• Journal of Biosystems Engineering
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• 제23권1호
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• pp.67-74
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• 1998

A nondestructive quality inspection technique using acoustic impulse response method was developed for eggshell inspection. An experimental system was built to generate the impact force, to measure the response signal and to analyze the frequency spectrum. This system includes an impulse generating unit, an egg holding seal a microphone with preamplifier, and a DSP board installed on Personal Computer. A simple algorithm .was developed for crack detection. Using the developed system with algorithm, crack detection ability was evaluated and the error rate to estimate the normal egg as cracked was found to be 4% and the error rate to estimate the cracked egg as normal was also found to be 4%. This system could be adopted in industry with some modification.

• 한국소음진동공학회논문집
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• 제13권7호
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• pp.499-507
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• 2003

This paper presents a method for active noise control (ANC) in a duct by using a ring-tyPe smart foam. The ring-type smart foam consists of an elastic porous material of lining shape and a PVDF film embedded In the material. The PVDF element acts as a secondary sound source to reduce the noise. Active noise control using a ring-type smart foam is only effective locally because of the way to excite radially. To enlarge the quiet zone, the duct Is lined with additional acoustic foam between the smart foam and the error microphone. When cancellation path ks optimized by the LMS/RLS algorithm, the computation power is reduced while control performance Is maintained. The filtered-x LMS algorithm is used to minimize the error signal.

• 한국소음진동공학회:학술대회논문집
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• 한국소음진동공학회 2001년도 추계학술대회논문집 II
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• pp.1132-1137
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• 2001

The smart foam that is first proposed by Fuller(2) is not applicable to active noise control in a duct having flow. Thus. this paper presents the ring-type smart foam as an alternative. The ring-type smart foam consists of polyurethane acoustic foam of lining shape and PVDF film embedded along the mid-surface of the foam lining. A feedforward adaptive filtered-x LMS controller is used to minimize the signal from the error microphone. To enlarge quiet sound region. two error microphones are used to update system modeling filter (SIMO method). Sound intensity control using the ring-type smart foam is also discussed

• 한국음향학회지
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• 제14권3호
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• pp.28-36
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• 1995

본 논문에서는 능동소음제어를 위하여 망각형 지연 LMS(least mean square) 알고리듬을 이용한 이중루프제어 모델을 제안하였다. 제안한 이중루푸제어 모델은 계산양을 줄이기 위해 이득과 음향시간지연인자를 이용하여 온라인으로 라우드스피커 특성 및 오차음경로를 추정한다. 이중루프구조를 통한 오차신호의 제어는 보다 견실한 제어시스템이 된다. 음향시간지연을 추정하기 위한 필터의 입력신호는 입력 마이크로폰 신호와 적응필터의 차를 사용한다. 망각형 지연 LMS 알고리듬은 비정상상태에서 계수들의 발산을 방지하기 위해 사용한다. 실제 소음신호에 대하여 제안한 이중루프제어 모델은 소음레벨이 평균 12.9dB정도 감쇠되었다.

• 한국도로학회논문집
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• 제15권4호
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• pp.21-29
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• 2013

PURPOSES : The objective of this study is to provide for the overall SPL (Sound Pressure Level) prediction model by using the NCPX (Noble Close Proximity) measurement method in terms of regression equations. METHODS: Many methods can be used to measure the traffic noise. However, NCPX measurement can powerfully measure the friction noise originated somewhere between tire and pavement by attaching the microphone at the proximity location of tire. The overall SPL(Sound Pressure Level) calculated by NCPX method depends on the vehicle speed, and the basic equation form of the prediction model for overall SPL was used, according to the previous studies (Bloemhof, 1986; Cho and Mun, 2008a; Cho and Mun, 2008b; Cho and Mun, 2008c). RESULTS : After developing the prediction model, the prediction model was verified by the correlation analysis and RMSE (Root Mean Squared Error). Furthermore, the correlation was resulted in good agreement. CONCLUSIONS: If the polynomial overall SPL prediction model can be used, the special cautions are required in terms of considering the interpolation points between vehicle speeds as well as overall SPLs.

• 한국음향학회지
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• 제36권5호
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• pp.345-352
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• 2017

본 논문에서는 잔향과 잡음이 존재하는 실제 환경을 모델링하여 두 개의 마이크로폰을 이용한 음원 위치추정의 정확성을 향상시키는 방법을 제안하였다. 입력신호에 VAD(Voice Activity Detection)를 적용하여 묵음 구간을 제외한 음성 구간만을 사용하였고, 샘플링 주파수의 제한으로 인한 측정 범위를 벗어나는 프레임은 업샘플링(up-sampling)을 통해 지연시간을 다시 추정하였다. 여기서 계산된 도착 지연 시간은 Time-table을 참조해 주변 후보위치의 지연 값들과의 비교로 최대 파워 값을 갖는 지연 시간을 선택하여 음원 위치의 정확도를 높였다. 또한 프레임간의 상관성을 이용하여 연속된 음성 프레임의 경우 큰 추정 차가 발생하는 곳을 찾아 주변 프레임의 평균값으로 대체함으로써 음원의 위치 추정 성능을 향상시켰다.

• 한국음향학회지
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• 제20권4호
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• pp.22-28
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• 2001

본 논문에서는 마이크로폰 어레이를 통해 수신한 화자의 음성신호를 이용하여 추출된 공간정보를 통해 화자의 위치를 실시간으로 추적하는 알고리듬을 개선하고 이를 실시간으로 구현하였다. 기존의 대표적인 화자 위치 추정 알고리듬인 CPSP (Cross Power, Spectrum Phase) 함수는 상호 상관관계 (Cross Correlation)가 정규화 되어있는 형태를 갖는데, CPSP 함수의 최대값 인덱스로부터 화자의 공간정보인 TDOA(Time Difference Of Arrival)를 추출하게 된다. 그러나 CPSP함수를 이용한 공간정보 추정 알고리듬은 실내환경에서 심각하게 일어나는 반향신호에 대해서 취약한 단점을 갖고 있다. 본 논문에서 제안하는 저주파 위상 복원 알고리듬은 주파수 측면에서 반향신호가CPSP함수에 미치는 영향을 분석하여 반향으로 인하여 왜곡된 위상 성분을 복원함으로써 보다 신뢰도 있는 TDOA 추정을 가능하게 한다. 반향신호로 인한 CPSP의 위상은 저주파보다 고주파에서 심하게 왜곡되는데, 각각의 반향신호의 도달 시간을 기하학적 분포를 갖는 확률변수로 모델링하여 이를 수학적으로 증명하였다. 제안한 시스템의 성능분석을 위해 DSP를 이용한 실시간 시스템을 구현하여 기존 CPSP 알고리듬과 제안된 알고리듬을 적용한 시스템을 실제 환경에서 비교 실험을 수행한 결과 제안된 알고리듬을 적용한 시스템에서 약 15샘플 이상 TDOA 추정 오차가 줄어들고 있음을 확인하였다.

• 대한임베디드공학회논문지
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• 제8권4호
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• pp.185-194
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• 2013

Abstract- In distant-talking environments, speech recognition performance degrades significantly due to noise and reverberation. Recent work of Michael L. Selzer shows that in microphone array speech recognition, the word error rate can be significantly reduced by adapting the beamformer weights to generate a sequence of features which maximizes the likelihood of the correct hypothesis. In this approach, called Likelihood Maximizing Beamforming algorithm (Limabeam), one of the method to implement this Limabeam is an UnSupervised Limabeam(USL) that can improve recognition performance in any situation of environment. From our investigation for this USL, we could see that because the performance of optimization depends strongly on the transcription output of the first recognition step, the output become unstable and this may lead lower performance. In order to improve recognition performance of USL, some post-filter techniques can be employed to obtain more correct transcription output of the first step. In this work, as a post-filtering technique for first recognition step of USL, we propose to add a Wiener-Filter combined with Feature Weighted Malahanobis Distance to improve recognition performance. We also suggest an alternative way to implement Limabeam algorithm for Hidden Markov Network (HM-Net) speech recognizer for efficient implementation. Speech recognition experiments performed in real distant-talking environment confirm the efficacy of Limabeam algorithm in HM-Net speech recognition system and also confirm the improved performance by the proposed method.

• ETRI Journal
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• 제32권6호
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• pp.921-931
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• 2010

It has been shown that the principal subspace-based multichannel Wiener filter (MWF) provides better performance than the conventional MWF for suppressing interference in the case of a single target source. It can efficiently estimate the target speech component in the principal subspace which estimates the acoustic transfer function up to a scaling factor. However, as the input signal-to-interference ratio (SIR) becomes lower, larger errors are incurred in the estimation of the acoustic transfer function by the principal subspace method, degrading the performance in interference suppression. In order to alleviate this problem, a principal subspace modification method was proposed in previous work. The principal subspace modification reduces the estimation error of the acoustic transfer function vector at low SIRs. In this work, a frequency-band dependent interpolation technique is further employed for the principal subspace modification. The speech recognition test is also conducted using the Sphinx-4 system and demonstrates the practical usefulness of the proposed method as a front processing for the speech recognizer in a distant-talking and interferer-present environment.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2000년도 하계종합학술대회 논문집(4)
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• pp.92-97
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• 2000

An AMR(Adaptive Multi-Rate) speech coding algorithm has been adopted as a standard speech codec for IMT-2000. It is based on the algebraic CELP, and consists of eight speech coding modes having the bit rate from 4.75 kbit/s to 12.2 kbit/s. It also contains the VAD(Voice Activity Detector), SCR (Source Controlled Rate) operation, and error concealment scheme for robustness in a radio channel. The bit rate of AMR is changed on a frame basis depending on the channel condition. In this paper, we introduced AMR speech coding algorithm and performed the real-time implementation using TMS320C6201, i.e., a Texas Instrument's fixed-point DSP. With the ANSI C source code released from ETSI and 3GPP, we convert and optimize the program to make it run in real time using the C compiler and assembly language. It is verified that the decoded result of the implemented speech codec on the DSP is identical with the PC simulation result using ANSI C code for test sequences. Also, actual sound input/output test using microphone and speaker demonstrates its proper real-time operation without distortions or delays.