• ETRI Journal
• /
• 제22권2호
• /
• pp.11-19
• /
• 2000

If the right and left signals of a binaural sound recording are reproduced through loudspeakers instead of a headphone, they are inevitably mixed during their transmission to the ears of the listener. This degrades the desired realism in the sound reproduction system, which is commonly called 'cross-talk.' A 'cross-talk canceler' that filters binaural signals before they are sent to the sound sources is needed to prevent cross-talk. A cross-talk canceler equalizes the resulting sound around the listener's ears as if the original binaural signal sound is reproduced next to the ears of listener. A cross-talk canceler is also a solution to the problem-how binaural sound is distributed to more than 2 channels that drive sound sources. This paper presents an effective way of building a cross-talk canceler in which geometric information, including locations of the listener and multiple loudspeakers, is divided into angular information and distance information. The presented method makes a database in an off-line way using an adaptive filtering technique and Head Related Transfer Functions. Though the database is mainly concerned about the situation where loudspeakers are located on a standard radius from the listener, it can be used for general radius cases after a distance compensation process, which requires a small amount of computation. Issues related to inverting a system to build a cross-talk canceler are discussed and numerical results explaining the preferred configuration of a sound reproduction system for stereo loudspeakers are presented.

• 전자공학회논문지B
• /
• 제31B권3호
• /
• pp.24-38
• /
• 1994

In this paper we develop the software simulator written in a C language for a frequency modulation synthesis and the approximate range of parameters, for a musically satisfactory timbre, obtained by using the software simulator will be applied to develop an algorithm for parameter extraction. For a frequency modulation synthesis, we also develop an algorithm for parameter extraction through waveform analysis in the time domain as well as spectrum analysis using a FFT in the frequency domain. To verify the validity of the developed algorithm as well as software simulator experimentally, we extract parameters for the several music instruments using the suggested algorithm and analyze the synthesized sound by applying the parameters to the software simulator. The evaluation of the synthesized sound is first done by listening the sound directly as a subjective testing. Secondly, to evaluate the synthesized sound objectively with an engineering sense, we compare the synthesized sound with an original one in a time domain and a frequency domain.

• 전자공학회논문지B
• /
• 제32B권7호
• /
• pp.1025-1035
• /
• 1995

In this paper, we realize a real time digital FM synthesizer based on genetic algorithm using a general purpose digital signal processor. Especially, we synthesize diverse music sounds nicely using a synthesis model consisting of a single modulator and multiple carriers. Also we present genetic algorithm-based technique which determines optimal parameters for reconstruction through FM synthesis of a sound after analyzing the spectrum of PCM data as a standard music sound using FFT. Using the suggested parameter extractiuon algorithm, we extract parameters of several instruments and then synthesize digital FM sounds. To verify the validity of the parameter extraction algorithm as well as realization of a real time digital music synthesizer, the evaluation is first done by listening the sound directly as subjective test. Secondly, to evaluate the synthesized sound objectively with an engineering sense, we compare the synthesized sound with an original one in a time domain and a frequency domain.

• 한국광학회지
• /
• 제32권3호
• /
• pp.108-113
• /
• 2021

본 논문에서는 어떤 음원에 의하여 진동하는 물체로부터 그 음원의 소리를 레이저 도플러 간섭계를 이용하여 원거리에서 복원하는 방법을 고안하고 실험적으로 시연하였다. 어떤 음파에 의하여 진동하는 물체를 간섭계를 통하여 측정할 경우, 측정되는 간섭계의 주파수는 도플러 효과에 의하여 그 소리의 주파수와 동일하게 변한다. 이 현상을 이용하여 어떤 소리에 영향을 받는 대상의 진동 주파수를 레이저 도플러 간섭계를 통해 원거리에서 실시간으로 측정하고, 간섭계 출력의 최대 주파수를 추적하는 신호처리를 통하여 얻은 결과가 음원의 소리와 같은 주파수 특성을 갖는다는 것을 실험적으로 확인하였다. 또한, 각각의 단일 톤 음원뿐만 아니라 여러 가지 주파수가 혼합된 음원의 복원도 가능함을 확인하였다.

• 한국소음진동공학회:학술대회논문집
• /
• 한국소음진동공학회 2005년도 추계학술대회논문집
• /
• pp.200-204
• /
• 2005

The use of beamforming method and de-Dopplerization technique was applied in studying the rotating sound sources. Acoustic analysis of a moving sound source required that the measured sound signals be do-Dopplerized and restored as of the original emission signals. Two main issues of the signal reconstruction in time domain are addressed herein: First, to remove Doppler effect from the measured data and to restore the original emission data of the moving source. The difference of the time domain beamforming from the frequency domain beamforming was mentioned. Also, the time domain beamforming method is deployed in the test and the comparisons were made to the frequency domain results. The time domain signal reconstruction was numerically simulated prior to the application. To validate the de-Dopplerization Performance, the rotating Point sources were examined and localized by the use of a phased array of microphone. The application of prop-rotor was conducted in a hovering condition. The results of reconstructing time signals of rotating sources and its locations were shown in the power distribution maps. In the prop-rotor measurements, the acoustic source locations were successfully verified in varying positions for different frequencies of interest.

• 인문언어
• /
• 제6권
• /
• pp.7-31
• /
• 2004

This paper argues that (1) word-for-word literary translation is not possible; all it does is achieve what Chukovsky characterized as 'imprecise precision' (1984:47), (2) contra to Nida (1969) and others, translation does not just mean translating meaning, and (3) therefore, a translator must negotiate an uneasy but inevitable compromise between accuracy and elegance. To make the translated passage just as pleasing, moving, and cathartic as the original passage as much as possible, a great deal of literary skill is required on the part of the translator. The iniquity of translators is not so much infidelity as infertility to produce an offspring worthy of an heir to the original writer. Translators are not traitors; they are traders, or literary merchants, who trade one form of linguistic unit for another, often meaning for form, or sense for sound, but sometimes form for meaning. A translator then is not a man of treason but is a tradesman.

• 한국항해학회지
• /
• 제16권4호
• /
• pp.55-63
• /
• 1992

This computer simulation is the basic research for realize a real-time hardware of the reproduction system in original sound field with two loudspeakers based on the OSS(Ortho Stereophonic System) method which was proposed by Hamada of Japan in 1983. Through the computer simulation, presumed the system function of OSS equalizer using HRTF(Head Related Transfer Function), constructed the model of OSS equalizer and , evaluated the modelling OSS equalizer by evaluation formula. The obtained results are summarized as follows : 1) By the modelling OSS equalize operate as inverse filter of HRTF, an input signal reproduced effectively. 2) Known that the real-time hardware of OSS equalizer can be made by the fast convolution between the impulse response of OSS equalizer and input speech signal. 3) Since the system function of OSS equalizer presumed from HRTF, the study on the measuring of HRTF have to proceed.

• 한국음향학회지
• /
• 제28권7호
• /
• pp.653-660
• /
• 2009

본 논문에서는 오디오 신호의 저작권 보호와 효과적인 음질 개선을 위한 새로운 워터마킹 알고리듬을 제안한다. 제안한 방법은 원 신호에 푸리에 변환을 하여 주파수 영역으로 변환하고 n개의 서브밴드로 균등 분할한다. 각 밴드별 에너지를 계산하여 에너지가 큰 것부터 k개를 선택하고 해당 밴드에서 p개의 주요 피크 성분을 검출하여 길이 m의 워터마크를 삽입한다. 워터마크된 오디오 신호를 청자에게 들려주었을 때 워터마크 삽입으로 인한 오디오 신호의 왜곡을 느끼지 못하였다. 또한, 제안한 방법은 Cox 방법만큼 MP3 압축, 잘라내기 (cropping),주파수 변환 (FFT), 반향 (echo)과 같은 워터마크 공격에 강인하였고 신호 대 잡음비 측면에서는 10 dB이상 우수함을 실험을 통해 확인할 수 있었다.

• 한국전자통신학회논문지
• /
• 제10권1호
• /
• pp.1-6
• /
• 2015

최근 원음 반주기에 대한 관심이 증가됨에 따라 고가의 스튜디오 직접 녹음 방법 대신 보다 저렴한 방법을 시도하고 있다. 그 구체적인 방법으로는 가수의 음악 앨범에서 가수의 목소리만 제거하여 원음 반주 음원을 만드는 것이다. 본 논문에서는 스테레오로 녹음된 반주음악에서 보컬을 분리하는 시스템을 제안한다. 제안하는 시스템은 두 단계로 구성된다. 첫 단계는 보컬을 검출하는 단계이다. 이 단계에서는 MFCC를 가지고 SVM 방법을 이용하여 입력 신호를 보컬 부분과 비보컬 부분으로 분리한다. 두 번째 단계에서는 보컬 부분에 대해 각 주파수 빈별로 선택적 주파수 차감을 수행한다. 제안하는 방법으로 보컬을 제거한 음악에 대한 청취실험에서 상대적으로 높은 만족도를 보여준다.

• 말소리와 음성과학
• /
• 제8권4호
• /
• pp.131-138
• /
• 2016

This study analyzes the characteristics of good impression using speech manipulation scripts and investigates the characteristics of preferred speech voice. Fourty male and female college students participated in this study. They have been exposed to the Gyeongsang dialect spoken by their friends and family for more than 15 years. Two sample voices(1 male and 1 female), considered as giving good impression, were subject to voice analysis. Two students were asked to read the sample paragraph of 'Walking' and their voice samples were analyzed through Praat. The collected speech data were manipulated into 4 different sets by changing pitch level, degree of loudness and speech rate. First, both men and women received good impression more from pitch-lowered sound than from the original one. Second, men tended to receive good impression more from slightly louder voice than from the natural-pitched one. Third, it was shown that men often felt more drowned to a voice at slightly faster speech rate than at the original speech rate. Overall, both male and female listeners favored lower pitch over the original pitch. Men tended to prefer louder voice sound while women preferred less loud one. Men received better impression at a lower speech rate but women at a faster speech rate.