• The Journal of the Acoustical Society of Korea
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• v.23 no.6
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• pp.437-445
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• 2004

In this paper. we describe a novel method for transforming audio contents according to user's preference on sound field. Sound field effect technologies. which transform or simulate acoustic environments as user's preference, are very important for enlarging the reality of acoustic scene. However huge amount of computational power is required to process sound field effect in real time. so it is hard to implement this functionality at the portable audio devices such as MP3 player. In this paper, we propose an efficient method for providing sound field effect to audio contents independent of terminal's computational power through processing this functionality at the server using user's sound field preference, which is transfered from terminal side. To describe sound field preference, user can use perceptual acoustic parameters as well as the URI address of room impulse response signal. In addition, a novel fast convolution method is presented to implement a sound field effect engine as a result of convoluting with a room impulse response signal at the realtime application. and verified to be applicable to real-time applications through experiments. To verify the evidence of benefit of proposed method we performed two subjective listening tests about sound field descrimitive ability and preference on sound field processed sounds. The results showed that the proposed sound field preference can be applicable to the public.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.14 no.12
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• pp.2798-2804
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• 2010

The audio and image signal are corrupted by various noises in signal processing, many studies are being accomplished to restore those signals. In this paper, the algorithm is proposed to remove additive Gaussian noise and impulse noise at one dimension signal like an speech signal. The algorithm is composed to remove Gaussian noise after removing impulse noise. And the method using wavelet coefficient accumulation is used to remove the Gaussian noise, and the median filter based on element deviation is applied to remove the impulse noise. Also we compare existing methods using SNR(signal-to-noise ratio) as the standard of judgement of improvemental effect.

• International Journal of Highway Engineering
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• v.15 no.6
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• pp.41-47
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• 2013

PURPOSES : The purpose of this study is about noise which is generated from roads and is consist of irregular frequency variation from low frequency to various band. The existing methods of noise reduction are sound barrier that uses insulation material and absorbing material or have applied passive technology of noise reduction by devices. The total frequency band is needed to apply active noise control. METHODS : In this study applies to the field of road traffic environment, signal processing controller and various analog signal input/output, the amplifier module is based on parallel-core embedded processor designed. DSP performs the control algorithm of the road traffic noise. Noise sources in the open space performance of evaluation were applied. In this study, controller of active signal processor was designed based on the module of audio input/output and main controller of embedded process. The controller of active signal processor operates noise reduction algorithm and performance tests of noise reduction in inside and outside environment were executed. RESULTS : The signal processing controller with OMAP-L137 parallel-core processors as the center, DSP processors in the active control operations dealt with quickly. To maximize the operation speed of an object and ARM processor is external function keys and display for functions and evaluating the performance management system was designed for the purpose of the interface. Therefore the reduction of road traffic noise has established an electronic controller-based noise reduction. CONCLUSIONS : It is shown that noise reduction is effective in the case of pour tonal sound and complex tonal sound below 500Hz by appling to Fx-LMS.

• Proceedings of the IEEK Conference
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• 1998.10a
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• pp.1227-1230
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• 1998

Phase control methods are used to expand the sound image in general AV system. However, these methods are effective only to the signal under 1kHz, and the listener must be located in front center of the speaker system. In this paper, we realize the realtime processing system in which phase shifting method is dominant at low frequency and precedence effect is dominant at high frequency. Two sound cards are used to process the audio signal in realtime with 16 bits stereo channel of 44.1 kHz sampling frequency. And the analog circuit is designed to process the phase shifting. In experiments the usefulness of the proposed stereo system is confirmed.

• The Journal of Information Technology
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• v.9 no.1
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• pp.9-19
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• 2006

Medications or conitive-behavior methods have been mainly used as a treatment of alcoholism. lately the virtualy technology has been applied to the kink of alcoholic disorders. A virtual environment makes him having avility to over come the drink. In this study, we were implemented by making panorama images and 3D object modules using 3D MAX, VRML, JAVA. And the BAR stimulator that composed with a position sensor, head mount display, and audio system, is suggested. To illustrate the physiological difference between a person who has a alcoholism and without a liquor bottle, heart rate was measured during experiment, and also measured a person's HR after the virtual reality training. The system measures the Physiological signals such as ECG, Temperature, analyzes those data automatically. The system has two parts, one is physiological data acquisition part which gets the body signal, and the other one is mobile nuit which includes signal processing and transmission functions, And Bluetooth allows two parts to communicate with each other. we demonstrated the subjective effectiveness of virtual reality psychotherapy through the clinical experiment.

• The Journal of Korea Robotics Society
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• v.17 no.3
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• pp.273-281
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• 2022

In this paper, we propose a robot sound recognition system that detects various sound events. The proposed system is designed to detect various sound events in real-time by using a microphone on a robot. To get real-time performance, we use a VGG11 model which includes several convolutional neural networks with real-time normalization scheme. The VGG11 model is trained on augmented DB through 24 kinds of various environments (12 reverberation times and 2 signal to noise ratios). Additionally, based on random forest algorithm, a decision logic is also designed to generate event signals for robot applications. This logic can be used for specific classes of acoustic events with better performance than just using outputs of network model. With some experimental results, the performance of proposed sound recognition system is shown on real-time device for robots.

• The KIPS Transactions:PartB
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• v.17B no.2
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• pp.107-114
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• 2010

A guitar, a piano, and a violin are popular musical instruments for User Created Contents(UCC). However the patterns of audio signal generated by a guitar and a piano are too similar to differentiate. The difference between two musical instruments can be found by analyzing the frequency variation per each band near signal peaks. The distribution of probability on the existence of signal peaks based on Cumulative Histogram were applied to musical instrument recognition. Experiments with statistical models of the frequency variation per each band near signal peaks showed the 14% improvement of musical instrument recognition.

• Journal of the Institute of Convergence Signal Processing
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• v.5 no.1
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• pp.79-83
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• 2004

The purpose of ATM traffic management is to protect the network using minimum resources and guarantee the requred QoS and also it is desirable to provide fairness between input channels. In this paper, we propose the TBBM(threshold based buffer management) algorithm to improve fairness between input channels and utilization of ATM networks. TBBM algorithm controls output cell rate dynamically based on threshold. The result shows that the required bandwidth of the TBBM algorithm is 14.3% lower in audio traffic and 41.8% lower in video traffic than that of theoretically calculated equivalent capacity method. and also reveals that the TBBM algorithm provide improved CLR fairness between input channels

• Proceedings of the Korea Information Processing Society Conference
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• 2014.11a
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• pp.693-694
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• 2014

This short paper briefly describes the proposed music recommendation method that provides suitable music pieces to a listener depending on both listeners' ratings and content of music pieces. The proposed method consists of two methods. First, listeners' ratings prediction method is a combination the traditional user-based and item-based collaborative filtering methods. Second, genre classification method is a combination of feature extraction and classification procedures. The feature extraction step obtains audio signal information and stores it in data structure, while the second one classifies the music pieces into various genres using decision tree algorithm.

• Journal of the Korean Institute of Telematics and Electronics
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• v.23 no.6
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• pp.944-949
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• 1986

By improving the ability of arithmatic processing with an arithmatic processor in a microcomputer and realizing the data input system for real time analysis, an FFT analyzer that is usable within the range of audio frequency is designed and constructed. The input signal passes through a gain programmable pre-amplifier and anti-aliasing lowpass filter into an analogditital converter to be converted into digital form. The converted input data is processed by an Apple II microcomputer. The results of the processing are displayed using a microcomputer display unit and can be copied on a printer or stored in a floppy disk.