• Journal of Satellite, Information and Communications
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• v.8 no.4
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• pp.105-110
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• 2013

After analog broadcast changed to digital, the dynamic range of the broadcast audio became wider. As there are no regulation for digital broadcast audio level, the audio level of digital TV is gradually increasing, and this phenomenon is getting more serious. because of competition between broadcasters and programmes. To solve this problem, ITU-R legislated technical recommendations for digital TV audio level. In this paper, the audio levels of domestic TV channels are measured according to the algorithm of the ITU-R, and analyzed management method is suggested.

• Journal of Satellite, Information and Communications
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• v.8 no.1
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• pp.31-34
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• 2013

This paper deals with the interfering audio noise caused by LTE(Long Term Evolution) UE(User Equipment) which is 4th generation mobile communications on audio devices. At first, we realized that the interfering signal of the LTE UE is determined by the transmit power of the LTE UE through analysis and measurement. Then, we performed to measure audio noise level according to the variation of transmitting power level and separation distance between the LTE UE and an audio device. As a result, it is required that minimum separation distance should be 25 cm and above in order to protect audio device from the interference noise of the LTE UE with the maximum transmit power level of 22 dBm.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.23 no.1
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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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• v.2 no.1
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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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• v.5 no.3
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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.

• The Journal of Korean Academic Society of Nursing Education
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• v.17 no.1
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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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• v.8 no.1
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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.

• MALSORI
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• no.53
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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.

• The Journal of the Acoustical Society of Korea
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• v.32 no.6
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• pp.541-547
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• 2013

The speech quality of multi-party audio telephony between two, three or more participants is decreased by audio volume imbalance, audio volume saturation and noise level increase. To solve this issue, this paper proposes an advanced audio mixing algorithm for software-based multi-point control unit. Our approach is based on the combined voice activity detection and gain control technique that consists of a set of algorithms that classify audio signals, estimate audio volumes, adjust gain factors and mix audio signals of all channels. The proposed audio mixing algorithm is computationally efficient, delivers high-quality speech, and is suitable for use in any practical multi-party audio telephony.

• ETRI Journal
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• v.40 no.1
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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.