• 한국산학기술학회논문지
• /
• 제14권10호
• /
• pp.4753-4761
• /
• 2013

본 연구는 국악기 방사특성을 고려하여 다양한 마이크로폰의 설치 각도와 거리를 설정하여 국악기소리를 녹음한 후 녹음된 데이터를 주파수 스펙트럼과 반복청취를 통하여 음색특징을 분석, 가장 적합한 마이크로폰의 위치 및 녹음위치를 도출하였다. 본 연구의 실험은 대중음악에 사용되어지는 근접 마이크로폰 테크닉을 위한 실험이며 대중음악 녹음 스튜디오에서 이루어 졌다. 이 연구는 녹음 시 가장 적합한 마이크로폰의 위치 제시와 함께 다양한 위치에서 나타나는 음색의 차이를 설명하였다. 연구의 결과는 아름다운 우리 국악기 소리를 정확하게 표현하고 다양한 음색을 표현하기 위한 국악기 근접 마이크로폰 테크닉스로 사용될 수 있을 것이다.

• 한국소음진동공학회:학술대회논문집
• /
• 한국소음진동공학회 2005년도 춘계학술대회논문집
• /
• pp.334-337
• /
• 2005

The speech intelligibility is one of the most important factors in communication system. The speech intelligibility is related with speech to noise ratio. To enhance the speech to noise ratio, background noise reduction techniques are being developed. As a part of solution to noise reduction, this paper introduces directional microphone using beamforming method and speech filtering method. The directional microphone narrows the spatial range of processing signal into the direction of the target speech signal. The noise signal located in the same direction with speech still remains in the processing signal. To sort this mixed signal into speech and noise, as a following step, a speech-filtering method is applied to pick up only the speech signal from the processed signal. The speech filtering method is based on the characteristics of speech signal itself. The combined directional microphone and speech filtering method gives enhanced performance to speech intelligibility in communication system.

• 한국수소및신에너지학회논문집
• /
• 제29권2호
• /
• pp.163-169
• /
• 2018

The present article discusses about the measurement techniques of acoustic impedance that becomes one of the important acoustic characteristics of various boundaries found inside of propulsion systems. Acoustic characteristics including acoustic impedance and reflection coefficient can be often assessed and estimated by use of the two-microphone method. Theoretical expressions of acoustic impedance and reflection coefficient measured in an impedance tube are presented for both cases with mean flow and without flow, and the practical application of the method through calibration is also provided. The acoustic impedance and the reflection coefficient are related with axial locations of microphones, thermodynamic characteristics of gas inside, and the transfer function between the pressure wave measurements at multiple locations.

• 말소리와 음성과학
• /
• 제4권4호
• /
• pp.87-92
• /
• 2012

Speech enhancement aims to improve speech quality by removing background noise from noisy speech. Independent vector analysis is a type of frequency-domain independent component analysis method that is known to be free from the frequency bin permutation problem in the process of blind source separation from multi-channel inputs. This paper proposed a new method of microphone array based speech enhancement that combines independent vector analysis and beamforming techniques. Independent vector analysis is used to separate speech and noise components from multi-channel noisy speech, and delay-sum beamforming is used to determine the enhanced speech among the separated signals. To verify the effectiveness of the proposed method, experiments for computer simulated multi-channel noisy speech with various signal-to-noise ratios were carried out, and both PESQ and output signal-to-noise ratio were obtained as objective speech quality measures. Experimental results have shown that the proposed method is superior to the conventional microphone array based noise removal approach like GSC beamforming in the speech enhancement.

• 한국소음진동공학회:학술대회논문집
• /
• 한국소음진동공학회 2001년도 추계학술대회논문집 II
• /
• pp.1335-1340
• /
• 2001

The information technology brought us many kinds of conveniences to our life, but it also caused social problems such as privacy interference, unexpected personal information leaks, and nose generation by telephonic talks, etc. In this paper, the microphone system of the cellular phone is developed to prevent these problems caused by progress of information technology. The developed system was designed to detect only acoustic signals from a human being in the presence of various kinds of background noises. A windscreen was designed by use of micro-channels to eliminate the popping noise by the wind from the mouth of a speaker and four microphone array and signal processing techniques are applied to reduce background noise. The impact of the developed system was evaluated by experimental tests. The results show that the system can improve the required functions considerably.

• 한국소음진동공학회:학술대회논문집
• /
• 한국소음진동공학회 2009년도 추계학술대회 논문집
• /
• pp.225-228
• /
• 2009

The simulated indoor pass-by noise measurement system is the tool to measure and evaluate the pass-by noise at the test laboratory, without doing measurement at the field. This measurement system can realize the precision measurement under the specific condition and overcome the limitations of the field measurement, i.e. weather conditions, repeatability, .. This measurement system is done in time domain process using the array techniques, which synchronizes the time signals. The reliability of the obtained result depends on the array shapes, which can generate the moving source effect. In this paper, the validations are checked focusing the time domain synchronization of the signals with the optimum microphone array shape.

• 한국소음진동공학회:학술대회논문집
• /
• 한국소음진동공학회 2005년도 춘계학술대회논문집
• /
• pp.164-169
• /
• 2005

Microphone array techniques are being used widely in wind tunnel measurements for identification of the distributed aerodynamic noise sources on the model being tested. Depending on the frequencies and sound levels, conventional beamforming algorithm has limitation in separating two adjacent sources. Several modifications to the classical beamforming have been developed to enhance way resolution and reduce sidelobe levels. In this Paper the robust adaptive beamforming and the CLEAN algorithm are used to compare to the result of conventional beamforming method. It is found that the CLEAN algorithm is capable of pin-pointing locations of multiple sources nearby, while these sources are unidentifiable with robust adaptive or conventional beamforming techniques.

• 대한의용생체공학회:의공학회지
• /
• 제32권3호
• /
• pp.198-206
• /
• 2011

Hearing devices like cochlear implant, vibrant soundbridge, etc. try to offer better sound for people. In hearing devices, several beamformers including conventional directional microphone are applicable to noise reduction. Each beamformer has different directional response and it could change sound intelligibility or quality for listeners. Therefore, we investigated the performance of three beamformers, which are first and second order directional microphone, and broadband beamformer(BBF) with a computer simulation assuming hearing device microphone configuration. We also calculated objective measurements which have been used to evaluate speech enhancement algorithms. In the simulation, a single speech and a single babble noisewere propagated from the front and $135^{\circ}$ azimuth degrees respectively. Microphones were configured in an end-fire array and the spacing was varied in comparison. With 3 cm spacing, BBF had about 3 dB higher enhanced SNR than that of directional microphones. However, enhancement of segmental SNR and frequency weighted segmental SNR were similar between the first order directional microphone and broadband beamformer. In addition when steady state noise was used, broadband beamformer showed the increased performance and had the highest enhanced SNR, and segmental SNR.

• 대한음성학회지:말소리
• /
• 제50호
• /
• pp.111-122
• /
• 2004

Automatic detection of laryngeal diseases by voice is attractive because of its non-intrusive nature. Cepstrum based approach to detect laryngeal cancer shows reliable performance even when the periodicity of voice signals is severely lost, but it has a drawback that it is not robust to channel mismatch due to different microphone characteristics. In this paper, to deal with mismatched training and test microphone conditions, we investigate channel compensation techniques such as Cepstral Mean Subtraction (CMS) and Pole Filtered CMS (PFCMS). According to our experiments, PFCMS yields better performance than CMS. By using PFCMS, we obtained 12% and 40% error reduction over baseline and CMS, respectively.

• 대한음성학회지:말소리
• /
• 제66호
• /
• pp.73-86
• /
• 2008

Beamforming is one of the spatial filtering techniques which extract only desired signals from noisy environments using microphone arrays. Fixed beamforming is a simple concept and easy to implement. However, it does not show good performance in real noisy conditions. As an adaptive beamforming, Frost algorithm can be a good candidate. It uses the concept of the linearly constrained minimum variance (LCMV) algorithm. The difference between the Frost and the LCMV algorithm is the error correction scheme which is very effective feature in the aspect of performance. In this paper, as quadrature mirror filtering (QMF)-based filterbank is utilized as the pre-processing of the Frost beamformning, the filter length and the learning rate of each band is optimized to improve the performance. The performance is measured by the signal-to-noise ratio (SNR) and the Bark's scale spectral distortion (BSD).