• International journal of advanced smart convergence
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• v.2 no.2
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• pp.16-18
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• 2013

In this paper, we develop a method to design a flexible thin film audio systems utilizing Polyvinylidene fluoride. The system we designed showed the properties of increased transparency and sound pressure levels. As an input terminal transparent oxide thin film is adopted. In order to provide dielectric insulation, a transparent insulating oxide thin film is coated to obtain double -layered structure. In the range of visible light, the output from the output of the system showed an increased and improved sound pressure level. The piezoelectric polymer film of polyvinylidene fluoride (PVDF) is used to produce mechanical vibration due to the applied electrical voltage signal. An analog electric voltage signal is transformed into sound waves in the audio system.

• Proceedings of the KIEE Conference
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• 2007.11c
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• pp.179-182
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• 2007

The Design and Optimization of a low-profile antenna for the Satellite Digital Audio Radio System(SDARS) using a Non-Dominated Sorting Genetic Algorithm(NSGA) is presented. a fairly omni-directional elevation gain pattern over 60 degrees < q <60 degrees are obtained. The average value of the LCP gain pattern is approximately 2 dBic. The heights of the antennas are less than 0.251. The cross polarized signal level is approximately 12 dBic less than the co-polarized signal level.

• International Journal of Internet, Broadcasting and Communication
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• v.6 no.1
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• pp.13-15
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• 2014

We develop a method to fabricate a flexible thin film audio systems using polyvinylidene fluoride(PVDF). The system we designed showed the properties of increased flexibility, transparency, and sound pressure levels. As an input port of two terminals, transparent oxide thin film with a low resistivity is adopted. In order to provide dielectric insulation, a transparent insulating oxide thin film is coated to obtain double-layered structure. In the range of visible light, the output from the output of the system showed a increased and improved sound pressure level. The piezoelectric polymer film of PVDF is used to produce mechanical vibration due to the applied electrical voltage signal. An analog electric voltage signal is transformed into sound waves in the audio system.

• Convergence Security Journal
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• v.8 no.4
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• pp.91-96
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• 2008

An increasing in quantity of multimedia data brought a new problem that expected data should be retrieved fast and exactly. The adequate representation is a key element for the efficient retrieval. For this reason, MPEG-7 standard was established for description of multimedia data in 2001. In this paper, we designed a Audio/Image Retrieval System based on MPEG-7 that can retrieve multimedia data like audio, image efficiently. And we integrated high-level and low-level schemas to retrieve datas for users.

• The Journal of the Acoustical Society of Korea
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• v.29 no.3E
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• pp.131-139
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• 2010

This paper proposes a novel spectral hole substitution technique for low bit-rate audio coding. The spectral holes frequently occurring in relatively weak energy bands due to zero bit quantization result in severe quality degradation, especially for harmonic signals such as speech vowels. The enhanced aacPlus (EAAC) audio codec artificially adjusts the minimum signal-to-mask ratio (SMR) to reduce the number of spectral holes, but it still produces noisy sound. The proposed method selectively predicts the spectral shapes of hole bands using either intra-band correlation, i.e. harmonically related coefficients nearby or inter-band correlation, i.e. previous frames. For the bands that have low prediction gain, only the energy term is quantized and spectral shapes are replaced by pseudo random values in the decoding stage. To minimize perceptual distortion caused by spectral mismatching, the criterion of the just noticeable level difference (JNLD) and spectral similarity between original and predicted shapes are adopted for quantizing the energy term. Simulation results show that the proposed method implemented into the EAAC baseline coder significantly improves speech quality at low bit-rates while keeping equivalent quality for mixed and music contents.

• Proceedings of the IEEK Conference
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• 2006.06a
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• pp.241-242
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• 2006

MPEG Surround is an audio coding technology that represents multi-channel audio signal with downmixed audio signal(s) and very low bitrate side information based on Binaural Cue Coding. The side information consists of Inter-Channel Level Difference, Inter-Channel Correlation, and payloads. These two parameters are correspondent to the well-known spatial parameters in psycho-acoustics, Inter-aural Level Difference (ILD) and Inter-Aural Cross Correlation (IACC). Though ICLD is to provide perceptually equivalent ILD to the listener, however, the ILD of the original multi-channel audio signal and that of the MPEG Surround encoded signal was different. The difference between two ILD values is defined as ILD Distortion (ILDD). This paper provides how ILDD can be applied to enhance sound quality in MPEG Surround and how much ILDD is decreased.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.8A
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• pp.1169-1178
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• 1999

In this paper, the method to improve audio noise when GMS(Global System of Mobile Communication) is built in HPC(Handheld PC)is provided. The biggest problem with quality of audio is improved by connecting GSM module with wire to the widest erea near by HPC Power Ground to reduce impedance difference, and designing amplifier and earpiece, Result of measurement is satisfied with GSM Acoustic Standard. Standard of GSM Audio is declared in GSM 11.10 ETS 300 607-1. Measuring items corresponds to that standard and B&K Type 6712 is used for measurement. The list of measurement presented in this paper is Sending Sensitivity Frequency Response and SLR, Sending Loudness Rating( SLR level), Receiving Sensitivity Loudness Rating(RLR level), Talker Sidetone(STMR), Stability margin, and Echo Loss(ERL)

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.49 no.4
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• pp.16-22
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• 2012

In this paper, the forensic marking algorithm is proposed using psychoacoustic model II and MDCT for high-quality audio. The proposed forensic marking method, that inserts the user fingerprinting code of the audio content into the selected sub-band, in which audio signal energy is lower than the spectrum masking level. In the range of the one frame which has 2,048 samples for FFT of original audio signal, the audio forensic marking is processed in 3 sub-bands. According to the average attack of the fingerprinting codes, one frame's SNR is measured on 100% trace ratio of the collusion codes. When the lower strength 0.1 of the inserted fingerprinting code, SNR is 38.44dB. And in case, the added strength 0.5 of white gaussian noise, SNR is 19.09dB. As a result, it confirms that the proposed audio forensic marking algorithm is maintained the marking robustness of the fingerprinting code and the audio high-quality.

• ETRI Journal
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• v.27 no.5
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• pp.608-616
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• 2005

Watermarking has become a technology of choice for a broad range of multimedia copyright protection applications. This paper proposes an audio watermarking scheme that uses the modified tonal masker as an embedding carrier for imperceptible and robust audio watermarking. The method of embedding is to select one of the tonal maskers using a secret key, and to then modify the frequency signals that consist of the tonal masker without changing the sound pressure level. The modified tonal masker can be found using the same secret key without the original sound, and the embedded information can be extracted. The results show that the frequency signals are stable enough to keep embedded watermarks against various common signal processing types, while at the same time the proposed scheme has a robust performance.

• ETRI Journal
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• v.27 no.3
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• pp.330-333
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• 2005

The Inter-Channel Level Difference (ICLD) is a cue parameter to estimate spectral information in a binaural cue coding that has been recently in the spotlight as a multichannel audio signal compression technique. Even though the ICLD is an essential parameter, it is generally distorted by quantization. In this paper, a new modified ICLE representation method to minimize the quantization distortion is proposed by adopting a flexible determination of the reference channel and the unidirectional quantization. Our experimental result confirms that the proposed method improves the multichannel audio output quality even with the reduced bit-rate.