• Journal of the Institute of Convergence Signal Processing
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• v.20 no.4
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• pp.245-251
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• 2019

Sonar bearing accuracy is the correspondence between the target orientation predicted by sonar and actual target orientation, and is obtained from measurements. However, when measuring sonar bearing accuracy, many errors are included in the results because they are made at sea, where complex and diverse environmental factors are applied. In particular, parallax error caused by the difference between the position of the GPS receiver and the sonar sensor, and the time delay error generated between the speed of underwater sound waves and the speed of electromagnetic waves in the air have a great influence on the accuracy. Correcting these parallax errors and time delay errors without an automated tool is a laborious task. Therefore, in this study, we propose a sonar bearing accuracy measurement equipment with parallax error and time delay error correction. The tests were carried out through simulation data and real data. As a result of the test it was confirmed that the parallax error and time delay error were systematically corrected so that 51.7% for simulation data and more than 18.5% for real data. The proposed method is expected to improve the efficiency and accuracy of sonar system detection performance verification in the future.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.8B
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• pp.1460-1468
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• 2000

The goal of this research is the real-time implementation of MPEG-1 Layer III audio decoder using the fixed-point digital signal processor of TMS320C6201 The main job for this work is twofold: one is to convert floating-point operation in the decoder into fixed-point operation while maintaining the high resolution, and the other is to optimize the program to make it run in real-time with memory size as small as possible. We, especially, devote much time to the descaling module in the decoder for conversion of floating-point operation into fixed-point operation with high accuracy. The inverse modified cosine transform(IMDCT) and synthesis polyphase filter bank modules are optimized in order to reduce the amount of computation and memory size. After the optimization process, in this paper, the implemented decoder uses about 26% of maximum computation capacity of TMS320C6201. The program memory, data ROM, data RAM used in the decoder are about 6.77kwords, 3.13 kwords and 9.94 kwords, respectively. Comparing the PCM output of fixed-point computation with that of floating-point computation, we achieve the signal-to-noise ratio of more than 60 dB. A real-time operation is demonstrated on the PC using the sound I/O and host communication functions in the EVM board.

• Journal of the Korean Society for Nondestructive Testing
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• v.20 no.6
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• pp.538-544
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• 2000

A phased array ultrasonic nondestructive inspection system is being developed to obtain images of the interior of steel structures by modifying a medical ultrasound imaging system. The medical system consists of 64 individual transceiver channels that can drive 128 array elements. Several modifications of the system were required mainly due to the change of sound speed. It was necessary to fabricate array transducers for steel structure and to obtain A-scan signal that is necessary for the nondestructive testing. Boundary diffraction wave model was used for the prediction of radiation beam field from array transducers, which provided guidelines to design array transducers. And a RF data acquisition board was fabricated for the A-scan signal acquisition along a selected un line within an image. For the proper beam forming in the transmission and reception for steel structure, delay time was controlled. To demonstrate the performance of the developed system and fabricated transducers, images of artificial specimens and A-scan signals for selected scan lines were obtained in a real time fashion.

• Journal of Intelligence and Information Systems
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• v.29 no.1
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• pp.233-247
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• 2023

Recently, research on detecting objects in hidden spaces beyond the direct line-of-sight of observers has received attention. Most studies use optical equipment that utilizes the directional of light, but sound that has both diffraction and directional is also suitable for non-line-of-sight(NLOS) research. In this paper, we propose a novel method of detecting objects in non-line-of-sight (NLOS) areas using acoustic signals in the audible frequency range. We developed a deep learning model that extracts information from the NLOS area by inputting only acoustic signals and predicts the properties and location of hidden objects. Additionally, for the training and evaluation of the deep learning model, we collected data by varying the signal transmission and reception location for a total of 11 objects. We show that the deep learning model demonstrates outstanding performance in detecting objects in the NLOS area using acoustic signals. We observed that the performance decreases as the distance between the signal collection location and the reflecting wall, and the performance improves through the combination of signals collected from multiple locations. Finally, we propose the optimal conditions for detecting objects in the NLOS area using acoustic signals.

• Journal of the Institute of Convergence Signal Processing
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• v.23 no.3
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• pp.115-122
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• 2022

With the outbreak of COVID-19 a few years ago, video conferencing and electronic document work have increased, and for this reason, the proportion of computer work among modern people's daily routines is increasing. However, as more and more people work on computers in the wrong posture for a long time, the number of patients with poor eyesight and text neck is increasing. Until recently, many studies have been published to correct posture, but most of them have limitations that users may experience discomfort because they have to correct posture by wearing equipment. A posture correction sensor algorithm is proposed to prevent access to the minimum distance between a computer monitor and a person using an ultrasonic sensor device. At this time, an algorithm for minimizing false alarms among warning alarms that sound at the minimum distance is also proposed. Because the ultrasonic sensor device is used, posture correction can be performed without attaching a device to the body, and the user can relieve discomfort. In addition, experimental results showed that accuracy can be improved by reducing false alarms by removing more than half of the noise generated during distance measurement.

• The Journal of the Acoustical Society of Korea
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• v.28 no.2
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• pp.119-126
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• 2009

This paper describes the non-stringed guitar composed of laser strings, frets, sound synthesis algorithm and a processor. The laser strings that can depict stroke and playing arpeggios comprise laser modules and photo diodes. Frets are implemented by voltage divider. The guitar body does not need to implement physically because commuted waveguide synthesis is used. The proposed frets enable; players to represent all of chords by the chord glove as well as guitar solo. Sliding, hammering-on and pulling-off sounds are synthesized by using parameters from the voltage divider. Because the pitch shifting corresponds to the time-varying propagation speed in the digital waveguide model, the proposed model can synthesize vibrato as well. After transformation of signals from the laser strings and frets into parameters for synthesis algorithm, the digital signal processor, TMS320F2812, performs the real-time synthesis algorithm and communicates with the DAC. The demonstration movieclip available via the Internet shows one to play a song, 'Arirang', synthesized by proposed algorithm and interfaces in real-time. Consequently, we can conclude that the proposed synthesis algorithm is efficient in guitar solo and there is no problem to play the non-stringed guitar in real-time.

• The Journal of the Acoustical Society of Korea
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• v.43 no.2
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• pp.225-233
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• 2024

Active sonar transmits sound waves to detect covertly maneuvering underwater objects and detects the signals reflected back from the target. However, in addition to the target's echo, the active sonar's received signal is mixed with seafloor, sea surface reverberation, biological noise, and other noise, making target recognition difficult. Conventional techniques for detecting signals above a threshold not only cause false detections or miss targets depending on the set threshold, but also have the problem of having to set an appropriate threshold for various underwater environments. To overcome this, research has been conducted on automatic calculation of threshold values through techniques such as Constant False Alarm Rate (CFAR) and application of advanced tracking filters and association techniques, but there are limitations in environments where a significant number of detections occur. As deep learning technology has recently developed, efforts have been made to apply it in the field of underwater target detection, but it is very difficult to acquire active sonar data for discriminator learning, so not only is the data rare, but there are only a very small number of targets and a relatively large number of non-targets. There are difficulties due to the imbalance of data. In this paper, the image of the energy distribution of the detection signal is used, and a classifier is learned in a way that takes into account the imbalance of the data to distinguish between targets and non-targets and added to the existing technique. Through the proposed technique, target misclassification was minimized and non-targets were eliminated, making target recognition easier for active sonar operators. And the effectiveness of the proposed technique was verified through sea experiment data obtained in the East Sea.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.14 no.7
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• pp.3218-3222
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• 2013

Pansori is our traditional vocal sound, originality and excellence in the art of conversation, gesture general became a globally recognized world intangible heritage. Especially, Pansori as shrews and humorous representation of audience participation with a high degree of artistic value and enjoy the arts throughout all layers to be responsible for the social integration of functions is evaluated. Therefore, in this paper, Pansori five yard target speech signal analysis techniques applied to analyze the Pansori acoustic features of a representation of a society and era correlation extraction studies were performed. Pansori on the five yard spectrogram, pitch, stability and strength analysis for this experiment. Pansori through experimental results Comical story while keeping the audience focused and interested to better reflect the characteristics of energy for the wave of voice and vocal cord tremor change the width of a large, stable and voice with a loud voice, that expresses were analyzed.

• KIPS Transactions on Computer and Communication Systems
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• v.4 no.9
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• pp.321-326
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• 2015

In this paper, we proposed an algorithm which could transmit reliable data between smart devices by using inaudible high frequency of audible frequency range and cyclic redundancy check method. The proposed method uses 18 kHz~22 kHz as high frequency which inner speaker of smart device can make a sound in audible frequency range (20 Hz~22 kHz). To increase transmission quantity of data, we send mixed various frequencies at high frequency range 1 (18.0 kHz~21.2 kHz). At the same time, to increase accuracy of transmission data, we send some mixed frequencies at high frequency range 2 (21.2 kHz~22.0 kHz) as checksum. We did experiments about data transmission between smart devices by using the proposed method to confirm data transmission speed and accuracy of the proposed method. From the experiments, we showed that the proposed method could transmit 32 bits data in 235 ms, the transmission success rate was 99.47%, and error detection by using cyclic redundancy check was 0.53%. Therefore, the proposed method will be a useful for wireless transmission technology between smart devices.

• The Journal of the Acoustical Society of Korea
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• v.14 no.4
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• pp.89-95
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• 1995

The waveform coding technique has concerned with simply preserving the waveform shape of speech signal through a redundancy reduction process. In the case of speech synthesis, the waveform coding with high sound quality is mainly used to the synthesis by analysis. However, since the parameters of this coding are not classified into either excitation or vocal tract parameters, it is difficult to applying the waveform coding to the synthesis by rule. In order to apply the waveform coding to the synthesis by rule, the pitch alteration technique is required in prosody control. In this paper, we propose a new pitch alteration method that can change the pitch period in waveform coding by scaling the time-axis and compensating the spectrum. This is relevant to the time-frequency domain method were the phase components of the waveform is preserved with a little spectrum distortion of 2.5 % and less for 50% pitch change.