• 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.

• Journal of Korea Multimedia Society
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• v.21 no.9
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• pp.1068-1075
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• 2018

We propose a method for driving loudspeakers from class D audio power amplifiers in current mode, instead of in conventional voltage mode, which was impossible with the feedback circuitry. Unlike analog audio amplifiers, Class D audio power amplifiers have signal delay between the input and output signals, which makes it difficult to apply the feedback circuitry for current-mode driving. The idea of the pre-distortion scheme used for the compensation of the non-linearity of RF power amplifiers is adapted to remedy the impedance variation effect of the loudspeakers for current driving. The method uses the speaker model for the pre-distorter to compensate for the speaker impedance variation with frequency. The simulation and test results confirms the validity of the proposed method.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.283-286
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• 2004

Due to advances in electronics very high quality audio reproduction is today possible. But the listening environment causes deviation of the audio system from the expected behavior. Firstly the listening Room significantly changes the audio signal frequencies and their phase. Secondly the position of the user in the room affects the perceived sound. With existing DSP technology it is possible to adequately correct these effects. In our work we developed a room correction system, correcting up to 7.1 channels using dual Motorola 56367 fixed point DSP's, implementing position dependent room effects measurement, real time compensation filter design and equalization filtering procedures.

• The Journal of the Acoustical Society of Korea
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• v.19 no.3
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• pp.63-71
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• 2000

A new type of second-order digital peaking filters for professional-quality digital audio system is proposed whose frequency response can be elaborately controlled throughout the composite structure of a standard band-pass filter and a 0-dB bypass gain. The proposed method for designing the peaking filter uses the Q-compensation technique to prevent the Q-distortion caused by the variation of the gain factor and is reduced into a compact form which is proper to the real-time implementation. Methods are examined for computing its coefficients, which are exact and very straightforward to compute with small amount of the system resources.

• The Journal of the Acoustical Society of Korea
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• v.23 no.1
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• pp.54-60
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• 2004

In order to mitigate the problems in storage space and network bandwidth for the full CD quality audio, current digital audio is always restricted by sampling rate and bandwidth. This restriction normally results in low sample rate audio or calls for the data compression scheme such as MP3. However, they can only reproduce a lower frequency range than a regular CD quality because of the Nyquist sampling theory. Consequently they lose rich spatial information embedded in high frequency. The propose of this paper is to propose efficient high frequency enhancement of low sample rate audio using n adaptive filtering and DWT analysis and synthesis. The proposed algorithm uses the LMS adaptive algorithm to estimate the missing high frequency contents in DWT domain and it then reconstructs the spectrally enhanced audio by using the DWT synthesis procedure. Several experiments with real speech and audio are performed and compared with other algorithm. From the experimental results of spectrogram and sonic test, we confirm that the proposed algorithm outperforms the other algorithm and reasonably works well for the most of audio cases.

• Journal of the Korea Convergence Society
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• v.7 no.3
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• pp.77-85
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• 2016

We propose a digital signal processing technic that can compensate the non-linearity inherent in audio amplifiers, and present the result of the simulation. The inherent non-linearity of the audio power amplifier arising from analog devices is compensated via a digital signal processing technic consisting of indirect learning architecture and an adaptive filter. The simulation results show that the compensator can be realized using a third-order polynomial and compensates odd-order non-linearity efficiently. The even-oder non-linearity is mainly due to the dc offset at the output, which is difficult to eliminate with the proposed method. Care must be taken in designing the bias circuit to avoid the DC offset at the output. The proposed technic has significance in that digital signal processing technic can compensate for the impairment that is an inherent characteristic of an analog system.

• Journal of Broadcast Engineering
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• v.6 no.1
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• pp.21-31
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• 2001

In the PN-sequence based audio watermarking system, the PN sequence is shaped by a filter derived from the psychoacoustic model to increase robustness and inaudibility The psychoacoustic model calculated in each audio segment, however, requires heavy computational loads. In this paper, we propose an efficient watermarking system adopting a fixed-shape perceptual filter that substitutes psychoacoustic model derived filter. The proposed filter can shape the PN-sequence to be inaudible and enable to embed the robust watermark in a simple manner. Moreover, we propose an anchitecture for the PN-sequence compensation fitter In the watermark detecter to increase correlation between the watermark and the PN-sequence. With the proposed architecture, the blind watermark detection performance has been enhanced.

• Journal of the Korean Chemical Society
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• v.46 no.6
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• pp.569-580
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• 2002

The purpose of this study was to analyze the problem-solving process of student's compensation con-cept.For this purpose, verbal interactions during activities were audio-taped, transcribed, and analyzed. And classroom observation and interview with students were carried out. Students who were superior in mathematical operations tended to explain compensation concept using proportionality. On the other hand, students who had low level of conservation concept can not connect 'relation of two variables' with 'conservation of equilibrium' at the formation process of com-pensation concept. Students who succeed in the formation of compensation concept showed high level of conservation concept. To promote the formation of compensation concept, it is necessary that how to develop proportional concept and conservation concept as closely related with compensation concept should be studied.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 1995.06a
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• pp.165-170
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• 1995

In this paper, it is attempted to apply the adaptive digital signal processing to compensate for a linear distortion of a loudspeaker and implement a real time hardware for that purpose. The real time system is implemented by using the DSP56001, a general purpose signal processor, as a host processor and the DSP56200, a cascadable adaptive FIR filter peripheral chip, as an adaptive digital filter. The system has 1000 taps at a 44.1kHz. After inverse modeling of under_compensation_speaker, the system reduces loudspeaker's linear distortions by pre-processing an input audio signal to loudspeaker. The experiment shows satisfactory results; after adaption with white noise as input signal for 60sec, the flat amplitude and linear phase frequency characteristics is found to lie over a wide frequency range of 100Hz to 20kHz.

• Proceedings of the IEEK Conference
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• 2002.07c
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• pp.1717-1720
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• 2002

Objective quality measurement schemes that in- corporate properties of the human auditory system. The basilar membrane(BM) acts as a spectrum analyzer, spatially decomposing the signal into frequency components. Each filterbank is an implementation of the ERB, gam-machirp function. This filterbank is level-dependent asymmetric compensation filters. And for the validation of the auditory model, we calculate the CPD. Quality measurement is obtained from the result.