• Title/Summary/Keyword: Microphone Signal

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Active noise control with the active muffler in automotive exhaust system (액티브 머플러를 이용한 자동차 배기계의 능동소음제어)

  • Kim, Heung-Seob;Hong, Jin-Seok;Oh, Jae-Eung
    • Transactions of the Korean Society of Mechanical Engineers A
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    • v.21 no.11
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    • pp.1837-1843
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    • 1997
  • This study experimentally demonstrates the use of active muffler attached to the automotive exhaust system to reduce exhaust noise. For improving the signal to noise ratio in the process of estimation of secondary path transfer functions, the on-line algorithm that conventional inverse modeling is combined with adaptive line enhancer is used as the control algorithm. Active muffler is designed that the primary noise and the control sound are propagated as a plane wave in the outlet. Therefore, the error microphone could be placed out of the tail pipe center of a high temperature and the radiation noise to the outside could be reduced in the whole area around the outlet. The control experiment for reducing exhaust noise with active muffler is implemented during run-up at no load. From the experimental results presented, compared with the conventional off-line method, the proposed on-line method is capable to acquire a reduction of exhaust noise above 5 dB in overall sound power level.

A Study on Precise Control of Autonomous Travelling Robot Based on RVR (RVR에 의한 자율주행로봇의 정밀제어에 관한연구)

  • Shim, Byoung-Kyun;Cong, Nguyen Huu;Kim, Jong-Soo;Ha, Eun-Tae
    • Journal of the Korean Society of Industry Convergence
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    • v.17 no.2
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    • pp.42-53
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    • 2014
  • Robust voice recognition (RVR) is essential for a robot to communicate with people. One of the main problems with RVR for robots is that robots inevitably real environment noises. The noise is captured with strong power by the microphones, because the noise sources are closed to the microphones. The signal-to-noise ratio of input voice becomes quite low. However, it is possible to estimate the noise by using information on the robot's own motions and postures, because a type of motion/gesture produces almost the same pattern of noise every time it is performed. In this paper, we propose an RVR system which can robustly recognize voice by adults and children in noisy environments. We evaluate the RVR system in a communication robot placed in a real noisy environment. Voice is captured using a wireless microphone. Navigation Strategy is shown Obstacle detection and local map, Design of Goal-seeking Behavior and Avoidance Behavior, Fuzzy Decision Maker and Lower level controller. The final hypothesis is selected based on posterior probability. We then select the task in the motion task library. In the motion control, we also integrate the obstacle avoidance control using ultrasonic sensors. Those are powerful for detecting obstacle with simple algorithm.

Design of u-Healthcare RF-Tag Based on Heart Sounds of Chest (흉부 심음을 기반한 u-헬스케어용 RF-Tag설계)

  • Lee, Ju-Won;Lee, Byeong-Ro
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.13 no.4
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    • pp.753-758
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    • 2009
  • This paper is proposed the hardware structure and signal processing method of the RF-Tag based on heart sound to develop the mobile biomedical information device for the u-healthcare system. The RF-Tag in this study is consisted of a skin temperature sensor, a dynamic microphone for heart sound detection, Bluetooth communication to transmute healthcare data, and pulse period detection algorithm with adaptive gain controller. We experimented to evaluate performance of the RF-Tag in noisy environments. In addition, the RF-Tag has shown the good performance in the results of experiment. If the proposed methods in this study apply to design the u-healthcare device, we will be able to get the exact health data on real time in mobile environments.

Implementation of Environmental Noise Remover for Speech Signals (배경 잡음을 제거하는 음성 신호 잡음 제거기의 구현)

  • Kim, Seon-Il;Yang, Seong-Ryong
    • 전자공학회논문지 IE
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    • v.49 no.2
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    • pp.24-29
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    • 2012
  • The sounds of exhaust emissions of automobiles are independent sound sources which are nothing to do with voices. We have no information for the sources of voices and exhaust sounds. Accordingly, Independent Component Analysis which is one of the Blind Source Separaton methods was used to segregate two source signals from each mixed signals. Maximum Likelyhood Estimation was applied to the signals came through the stereo microphone to segregate the two source signals toward the maximization of independence. Since there is no clue to find whether it is speech signal or not, the coefficients of the slope was calculated by the autocovariances of the signals in frequcency domain. Noise remover for speech signals was implemented by coupling the two algorithms.

Desgin of Low-power, Low-noise Preamplifier for Digital Hearing-Aids (디지털 보청기를 위한 저전력, 저잡음 전치증폭기 설계)

  • Im, Saemin;Park, Sang-Gyu
    • Journal of the Institute of Electronics and Information Engineers
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    • v.49 no.12
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    • pp.219-225
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    • 2012
  • A low-power, low-noise pre-amplifier for digital hearing-aid application is designed. This pre-amplifier amplifies single-ended signal from an electret microphone, and produces differential output to be delivered to an ADC. It has a variable gain of 3.6, 7.2, 14.4 and 28.8 with a bandwidth between 100Hz~10kHzon. The measurement results show 85 dB of SNR, 0.05 % of harmonic distortion and $200{\mu}W$ of power consumption with 1.2V supply.

Spectrum Sensing Scheme Using the Ratio of the Maximum and the Minimum of Power Spectrum (전력 스펙트럼의 최대 최소 비율을 이용한 스펙트럼 감지 방식)

  • Lim, Chang Heon
    • Journal of the Institute of Electronics and Information Engineers
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    • v.51 no.6
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    • pp.3-8
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    • 2014
  • Recently, a spectrum sensing technique employing the maximum value of a received power spectrum as a test statistic has been presented in the literature for the purpose of detecting a wireless microphone signal in TV bands This detects the presence of a primary user by comparing the test statistic with some threshold, which depends on the background noise power level as well as a target false alarm rate. Therefore its performance may deteriorate when the noise power uncertainty occurs. As a means to mitigate this difficulty, we present a spectrum sensing strategy adopting the ratio of the maximum and the minimum value of the power spectrum as a test statistic and analyze its performance of spectrum sensing.

Multi-channel Speech Enhancement Using Blind Source Separation and Cross-channel Wiener Filtering

  • Jang, Gil-Jin;Choi, Chang-Kyu;Lee, Yong-Beom;Kim, Jeong-Su;Kim, Sang-Ryong
    • The Journal of the Acoustical Society of Korea
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    • v.23 no.2E
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    • pp.56-67
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    • 2004
  • Despite abundant research outcomes of blind source separation (BSS) in many types of simulated environments, their performances are still not satisfactory to be applied to the real environments. The major obstacle may seem the finite filter length of the assumed mixing model and the nonlinear sensor noises. This paper presents a two-step speech enhancement method with multiple microphone inputs. The first step performs a frequency-domain BSS algorithm to produce multiple outputs without any prior knowledge of the mixed source signals. The second step further removes the remaining cross-channel interference by a spectral cancellation approach using a probabilistic source absence/presence detection technique. The desired primary source is detected every frame of the signal, and the secondary source is estimated in the power spectral domain using the other BSS output as a reference interfering source. Then the estimated secondary source is subtracted to reduce the cross-channel interference. Our experimental results show good separation enhancement performances on the real recordings of speech and music signals compared to the conventional BSS methods.

A Study on the Wear Estimation of End Mill Using Sound Frequency Analysis (음향주파수 분석에 의한 엔드밀의 마모상태 추정에 관한 연구)

  • Lee, Chang-Hee;Cho, Taik-Dong
    • Transactions of the Korean Society of Mechanical Engineers A
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    • v.27 no.8
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    • pp.1287-1294
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    • 2003
  • The wear process of end mill is so complicated process that a more reliable technique is required for the monitoring and controlling the tool life and its performance. This research presents a new tool wear monitoring method based on the sound signal generated on the machining. The experiment carried out continuous-side-milling for 4 cases using the high-speed-steel end mill under wet condition. The sound pressure was measured at 0.5m from the cutting zone by a dynamic microphone, and was analyzed at frequency domain. As the cutter impacts the workpiece surface, a situation of farced vibration arises in which the dominant forcing frequency is equal to the tooth passing frequency of the cutter. The tooth passing frequency appears as a harmonics form, and end mill flank wear is related with the first harmonic. It is possible to detect end . mill flank wear. This paper proposed the new method of the end mill wear detection.

Active Noise Control in a Duct System Using the Hybrid Control Algorithm (하이브리드 제어 알고리즘을 이용한 덕트내 능동소음제어)

  • Lee, You-Yub;Park, Sang-Gil;Oh, Jae-Eung
    • Transactions of the Korean Society for Noise and Vibration Engineering
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    • v.19 no.3
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    • pp.288-293
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    • 2009
  • This study presents the active noise control of duct noise. The duct was excited by a steady-state harmonic and white noise force and the control was performed by one control speaker attached to surface of the duct. An adaptive controller based on filtered x LMS(FXLMS) algorithm was used and controller was defined by minimizing the square of the response of the error microphone. The assemble controller, which is called a hybrid ANC(active noise control) system, was combined with feedforward and feedback controller. The feedforward ANC attenuates primary noise that is correlated with the reference signal, while the feedback ANC cancels the narrowband components of the primary noise that are not observed by the reference sensor. Furthermore, in many ANC applications, the periodic components of noise are the most intense and the feedback ANC system has the effect of reducing the spectral peaks of the primary noise, thus easing the burden of the feedforward ANC filter.

Impulsive sound localization using crest factor of the time-domain beamformer output (빔형성기 출력의 파고율을 이용한 충격음의 방향 추정)

  • Seo, Dae-Hoon;Choi, Jung-Woo;Kim, Yang-Hann
    • Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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    • 2014.10a
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    • pp.713-717
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    • 2014
  • This paper presents a beamforming technique for locating impulsive sound source. The conventional frequency-domain beamformer is advantageous for localizing noise sources for a certain frequency band of concern, but the existence of many frequency components in the wide-band spectrum of impulsive noise makes the beamforming image less clear. In contrast to a frequency-domain beamformer, it has been reported that a time-domain beamformer can be better suited for transient signals. Although both frequency- and time-domain beamformers produce the same result for the beamforming power, which is defined as the RMS value of its output, we can use alternative directional estimators such as the peak value and crest factor to enhance the performance of a time-domain beamformer. In this study, the performance of three different directional estimators, the peak, crest factor and RMS output values, are investigated and compared with the incoherent interfering noise embedded in multiple microphone signals. The proposed formula is verified via experiments in an anechoic chamber using a uniformly spaced linear array. The results show that the peak estimation of beamformer output determines the location with better spatial resolution and a lower side lobe level than crest factor and RMS estimation in noise free condition, but it is possible to accurately estimate the direction of the impulsive sound source using crest factor estimation in noisy environment with stationary interfering noise.

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