• Proceedings of the IEEK Conference
• /
• 2000.09a
• /
• pp.609-612
• /
• 2000

본 논문에서는 음원에 관한 정보가 미지의 상황에서 마이크로폰 어레이를 사용하여 두 음원신호를 분리하는 ,시스템을 제안한다 이 시스템은 두 단계로 구성되어 있으며, 첫 번째 단계에서는 파워가 큰 제 1음원의 DOA(Direction Of Arrival)를 추정하고, AMUSE(Algorithm for Multiple Unknown Signals Extraction)법을 사용한 Blind Deconvolution에 의해 음원신호의 분리를 행한다 두 번째 단계에서는 파워가 낮은 제 2음원의 강조신호를 사용하여 DSA(Delay and Sum Array)법에 의해 제 2음원의 DOA를 추정하고,AMUSE법의 출력신호와 두 음원의 DOA를 이용하여 ANF(Adaptive Notch Filter)를 구성하고, 두 음원신호의 재 분리를 행한다. 그리고, 시뮬레이션을 통해 제안한 방법의 유효성을 검토한 결과 두 음원 신호가 분리 가능한 것이 확인되었다.

• Journal of Korea Multimedia Society
• /
• v.8 no.3
• /
• pp.328-335
• /
• 2005

This paper proposes a frequency domain independent component analysis (ICA) algorithm to separate the mixed speech signals using a multiple microphone array By estimating the delay timings using a input cross-correlation, even in the delayed mixture case, we propose a good initial value setting method which leads to optimal convergence. To reduce the calculation, separation process is performed at frequency domain. The results of simulations confirms the better performances of the proposed algorithm.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
• /
• 2000.06a
• /
• pp.1479-1486
• /
• 2000

This paper introduces the improved moving frame acoustic holography (MFAH) method and its application. MFAH allows us to visualize the noise generated by moving noise sources by employing a vertical line array of microphones affixed to the ground. The sound field generated by moving sources is different from that of stationary ones due to the movement of the sources. Therefore the measured sound pressure by the microphone on the ground has to be processed so that it cooperates the effect of the movement. This paper discusses the effect of moving noise sources on the obtained hologram by MFAH. This assures the applicability of MFAH to the visualization of moving sources. This paper also reviews the improved MFAH that can visualize a coherent narrow band noise and a pass-by noise. The practical applicability of the improved MFAH was demonstrated by visualizing tire noise during a pass-by test.

• The Journal of the Acoustical Society of Korea
• /
• v.36 no.5
• /
• pp.345-352
• /
• 2017

We propose a methods which is reducing direction estimation error of sound source in the reverberant and noisy environments. The proposed algorithm divides speech signal into voice and unvoice using VAD. We estimate the direction of source when current frame is voiced. TDOA (Time-Difference of Arrival) between microphone array using the GCC-PHAT (Generalized Cross Correlation with Phase Transform) method will be estimated in that frame. Then, we compare the peak value of cross-correlation of two signals applied to estimated time-delay with other time-delay in time-table in order to improve the accuracy of source location. If the angle of current frame is far different from before and after frame in successive voiced frame, the angle of current frame is replaced with mean value of the estimated angle in before and after frames.

• The Journal of the Acoustical Society of Korea
• /
• v.20 no.4
• /
• pp.22-28
• /
• 2001

This paper describes the implementation of a robust speaker position location system using the voice signal received by microphone array. To be robust to the reverberation which is the major factor of the performance degradation, low-frequency phase restoration algorithm which eliminates the influence of reverberations using the low-frequency information of the CPSP function is proposed. The implemented real-time system consists of a general purpose DSP (TMS320C31 of Texas instruments), analog part which contains amplifiers and filters, and digital part which is composed of the external memory and 12-bit A/D converter. In the real conference room environment, the implemented system that was constructed by the proposed algorithms showed better performance than the conventional system. The error of the TDOA estimation reduced more than 15 samples.

• IEMEK Journal of Embedded Systems and Applications
• /
• v.8 no.4
• /
• pp.185-194
• /
• 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.

• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.19 no.1
• /
• pp.61-68
• /
• 2015

Noise reduction is necessary to compensate for the degradation of recognition performance by various types of noises. Among many noise reduction techniques using microphone array, generalized sidelobe canceller (GSC) has been widely applied to reduce nonstationary noises. The performance of GSC is directly affected by its adaptation mode controller (AMC). That is, accurate target speech detection is essential to guarantee the sufficient noise reduction in pure noise intervals and the less distortion in target speech intervals. Thus, this paper proposes an improved AMC design technique in which the power ratio of the output of fixed beamforming to that of blocking matrix is calculated frequency bandwise and probabilistically modeled by mixture Gaussians for each class. Experimental results show that the proposed algorithm outperforms conventional AMCs in receiver operating curves (ROC) and output SNRs.

• Journal of KSNVE
• /
• v.10 no.4
• /
• pp.669-678
• /
• 2000

This paper reviews the improved moving frame acoustic holography (MFAH) method and its application. Moving frame acoustic holography was originally proposed to increase the aperture size and the spatial resolution of hologram by using a moving line array of microphones. The hologram of scanned plane can be obtained by assuming the sound field to be product of spatial and temporal information. Although conventional MFAH was only applied to sinusoidal signals, it allows us to visualize the noise generated by moving noise sources by employing a vertical line array of microphones affixed to the ground. However, the sound field generated by moving sources becomes different from that of stationary ones due to the movement of the sources. Firstly, this paper introduces the effect of moving noise sources on the obtained hologram by MFAH and the applicability of MFAH to the visualization of moving sources. Secondly, this paper also reviews improved MFAH that can visualize a coherent narrow band noise and a pass-by noise. The practical applicability of the improved MFAH was demonstrated by visualizing tire noise during a pass-by test.

• The Journal of the Acoustical Society of Korea
• /
• v.13 no.5
• /
• pp.15-23
• /
• 1994

We developed an electronic lens for acoustic imaging systems, which is linear array with 31 microphones equally spaced with distance 34mm. Resonant frequency fo receiver circuit coupled to microphone is 20 kHz. We arranged 16 microphones horizontally and 15 microphones vertically, so that the array allows us to obtain a 2 dimensional angle of source, and to track the motion of source in real time. Due to the problem of aliasing in discrete Fourier Transfrom, the maximum observable angle of the lens is limited to 15${\circ}$. We also employed quadrature phase detection scheme to adjust the focus. We have tested the acoustic lens with a personal computer in an anechoic room and obtained the results agreed with the acoustic imaging theory.

• The Journal of the Acoustical Society of Korea
• /
• v.20 no.4E
• /
• pp.52-60
• /
• 2001

This paper describes real time implementation of subband sound localization system on a floating-point DSP TI TMS320C31. The system determines two dimensional location of an active speaker in a closed room environment with real noise presents. The system consists of an two microphone array connected to TI DSP hosted by PC. The implemented sound localization algorithm is Subband CPSP which is an improved version of traditional CPSP (Cross-Power Spectrum Phase) method. The algorithm first split the input speech signal into arbitrary number of subband using subband filter banks and calculate the CPSP in each subband. It then averages out the CPSP results on each subband and compute a source location estimate. The proposed algorithm has an advantage over CPSP such that it minimize the overall estimation error in source location by limiting the specific band dominant noise to that subband. As a result, it makes possible to set up a robust real time sound localization system. For real time simulation, the input speech is captured using two microphone and digitized by the DSP at sampling rate 8192 hz, 16 bit/sample. The source location is then estimated at once per second to satisfy real-time computational constraints. The performance of the proposed system is confirmed by several real time simulation of the speech at a distance of 1m, 2m, 3m with various speech source locations and it shows over 5% accuracy improvement for the source location estimation.