• Proceedings of the Korean Institute of Communication Sciences Conference
• /
• 1983.04a
• /
• pp.16-17
• /
• 1983

In Speech signal processing, ARMA spectral estimation method is used. It has been demonstrated that the ARMA model provides better spectral estimation then the more specialized AR model and MA model. Dynamic program is used to achieve time algnment. Speech sound similarity is defined to be proportional to the distance seperating to sound in a vector space defined by ARMA model. AS a result, the recognition rate of 97.3% for three speaker is obtained.

• Proceedings of the KIEE Conference
• /
• 2006.10c
• /
• pp.419-421
• /
• 2006

The importance of automatic discrimination image data has evolved as a research topic over recent years. We have used forward neural network as a classifier using sound data features within image data, our initial tests have shown encouraging results that indicate the viability of our approach.

• Proceedings of the Korean Society of Machine Tool Engineers Conference
• /
• 1996.10a
• /
• pp.85-90
• /
• 1996

In order to detect and suppress chatter in turning processes a stability control methodology was studied through manipulation of spindle speeds regarding to chatter frequencies. The chatter frequency was identified by monitoring and signal processing of sound pressure during turning on a lathe. The stability control methodology can select stable spindle speeds without knowing a prior knowledge of machine compliances and cutting dynamics. Teliability of the developed stability control methodology was verified through turning experiments on an engine lathe. Experimental results show that a microphone is an excellent sensor for chatter detection and control

• Proceedings of the KIEE Conference
• /
• 1999.07g
• /
• pp.2905-2907
• /
• 1999

The meaningful speech sound block classification provides very important information in the speech recognition. The following technique of the classification is based on the DWT (discrete wavelet transform), which will provide a more fast algorithm and a useful, compact solution for the pre-processing of speech recognition. The algorithm is implemented to the unvoiced/voiced classification and the denoising.

• The Journal of the Acoustical Society of Korea
• /
• v.21 no.3
• /
• pp.245-255
• /
• 2002

This thesis deals a method to deliver more realistic sound by cancelling the cross-talk which is inherent to the 5.1 channel speaker system. The acoustical model for cross-talk cancellation is the free field model. This model minimizes distortion of sound. I used the bark scale sound quality compensation which based on psycho-acoustic. For the surround channels, band-limited sound quality compensation is performed in the frequency domain. I also performed the sound quality assessment test on the traditional 2 channel stereo and 5.1 channel system. This test is performed in the test chamber which satisfies the ITU-R specifications. I uses the IACC (Inter-Aural Cross-Correlation) to determine the preferences of the amateur and the golden ear experts to asses the trans-aural filter. According to the result from the proposed method, I got more the 38 dB separation rates with the Dolby standard speaker array. The results on the diffusion by the subjective test with the experts shows 0.4 point increased then before.

• Journal of the Institute of Electronics Engineers of Korea SP
• /
• v.41 no.2
• /
• pp.99-108
• /
• 2004

This thesis deals a method to deliver more realistic sound by cancelling the cross-talk which is inherent to the 5.1 channel speaker system. The acoustical model for cross-talk cancellation is the free field model. This model minimizes distortion of sound. I used the bark scale sound quality compensation which based on psycho-acoustic. For the surround channels, band-limited sound quality compensation is performed in the frequency domain. I also performed the sound quality assessment test on the traditional 2 channel stereo and 5.1 channel system. This test is performed in the test chamber which satisfies the ITU-R specifications. I uses the IACC(Inter-Aural Cross-Correlation) to determine the preferences of the amateur and the golden ear experts to asses the trans-aural filter. According to the result from the proposed method, I got more the 38㏈ separation rates with the Dolby standard speaker array. The results on the diffusion by the subjective test with the experts shows 0.4 point increased then before.

• The Journal of the Acoustical Society of Korea
• /
• v.43 no.1
• /
• pp.29-38
• /
• 2024

The Matched Field Processing (MFP) is an estimation method for a source range and depth based on the prediction of sound propagation. However, as the frequency increases, the prediction inaccuracy of sound propagation increases, making it difficult to estimate the source position. Recently proposed, the Frequency-Difference Matched Field Processing (FD-MFP) is known to be robust even if there is a mismatch by applying a frequency-difference autoproduct extracted from the auto-correlation of a high frequency signal. In this paper, in order to evaluate the performance of the FD-MFP using a horizontal line array, simulations were conducted in the environment of the East Sea of Korea. In the area of Bottom Bounce (BB) and Convergence Zone (CZ) where detection of a sound source is possible at a long range, and the results of localization were analyzed. According to the the FD-MFP simulations of horizontal line array, the accuracy of localization is similar or degraded compared to the conventional MFP due to diffracted field and mismatch of sound speed. There was no clear result from the simulations conforming that the FD-MFP was more robust to mismatch than the conventional MFP.

• 한국HCI학회:학술대회논문집
• /
• 2009.02a
• /
• pp.158-162
• /
• 2009

Sound effects coming with appropriate tactile stimuli can strengthen its reality. For example, gunfire in games and movies, if it is accompanied by vibrating effects, can enhance the impressiveness. On a similar principle, adding the vibration information to existing sound data file and playing sound while generating vibration effects through haptic interfaces can augment the sound experience. In this paper, we propose a method to generate vibration information by analyzing the sound. The vibration information consists of vibration patterns and the timing within a sound file. Adding the vibration information is labor-intensive if it is done manually. We propose a sound detection algorithm to search the moments when specific sounds occur in a sound file and a method to create vibration effects at those moments. The sound detection algorithm compares the frequency characteristic of specific sounds and finds the moments which have similar frequency characteristic within a sound file. The detection ratio of the algorithm was 98% for five different kinds of gunfire. We also develop a GUI based vibrating pattern editor to easily perform the sound search and vibration generation.

• Journal of the Korean Institute of Telematics and Electronics B
• /
• v.33B no.4
• /
• pp.156-163
• /
• 1996

In this paper, we described the implementation and the sound quality assessment of a real-time stereo audio codec using TMS320C40 DSP (digital signal processing) chip for low bitrte and high quality audio. We implemented hardware and software in order to overcome a real-time processing problem of audio compression algorithm that can be produced by largely recursive computing and complexity of the process. We have studied five types of distortion that can be produced by perceptual coding and the codec was evaluated by eight test musics that are selected in SQAM (sound quality assessment material) 422-2-4-2 produced by EBU (european broadcast union). The subjective listening tests were carried out on the codec quality and preformance by double blind method in a listening room with eleven listeners. As a result, 5 grade-impairment scale was scored under minus one and the codec quality was evaluated to be perceptible, but not annoying.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
• /
• 1996.04a
• /
• pp.291-297
• /
• 1996

To visualize sound field or to identify noise sources, we can use many methods such as intensity method, acoustic holographic method, source identification method using line array, etc. Conventionally all these methods are performed with the assumption of stationary condition in space and time. But for moving source, spatial characteristics and frequency components are changing, so we need another processing algorithm. This paper shows some experimental results - sound field by moving noise sources. In the experiment cross type microphone line array is used for sensing pressure and cars and a motorcycle are used as moving sources that are assumed to have constant speed. The processing methods are acoustic holographic method, spherical beamforming and spectrogram.