• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
• /
• 2009.10a
• /
• 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.

• The Journal of the Institute of Internet, Broadcasting and Communication
• /
• v.14 no.1
• /
• pp.31-36
• /
• 2014

A study on the indoor positioning system has been active recently for the location based service indoors. In the 3 dimensional positioning system based on the acoustic signal and TDOA technology, the error characteristics of the estimated source position would be changed depending on the number of microphones and the pattern of the microphone array. In this paper, the estimated position error according to the measured distance error between the microphones and the signal source is analyzed, and the optimum microphone array is decided considering the estimated position error patterns and the total amount of the estimated position error.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
• /
• 1999.11a
• /
• pp.173-176
• /
• 1999

Recently, speaker localizing estimation technique has been rising in teleconference systems. In this system, it is recognized speaker location using microphone array and camera is directed to speaker location automatically. In this paper, it was described to be able to enhance the speech qualify through microphone array, decrease computational loads using IIR filter as inverse filter, and confirmed to implement hardware using DSP processor.

• Proceedings of the KSPS conference
• /
• 2002.11a
• /
• pp.187-190
• /
• 2002

In this paper, a speech enhancement system using microphone array with MMSE-STSA (Minimum Mean Square Error-Short Time Spectral Amplitude) estimator based post-processing is proposed. Speech enhancement is first carried out by conventional delay-and-sum beamforming (DSB). A new MMSE-STSA estimator is then obtained by refining MMSE-STSA estimators from each microphone, which is applied to the output of conventional DSB to obtain additional speech enhancement. Computer simulation for white and pink noises show that the proposed system is superior to other approaches.

• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.26 no.5
• /
• pp.808-811
• /
• 2022

In this paper, a wide-range microphone system for lectures using dual 3D sensors is proposed. A previous work using a single sensor had lowering the detecting threshold to support wide-area. However it was found that an error occurred when lecturer wears clothes with low reflectivity or has small body size. When multiple sensors are used to expand the coverage it could be cause various problems. Each sensor could show different distance to the same target. We derive the rotation angle and and compensate for lecturing microphone system using sensors on the line. The proposed method shows a little improvement in performance by about 1dB compared to the previous works but the performance is uniform in all areas regardless of reflectivity.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
• /
• 2009.10a
• /
• pp.723-726
• /
• 2009

In this research, We simulated performances of error reduction algorithm for the speech signal based on the microphone array-based beamforming method in speech recognition system and analyzed its performance. Also, we processed speech signal adopted from microphone array and maximum signal to noise ratio for each channel, and then compared them with signal to noise ratio of speech signal. Speech recognition rate is improved from 54.2% to 61.4% in case 1 and is improved from 41.2% to 50.5% in case 2 of the lower signal to noise ratio. Therefore the average reduction rates are showed 15.7% in case 1.

• The Journal of the Acoustical Society of Korea
• /
• v.17 no.2E
• /
• pp.38-46
• /
• 1998

In this paper, we propose a remote speech input device, a new method of user-friendly speech input in spontaneous speech recognition system. We focus the user friendliness on hands-free and microphone independence in speech recognition applications. Our method adopts two algorithms, the automatic speech detection and the microphone array delay-and-sum beamforming (DSBF)-based speech enhancement. The automatic speech detection algorithm is composed of two stages; the detection of speech and nonspeech using the pitch information for the detected speech portion candidate. The DSBF algorithm adopts the time domain cross-correlation method as its time delay estimation. In the performance evaluation, the speech detection algorithm shows within-200 ms start point accuracy of 93%, 99% under 15dB, 20dB, and 25dB signal-to-noise ratio (SNR) environments, respectively and those for the end point are 72%, 89%, and 93% for the corresponding environments, respectively. The classification of speech and nonspeech for the start point detected region of input signal is performed by the pitch information-base method. The percentages of correct classification for speech and nonspeech input are 99% and 90%, respectively. The eight microphone array-based speech enhancement using the DSBF algorithm shows the maximum SNR gaing of 6dB over a single microphone and the error reductin of more than 15% in the spontaneous speech recognition domain.

• Journal of the Korean Society for Precision Engineering
• /
• v.26 no.11
• /
• pp.20-28
• /
• 2009

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
• /
• 2004.11a
• /
• pp.1032-1035
• /
• 2004

Pressure distribution is measured by way microphones to identify noise sources in the space. For example, beam-forming method or acoustic holography use phase information to identify the source. Therefore, the phase is significant information to correctly identify the source position. However, due to the microphone characteristics and measuring systems, measured signals always have errors, which make the identification difficult. Therefore, phase calibration of microphones is needed. Duct and speaker systems are generally used as calibrators. Acoustic characteristics of the calibrator are, of course, functions of many Parameters of the system: i.e. duct size, frequency, and microphone spacing. In this paper, design parameters which effect on the performance and size of the calibrators are considered. Then the parameters would be applied to design and real product of the phase-calibrator.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.29 no.11C
• /
• pp.1573-1580
• /
• 2004

In this paper, an adaptation mode controller for adaptive microphone array system is proposed for high-quality speech acquisition in real environments. To ensure proper adaptation of the adaptive array algorithm, the proposed adaptation mode controller uses not only temporal information, but also spatial information. The proposed adaptation mode controller is constructed with two processing stages: an initialization stage and a running stage. In the initialization stage, a sound source localization technique is adopted, and a signal correlation characteristic is used in the running stage. For the adaptive may algorithm, a generalized sidelobe canceller with an adaptive blocking matrix is used. The proposed adaptation mode controller can be used even when the adaptive blocking matrix is not adapted, and is much stable than the power ratio method. The proposed algorithm is evaluated in real environment, and simulation results show 13dB SINR improvement with the speaker sitting 2m distance from the may.