Dae Hee Youn

According to our database1, Dae Hee Youn authored at least 105 papers between 1980 and 2015.

Collaborative distances:
  • Dijkstra number2 of five.
  • Erdős number3 of four.

Timeline

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Bibliography

2015
Auditory Distance Rendering Based on ICPD Control for Stereophonic 3D Audio System.
IEEE Signal Process. Lett., 2015

Scalable Multiband Binaural Renderer for MPEG-H 3D Audio.
IEEE J. Sel. Top. Signal Process., 2015

2014
Stereo upmix-based binaural auralization for mobile devices.
IEEE Trans. Consumer Electron., 2014

Numerical Synthesis of an Optimal Low-Sidelobe Beam Pattern for a Microphone Array.
IEEE Signal Process. Lett., 2014

On Improving the Performance of a Speech Model-Based Blind Reverberation Time Estimation in Noisy Environments.
IEICE Trans. Fundam. Electron. Commun. Comput. Sci., 2014

Approximated Virtual Source Imaging System for a Pair of Closely Spaced Loudspeakers.
IEICE Trans. Inf. Syst., 2014

Binaural noise suppression based on an unbiased estimator of target PSD in complex noise environments.
Proceedings of the IEEE International Conference on Acoustics, 2014

2013
Virtual bass system based on a multiband harmonic generation.
Proceedings of the IEEE International Conference on Consumer Electronics, 2013

2012
LP/WLP Hybrid Scheme for Quality Improvement of TCX Coders Operating at Low Bit Rates.
IEICE Trans. Inf. Syst., 2012

Estimation and quantization of ICC-dependent phase parameters for parametric stereo audio coding.
EURASIP J. Audio Speech Music. Process., 2012

Acoustic depth rendering for 3D multimedia applications.
Proceedings of the IEEE International Conference on Consumer Electronics, 2012

A variable step-size filtered-x gradient adaptive lattice algorithm for active noise control.
Proceedings of the 2012 IEEE International Conference on Acoustics, 2012

Blind depth estimation based on primary-to-ambient energy ratio for 3-D acoustic depth rendering.
Proceedings of the Asia-Pacific Signal and Information Processing Association Annual Summit and Conference, 2012

2011
Virtual Sound Rendering in a Stereophonic Loudspeaker Setup.
IEEE Trans. Speech Audio Process., 2011

Efficient Windowing Scheme for MDCT-Based TCX in AMR-WB+.
IEICE Trans. Inf. Syst., 2011

Improved phase parameter analysis and synthesis for parametric stereo audio coding.
Proceedings of the IEEE International Conference on Acoustics, 2011

Virtual source panning using multiple-wise vector base in the multispeaker stereo format.
Proceedings of the 19th European Signal Processing Conference, 2011

2010
A Robust Room Inverse Filtering Algorithm for Speech Dereverberation Based on a Kurtosis Maximization.
IEICE Trans. Inf. Syst., 2010

Enhancement of principal to ambient energy ratio for PCA-based parametric audio coding.
Proceedings of the IEEE International Conference on Acoustics, 2010

2009
Efficient FFT Algorithm for Psychoacoustic Model of the MPEG-4 AAC.
IEICE Trans. Inf. Syst., 2009

A GMM-Based Feature Selection Algorithm for Multi-Class Classification.
IEICE Trans. Inf. Syst., 2009

2008
Effective bass enhancement using second-order adaptive notch filter.
IEEE Trans. Consumer Electron., 2008

Design of Time-Varying Reverberators for Low Memory Applications.
IEICE Trans. Inf. Syst., 2008

2007
Speech quality estimation using packet loss effects in CELP-type speech coders.
Proceedings of the 8th Annual Conference of the International Speech Communication Association, 2007

2006
On the Use of Voting Methods for Speaker Identification Based on Various Resolution Filterbanks.
Proceedings of the 2006 IEEE International Conference on Acoustics Speech and Signal Processing, 2006

Audio Transcoding Algorithm For Mobile Multimedia Application.
Proceedings of the 2006 IEEE International Conference on Acoustics Speech and Signal Processing, 2006

2005
Improving the Performance of the Minimum Statistics Noise Estimator for Single Channel Speech Enhancement.
IEICE Trans. Fundam. Electron. Commun. Comput. Sci., 2005

Adaptive Microphone Array System with Two-Stage Adaptation Mode Controller.
IEICE Trans. Fundam. Electron. Commun. Comput. Sci., 2005

A noise-robust pitch synchronous feature extraction algorithm for speaker recognition systems.
Proceedings of the 9th European Conference on Speech Communication and Technology, 2005

2004
An efficient transcoding algorithm for G.723.1 and G.729A speech coders: interoperability between mobile and IP network.
Speech Commun., 2004

Temporal normalization techniques for transform-type speech coding and application to split-band wideband coders.
Proceedings of the 8th International Conference on Spoken Language Processing, 2004

Performance analysis of transcoding algorithms in packet-loss environments.
Proceedings of the 8th International Conference on Spoken Language Processing, 2004

A pitch synchronous feature extraction method for speaker recognition.
Proceedings of the 2004 IEEE International Conference on Acoustics, 2004

Improvement issues on transcoding algorithms: for the flexible usage to the various pairs of speech codec.
Proceedings of the 2004 IEEE International Conference on Acoustics, 2004

2003
An efficient DMT modem for the G.LITE ADSL transceiver.
IEEE Trans. Very Large Scale Integr. Syst., 2003

Performance Comparison of Single and Multi-Stage Algebraic Codebooks.
IEICE Trans. Fundam. Electron. Commun. Comput. Sci., 2003

Transcoding algorithm for g.723.1 and AMR speech coders: for interoperability between voIP and mobile networks.
Proceedings of the 8th European Conference on Speech Communication and Technology, EUROSPEECH 2003, 2003

A new training symbol structure to enhance the performance of channel estimation for MIMO-OFDM systems.
Proceedings of the 2003 IEEE International Conference on Acoustics, 2003

A packet loss concealment algorithm based on time-scale modification for CELP-type speech coders.
Proceedings of the 2003 IEEE International Conference on Acoustics, 2003

A cascaded algebraic codebook structure to improve the performance of speech coder.
Proceedings of the 2003 IEEE International Conference on Acoustics, 2003

2002
Software optimization of the MPEG-audio decoder using a 32-bit MCU RISC processor.
IEEE Trans. Consumer Electron., 2002

Design and VLSI implementation of a digital audio-specific DSP core for MP3/AAC.
IEEE Trans. Consumer Electron., 2002

A residual echo cancellation scheme for hands-free telephony.
IEEE Signal Process. Lett., 2002

Acoustic interference cancellation for hands-free terminals.
Proceedings of the 14th International Conference on Digital Signal Processing, 2002

A new bandwidth scalable wideband speech/audio coder.
Proceedings of the IEEE International Conference on Acoustics, 2002

Design optimization of a dual MP3/AAC decoder.
Proceedings of the IEEE International Conference on Acoustics, 2002

2001
Low power MPEG/audio encoders using simplified psychoacoustic model and fast bit allocation.
IEEE Trans. Consumer Electron., 2001

An architecture and implementation of MPEG audio layer III decoder using dual-core DSP.
IEEE Trans. Consumer Electron., 2001

Design optimization of MPEG-2 AAC decoder.
IEEE Trans. Consumer Electron., 2001

A delayless subband active noise control system for wideband noise control.
IEEE Trans. Speech Audio Process., 2001

LMS based forward-link beamforming using reverse-link channel estimator in FDD/CDMA system.
Proceedings of the 54th IEEE Vehicular Technology Conference, 2001

An efficient transcoding algorithm for G.723.1 and EVRC speech coders.
Proceedings of the 54th IEEE Vehicular Technology Conference, 2001

High quality MPEG-audio layer III algorithm for a 16-bit DSP.
Proceedings of the 2001 International Symposium on Circuits and Systems, 2001

An efficient transcoding algorithm for g.723.1 and g.729a speech coders.
Proceedings of the EUROSPEECH 2001 Scandinavia, 2001

Speech quality measure for voIP using wavelet based bark coherence function.
Proceedings of the EUROSPEECH 2001 Scandinavia, 2001

Efficient implementation of ITU-t g.723.1 speech coder for multichannel voice transmission and storage.
Proceedings of the EUROSPEECH 2001 Scandinavia, 2001

Fast harmonic estimation using a low resolution pitch for low bit rate harmonic coding.
Proceedings of the EUROSPEECH 2001 Scandinavia, 2001

Transparent And Robust Audio Watermarking With A New Echo Embedding Technique.
Proceedings of the 2001 IEEE International Conference on Multimedia and Expo, 2001

An area-efficient interpolation filter using block structure.
Proceedings of the 2001 8th IEEE International Conference on Electronics, 2001

On integrating acoustic echo and noise cancellation systems for hands-free telephony.
Proceedings of the IEEE International Conference on Acoustics, 2001

New echo embedding technique for robust and imperceptible audio watermarking.
Proceedings of the IEEE International Conference on Acoustics, 2001

Design optimization of main-profile MPEG-2 AAC decoder.
Proceedings of the IEEE International Conference on Acoustics, 2001

2000
Adaptive processing technique for enhanced CFAR detecting performance in active sonar systems.
IEEE Trans. Aerosp. Electron. Syst., 2000

Evaluation of wavelet filters for speech recognition.
Proceedings of the IEEE International Conference on Systems, 2000

A bark coherence function for perceived speech quality estimation.
Proceedings of the Sixth International Conference on Spoken Language Processing, 2000

Speaker dependent emotion recognition using speech signals.
Proceedings of the Sixth International Conference on Spoken Language Processing, 2000

An efficient codebook search algorithm for EVRC.
Proceedings of the Sixth International Conference on Spoken Language Processing, 2000

Efficient harmonic-CELP based hybrid coding of speech at low bit rates.
Proceedings of the Sixth International Conference on Spoken Language Processing, 2000

A new adaptive algorithm for stereophonic acoustic echo canceller.
Proceedings of the IEEE International Conference on Acoustics, 2000

1999
Image enhancement based on signal subspace approach.
IEEE Trans. Image Process., 1999

On compensating nonlinear distortions of an OFDM system using an efficient adaptive predistorter.
IEEE Trans. Commun., 1999

On using formants to improve SCHMM speaker adaptation.
IEEE Trans. Speech Audio Process., 1999

New implementation techniques of a real-time MPEG-2 audio encoding system.
Proceedings of the 1999 IEEE International Conference on Acoustics, 1999

1998
An adaptive nonlinear prefilter for compensation of distortion in nonlinear systems.
IEEE Trans. Signal Process., 1998

Adaptive precompensation of Wiener systems.
IEEE Trans. Signal Process., 1998

Duration modeling using cumulative duration probability and speaking rate compensation.
Proceedings of the 5th International Conference on Spoken Language Processing, Incorporating The 7th Australian International Speech Science and Technology Conference, Sydney Convention Centre, Sydney, Australia, 30th November, 1998

An optimum space-time MTI processor for airborne radar.
Proceedings of the 1998 IEEE International Conference on Acoustics, 1998

4-way superscalar DSP processor for audio codec applications.
Proceedings of the 1998 IEEE International Conference on Acoustics, 1998

Stereophonic acoustic echo canceller using single adaptive filter per channel.
Proceedings of the 9th European Signal Processing Conference, 1998

1997
Improved regular pulse VSELP coding of speech at low bit-rates.
Proceedings of the Fifth European Conference on Speech Communication and Technology, 1997

A 3 channel digital CVSD bit-rate conversion system using a general purpose DSP.
Proceedings of the Fifth European Conference on Speech Communication and Technology, 1997

Subband active noise control algorithm based on a delayless subband adaptive filter architecture.
Proceedings of the 1997 IEEE International Conference on Acoustics, 1997

1996
A codebook adaptation algorithm for SCHMM using formant distribution.
Proceedings of the 4th International Conference on Spoken Language Processing, 1996

A new voice transformation method based on both linear and nonlinear prediction analysis.
Proceedings of the 4th International Conference on Spoken Language Processing, 1996

Image enhancement based on signal subspace approach.
Proceedings of the Proceedings 1996 International Conference on Image Processing, 1996

A fast VSELP speech coder based on mutually orthonormal regular pulse vectors.
Proceedings of the 1996 IEEE International Conference on Acoustics, 1996

1995
A Lattice/Transversal Joint (LTJ) Structure for an Acoustic Echo Canceller.
Proceedings of the 1995 IEEE International Symposium on Circuits and Systems, ISCAS 1995, Seattle, Washington, USA, April 30, 1995

Voice personality transformation using an orthogonal vector space conversion.
Proceedings of the Fourth European Conference on Speech Communication and Technology, 1995

A low bit-rate speech coder using the perceptual properties of the human ear.
Proceedings of the Fourth European Conference on Speech Communication and Technology, 1995

1994
HMM with global path constraint in Viterbi decoding for isolated word recognition.
Proceedings of ICASSP '94: IEEE International Conference on Acoustics, 1994

1993
Adaptive implementation of optimum MTI processor.
Proceedings of the IEEE International Conference on Acoustics, 1993

1992
Adaptive multichannel lattice-escalator filter structure: an application to generalized sidelobe canceler.
IEEE Trans. Signal Process., 1992

On detecting the presence of fetal R-wave using the moving averaged magnitude difference algorithm.
IEEE Trans. Biomed. Eng., 1992

1990
Adaptive multichannel digital filter with lattice-escalator hybrid structure.
Proceedings of the 1990 International Conference on Acoustics, 1990

1987
A unified approach to nonparametric spectrum estimation algorithms.
IEEE Trans. Acoust. Speech Signal Process., 1987

1986
Multichannel lattice filter for an adaptive array processor with linear constraints.
Proceedings of the IEEE International Conference on Acoustics, 1986

1985
Analysis of the short-time unbiased spectrum estimation algorithm.
IEEE Trans. Acoust. Speech Signal Process., 1985

An efficient algorithm for lattice filter/Predictor.
Proceedings of the IEEE International Conference on Acoustics, 1985

1984
On realizations and related algorithms for adaptive linear phase filtering.
Proceedings of the IEEE International Conference on Acoustics, 1984

Adaptive realization of the phase transform for time delay estimation.
Proceedings of the IEEE International Conference on Acoustics, 1984

1983
Comparison of two adaptive methods for time delay estimation.
Proceedings of the IEEE International Conference on Acoustics, 1983

1982
Estimation of magnitude-squared coherence function: An adaptive approach.
Proceedings of the IEEE International Conference on Acoustics, 1982

On the scot and roth algorithms for time delay estimation.
Proceedings of the IEEE International Conference on Acoustics, 1982

1981
A method for generating a class of time-delayed signals.
Proceedings of the IEEE International Conference on Acoustics, 1981

1980
Frequency domain considerations of an adaptive escalator predictor.
Proceedings of the IEEE International Conference on Acoustics, 1980


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