• The Journal of the Acoustical Society of Korea
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• v.16 no.1E
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• pp.15-23
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• 1997

Zero-Crossings with Peak Amplitudes(ZCPA) model motivated by human auditory periphery was proposed to extract reliable features speech signals even in noisy environments for robust speech recognition. In this paper, some practical considerations for digital hardware implementations of the ZCPA model are addressed and evaluated for recognition of speech corrupted by several real world noises as well as white Gaussian noise. Infinite impulse response(IIR) filters which constitute the cochliar filterbank of the ZCPA are replaced by hamming bandpass filters of which frequency responses are less similar to biological neural tuning curves. Experimental results demonstrate that the detailed frequency response of the cochlear filters are not critical to performance. Also, the sensitivity of the model output to the variations in microphone gain is investigated, and results in good reliability of the ZCPA model.

• Proceedings of the KOSOMBE Conference
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• v.1998 no.11
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• pp.163-164
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• 1998

In this paper the digital speech processing part of cochlear implant for sensorineural disorderly patients is implemented and simulated. We implement the speech processing part by dividing into three small parts - Filterbank, Pitch Detect, and Bandmapping parts. With the result, we conclude digital speech processing algorithm is implemented in FPGA perfectly. This means that cochlear implant can be made very small size.

• Journal of Communications and Networks
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• v.4 no.3
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• pp.189-198
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• 2002

Precoder has been shown to be able to provide source diversity and design flexibility. In this paper we employ precoding techniques for block transmission based on a multirate filterbank structure. To accommodate multiuser communication with variable data rates, different precoders with corresponding coefficients and up/down sampling rates are used. However, due to unknown multipath distortion, different interferences may exist in the received data, such as multiuser interference, intersymbol interference and interblock interference. To estimate channel parameters for a desired user, we employ all structured signature waveforms associated with different symbols of that user and apply subspace techniques. Therefore better performance of channel estimator can be achieved than the conventional subspace method based only on the signature of the current symbol. The delay for that user can also be jointly estimated. Channel identifiability conditions and asymptotic channel estimation error are investigated in detail. Numerical examples are provided to justify the proposed method. gest either multicode (MC) or multiple processing gain (MPG) mechanism [2], while requiring data rates to be integral multiples of some basic low-rate. In order to support variable rate transmission however, a comprehensive scheme needs to be investigated.

• Journal of the Korea Society of Computer and Information
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• v.5 no.3
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• pp.91-98
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• 2000

In this paper we design implement the speaker verification system for verifying personal identification using voice. Filter bank magnitude was used as a feature parameter and code-book was made using LBG a1gorithm. The code book convert feature parameters into code sequence. The difference between reference pattern and input pattern measures using DTW(Dynamic Time Warping). The similarity measured using DTW and threshold value derived from deviation were used to discriminate impostor from client speaker.

• Proceedings of the Korea Information Processing Society Conference
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• 2013.11a
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• pp.1450-1451
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• 2013

본 논문에서는 음향 확성 환경과 음향신호의 입 출력 조건에 최적화된 음향 시스템을 실시간으로 생성 및 적용 가능한 음향 신호처리용 시스템으로서, 가변형 필터뱅크 기술 및 상황 적응적 필터 재조합 재배열 기술을 음향 신호처리용 DSP에 적용함으로서 Pro-Audio 장비, 방송 음향장비, 산업 음향장비와 같은 다양한 음향관련 장비에서 고품질 음향 서비스를 제공하기위한 핵심 기술인 가변형 필터뱅크 기반 고품질 음향 DSP 시스템 개발을 제안한다.

• Journal of Korea Multimedia Society
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• v.24 no.12
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• pp.1606-1613
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• 2021

Recently various attempts to control hardware through integration of sensors and artificial intelligence have been made. This paper proposes a smart hazardous sound detection at home. Previous sound recognition methods have problems due to the processing of background sounds and the low recognition accuracy of high-frequency sounds. To get around these problems, a new MFCC(Mel-Frequency Cepstral Coefficient) algorithm using Wiener filter, modified filterbank is proposed. Experiments for comparing the performance of the proposed method and the original MFCC were conducted. For the classification of feature vectors extracted using the proposed MFCC, DNN(Deep Neural Network) was used. Experimental results showed the superiority of the modified MFCC in comparison to the conventional MFCC in terms of 1% higher training accuracy and 6.6% higher recognition rate.

• Journal of the Korean Society for Aeronautical & Space Sciences
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• v.43 no.7
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• pp.641-647
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• 2015

In this paper, Analyzed the multi-channel signal processing algorithms for digital-transponder. To analyze suitable multi-channel signal processing algorithms, compare algorithms about four criteria. Four criteria are as follows, perfect reconstruction, interference rejection, resource usage and power consumption. Analysis for each algorithm in accordance with these four criteria. then propose the multi-channel signal processing algorithms for digital satellite communication system.

• Journal of the Korean Institute of Intelligent Systems
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• v.14 no.2
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• pp.150-155
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• 2004

In this paper, an emotion recognition method using speech signal is presented. Six basic human emotions including happiness, sadness, anger, surprise, fear and dislike are investigated. The proposed recognizer have each codebook constructed by using the wavelet transform for the emotional state. Here, we first verify the emotional state at each filterbank and then the final recognition is obtained from a multi-decision method scheme. The database consists of 360 emotional utterances from twenty person who talk a sentence three times for six emotional states. The proposed method showed more 5% improvement of the recognition rate than previous works.

• The Journal of the Acoustical Society of Korea
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• v.19 no.4
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• pp.52-57
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• 2000

In this paper, we designed the 16 Kbps speech coder that has compatibility with CS-ACELP algorithm(G.729). The speech signal is sampled at rate of 16 KHz, divided into two narrowband signal by QMF filterbank, and decimated to rate of 8 KHz. The lower-band signal is encoded by CS-ACELP and the upper-band signal is encoded by Adaptive Transform Coding(ATC) algorithm. At the receiver, two band signals are synthesized by decoder of CS-ACELP and ATC, respectively. The reconstructed output is obtained by passing the QMF synthesis bank. The proposed wideband coder is evaluated with ITU-T G.722 coder through the Mean Opinion Score(MOS) test.

• Proceedings of the Korea Information Processing Society Conference
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• 2007.11a
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• pp.113-116
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• 2007

In this paper, our proposed Nonlinear Autoregressive Exogenous (NARX) based on Least Square-Support Vector Regression (LS-SVR) is introduced and tested for producing natural sounds. This nonlinear synthesizer perfectly reproduce voiced sounds, and also conserve the naturalness such as jitter and shimmer, compared to LPC does not keep these naturalness. However, the results of some phonation are quite different from the original sounds. These results are assumed that single-band model can not afford to control and decompose the high frequency components. Therefore multi-band model with wavelet filterbank is adopted for substituting single band model. As a results, multi-band model results in improved stability. Finally, nonlinear speech modeling using NARX based on LS-SVR can successfully reconstruct synthesized sounds nearly similar to original voiced sounds.