• Journal of Korea Multimedia Society
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• v.20 no.9
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• pp.1551-1558
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• 2017

Audio power amplifiers have been designed based on the premise that the impedance of loudspeakers is fixed at nominal 4 ohms or 8 ohms. However, it is known that the impedance varies with frequency and takes on the nominal value at some limited frequencies. The principle of the loudspeaker operation reveals that the sound pressure produced by the loudspeaker is proportional to the current flowing in the voice coil, not the voltage between the two terminals. We take the characteristics of the loudspeaker into account and compare the frequency responses of the loudspeaker in voltage-drive mode and current-drive mode via computer simulations, to conclude that the audio amplifier drive mode should be re-considered in an effort to improve the sound quality.

• Proceedings of the IEEK Conference
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• 2008.06a
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• pp.1175-1176
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• 2008

This paper deals with the study on the development of Audio SMPS with reduced noise. The noise defined here is in the range between 20 - 20 kHz frequency that a human can listen. We proposed a design method for Audio SMPS, which reduce noise and raise audio performances. This was achieved by new electronic elements, grounding, pattern, and shielding methods.

• The Journal of the Acoustical Society of Korea
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• v.21 no.6
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• pp.566-575
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• 2002

Low-titrate audio coding enables a number of Internet and mobile multimedia streaming service more efficiently. For the help of next-generation mobile telephone technologies and digital audio/video compression algorithm, we can enjoy the real-time multimedia contents on our mobile devices (cellular phone, PDA notebook, etc). But the limited available bandwidth of mobile communication network prohibits transmitting high-qualify AV contents. In addition, most bandwidth is assigned to transmit video contents. In this paper, we design a novel and simple method for reproducing high frequency components. The spectrum of high frequency components, which are lost by down-sampling, are modeled by the energy rate with low frequency band in Bark scale, and these values are multiplexed with conventional coded bitstream. At the decoder side, the high frequency components are reconstructed by duplicating with low frequency band spectrum at a rate of decoded energy rates. As a result of segmental SNR and MOS test, we convinced that our proposed method enhances the subjective sound quality only 10%∼20% additional bits. In addition, this proposed method can apply all kinds of frequency domain audio compression algorithms, such as MPEG-1/2, AAC, AC-3, and etc.

• The Journal of the Acoustical Society of Korea
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• v.24 no.8
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• pp.492-497
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• 2005

In this paper, we suggest that the new volume of robust audio watermarking method protects form the MP3 compression attack, mostly used in audio field. Watermarks were inserted at the coefficients repeatedly from low frequencies to high frequencies after DCT transform in commonly used Cox's spread spectrum method. However, our method could be investigated losing frequency when MP3 attacks and make order by small amount of losing frequency and then insert watermarks rightly through statistical analysis. In the result about various sound clips, the suggested method have better performances than Cox's method by preserving watermarks and reducing distortions of the original sounds.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.36 no.10B
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• pp.1252-1257
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• 2011

In this paper, efficient functions for audio spectrum mapping with visible spectrum are proposed. Through mapping overall hearing frequency band with visible frequency band, emotional lighting might be possible. We propose a basic linear mapping function and non-linear mapping functions emphasizing specific audio frequency bands. For the algorithm implementation, spectrum analysis method and filter method are introduced. Especially, in this paper, a prototype LED lighting equipment using the digital filter method is implemented. The proposed lighting method can be applied to many LED lighting area using music.

• The Journal of the Acoustical Society of Korea
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• v.32 no.1
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• pp.71-78
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• 2013

This paper describes a multiple audio watermarking using Quantization Index Modulation (QIM) on frequency phase and magnitude response. Proposed embedding procedure is composed of two stage. At the first stage, the watermark is embedded on the frequency phase response using QIM. In the second stage, the watermark is embedded using adaptive QIM with the step-size that is adaptively determined using the maximum value of the frequency magnitude response of every frame. The watermark is extracted by calculating the Euclidean distance as the blind detection. The proposed method is robust against most of attacks of audio watermark benchmarking. For the Fourier attacks, the proposed method shows over 95% recovery rate.

• The Journal of the Acoustical Society of Korea
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• v.30 no.6
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• pp.303-307
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• 2011

This paper describes an audio watermarking using Quantization Index Modulation (QIM) on significant peaks in frequency domain. The audio signal is broken up into L samples length frames with non-overlapping and rectangular window. The zero-crossing rate of each frame is calculated for decision whether it is proper to be watermarked or not. If the frame is legitimate, frequency magnitude response is computed by discrete Fourier transform. For the QIM, we set the quantization step size based on maximum value of frequency magnitude response and select n significant peaks with w samples around them in frequency domain, totally $n{\times}(w+1)$ samples. Finally, watermark embedding is performed. Decoder extract watermarks based on Euclidean distance, that is a blind detection. The proposed method is robust against many attacks of watermark benchmark.

• The Journal of the Acoustical Society of Korea
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• v.18 no.2
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• pp.77-82
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• 1999

A sinusoidal model has been widely used in the analysis and synthesis of speech and audio signals, and becomes one of the efficient candidates for high quality low bit rate audio coders. One of the crucial steps in the analysis and synthesis using a sinusoidal model is the detection of tonal components. This paper proposes an efficient method for the analysis and synthesis of audio signals using a sinusoidal model, which uses psychoacoustic criteria such as masking effect, masking index, and JNDf(Just Noticeable Difference in Frequency). Simulation results show that the proposed method reduces the number of sinusoids significantly without degrading the quality of the synthesized audio signals.

• The Journal of the Acoustical Society of Korea
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• v.28 no.2
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• pp.107-111
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• 2009

Acoustic OFDM is used for audible frequency band acoustic communication which employs loudspeaker as transmitter and microphone as the receiver antenna. Since acoustic OFDM can transmit about 1 kbps using 1600 Hz band. acoustic OFDM signal is inserted into the audio signal like music or speech, However. audio quality deteriorates definitely during the inserting process. This paper introduces a reason for audio quality deterioration and discuss how to reduce this phenomenon.

• Proceedings of the KIEE Conference
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• 2004.11c
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• pp.605-607
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• 2004

A class D digital audio amplifier with small size, low cost, and high quality is positively necessary in the multimedia era made of home theater system and the digital audio broadcasting (DAB). It is impossible to analyze the stability of the digital audio amplifier, which is based on the PWM signal processing. To solve this problem, the digital audio amplifier is analyzed using variable structure control theory which is one of nonlinear system theories. Moreover, the magnitude and the frequency of ripple signal, which generated by hysteresis in the comparator, is obtained using describing function which is useful to represent the input-output relation of nonlinear system.