• The Journal of the Acoustical Society of Korea
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• v.28 no.3
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• pp.279-284
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• 2009

Speaker verification determines whether the claimed speaker is accepted based on the score of the test utterance. In recent years, methods based on Gaussian mixture models and universal background model have been the dominant approaches for text-independent speaker verification. These speaker verification systems based on these methods provide very good performance under laboratory conditions. However, in real situations, the performance of speaker verification system is degraded dramatically. For overcoming this performance degradation, the feature recombination method was proposed, but this method had a drawback that whole sub-band feature vectors are used to compute the likelihood scores. To deal with this drawback, a modified feature recombination method which can use each sub-band likelihood score independently was proposed in our previous research. In this paper, we propose a sub-band weighting method based on sub-band signal-to-noise ratio which is combined with previously proposed modified feature recombination. This proposed method reduces errors by 28% compared with the conventional feature recombination method.

• The Journal of the Acoustical Society of Korea
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• v.21 no.6
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• pp.552-558
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• 2002

Recently, based on the human speech recognition (HSR) model of Fletcher, the multi-band speech recognition has been intensively studied by many researchers. As a new automatic speech recognition (ASR) technique, the multi-band speech recognition splits the frequency domain into several sub-bands and recognizes each sub-band independently. The likelihood scores of sub-bands are weighted according to reliabilities of sub-bands and re-combined to make a final decision. This approach is known to be robust under noisy environments. When the noise is stationary a sub-band SNR can be estimated using the noise information in non-speech interval. However, if the noise is non-stationary it is not feasible to obtain the sub-band SNR. This paper proposes the inverse sub-band distance (ISD) weighting, where a distance of each sub-band is calculated by a stochastic matching of input feature vectors and hidden Markov models. The inverse distance is used as a sub-band weight. Experiments on 1500∼1800㎐ band-limited white noise and classical guitar sound revealed that the proposed method could represent the sub-band reliability effectively and improve the performance under both stationary and non-stationary band-limited noise environments.

• Journal of the Korean Institute of Electrical and Electronic Material Engineers
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• v.15 no.4
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• pp.325-331
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• 2002

In this paper, surface acoustic wave(SAW) bandpass filters using slanted finger interdigital electrode transducers(IDTs) are investigated. The slanted finger IDTs are used to design SAW filters with good shape factor, a flat passband, and good out-of-band rejection characteristic. For the filter design, input-output IDT structure was simulated with modified impulse model; uniform-withdrawal weighting IDTs, withdrawal-withdrawal weighting IDTs. SAW filters of uniform-withdrawal weighting IDTs structure were designed and fabricated on $128^{\circ}LiNbO_3$ piezoelectric substrates. Implemented SAW filter has a fractional bandwidth of 30%, center frequency of 70MHz and shape factor of $1.12\pm0.01$.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.10C
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• pp.965-975
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• 2005

In this paper, we propose and evaluate joint carrier and sampling frequency offset estimation schemes based on the channel estimation sequences in PLCP (Physical Layer Convergence Procedure) preamble for the proper and effcient synchronization of the MB-OFDM WB (Multi-Band Orthogonal Frequency Division Multiplexing Ultra Wide Band) systems which have recently drawn explosive attention for future W-PAN (Wireless Personal Area Network) applications. In the joint estimation schemes, we first estimate the sampling frequency offset, and then estimate the carrier frequency offset using the estimated sampling frequency offset. Moreover, to improve the reliability of the estimated offset values, each process uses a combination scheme based on weighting factors. Simulation results using IEEE 802.15 Task Group 3a UWB channel models reveal that the estimation scheme using the simple weighting factors based on easily-measurable received signal power of each sub-channel shows favorably comparable performance to the ideal scheme using the weighting factors based on the perfectly-estimated frequency response of the channel.

• The Journal of the Acoustical Society of Korea
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• v.18 no.8
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• pp.61-67
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• 1999

This paper proposes a new beamforming algorithm for acoustic measurement by using the nested linear array. In this algorithm, the weighting is optimized by minimizing the LMS error with the initial value obtained by FIR filter design algorithm. The optimization process is applied to each sub-band, which is divided from the octave band, to produce the uniform directivity index. For the optimization pseudo inverse matrix is used for the transfer matrix. As the simulation results, it is found that the proposed algorithm can get the desired beam pattern and unform directivity index so as to be used efficiently for the acoustic measurement by using a nested linear array.

• The Journal of the Acoustical Society of Korea
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• v.29 no.1
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• pp.62-68
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• 2010

This paper proposes a new masking threshold adjustment strategy to improve the performance for speech signals in low bit-rate audio coding. After determining formant regions, the masking threshold is adjusted by using the energy ratio of each sub-band to the average energy of each formant. More quantization noises are added to the bands that have relatively large energy, but less distortion is allowed in spectral valley regions by allocating more bits, which reflects the concept of perceptual weighting widely used in speech coding. From the results of objective speech quality measure, we verified that the proposed method improves quality for the speech input signals compared to the conventional one.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.47 no.4
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• pp.69-78
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• 2010

The irreversible 9/7 JPEG2000, which is one of sub-band coding techniques, has a problem of severe picture quality distortion at the edge and the background caused by the quantization error below 0.15bpp. In this paper, to solve such problems we propose a VQ(Visual Quantizer) based on L-pdf(Laplace probability density function) statistical characteristics of high frequency sub-bands. The proposed VQ is designed by visual parameter for improving the subjective quality and weighting parameter for increasing the compression ratio. A proposed method, based on 9/7 JPEG2000 scheme, gives the high subjective quality to reconstructed images below 0.15bpp and provides minimum MSE(Mean-Squared Error) regardless of the compression ratio.

• Journal of the Institute of Electronics and Information Engineers
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• v.51 no.6
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• pp.221-227
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• 2014

In this paper, we propose a synthesis method that can effectively suppress the ambience which affects tonal components in the PS decoder. Ambience component was obtained by using decorrelation filter and the weighting of the ambience in the decoder was determined through IC parameter. However, since the parameters are extracted in the sub-band domain, a low IC value could be analyzed even if the tonal component is dominant. The quality of the output signal may be degraded. To prevent this problem, the tonality was measured in the downmixed signal and the weighting of the ambience components were adjusted appropriately according to the measured tonality index. The performance of the proposed method was evaluated by simulations. Furthermore, the subjective test was performed and the results confirmed that the proposed method offers improved quality.