Hong Kook Kim

According to our database1, Hong Kook Kim authored at least 120 papers between 1997 and 2019.

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

Timeline

Legend:

Book 
In proceedings 
Article 
PhD thesis 
Other 

Links

On csauthors.net:

Bibliography

2019
Non-linear Acoustic Echo Cancellation Based on Mel-Frequency Domain Volterra Filtering.
Proceedings of the IEEE International Conference on Consumer Electronics, 2019

Multi-Channel Audio Source Separation Using Azimuth-Frequency Analysis and Convolutional Neural Network.
Proceedings of the International Conference on Artificial Intelligence in Information and Communication, 2019

2018
Coordinate-based direction-of-arrival estimation method using distributed microphones.
Proceedings of the IEEE International Conference on Consumer Electronics, 2018

Single-channel speech dereverberation based on block-wise weighted prediction error and nonnegative matrix factorization.
Proceedings of the IEEE International Conference on Consumer Electronics, 2018

2017
A lossless compression method incorporating sensor fault detection for underwater acoustic sensor array.
IJDSN, 2017

Audio enhancement using local SNR-based sparse binary mask estimation and spectral imputation.
Digital Signal Processing, 2017

Low-Frequency Ultrasonic Communication for Speech Broadcasting in Public Transportation.
Proceedings of the Interspeech 2017, 2017

Application of low-frequency ultrasonic communication to audio marker for augmented reality.
Proceedings of the IEEE International Conference on Consumer Electronics, 2017

Speech emotion recognition based on multi-task learning using a convolutional neural network.
Proceedings of the 2017 Asia-Pacific Signal and Information Processing Association Annual Summit and Conference, 2017

2016
Noncoherent Low-Frequency Ultrasonic Communication System with Optimum Symbol Length.
IJDSN, 2016

Underwater acoustic sensor fault detection for passive sonar systems.
Proceedings of the First International Workshop on Sensing, 2016

Local Sparsity Based Online Dictionary Learning for Environment-Adaptive Speech Enhancement with Nonnegative Matrix Factorization.
Proceedings of the Interspeech 2016, 2016

Subband-based upmixing of stereo to 5.1-channel audio signals using deep neural networks.
Proceedings of the International Conference on Information and Communication Technology Convergence, 2016

A discriminative training method incorporating pronunciation variations for dysarthric automatic speech recognition.
Proceedings of the Asia-Pacific Signal and Information Processing Association Annual Summit and Conference, 2016

2015
Adaptive Speech Streaming Based on Speech Quality Estimation and Artificial Bandwidth Extension for Voice over Wireless Multimedia Sensor Networks.
IJDSN, 2015

Conversion of nearly monaural audio to 5.1-channel audio for portable multimedia devices.
Proceedings of the IEEE International Conference on Consumer Electronics, 2015

Two-stage lexicon optimization of G2P-converted pronunciation dictionary based on statistical acoustic confusability measure.
Proceedings of the Asia-Pacific Signal and Information Processing Association Annual Summit and Conference, 2015

Lexicon Optimization for WFST-Based Speech Recognition Using Acoustic Distance Based Confusability Measure and G2P Conversion.
Proceedings of the Natural Language Dialog Systems and Intelligent Assistants, 2015

2014
Multi-channel audio recording based on superdirective beamforming for portable multimedia recording devices.
IEEE Trans. Consumer Electronics, 2014

Direction-of-arrival based SNR estimation for dual-microphone speech enhancement.
IEEE/ACM Trans. Audio, Speech & Language Processing, 2014

A Packet Loss Concealment Technique Improving Quality of Service for Wideband Speech Coding in Wireless Sensor Networks.
IJDSN, 2014

Nonnegative Matrix Factorization Based Adaptive Noise Sensing over Wireless Sensor Networks.
IJDSN, 2014

Reducing Speech Noise for Patients with Dysarthria in Noisy Environments.
IEICE Transactions, 2014

Noise variance estimation based on dual-channel phase difference for speech enhancement.
Digital Signal Processing, 2014

Hybrid probabilistic adaptation mode controller for generalized sidelobe cancellers applied to multi-microphone speech enhancement.
Digital Signal Processing, 2014

Single-channel speech enhancement based on non-negative matrix factorization and online noise adaptation.
Proceedings of the INTERSPEECH 2014, 2014

Feasibility Study for Objective Measurement on Sound Localization Using Auditory Evoked Potential.
Proceedings of the 2014 Tenth International Conference on Intelligent Information Hiding and Multimedia Signal Processing, 2014

Audio restoration based on multi-band spectral subtraction and missing data imputation.
Proceedings of the IEEE International Conference on Consumer Electronics, 2014

2013
An MDCT-domain audio denoising method with a block switching scheme.
IEEE Trans. Consumer Electronics, 2013

Mechanical noise suppression based on non-negative matrix factorization and multi-band spectral subtraction for digital cameras.
IEEE Trans. Consumer Electronics, 2013

Ultrasonic Sensor-Based Personalized Multichannel Audio Rendering for Multiview Broadcasting Services.
IJDSN, 2013

Target-to-non-target directional ratio estimation based on dual-microphone phase differences for target-directional speech enhancement.
Proceedings of the INTERSPEECH 2013, 2013

Multi-band spectral subtraction based zoom-noise suppression for digital cameras.
Proceedings of the IEEE International Conference on Consumer Electronics, 2013

2012
A user voice reduction algorithm based on binaural signal separation for portable digital imaging devices.
IEEE Trans. Consumer Electronics, 2012

Dysarthric Speech Recognition Error Correction Using Weighted Finite State Transducers Based on Context-Dependent Pronunciation Variation.
Proceedings of the Computers Helping People with Special Needs, 2012

Adaptation mode control with residual noise estimation for beamformer-based multi-channel speech enhancement.
Proceedings of the 2012 IEEE International Conference on Acoustics, 2012

Non-negative Matrix Factorization Based Noise Reduction for Noise Robust Automatic Speech Recognition.
Proceedings of the Latent Variable Analysis and Signal Separation, 2012

2011
Probabilistic spectral gain modification applied to beamformer-based noise reduction in a car environment.
IEEE Trans. Consumer Electronics, 2011

A smart background music mixing algorithm for portable digital imaging devices.
IEEE Trans. Consumer Electronics, 2011

Sound source elevation using spectral notch filtering and directional band boosting in stereo loudspeaker reproduction.
IEEE Trans. Consumer Electronics, 2011

Burst Packet Loss Concealment Using Multiple Codebooks and Comfort Noise for CELP-Type Speech Coders in Wireless Sensor Networks.
Sensors, 2011

Adaptive Redundant Speech Transmission over Wireless Multimedia Sensor Networks Based on Estimation of Perceived Speech Quality.
Sensors, 2011

Phonetically Balanced Text Corpus Design Using a Similarity Measure for a Stereo Super-Wideband Speech Database.
IEICE Transactions, 2011

MDCT-Domain Packet Loss Concealment for Scalable Wideband Speech Coding.
Proceedings of the Ubiquitous Computing and Multimedia Applications, 2011

A Smart Error Protection Scheme Based on Estimation of Perceived Speech Quality for Portable Digital Speech Streaming Systems.
Proceedings of the Ubiquitous Computing and Multimedia Applications, 2011

Audio Effect for Highlighting Speaker's Voice Corrupted by Background Noise on Portable Digital Imaging Devices.
Proceedings of the Ubiquitous Computing and Multimedia Applications, 2011

High-Quality and Low-Complexity Real-Time Voice Changing with Seamless Switching for Digital Imaging Devices.
Proceedings of the Ubiquitous Computing and Multimedia Applications, 2011

Complexity Reduction of Virtual Reverberation Filtering Based on Index-Based Convolution for Resource-Constrained Devices.
Proceedings of the Ubiquitous Computing and Multimedia Applications, 2011

Hybrid probabilistic adaptation mode controller for generalized sidelobe canceller-based target-directional speech enhancement.
Proceedings of the IEEE International Conference on Acoustics, 2011

Artificial Bandwidth Extension of Narrowband Speech Signals for the Improvement of Perceptual Speech Communication Quality.
Proceedings of the Communication and Networking, 2011

Discrimination of Speech Activity and Impact Noise Using an Accelerometer and a Microphone in a Car Environment.
Proceedings of the Communication and Networking, 2011

Quality-Aware Loss-Robust Scalable Speech Streaming Based on Speech Quality Estimation.
Proceedings of the Communication and Networking, 2011

Crosstalk Cancellation for Spatial Sound Reproduction in Portable Devices with Stereo Loudspeakers.
Proceedings of the Communication and Networking, 2011

Perceptual Enhancement of Sound Field Reproduction in a Nearly Monaural Sensing System.
Proceedings of the Communication and Networking, 2011

2010
Entropy coding of compressed feature parameters for distributed speech recognition.
Speech Communication, 2010

Despeckling of medical ultrasound images using Daubechies complex wavelet transform.
Signal Processing, 2010

Acoustic Model Combination Incorporated With Mask-Based Multi-Channel Source Separation for Automatic Speech Recognition.
J. Sel. Topics Signal Processing, 2010

A Hybrid Acoustic and Pronunciation Model Adaptation Approach for Non-native Speech Recognition.
IEICE Transactions, 2010

Immersive modeling system (IMMS) for personal electronic products using a multi-modal interface.
Computer-Aided Design, 2010

An Integrated Approach of 3D Sound Rendering Techniques for Sound Externalization.
Proceedings of the Advances in Multimedia Information Processing - PCM 2010, 2010

SNR-based mask compensation for computational auditory scene analysis applied to speech recognition in a car environment.
Proceedings of the INTERSPEECH 2010, 2010

On the use of feature-space MLLR adaptation for non-native speech recognition.
Proceedings of the IEEE International Conference on Acoustics, 2010

Statistical Model-Based Voice Activity Detection Using Spatial Cues and Log Energy for Dual-Channel Noisy Speech Recognition.
Proceedings of the Communication and Networking, 2010

Design and Implementation of a Video-Zoom Driven Digital Audio-Zoom System for Portable Digital Imaging Devices.
Proceedings of the Signal Processing and Multimedia, 2010

A Packet Loss Concealment Algorithm Robust to Burst Packet Loss Using Multiple Codebooks and Comfort Noise for CELP-Type Speech Coders.
Proceedings of the Communication and Networking, 2010

Duration Model-Based Post-processing for the Performance Improvement of a Keyword Spotting System.
Proceedings of the Communication and Networking, 2010

Complexity Reduction of WSOLA-Based Time-Scale Modification Using Signal Period Estimation.
Proceedings of the Communication and Networking, 2010

3D Sound Techniques for Sound Source Elevation in a Loudspeaker Listening Environment.
Proceedings of the Communication and Networking, 2010

A Real-Time Audio Upmixing Method from Stereo to 7.1-Channel Audio.
Proceedings of the Communication and Networking, 2010

2009
Cepstrum-Domain Model Combination Based on Decomposition of Speech and Noise Using MMSE-LSA for ASR in Noisy Environments.
IEEE Trans. Audio, Speech & Language Processing, 2009

Bandwidth-Scalable Stereo Audio Coding Based on a Layered Structure.
IEICE Transactions, 2009

A media-specific FEC based on huffman coding for distributed speech recognition.
Proceedings of the INTERSPEECH 2009, 2009

Acoustic model combination to compensate for residual noise in multi-channel source separation.
Proceedings of the IEEE International Conference on Acoustics, 2009

Class-dependent and differential Huffman coding of compressed feature parameters for distributed speech recognition.
Proceedings of the IEEE International Conference on Acoustics, 2009

Upmixing Stereo Audio into 5.1 Channel Audio for Improving Audio Realism.
Proceedings of the Signal Processing, Image Processing and Pattern Recognition, 2009

MLLR/MAP adaptation using pronunciation variation for non-native speech recognition.
Proceedings of the 2009 IEEE Workshop on Automatic Speech Recognition & Understanding, 2009

2008
Cepstral domain interpretations of line spectral frequencies.
Signal Processing, 2008

HMM-Based Mask Estimation for a Speech Recognition Front-End Using Computational Auditory Scene Analysis.
IEICE Transactions, 2008

Gammatone-domain model combination for consonant recognition in noisy environments.
Proceedings of the INTERSPEECH 2008, 2008

Mask estimation incorporating time-frequency trajectories for a CASA-based ASR front-end.
Proceedings of the INTERSPEECH 2008, 2008

Acoustic and pronunciation model adaptation for context-independent and context-dependent pronunciation variability of non-native speech.
Proceedings of the IEEE International Conference on Acoustics, 2008

2007
Acoustic model adaptation based on pronunciation variability analysis for non-native speech recognition.
Speech Communication, 2007

A MFCC-Based CELP Speech Coder for Server-Based Speech Recognition in Network Environments.
IEICE Transactions, 2007

A Statistical Approach to Error Compensation in Spectral Quantization.
IEICE Transactions, 2007

Bandwidth Extension of a Narrowband Speech Coder for Music Streaming Services Over IP Networks.
Proceedings of the IEEE Workshop on Signal Processing Systems, 2007

Non-native pronunciation variation modeling using an indirect data driven method.
Proceedings of the IEEE Workshop on Automatic Speech Recognition & Understanding, 2007

2006
Bandwidth Extension of a Narrowband Speech Coder for Music Delivery over IP.
Proceedings of the Advances in Hybrid Information Technology, 2006

Acoustic Model Adaptation Based on Pronunciation Variability Analysis for Non-Native Speech Recognition.
Proceedings of the 2006 IEEE International Conference on Acoustics Speech and Signal Processing, 2006

A Highly Adaptive Acoustic Echo Cancellation Solution for VoIP Conferencing Systems.
Proceedings of the 2006 IEEE/ACS International Conference on Computer Systems and Applications (AICCSA 2006), 2006

2005
Procedural Constraints in the Extended RBAC and the Coloured Petri Net Modeling.
IEICE Transactions, 2005

A CELP coder using MFCC for server-based speech recognition in mobile.
Proceedings of the Signal and Image Processing (SIP 2005), 2005

A MFCC-based CELP speech coder for server-based speech recognition in network environments.
Proceedings of the INTERSPEECH 2005, 2005

Error Prediction in Spoken Dialog: From Signal-to-Noise Ratio to Semantic Confidence Scores.
Proceedings of the 2005 IEEE International Conference on Acoustics, 2005

2004
Harmonic Model Based Excitation Enhancement for Low-Bit-Rate Speech Coding.
IEICE Transactions, 2004

Compensation of Speech Coding Distortion for Wireless Speech Recognition.
IEICE Transactions, 2004

A Forward-Backward Voice Packet Loss Concealment Algorithm for Multimedia over IP Network Services.
Proceedings of the Advances in Multimedia Information Processing - PCM 2004, 5th Pacific Rim Conference on Multimedia, Tokyo, Japan, November 30, 2004

Robust speech recognition in client-server scenarios.
Proceedings of the INTERSPEECH 2004, 2004

Why speech recognizers make errors ? a robustness view.
Proceedings of the INTERSPEECH 2004, 2004

2003
Improving the transcoding capability of speech coders.
IEEE Trans. Multimedia, 2003

Cepstrum-domain acoustic feature compensation based on decomposition of speech and noise for ASR in noisy environments.
IEEE Trans. Speech and Audio Processing, 2003

2002
Performance improvement of a bitstream-based front-end for wireless speech recognition in adverse environments.
IEEE Trans. Speech and Audio Processing, 2002

An adaptive short-term postfilter based on pseudo-cepstral representation of line spectral frequencies.
Speech Communication, 2002

Algorithms for distributed speech recognition in a noisy automobile environment.
Proceedings of the 7th International Conference on Spoken Language Processing, ICSLP2002, 2002

Cepstrum-domain model combination based on decomposition of speech and noise for noisy speech recognition.
Proceedings of the IEEE International Conference on Acoustics, 2002

A phase generation method for speech reconstruction from spectral envelope and pitch intervals.
Proceedings of the IEEE International Conference on Acoustics, 2002

2001
A bitstream-based front-end for wireless speech recognition on IS-136 communications system.
IEEE Trans. Speech and Audio Processing, 2001

A new distortion measure for spectral quantization based on the LSF intermodel interlacing property.
Speech Communication, 2001

Robust speech recognition techniques applied to a speech in noise task.
Proceedings of the EUROSPEECH 2001 Scandinavia, 2001

Acoustic feature compensation based on decomposition of speech and noise for ASR in noisy environments.
Proceedings of the EUROSPEECH 2001 Scandinavia, 2001

Feature enhancement for a bitstream-based front-end in wireless speech recognition.
Proceedings of the IEEE International Conference on Acoustics, 2001

2000
On approximating line spectral frequencies to LPC cepstral coefficients.
IEEE Trans. Speech and Audio Processing, 2000

Speech recognition using quantized LSP parameters and their transformations in digital communication.
Speech Communication, 2000

Bitstream-based feature extraction for wireless speech recognition.
Proceedings of the IEEE International Conference on Acoustics, 2000

1999
Use of spectral autocorrelation in spectral envelope linear prediction for speech recognition.
IEEE Trans. Speech and Audio Processing, 1999

Interlacing properties of line spectrum pair frequencies.
IEEE Trans. Speech and Audio Processing, 1999

A 4 kbps adaptive fixed code-excited linear prediction speech coder.
Proceedings of the 1999 IEEE International Conference on Acoustics, 1999

LSP weighting functions based on spectral sensitivity and mel-frequency warping for speech recognition in digital communication.
Proceedings of the 1999 IEEE International Conference on Acoustics, 1999

1998
Adaptive encoding of fixed codebook in CELP coders.
Proceedings of the 1998 IEEE International Conference on Acoustics, 1998

1997
Joint estimation of pitch, band magnitudes, and v\UV decisions for MBE vocoder.
Proceedings of the Fifth European Conference on Speech Communication and Technology, 1997

A 4 kbit/s renewal code excited linear prediction speech coder.
Proceedings of the 1997 IEEE International Conference on Acoustics, 1997


  Loading...