Keunsang Lee

Feasibility of bone assessment with leaky Lamb waves in bone phantoms and a bovine tibia

In this study, the effect of cortical thickness variation on the propagation of leaky Lamb waves is investigated by using an axial transmission technique commonly used to characterize long bones. Three Lucite plates with thicknesses of 1, 3, and 5 mm as bone phantoms and one bovine tibia with a cortical thickness of 2 mm were used at various low frequencies. Experimental measurements in bone phantoms show that the peak frequency and amplitude of excited Lamb modes strongly depend on the thickness of the Lucite plate. In the bovine tibia, the S0 and A0 Lamb modes are consistently observed in the frequency-thickness region from 0.2 to 1.0 MHz mm, and can be effectively launched at a frequency of 200 kHz, suggesting 200 kHz to be the optimal signal frequency for in vivo clinical applications. It can be also seen that both modes are affected by the frequency-thickness product, but the effect is greater for the A0 mode. Hence, the A0 Lamb mode seems more sensitive to cortical thickness change due to aging and osteoporosis. This study suggests that the use of leaky Lamb waves is feasible for ultrasonic bone assessment.

Robust adaptive feedback canceller based on modified pseudo affine projection algorithm

This paper presents an adaptive feedback cancellation (AFC) algorithm with robust and stable performance. The proposed algorithm is based on the pseudo affine projection (PAP) algorithm which approximates the affine projection (AP) with a complexity comparable to the NLMS. Direct application of the PAP to AFC, however, often exhibits instability because of the delayed estimate of the linear prediction (LP) coefficients. This problem is solved in the proposed algorithm by utilizing an inverse gain filter (IGF) before the update of adaptive filter and by estimating the LP coefficients from the input signal without delay. Simulation results confirmed robustness and stability of the proposed algorithm

Nonlinear Acoustic Echo Cancellation Using a Nonlinear Postprocessor With a Linearly Constrained Affine Projection Algorithm

This brief presents a nonlinear acoustic echo cancellation algorithm comprising a postprocessor that can model the saturation-type nonlinear distortion of a microphone circuit. To support fast adaptation to the acoustic echo environment, the linear filter is adapted using an affine projection algorithm where the reference input vectors subject to linear operation are dynamically selected to update the coefficient vector. We consider the hard-clipper and sigmoid functions for the postprocessor, and their parameters are adaptively updated using the least-mean-square algorithm. The simulation results confirm that the proposed algorithm achieves fast and stable convergence under saturation-type microphone distortion, and it can have better performance than conventional algorithms.

Low-Frequency Active Echo Reduction Using a Tile Projector

ABSTRACT: With the advent of submarine detection technology using low-frequency active sonar there is a call for a new submarine stealth device that can replace the exis ting passive anechoic tiles. Proposed in this study is a low-frequency echo reduction technique based on active impedance matching, which employs a tile projector designed to cover a wide area such as the surface of a ship. To judge the feasibility of the active impedance matching technique finite-element simulations of low-frequency echo reduction are performed. Based on the analysis, a tile projector is designed, fabricated, and tested in an acoustic tank for its low-frequency echo reduction performance. Keywords: Low-frequency, Echo reduction, Tile projector, Active impedance matching PACS numbers: 43.50.Ki, 43.20.El† Corresponding author: Won-Suk Ohm (ohm@yonsei.ac.kr) School of Mechanical Engineering, Yonsei University, 50 Yonsei-Ro, Seodaemun-Gu, Seoul 120-749, Republic of Korea(Tel: 82-2-2123-5819, Fax: 82-2-312-2159)

Partitioned Block Frequency Domain Adaptive Filtering Algorithm for Nonlinear Acoustic Echo Cancellation

This paper proposes a robust nonlinear acoustic echo canceller (NAEC) which is effective for modeling the nonlinearity of a speaker module and the long acoustic echo path within a speech communication environment. The proposed NAEC utilizes a sigmoid pre-processor for modeling the speaker nonlinearity and a partitioned block frequnecy-domain adaptive filter for identifying the acoustic echo path with small delay. Simulation results confirmed that the proposed algorithm achieves excellent performance with much lower computational complexity than the previous NAEC.

Secondary Path Estimation Algorithm Based on Residual Music Canceller for Noise Cancelling Headphone

An active noise control (ANC) algorithm for noise canceling headphone is proposed. In this study, the feedback ANC operated with the filtered-x least mean square algorithm (FxLMS) algorithm is used to attenuate the undesired noise. Also an adaptive residual music canceller (RMC) is proposed for enhancing the accuracy of the reference signal of the feedback ANC. Simulation results show that a high quality of music sound can be consistently achieved in a time-varying secondary path situation.

A feedback active noise control for in-ear headphone with robust on-line secondary path modeling

The active noise control (ANC) technique can be used to improve the comfortness and quality of the music listening with an in-ear headphone. For a compact headphone design, the feedback ANC operated with the filtered-x LMS (FxLMS) algorithm is preferred. In practice, the error in the estimated secondary path can be problematic for the FxLMS algorithm, since it can lead the algorithm to an unstable state, and thus cause degradation of sound quality with the in-ear headphone. In this study, an adaptive residual music canceler (RMC) is proposed for enhancing the accuracy of the reference signal of the feedback ANC. Since RMC is designed to track the bias of the current secondary path estimate, the secondary path is continuously updated by combining the previous estimate of the secondary path with the current weight vector of RMC. In addition, variable step-size schemes are developed for both the control and secondary path estimation filters, which enable the ANC system to adapt quickly and robustly to the va...

Channel-prediction based noise reduction algorithm for dual microphone smartphones

In this paper, a noise reduction algorithm for dual-microphone mobile phones is proposed, in which a frequency-domain channel prediction filter combined with a variable step-size scheme is utilized to adaptively estimate and suppress the coherent noise component in low-frequency region. To suppress the incoherent noise component, the proposed algorithm estimates a noise power spectral density (PSD) from the error signal of the channel prediction filter using a voice activity detection (VAD) scheme based on a SNR ratio between the two microphone signals. Computer simulations were performed using synthesized and recorded noises, and the results confirmed that the proposed algorithm obtains more accurate noise PSD especially in low-frequency band and better overall speech quality compared with the conventional algorithms.

Novel Variable Step-Size Gradient Adaptive Lattice Algorithm for Active Noise Control

In this paper, a novel variable step-size filtered-x gradient adaptive lattice (NVSS-FxGAL) algorithm for active noise control system is proposed. The gradient adaptive lattice (GAL) algorithm is capable of controlling the narrow band noise effectively. The GAL algorithm can achieve both fast convergence rate and low steady-state level using the variable step-size. However, it suffers from the convergence performance for varying signal characteristic since the global variable step-size is equally applied to all lattice stages. Therefore, the proposed algorithm guarantees the stable and consistency convergence performance by using the local variable step-size for the suitable each lattice stage. Simulation results confirm that the proposed algorithm can obtain the fast convergence rate and low steady-state level compared to the conventional algorithms.

Development and Implementation of Noise-Canceling Technology for Digital Stethoscope

In this paper, an algorithm for suppressing acoustic noises contained in stethoscope sound is proposed and implemented in real-time using an embedded DSP system. Sound collected by stethoscope is down-sampled and band-pass filtered, and later an NLMS adaptive filter is used to cancel the acoustic noise induced from external noise sources. Also, the unpredictable impulsive noises due to fabric friction and instantaneous tapping are detected using the SD-ROM algorithm, and suppressed using an algorithm approximating the morphology filter. The proposed algorithm was tested using signals collected with a digital stethoscope mockup, and implemented on an ARM920T-based DSP system.