• 한국위성정보통신학회논문지
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• 제8권4호
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• pp.105-110
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• 2013

방송이 아날로그방식에서 디지털방식으로 전환되면서 방송 음량의 다이나믹 레인지가 확대되나, 방송 음량에 대해 어떤 규제가 없는 상황에서 방송사 간 혹은 프로그램 간의 경쟁 등에 방송 음량이 점차 커지고 있다. 방송 시청 중 채널 간에 음량 변화가 큰 경우, 시청자들의 정서장애 등이 유발될 수도 있으며, 이를 해결하기 위해 ITU에서는 방송 음량 기준을 제정했다. 본 연구에서는 ITU-R에서 제시하는 음량 측정 알고리즘을 적용하여, 국내 주요 방송채널들의 음량을 측정 분석하고 음량을 관리하기 위한 방안을 제시하고자 한다.

• 한국위성정보통신학회논문지
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• 제8권1호
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• pp.31-34
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• 2013

본 논문은 4세대 이동통신인 Long Term Evolution (LTE) 단말기에 의해 발생된 간섭 신호가 음향기기에 유입되었을 때 음향기기의 잡음영향에 대해 검토하였다. 먼저, LTE 송신 신호에 대한 분석 및 측정을 바탕으로 음향기기에 간섭을 주는 LTE 간섭 신호는 송신 전력의 크기에 의해 결정됨을 확인하였다. 또한, LTE 단말기의 송신전력 및 단말기와 음향기기간의 거리를 변화시키면서 발생하는 음향기기의 잡음을 측정하였다. 측정 결과, LTE 단말기가 최대 전력(22 dBm)을 송신하더라도 단말기와 음향기기의 거리를 25 cm 이상 이격시킴으로써 음향기기에서 발생하는 잡음을 방지할 수 있었다.

• 한국소음진동공학회논문집
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• 제23권1호
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• pp.3-8
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• 2013

Recently, the impact on hearing induced by using of portable audio equipment have been actively studied. In general, Because they turn the volume up with loud background noise, they may expose to louder noise. In this study, we investigated the maximum noise level of 20 the domestic potable audio equipment and estimated the impact of the hearing induced by portable audio equipment in according to exposure time. As a result, the use of portable audio equipment is assumed to be more three hours when the level of more than 50 % of volume is most likely to affect the hearing.

• IEIE Transactions on Smart Processing and Computing
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• 제2권1호
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• pp.11-19
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• 2013

A video-zoom driven audio-zoom algorithm is proposed to provide audio zooming effects according to the degree of video-zoom. The proposed algorithm is designed based on a super-directive beamformer operating with a 4-channel microphone array in conjunction with a soft masking process that uses the phase differences between microphones. The audio-zoom processed signal is obtained by multiplying the audio gain derived from the video-zoom level by the masked signal. The proposed algorithm is then implemented on a portable digital imaging device with a clock speed of 600 MHz after different levels of optimization, such as algorithmic level, C-code and memory optimization. As a result, the processing time of the proposed audio-zoom algorithm occupies 14.6% or less of the clock speed of the device. The performance evaluation conducted in a semi-anechoic chamber shows that the signals from the front direction can be amplified by approximately 10 dB compared to the other directions.

• Journal of information and communication convergence engineering
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• 제5권3호
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• pp.245-248
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• 2007

In this paper, we research the content-based analysis and classification according to the composition of the feature parameter data base for the audio data to implement the audio data index and searching system. Audio data is classified to the primitive various auditory types. We described the analysis and feature extraction method for the feature parameters available to the audio data classification. And we compose the feature parameters data base in the index group unit, then compare and analyze the audio data centering the including level around and index criterion into the audio categories. Based on this result, we compose feature vectors of audio data according to the classification categories, and simulate to classify using discrimination function.

• 한국간호교육학회지
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• 제17권1호
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• pp.100-109
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• 2011

Purpose: This study was conducted to investigate how a program using video-audio media will affect the knowledge level and compliance of the sick role of patients admitted for coronary angiography. Method: A non-equivalent control group non-synchronized design was used. Subjects were selected from patients admitted to the internal ward of a hospital in B city for coronary angiography between July 1 and September 31, 2010. Twenty subjects were assigned to the control and experimental group, respectively. Video-audio media developed by the authors was used as the experimental tool. The effects of the program were analyzed using a knowledge assessment tool and questionnaire for compliance of the sick role. The SPSS/WIN 14.0 program was used for data analysis. Result: The first hypothesis that the "experimental group receiving the program using video-audio media will report a higher level of knowledge compared to the control group" was supported. The second hypothesis that the "experimental group that received the program using video-audio media will report a higher level of compliance of sick role" was supported. Conclusion: The program using video-audio media was effective in enhancing patients' knowledge about the disease and compliance of the sick role. Thus, it can be an effective nursing intervention for patients with coronary artery disease.

• International Journal of Fuzzy Logic and Intelligent Systems
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• 제8권1호
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• pp.75-81
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• 2008

This paper presents a top-down approach to implement an MPEG-2 audio encoder in VLSI. As the algorithm of an MPEG-2 audio encoder is heavy-weighted and heterogeneous(to be mixture of several strategies), the encoder design process is undertaken carefully from the algorithmic level to the architectural level. Firstly, the encoding algorithm is analyzed and divided into sub-algorithms, called tasks, and the tasks are partitioned in the way of reusing the same designs. Secondly, the partitioned tasks are scheduled and synthesized to make the most efficient use of time and space. In the end, a real-time 5 channel MPEG-2 audio encoder is designed which is a heterogeneous multiprocessor system; two hardwired logic blocks and one specialized DSP processor.

• 대한음성학회지:말소리
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• 제53호
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• pp.157-169
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• 2005

This paper proposes an efficiently representing method of spatial cues on multichannel spatial audio coding. The Binaural Cue Coding (BCC) method introduced recently represents multichannel audio signals by means of Inter Channel Level Difference (ICLD) or Source Index (SI). We tried to express more efficiently ICLD and SI information based on Inter Channel Correlation in this paper. We adopt different spatial cues according to ICC and propose a compensation method of empty spectrums created by using SI. We performed a MOS test and measuring spectral distortion. The results show that the proposed method can reduce the bitrate of side information without large degradation of the audio quality.

• 한국음향학회지
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• 제32권6호
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• pp.541-547
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• 2013

두세 명 혹은 그 이상의 참가자간사이의 다자간통화 시 음량불균형, 음량포화, 잡음레벨상승으로 인해서 음질 저하가 발생한다. 이 문제를 해결하기 위해서 본 논문은 소프트웨어 기반의 다지점제어장치를 위한 향상된 오디오 믹싱 알고리즘을 제안한다. 제안된 방식은 음성구간검출과 게인콘트롤이 결합된 기술로서 음성신호 분류, 음량 추정, 게인값 적용, 모든 채널의 음성신호를 믹싱하는 알고리즘들로 구성되어 있다. 제안된 오디오 믹싱 알고리즘은 효율적인 연산과 고품질의 음성을 제공하며, 실질적인 다자간 음성 통화에 적합하다.

• ETRI Journal
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• 제40권1호
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• pp.133-145
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• 2018

Robustness, payload, and imperceptibility of audio watermarking algorithms are contradictory design issues with high-level security of the watermark. In this study, the major issue in achieving high payload along with adequate robustness against challenging signal-processing attacks is addressed. Moreover, a security code has been strategically used for secure transmission of data, providing tamper detection at the receiver end. The high watermark payload in this work has been achieved by using the complementary features of third-level detailed coefficients of discrete wavelet transform where the human auditory system is not sensitive to alterations in the audio signal. To counter the watermark loss under challenging attacks at high payload, Daubechies wavelets that have an orthogonal property and provide smoother frequencies have been used, which can protect the data from loss under signal-processing attacks. Experimental results indicate that the proposed algorithm has demonstrated adequate robustness against signal processing attacks at 4,884.1 bps. Among the evaluators, 87% have rated the proposed algorithm to be remarkable in terms of transparency.