Dae Hee Youn

Yonsei University, Sŏul, Seoul, South Korea

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Publications (153)161.05 Total impact

  • [Show abstract] [Hide abstract]
    ABSTRACT: To provide immersive 3D multimedia service, MPEG has launched MPEG-H, ISO/IEC 23008, “High Efficiency Coding and Media Delivery in Heterogeneous Environments.” As part of the audio, MPEG-H 3D Audio has been standardized based on a multichannel loudspeaker configuration (e.g., 22.2). Binaural rendering is a key application of 3D audio; however, previous studies focus on binaural rendering with low complexity such as IIR filter design for HRTF or pre-/post-processing to solve in-head localization or front-back confusion. In this paper, a new binaural rendering algorithm is proposed to support the large number of input channel signals and provide high-quality in terms of timbre, parts of this algorithm were adopted into the MPEG-H 3D Audio. The proposed algorithm truncates binaural room impulse response at mixing time, the transition point from the early-reflections to the late reverberation part. Each part is processed independently by variable order filtering in frequency domain (VOFF) and parametric late reverberation filtering (PLF), respectively. Further, a QMF domain tapped delay line (QTDL) is proposed to reduce complexity in the high-frequency band, based on human auditory perception and codec characteristics. In the proposed algorithm, a scalability scheme is adopted to cover a wide range of applications by adjusting the threshold of mixing time. Experimental results show that the proposed algorithm is able to provide the audio quality of a binaural rendered signal using full-length binaural room impulse responses. A scalability test also shows that the proposed scalability scheme smoothly compromises between audio quality and computational complexity.
    No preview · Article · Aug 2015 · IEEE Journal of Selected Topics in Signal Processing
  • Se-Woon Jeon · Young-Cheol Park · Dae Hee Youn
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    ABSTRACT: In this letter, we propose a new auditory distance rendering (ADR) algorithm based on the interchannel phase difference (ICPD) control. In the conventional ICPD control, distance perception of the sound image is nonlinearly controlled, and directional localization of the sound image can be biased by changes of the interaural cues. These problems are caused by applying the frequency-independent ICPD without considering the acoustic transfer paths of the system setup. To solve these problems, first, the interaural cues of ear signals are analyzed by binaural auditory simulations. Then, stereophonic ADR filters are designed that produce ear signals with a linearly controlled interaural cross-correlation (IACC) and a consistent interaural level difference (IALD) for sophisticated distance perception under the given stereo setup. Subjective test results show that the proposed algorithm can provide better distance controllability than the conventional method with reduced lateralization blur of the sound image.
    No preview · Article · Apr 2015 · IEEE Signal Processing Letters
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    Se-Woon Jeon · Dae Hee Youn · Young-cheol Park · Gun-Woo Lee
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    ABSTRACT: During a phone conversation, loud vocal emission from the far-end to the near-end space can disturb nearby people. In this paper, the possibility of actively controlling such unwanted sound emission using a control source placed on the mobile device is investigated. Two different approaches are tested: Global control, minimizing the potential energy measured along a volumetric space surface, and local control, minimizing the squared sound pressure at a discrete point on the phone. From the test results, both approaches can reduce the unwanted sound emission by more than 6 dB in the frequency range up to 2 kHz.
    Preview · Article · Apr 2015 · The Journal of the Acoustical Society of America
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    ABSTRACT: This paper describes a technique for designing a collection of beamformers, a "beamformer bank," that approximately produces a constant-amplitude panning law. Useful in multichannel audio recording scenarios, a point source will appear with energy above a specified sidelobe level in at most two adjacent beams, and the sum of all beam signals will approximate the source signal. A design method is described in which a specified sidelobe level determines beamwidth as a function of arrival direction and frequency, leading directly to the number and placement of beams at each frequency. Simulation results are presented verifying the proposed technique's performance.
    Preview · Article · Sep 2014 · The Journal of the Acoustical Society of America
  • Yoomi Hur · Y.-C. Park · Jonathan S. Abel · Dae Hee Youn
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    ABSTRACT: This letter describes a numerical algorithm for synthesizing optimal low-sidelobe beampatterns. The pattern synthesis problem is formulated as a constrained optimization that minimizes the spatially weighted energy arriving at the array subject to unit gain in the look direction and a constant sidelobe level. The weighting takes on the form of a sensor correlation matrix, parameterized between uncorrelated sensor noise and correlated signal arrival terms according to a balancing factor. Setting the balancing factor to its extreme values of 0 and 1, produces, respectively, the optimal Riblet-Chebyshev and Delay-and-Sum type beamformers. In between, the method generates a low-sidelobe beamformer with varying beamwidths that provides a trade-off between White Noise Gain and Directivity Index. Simulation examples demonstrate optimal and intermediate designs for uniform and non-uniform sensor spacings.
    No preview · Article · Aug 2014 · IEEE Signal Processing Letters
  • Tung chin Lee · Young-Cheol Park · Dae Hee Youn

    No preview · Article · Jun 2014
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    ABSTRACT: In this paper, we propose a high-order lattice adaptive notch filter (LANF) that can robustly track multiple sinusoids. Unlike the conventional cascade structure, the proposed high-order LANF has robust tracking characteristics regardless of the frequencies of reference sinusoids and initial notch frequencies. The proposed high-order LANF is applied to a narrowband adaptive noise cancellation (ANC) to mitigate the effect of the broadband disturbance in the reference signal. By utilizing the gradient adaptive lattice (GAL) ANC algorithm and approximately combining it with the proposed high-order LANF, a computationally efficient narrowband ANC system is obtained. Experimental results demonstrate the robustness of the proposed high-order LANF and the effectiveness of the obtained narrowband ANC system.
    Preview · Article · Jan 2014 · Journal on Advances in Signal Processing
  • Taegyu Lee · Yonghyun Baek · Young-cheol Park · Dae Hee Youn
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    ABSTRACT: This paper presents a new binaural auralization algorithm suitable for mobile devices. A stereo-tomultichannel virtual upmix technique was utilized to expand the spatial dimension of stereo audio to a multichannel speaker space. In order to enhance subjective preferences and listener envelopment, reflections of the primary sound component were artificially generated by modeling a virtual ceiling and sidewalls based on concert hall acoustics. The uncorrelated ambient components were then distributed over spatial points at different azimuth and elevation angles. Computational efficiency was achieved by removing the redundant operations in each stage of the algorithm. Subjective tests were performed, and the results showed that the proposed algorithm significantly improves spatial perceptions of stereo audio being played over headphones.
    No preview · Conference Paper · Jan 2014
  • T.C. Lee · Y.-C. Park · D.-H. Youn
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    ABSTRACT: In this paper, we present a new vocal suppression algorithm that can enhance the quality of music signal coded using spatial audio object coding (SAOC) in Karaoke mode. The remained vocal component in the coded music signal is estimated and suppressed by using a spectral subtraction method. Using the fact that the level of the remained vocal components is varied depending on the input object power, we propose a psychoacoustic rule where the suppression level is adapted according to the auditory masking property. Objective and subjective test were performed and the results confirm that the proposed algorithm offers an improved quality.
    No preview · Article · Jan 2014
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    Seong-woo Kim · Young-cheol Park · Dae Hee Youn
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    ABSTRACT: In passive sonar, adaptive algorithms can be used to cancel strong sinusoidal self-interferences. In order to correctly recover low-power target signals during the early stages of processing, these adaptive algorithms must provide fast convergence and, at the same time, narrow notches at the frequencies of the sinusoids. In this respect, the gradient adaptive lattice (GAL) algorithm is a very attractive choice. However, the GAL algorithm with a constant step-size parameter has to compromise between the convergence rate and notch bandwidths. Therefore, in this article, we propose a variable step-size scheme for the GAL algorithm that can achieve both a fast convergence rate and narrow notches. Simulation results demonstrate the efficiency of the proposed algorithm compared to both the conventional GAL algorithm and transversal adaptive filter combined with the variable step-size scheme.
    Full-text · Article · Dec 2013 · Journal on Advances in Signal Processing
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    Tung Chin Lee · Young-Cheol Park · Dae Hee Youn
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    ABSTRACT: In this paper, we present a vocal suppression algorithm that can enhance the quality of music signal coded using Spatial Audio Object Coding (SAOC) in Karaoke mode. The residual vocal component in the coded music signal is estimated by using a cross prediction method in which the music signal coded in Karaoke mode is used as the primary input and the vocal signal coded in Solo mode is used as a reference. However, the signals are extracted from the same downmix signal and highly correlated, so that the music signal can be severely damaged by the cross prediction. To prevent this, a psycho-acoustic disturbance rule is proposed, in which the level of disturbance to the reference input of the cross prediction filter is adapted according to the auditory masking property. Objective and subjective test were performed and the results confirm that the proposed algorithm offers improved quality.
    Preview · Article · May 2013
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    Dong-Il Hyun · Young-Cheol Park · Dae Hee Youn
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    ABSTRACT: In this paper, the problems of the conventional amplitude panning method for a stereophonic panning system are analyzed. We observed that the distortion showed a feedforward comb filter response. As a remedy to this distortion, we propose a stereophonic panning system using a feedback comb filter. The comb filter is designed to minimize the difference between interaural level difference(ILD) of the proposed system and that of HRTF because ILD is most salient cue for the perception of the source direction. The proposed system is configured to operate selectively for the frequency band related to the source direction. The performance of the proposed system is verified by subjective listening tests.
    Preview · Article · Mar 2013
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    Taegyu Lee · Seokjin Lee · Young-cheol Park · Dae Hee Youn
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    ABSTRACT: In this paper we propose a method of creating virtual bass based on a multiband harmonic generation. The proposed algorithm separately generates harmonics for each band whose bandwidth is adaptively adjusted according to the tonal distribution in the bass signal. Both even and odd harmonics are generated by combining two nonlinear devices. The proposed algorithm overcomes the intermodulation distortion and spectral smearing problem that are often encountered in the previous algorithms. Results of objective and subjective tests are presented to validate the proposed algorithm.
    Full-text · Conference Paper · Jan 2013
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    Dong-il Hyun · Young-cheol Park · Dae Hee Youn
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    ABSTRACT: Conventional parametric stereo (PS) audio coding employs inter-channel phase difference and overall phase difference as phase parameters. In this article, it is shown that those parameters cannot correctly represent the phase relationship between the stereo channels when inter-channel correlation (ICC) is less than one, which is common in practical situations. To solve this problem, we introduce new phase parameters, channel phase differences (CPDs), defined as the phase differences between the mono downmix and the stereo channels. Since CPDs have a descriptive relationship with ICC as well as inter-channel intensity difference, they are more relevant to represent the phase difference between the channels in practical situations. We also propose methods of synthesizing CPDs at the decoder. Through computer simulations and subjective listening tests, it is confirmed that the proposed methods produce significantly lower phase errors than conventional PS, and it can noticeably improve sound quality for stereo inputs with low ICCs.
    Preview · Article · Nov 2012 · EURASIP Journal on Audio Speech and Music Processing
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    ABSTRACT: Recently, handheld devices with sound functionality are popular. A problem encountered by using such devices is sound leakage due to inappropriate volume setting, which should be reduced not to disturb people other than the user. In this paper, we present an active control (ANC) technique to control such sound leakage in handheld devices. The least squares (LS) optimum control under various positions of error sensors is investigated. Based on the results, desirable and feasible microphone-loudspeaker setups are suggested. Later, simulations are conducted with microphones mounted on the cellular phone mockup. Test results indicate that the unwanted sound leakage can be effectively reduced by more than 6 dB for all directions in the frequency range up to 2 kHz.
    No preview · Article · Jan 2012
  • Se-Woon Jeon · Young-Cheol Park · Dae Hee Youn
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    ABSTRACT: A stereophonic system for acoustic depth rendering is presented. The proposed system employs spatial source decomposition and frequency-dependent phase adjustment algorithms. The system dynamically controls a sound distance perception of stereo signal to provide more dramatic 3D effects for a multimedia user.
    No preview · Article · Jan 2012
  • S. Choi · D.-I. Hyun · Y.-C. Park · S. Lee · D.H. Youn
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    ABSTRACT: In this paper, we present a method of synthesizing the height and wide channel signals for stereo upmix to multichannel format beyond 5.1. To provide an improved envelopment, reflections from ceiling and side-walls are considered for the height and wide channel synthesis. Early reflections (ERs) corresponding to the spatial sections covered by the height and wide channel speakers are separately synthesized using the image source method, and the parameters for the ER generation are determined from the primary-to-ambient ratio (PAR) estimated from the stereo signal. Later, the synthesized ERs are mixed with decorrelated ambient signals and transmitted to the respective channels. Subjective listening tests verify that listener envelopment can be improved by using the proposed method.
    No preview · Article · Jan 2012
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    Dong-il Hyun · Jeongil Seo · Young-cheol Park · Dae Hee Youn
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    ABSTRACT: This paper proposes an improved phase parameter analysis and synthesis method for parametric stereo audio coding. For stereo signal with low interchannel coherence (IC), conventional phase parameter analysis and synthesis methods show poor quality, since they overlook the effect of IC to the phase parameters defined. The proposed system defines new phase parameters reflecting IC information, so that we can alleviate the deterioration of audio quality due to the synthesis of excessive phase difference, especially when IC is low. Subjective listening test results are presented to show that the proposed method provides always better audio quality than the conventional methods.
    Preview · Conference Paper · Jun 2011
  • Yoomi Hur · Jonathan S. Abel · Y.-C. Park · Dae Hee Youn
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    ABSTRACT: Methods are described for synthetically reconfiguring a microphone array, that is, processing signals recorded at a microphone array so as to estimate the signals that would have appeared at the elements of a different microphone array at the same location. Two approaches are described: a nonparametric method in which a fixed, low-sidelobe beamformer applied to the source array drives virtual sources rendered at the target array, and a parametric technique in which constrained beamformers are used to estimate source arrival directions, with the source signals extracted and rendered at their respective estimated arrival directions. Finally a hybrid method is presented which separately renders extracted point sources and the spatially diffuse residuals. Experimental results using arrays of 2 mm-diameter microphones and human HRTFs are reported.
    No preview · Article · Jun 2011 · Journal of the Audio Engineering Society
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    ABSTRACT: In the last few decades, various panning algorithms have been proposed to generate virtual sound localization in loud-speaker systems. Vector base amplitude panning (VBAP) is the most widely adopted pair-wise amplitude panning method. However, pair-wise amplitude panning has a di-rectional discontinuity problem in the multichannel surround panning. In particular, sound localization for the virtual source does not smoothly vary in the direction of the near speaker. While coincident panning using multiple loud-speakers, such as Ambisonics, performs better, it has less stability in sound localization due to the precedence effect. In this paper, a multiple-wise vector base virtual source pan-ning algorithm is proposed. To generate more stable local-ization of panning sound, the proposed panning algorithm calculates the amplitude panning gains using multiple-wise vector base formulation. Additionally, the angle-dependent and nonnegative gain control function is applied to prevent artifacts caused by negative amplitude gains. The subjective listening tests using the method of adjustment (MOA) are performed to evaluate the sound localization of the proposed algorithm, which is compared with the conventional ampli-tude panning methods.
    Preview · Article · Jan 2011

Publication Stats

907 Citations
161.05 Total Impact Points

Institutions

  • 1986-2015
    • Yonsei University
      • Department of Electrical and Electronic Engineering
      Sŏul, Seoul, South Korea
  • 1995-2002
    • University of Seoul
      Sŏul, Seoul, South Korea
  • 2001
    • Samsung Thales
      Sŏul, Seoul, South Korea
  • 1999
    • Hyundai Engineering Co., Ltd.
      Sŏul, Seoul, South Korea
    • Kunsan National University
      • Department of Electrical Engineering
      Gunzan, Jeollabuk-do, South Korea
  • 1998
    • The Seoul Institute
      Sŏul, Seoul, South Korea
  • 1997
    • Soongsil University
      Sŏul, Seoul, South Korea
  • 1988
    • University of New Mexico
      Albuquerque, New Mexico, United States
  • 1983-1986
    • University of Iowa
      • Department of Electrical and Computer Engineering
      Iowa City, Iowa, United States
  • 1979-1984
    • Kansas State University
      • Department of Electrical and Computer Engineering
      Kansas, United States