• 한국HCI학회:학술대회논문집
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• 2009.02a
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• pp.150-157
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• 2009

In this paper, we propose a person tracking based multi-channel audio panning system for multi-view broadcasting services. Multi-view broadcasting is to render the video sequences that are captured from a set of cameras based on different viewpoints, and multi-channel audio panning techniques are necessary for audio rendering in these services. In order to apply such a realistic audio technique to this multi-view broadcasting service, person tracking techniques which are to estimate the position of users are also necessary. For these reasons, proposed methods are composed of two parts. The first part is a person tracking method by using ultrasonic satellites and receiver. We could obtain user's coordinates of high resolution and short duration about 10 mm and 150 ms. The second part is MPEG Surround parameter-based multi-channel audio panning method. It is a method to obtain panned multi-channel audio by controlling the MPEG Surround spatial parameters. A MUSHRA test is conducted to objectively evaluate the perceptual quality and measure localization performance using a dummy head. From the experiments, it is shown that the proposed method provides better perceptual quality and localization performance than the conventional parameter-based audio panning method. In addition, we implement the prototype of person tracking based multi-view broadcasting system by integrating proposed methods with multi-view display system.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.4 no.2
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• pp.77-82
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• 2011

In this paper, we investigate source separation algorithms for stereo audio mixed using amplitude panning method. This source separation algorithms can be used in various applications such as up-mixing, speech enhancement, and high quality sound source separation. The methods in this paper estimate the panning angles of individual signals using the principal component analysis being applied in time-frequency tiles of the input signal and independently extract each signal through directional filtering. Performances of the methods were evaluated through computer simulations.

• ETRI Journal
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• v.32 no.5
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• pp.831-834
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• 2010

This paper proposes a new joint channel coding algorithm based on principal component analysis. A conventional joint channel coder using passive downmixing undergoes a reduction of both the primary-to-ambient energy ratio (PAR) of the downmix signal and the panning gain ratio of the primary source. The proposed system preserves the PAR of the downmix signal by using active downmixing which reflects spatial characteristic. The proposed system also improves the accuracy of the panning gain ratio estimation. Computer simulations and subjective listening tests verify the performance of the proposed system.

• The Journal of the Acoustical Society of Korea
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• v.30 no.4
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• pp.197-206
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• 2011

In this paper, we proposes the virtual sound source panning algorithm in the multichannel system. Recently, High-definition (HD) and Ultrahigh-definition (UHD) video formats are accepted for the multimedia applications and they provide the high-quality resolution pixels and the wider view angle. The audio format also needs to generate the wider sound field and more immersive sound effects. However, the conventional stereo system cannot satisfy the desired sound quality in the latest multimedia system. Therefore, the various multichannel systems that can make more improved sound field generation are proposed. In the mutichannel system, the conventional panning algorithms have acoustic problems about directivity and timbre of the virtual sound source. To solve these problems in the arbitrary positioned multichannel loudspeaker system, we proposed the virtual sound source panning algorithm using multiple vectors base nonnegative amplitude panning gains. The proposed algorithm can be easily controlled by the gain control function to generate an accurate localization of the virtual sound source and also it is available for the both symmetric and asymmetric loudspeakers format. Its performance of sound localization is evaluated by subjective tests comparing with conventional amplitude panning algorithms, e.g. VBAP and MDAP, in the symmetric and asymmetric formats.

• Journal of Broadcast Engineering
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• v.17 no.6
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• pp.924-932
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• 2012

Decomposition of a stereo signal into the primary and ambient components is a key step to the stereo upmix and it is often based on the principal component analysis (PCA). However, major shortcoming of the PCA-based method is that accuracy of the decomposed components is dependent on both the primary-to-ambient power ratio (PAR) and the panning angle. Previously, a modified PCA was suggested to solve the PAR-dependent problem. However, its performance is still dependent on the panning angle of the primary signal. In this paper, we proposed a new PCA-based primary-ambient decomposition algorithm whose performance is not affected by the PAR as well as the panning angle. The proposed algorithm finds scale factors based on a criterion that is set to preserve the powers of the mixed components, so that the original primary and ambient powers are correctly retrieved. Simulation results are presented to show the effectiveness of the proposed algorithm.

• Proceedings of the IEEK Conference
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• 2005.11a
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• pp.447-450
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• 2005

Multi-channel artificial reverberation algorithm to control perceived direction and distance is described in this paper. In conventional algorithms using IIR filters, reverberation time is the only parameter to be controlled. Moreover, since the convolution-based conventional algorithms apply only same impulse responses, but not considering sound localization, it was not realistic enough. The new algorithm proposed in this paper utilizes early reflections segmented according to the azimuth from which direct sound comes and controls perceived direction by panning the direct sound, and controls perceived distance by adjusting Energy Decay Curve (EDC) of reverberation and gain of the direct sound. In addition, the algorithm enhances Listener Envelopment(LEV) to make late reverberation incoherent among channels.

• The Journal of the Acoustical Society of Korea
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• v.36 no.5
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• pp.338-344
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• 2017

In this paper, an interpolation method of HRTF (Head-Related Transfer Function) is proposed for small-sized measurement data set, especially. The proposed algorithm is based on acoustic modeling of HRTFs, and the algorithm tries to interpolate the HRTFs via estimation the model coefficients. However, the estimation of the model coefficients is hard if there is lack of measurement points, so the algorithm solves the problem by a data augmentation using the VBAP (Vector Based Amplitude Panning). Therefore, the proposed algorithm consists of two steps, which are data augmentation step based on VBAP and model coefficients estimation step by least squares method. The proposed algorithm was evaluated by a simulation with a measured data from CIPIC (Center for Image Processing and Integrated Computing) HRTF database, and the simulation results show that the proposed algorithm reduces mean-squared error by 1.5 dB ~ 4 dB than the conventional algorithms.