• The KIPS Transactions:PartB
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• v.17B no.1
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• pp.69-78
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• 2010

Spectral modeling synthesis (SMS) has been used as a powerful tool for musical sound modeling. This technique considers a sound as a combination of a deterministic plus a stochastic component. The deterministic component is represented by the series of sinusoids that are described by amplitude, frequency, and phase functions and the stochastic component is represented by a series of magnitude spectrum envelopes that functions as a time varying filter excited by white noise. These representations make it possible for a synthesized sound to attain all the perceptual characteristics of the original sound. However, sometimes considerable phase variations occur in the deterministic component by using the conventional SMS for the complex sound such as whale sounds when the partial frequencies in successive frames differ. This is because it utilizes the calculated phase to synthesize deterministic component of the sound. As a result, it does not provide a good spectrum matching between original and synthesized spectrum in higher frequency region. To overcome this problem, we propose a modified SMS that provides good spectrum matching of original and synthesized sound by calculating complex residual spectrum in frequency domain and utilizing original phase information to synthesize the deterministic component of the sound. Analysis and simulation results for synthesizing whale sounds suggest that the proposed method is comparable to the conventional SMS in both time and frequency domain. However, the proposed method outperforms the SMS in better spectrum matching.

• The Journal of the Acoustical Society of Korea
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• v.30 no.5
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• pp.249-254
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• 2011

Inquiring activities on the acoustic phenomena have been carried out by using a sound card installed in a personal computer. A sound card is cheaper and more accessible to the students than the precision equipment such as a function generator or an oscilloscope. The students record the sounds from various acoustic phenomena to the sound card. Then they analyze the frequency spectrums of that sounds by using a program. Inquired phenomena include beat by two tuning forks, sound from Rijke tube, pouring sound, breaking of a wine glass and pop-up sound of a wine bottle. Through these activities students perform quantitative analysis of various phenomena due to superposition, resonance and standing wave.

• Journal of Biomedical Engineering Research
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• v.20 no.6
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• pp.591-599
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• 1999

연구개에 결함이 있는 사람의 발음은 부적절한 비음이 섞이게 되어 과비음성 비음이 되어 연구개를 복원해주는 시술을 하게 되는데, 과비음성 비음을 정량적으로 측정할 수있다면 시술 결과를 객관화 할 수 있게 된다. 현재 임상적으로 사용되고 있는 방법들은 관혈적이거나 고가의 장비를 필요로 한다. 본 논문에서는 비음의 특징인 스펙트럼에서 zero 의 존재와 비강에 의한 포만트의 존재 사실, 그리고 선형 예측 모델을 이용하여 마이크로폰과 사운드 카드가 장착된 PC로 구현할 수 있는 새로운 과비음성 비음 추정 알고리즘을 제안하였다. 음성 신호의 스펙트럼에 zero가 존재하는 경우, 낮은 차수(order)의 선형 예측 모델이 그 음성을 발음한 성도 시스템에 정확히 적용되지 않는다는 점을 이용하여, 같은 음성에 대한 높은 차수의 선형 예측 모델과의 차이를 이용해서 과비음성의 정량화를 시도했다. 본 논문에서는 제안된 알고리즘은 기존의 Teager Operator를 이용한 알고리즘에 비해서 Nasonmeter 의 측정결과와 더 높은 통계적 상관관계를 보여주었다.

• The Journal of the Acoustical Society of Korea
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• v.28 no.3
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• pp.229-238
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• 2009

This paper describes a sound engine of Korean traditional instruments, which are the Gayageum and Taepyeongso, by using a TMS320F2812. The Gayageum and Taepyeongso models based on commuted waveguide synthesis (CWS) are required to synthesize each sound. There is an instrument selection button to choose one of instruments in the proposed sound engine, and thus a corresponding sound is produced by the relative model at every certain time. Every synthesized sound sample is transmitted to a DAC (TLV5638) using SPI communication, and it is played through a speaker via an audio interface. The length of the delay line determines a fundamental frequency of a desired sound. In order to determine the length of the delay line, it is needed that the time for synthesizing a sound sample should be checked by using a GPIO. It takes $28.6{\mu}s$ for the Gayageum and $21{\mu}s$ for the Taepyeongso, respectively. It happens that each sound sample is synthesized and transferred to the DAC in an interrupt service routine (ISR) of the proposed sound engine. A timer of the TMS320F2812 has four events for generating interrupts. In this paper, the interrupt is happened by using the period matching event of it, and the ISR is called whenever the interrupt happens, $60{\mu}s$. Compared to original sounds with their spectra, the results are good enough to represent timbres of instruments except 'Mu, Hwang, Tae, Joong' of the Taepyeongso. Moreover, only one sound is produced when playing the Taepyeongso and it takes $21{\mu}s$ for the real-time playing. In the case of the Gayageum, players usually use their two fingers (thumb and middle finger or thumb and index finger), so it takes $57.2{\mu}s$ for the real-time playing.

• Journal of the Korea Society of Computer and Information
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• v.24 no.11
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• pp.153-161
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• 2019

In this paper, we proposed a method to measure the feed intake of cattle using the cattle's gulp downing sounds. To measure the sound of cattle's gulp downing, the recording is performed through a wearable device attached to the cattle's neck. A lot of noises are recorded according to the ranching environment. This paper proposed a method for spectralizing raw gulping sound data containing white noise and removing white noise through the signal transformation using a filter. This allows the feed intake to be measured. Through the proposed white noise reduction method, it was possible to extract only the cattle's gulp downing sound, and through this, the number of cattle's gulp downing could be measured. The proposed method in this paper makes it possible to measure cattle's feed intake easily, so that estrus prediction, health care for cattle, and feed management can be done efficiently.

• Journal of the Institute of Electronics and Information Engineers
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• v.53 no.11
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• pp.82-88
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• 2016

In this paper, we propose an intelligibility enhancement algorithm for multimedia contents using spectral shaping. The dialogue signals is essential to understand the plot of audio-visual media contents such as movie and TV. However, the non-dialogue components as like sound effects and background music often degrade the dialogue clarity. To overcome this problem, this paper tries to improves the dialogue clarity of audio soundtracks which contain important cues for the visual scenes. In the proposed method, the dialogue components are first detected by soft masker based on speech presence probability (SPP) which is widely used in speech enhancement field. Then, extracted dialogue signals are applied to the spectral shaping method. It reallocate the spectral-temporal energy of speech to enhanced the intelligibility. The total energy is maintained as unchanged via a loudness normalization process to prevent saturation. The algorithm was evaluated using the modeled and real movie soundtracks and it was shown that the proposed algorithm enhances the dialogue clarity while preserving the total audio power.

• Journal of Korea Society of Industrial Information Systems
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• v.23 no.2
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• pp.11-18
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• 2018

Severe snoring can lead to OSA(Obstructive Sleep Apnea), which can lead to life-threatening cases, and snoring can lead to serious pernicious relationships. In order to solve these snoring problems, several types of smart pillows have recently been released. The core technology is snoring discrimination technology, ie, a technique for determining whether snoring is included in the input sound. In this paper, we propose a snoring detection method to apply to a smart pillow. After extracting the features of the snoring sound from the input signal, we discriminate the snoring using these features and SVM. In order to measure the performance of the proposed method, comparative experiments with the existing methods are performed. The experimental results show about 6% better discrimination performance than the existing method.

• Proceedings of the Korean Society of Computer Information Conference
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• 2018.07a
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• pp.439-440
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• 2018

본 논문에서는 제작한 가상현실 게임의 소개와 적용기술에 대해여 논한다. HTC VIVE를 이용하여 강력한 주요 게임 개발엔진 중 하나인 Unreal engine4를 이용하여 쉽게 접근 할 수 있는 VR 리듬게임 구현을 목적으로 한다. 플랫폼의 한계의 벗어나 HUD로 좀 더 현실감과 몰입을 요구하는 게임개발을 목표로 리듬감 증진 혹은 운동효과도 기대할 수 있다. 사운드 플러그를 이용하여 주파수별 스펙트럼을 시각화하여 재생되는 음원중 스펙트럼 값이 조건을 만족하면, 노트가 자동으로 생성되고 일정 시간이 경과한 후 사라지거나, 플레이어가 타격하여 점수를 획득 하는 방식으로 진행되며, 플레이어가 노트를 맞을 경우 체력값이 단계별로 낮아져 게임이 종료된다. 더 많은 곡과 맵을 추가하여 흥미소요를 늘릴 수 있으며, 단순타격이 아닌 조건이나 임무를 부여 다양성과 복잡함을 추가해 낮은 접근성과 높은 정복성을 가질 수 있다.

• The Journal of the Acoustical Society of Korea
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• v.28 no.5
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• pp.441-446
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• 2009

This paper proposes an automatic indexing algorithm of golf video using audio information. In the proposed algorithm, the input audio stream is demultiplexed into the stream of video and audio. By means of Adaboost-cascade classifier, the continuous audio stream is classified into announcer's speech segment recorded in studio, music segment accompanied with players' names on TV screen, reaction segment of audience according to the play, reporter's speech segment with field background, filed noise segment like wind or waves. And golf swing sound including drive shot, iron shot, and putting shot is detected by the method of impulse onset detection and modulation spectrum verification. The detected swing and applause are used effectively to index action or highlight unit. Compared with video based semantic analysis, main advantage of the proposed system is its small computation requirement so that it facilitates to apply the technology to embedded consumer electronic devices for fast browsing.

• The Journal of the Acoustical Society of Korea
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• v.30 no.5
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• pp.281-294
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• 2011

Physics-based sound synthesis usually requires high computational costs and this results in a restriction of its use in real-time applications. This motivates us to implement the sound synthesis algorithm of plucked-string instruments using multi-core processor architectures and determine the optimal processing element (PE) configuration for the target instruments. To determine the optimal PE configuration, we evaluate the impacts of a sample-per-processing element (SPE) ratio that is defined as the amount of sample data directly mapped to each PE on system performance and both area and energy efficiencies using architectural and workload simulations. For the acoustic guitar, the highest area and energy efficiencies are achieved at a SPE ratio of 5,513 and 2,756, respectively, for the synthesis of musical sounds sampled at 44.1 kHz. In the case of the classical guitar, the maximum area and energy efficiencies are achieved at a SPE ratio of 22,050 and 5,513, respectively. In addition, the synthetic sounds were very similar to original sounds in their spectra. Furthermore, we conducted MUSHRA subjective listening test with ten subjects including nine graduate students and one professor from the University of Ulsan, and the evaluation of the synthetic sounds was excellent.