• 한국소음진동공학회:학술대회논문집
• /
• 한국소음진동공학회 2006년도 추계학술대회논문집
• /
• pp.757-762
• /
• 2006

The method of promoing sound surroundings with the new concept named 'Soundscape' attracts the public attention. This new method is the way of active promoting acoustic surroundings to offer the vitality, comfort and identity of urban space and relative quietness by masking the noise with natural and signal sounds. Therefore, this research has the purpose of finding out the method of soundscape concept of urban public space for the comfort acoustic surroundings.

• 음성과학
• /
• 제4권2호
• /
• pp.5-15
• /
• 1998

This paper proposes a speech state-dependent spectral subtraction method to regulate the blind spectral subtraction for improved enhancement. In the proposed method, a modified subtraction rule is applied over the speech selectively contingent to the speech state being voiced or unvoiced, in an effort to incorporate the acoustic characteristics of phonemes. In particular, the objective of the proposed method is to remedy the subtraction induced signal distortion attained by two state-dependent procedures, spectrum sharpening and minimum spectral bound. In order to remove the residual noise, the proposed method employs a procedure utilizing the masking effect. Proposed spectral subtraction including state-dependent subtraction and residual noise reduction using the masking threshold shows effectiveness in compensation of spectral distortion in the unvoiced region and residual noise reduction.

• 한국소음진동공학회:학술대회논문집
• /
• 한국소음진동공학회 2007년도 추계학술대회논문집
• /
• pp.1037-1041
• /
• 2007

With the advent of sound field simulator, many sound fields have been reproduced by obtaining the impulse responses of specific acoustic spaces like famous concert hall, opera house. This sound field reproduction has been done by the linear convolution operation between the sound input signal and the impulse response of certain acoustic space. However, the conventional finite impulse response based linear convolution operation always makes real-time implementation of sound field generator impossible due to the large amount of computational burden. This paper introduces the fast convolution method using perceptual redundancy in the processed signals, input audio signal and room impulse response. Temporal and spectral psycho-acoustic filters considering masking effects are implemented in the proposed convolution structure. It reduces the computational burden of convolution methods for realtime implementation of a sound field generator. The conventional convolutions are compared with the proposed one in views of computational burden and sound quality. In the proposed method, a considerable reduction in the computational burden was realized with acceptable changes in sound quality.

• The Journal of the Acoustical Society of Korea
• /
• 제13권2E호
• /
• pp.32-40
• /
• 1994

본 논문은 디지털 망에 접속된 ISDN 전화기의 전송품질인 음량정격과 송화자 에코를 설정하는 방법을 기술한다. ISDN 전화기의 바람직한 음량정격 및 송화자 에코를 설계하기 위하여 주관평가를 위한 디지털 음성통신 모델 시스템을 개발하였고, 이 모델 시스템을 이용하여 최적의 코덱 입력레벨, 전체 음량정격의 범위, 그리고 송화자 에코 등을 결정하기 위하여 오피니언 테스트를 수행하였다. 실험결과 송화 음량정격은 6~8dB, 수화음량정격은 0~2dB, 측음마스킹정격은 8~12dB로 설정되었다. 또한, 에코프리의 전화통화를 위한 단말결합손실은 전체음량정격이 10dB 일때 적어도 40dB 이상이어야 한다는 결론을 얻었다.

• 대한후두음성언어의학회지
• /
• 제10권2호
• /
• pp.124-129
• /
• 1999

1) 음성 장애를 호소하지 않는 성악인 50명중 여자는 59% 남자는 50%에서 후두스트로보스코피상 성대 병변이 발견되었다. 2) 성종별로는 소프라노에서 가장 성대 병변의 발생률이 높았다(66.7 %). 3) 최장발성 지속시간은 성악인의 경우 성대 병변에 관계없이 일반 정상인에 비해서 증가되어 있었으며 특히 정상 성대를 가진 여자 성악인에서는 일반인에 비해 통계적으로 의의 있게 증가되어 있었다. 4) 기본진동수, 기본진동수 표준편차, 기본진동수변 이들은 성악인이 일반인에 비해 안정되어 있었고, jitter. PPQ는 성악인이 성대 병변이 있더라도 일반인에 비해 감소되어 있었으며, 특히 남자 성악인에서는 통계적으로 유의하게 감소되어 있었고, Shimmer. APQ 역시 성악인이 성대 병변이 있더라도 일반인에 비해 통계적으로 의의 있게 감소되어 있었다.

• 한국소음진동공학회:학술대회논문집
• /
• 한국소음진동공학회 2005년도 춘계학술대회논문집
• /
• pp.603-606
• /
• 2005

This study is to make good acoustic environment around Gwang-ju riverside and performed survey and measurement of soundscape. The survey was focused on the types of sound producing. The results of measurement and analysis are as follows: 1) The types of sound at reverside were twenty and the Leq(equivalent noise level) was ranging from 50 dB(A) to 76 dB(A). The dominant sound was road traffic noise. 2) In the upperstream and the downstream, the sound of wind, bird and water was observed. 3) In the inner city, the water sound was produced by the difference of the height of the weir to mask the road traffic noise.

• 한국수산과학회지
• /
• 제35권6호
• /
• pp.563-567
• /
• 2002

Developing base data on luring fish schools into netting position by the use of underwater audible sound on japanese parrot fish Oplegnathus fasciatus found in the coastal waters of Jeju Island, S. Korea. Auditory threshold was determined by the heartbeat condition technique using pure tones coupled with a delayed electric shock. The audible range of japanese parrot fish extended from 80 Hz to 500 Hz with a peak sensitivity at 200 Hz. The mean auditory thresholds at the frequencies of 80 Hz, 100 Hz, 200 Hz,300 Hz and 500 Hz were 104 dB, 95 dB, 91 dB, 99 dB and 113 dB, respectively. As the frequency became higher than 200 Hz, the auditory threshold increased almost linearly with increasing frequency. Critical ratios of fishes measured in the presence of masking noise in the spectrum level range of 69$\~$78 dB (0 dB re 1$\mu$Pa/$\sqrt{Hz}$) ranged from 21 dB to 40 dB at test frequencies. The noise spectrum level at the start of masking was about 70 dB within the test frequency range. The sound pressure level of 100$\~$200 Hz recognized by japanese parrot fish under the ambient noise is above 91 dB and the critical ratio for them is above 21 dB.

• International Journal of Computer Science & Network Security
• /
• 제22권9호
• /
• pp.175-182
• /
• 2022

In our daily life, we come across different types of information, for example in the format of multimedia and text. We all need different types of information for our common routines as watching/reading the news, listening to the radio, and watching different types of videos. However, sometimes we could run into problems when a certain type of information is required. For example, someone is listening to the radio and wants to listen to jazz, and unfortunately, all the radio channels play pop music mixed with advertisements. The listener gets stuck with pop music and gives up searching for jazz. So, the above example can be solved with an automatic audio classification system. Deep Learning (DL) models could make human life easy by using audio classifications, but it is expensive and difficult to deploy such models at edge devices like nano BLE sense raspberry pi, because these models require huge computational power like graphics processing unit (G.P.U), to solve the problem, we proposed DL model. In our proposed work, we had gone for a low complexity model for Audio Event Detection (AED), we extracted Mel-spectrograms of dimension 128×431×1 from audio signals and applied normalization. A total of 3 data augmentation methods were applied as follows: frequency masking, time masking, and mixup. In addition, we designed Convolutional Neural Network (CNN) with spatial dropout, batch normalization, and separable 2D inspired by VGGnet [1]. In addition, we reduced the model size by using model quantization of float16 to the trained model. Experiments were conducted on the updated dataset provided by the Detection and Classification of Acoustic Events and Scenes (DCASE) 2020 challenge. We confirm that our model achieved a val_loss of 0.33 and an accuracy of 90.34% within the 132.50KB model size.

• 한국음향학회지
• /
• 제30권5호
• /
• pp.255-264
• /
• 2011

3차원 공간에서 음원으로부터 사람의 귀로 음향적인 전달 과정을 표현하는 머리전달함수는 사람이 음원의 위치를 판단할 수 있는 중요한 정보를 포함하고 있으며, 이를 이용하여 실질적으로는 존재하지 않는 음원을 근사적으로 생성할 수 있다. 그러나 각 청자에 특화되지 않는 머리전달함수를 사용함으로 인해 전후 방향이나 상하 방향의 혼동이 발생하게 되어 음상정위 성능이 저하된다. 이 논문에서는 머리전달함수를 통해 생성된 입체음향을 인간의 청각 특성을 사용하여 개선하는 알고리즘을 제안한다. 머리전달함수가 적용된 사운드 신호의 전역 마스킹 값을 계산한 후, 청자에게 크게 영향을 미치는 주파수 대역만을 추출하여 이를 강조함으로서 각 방향에 해당하는 인지 특성을 부각시키는 방법을 제안하였으며, 제안된 방법의 성능을 청감테스트를 통해 증명하였다.

• 한국소음진동공학회:학술대회논문집
• /
• 한국소음진동공학회 2005년도 춘계학술대회논문집
• /
• pp.611-614
• /
• 2005

The introduction of the sound based on the soundscape concept has the effect to offer comfortable sound environments at the public spaces by masking undesired ones and to identify the spaces. Sound installation, sound sculptor and the soundscape are used for these purpose, but the most important factors to be considered therein are to determine what kind of sounds to offer and how to adjust them to the changing circumstances. But, installing, maintaining and adjusting the soundscape system directly in the field will ensue numerous problems as well as high costs. And, even if it was epochal and novel when the soundscape is first installed at a space, new different sound environment is necessary to continue the effectiveness as time goes. Thus, this study aims at devising the instrument system that has the artificial intelligence, enables to remote control, with a great ease, numerous variables, reproduce most agreeable sound sources, and can produce the proper sound fit to the space automatically and spontaneously.