and Photonics
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• v.13 no.6
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• pp.461-466
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• 2002

Fiber-optic comb filters using a side-polished fiber coupler are proposed as multi-channel isolation filters on wavelength division multiplexing systems. We have demonstrated that the coupling efficiency between two waveguides is improved by the intermediate coupling layer in spite of the decrease of the optical power transfer between two waveguides due to the high-order modes of the overlay waveguide coupled with the side-polished single-mode fiber. When LiNbO$_3$with a 200-${\mu}{\textrm}{m}$-thickness was applied as a planar-overlay-waveguide, the comb filtering characteristics with a 4 nm-channel-spacing were achieved and the maximum power coupling occurred at the 1-${\mu}{\textrm}{m}$-thickness and the refractive index in range 1.52 to 1.53 of an intermediate coupling layer. If the intermediate coupling layer is optimized, an extinction ratio with more than 20 dB can be obtained. These experimental results are in good agreement with the BPM simulation.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.49 no.5
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• pp.51-58
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• 2012

Accurate transmission and estimation of the channel statistics affected by high Doppler spread is one of the main issues of concern for the latest and future mobile communication systems. Therefore, it is important to research in novel channel estimation techniques that overcome the limitations of conventional methods. In this paper, we propose a novel channel estimation method that, after a simple estimation in the first OFDM symbol, uses Kalman filter to predict the channel in the rest of OFDM symbols with pilot subcarriers. Our method is designed considering the lattice-type arrangement of pilot subcarriers in LTE-Advanced, since most of the studies so far focus on block-type or comb-type pilot arrangements. In addition, to optimize the results, we use the filtering of channel impulse response and Wiener Filter for the estimation of the channel frequency response in the rest of the subcarriers.