Hong-Goo Kang

Orcid: 0000-0002-6554-0783

According to our database1, Hong-Goo Kang authored at least 170 papers between 1995 and 2024.

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

Timeline

Legend:

Book 
In proceedings 
Article 
PhD thesis 
Dataset
Other 

Links

On csauthors.net:

Bibliography

2024
Disentangled Representations in Local-Global Contexts for Arabic Dialect Identification.
IEEE ACM Trans. Audio Speech Lang. Process., 2024

SC-ERM: Speaker-Centric Learning for Speech Emotion Recognition.
Proceedings of the International Conference on Electronics, Information, and Communication, 2024

On the Disentanglement and Robustness of Self-Supervised Speech Representations.
Proceedings of the International Conference on Electronics, Information, and Communication, 2024

Contextual Learning for Missing Speech Automatic Speech Recognition.
Proceedings of the International Conference on Electronics, Information, and Communication, 2024

2023
Real-time neural speech enhancement based on temporal refinement network and channel-wise gating methods.
Digit. Signal Process., March, 2023

SC-CNN: Effective Speaker Conditioning Method for Zero-Shot Multi-Speaker Text-to-Speech Systems.
IEEE Signal Process. Lett., 2023

C2C: Cough to COVID-19 Detection in BHI 2023 Data Challenge.
CoRR, 2023

Pruning Self-Attention for Zero-Shot Multi-Speaker Text-to-Speech.
CoRR, 2023

Feature Normalization for Fine-tuning Self-Supervised Models in Speech Enhancement.
CoRR, 2023

HD-DEMUCS: General Speech Restoration with Heterogeneous Decoders.
CoRR, 2023

A High-Rate Extension to Soundstream.
Proceedings of the IEEE Workshop on Applications of Signal Processing to Audio and Acoustics, 2023

HappyQuokka System for ICASSP 2023 Auditory EEG Challenge.
Proceedings of the IEEE International Conference on Acoustics, 2023

End-to-End Neural Audio Coding in the MDCT Domain.
Proceedings of the IEEE International Conference on Acoustics, 2023

Style Modeling for Multi-Speaker Articulation-to-Speech.
Proceedings of the IEEE International Conference on Acoustics, 2023

Progressive Multi-Stage Neural Audio Codec with Psychoacoustic Loss and Discriminator.
Proceedings of the IEEE International Conference on Acoustics, 2023

Facetron: A Multi-Speaker Face-to-Speech Model Based on Cross-Modal Latent Representations.
Proceedings of the 31st European Signal Processing Conference, 2023

BrainTalker: Low-Resource Brain-to-Speech Synthesis with Transfer Learning using Wav2Vec 2.0.
Proceedings of the IEEE EMBS International Conference on Biomedical and Health Informatics, 2023

PDF-NET: Pitch-adaptive Dynamic Filter Network for Intra-gender Speaker Verification.
Proceedings of the Asia Pacific Signal and Information Processing Association Annual Summit and Conference, 2023

Consideration of Varying Training Lengths for Short-Duration Speaker Verification.
Proceedings of the Asia Pacific Signal and Information Processing Association Annual Summit and Conference, 2023

2022
Two-Stage Refinement of Magnitude and Complex Spectra for Real-Time Speech Enhancement.
IEEE Signal Process. Lett., 2022

ReCAB-VAE: Gumbel-Softmax Variational Inference Based on Analytic Divergence.
CoRR, 2022

AILTTS: Adversarial Learning of Intermediate Acoustic Feature for End-to-End Lightweight Text-to-Speech.
CoRR, 2022

SASV Challenge 2022: A Spoofing Aware Speaker Verification Challenge Evaluation Plan.
CoRR, 2022

Length-Normalized Representation Learning for Speech Signals.
IEEE Access, 2022

Learning Audio-Text Agreement for Open-vocabulary Keyword Spotting.
Proceedings of the Interspeech 2022, 2022

FluentTTS: Text-dependent Fine-grained Style Control for Multi-style TTS.
Proceedings of the Interspeech 2022, 2022

Light-Weight Speaker Verification with Global Context Information.
Proceedings of the Interspeech 2022, 2022

Adversarial Audio Synthesis Using a Harmonic-Percussive Discriminator.
Proceedings of the IEEE International Conference on Acoustics, 2022

Progressive Multi-Stage Neural Audio Coding with Guided References.
Proceedings of the IEEE International Conference on Acoustics, 2022

Phase Continuity: Learning Derivatives of Phase Spectrum for Speech Enhancement.
Proceedings of the IEEE International Conference on Acoustics, 2022

2021
Facetron: Multi-speaker Face-to-Speech Model based on Cross-modal Latent Representations.
CoRR, 2021

LiteTTS: A Lightweight Mel-Spectrogram-Free Text-to-Wave Synthesizer Based on Generative Adversarial Networks.
Proceedings of the Interspeech 2021, 22nd Annual Conference of the International Speech Communication Association, Brno, Czechia, 30 August, 2021

Self-supervised Complex Network for Machine Sound Anomaly Detection.
Proceedings of the 29th European Signal Processing Conference, 2021

A Fast and Lightweight Text-To-Speech Model with Spectrum and Waveform Alignment Algorithms.
Proceedings of the 29th European Signal Processing Conference, 2021

Disentangled Representations for Arabic Dialect Identification based on Supervised Clustering with Triplet Loss.
Proceedings of the 29th European Signal Processing Conference, 2021

Looking Into Your Speech: Learning Cross-Modal Affinity for Audio-Visual Speech Separation.
Proceedings of the IEEE Conference on Computer Vision and Pattern Recognition, 2021

Stacked U-Net with High-Level Feature Transfer for Parameter Efficient Speech Enhancement.
Proceedings of the Asia-Pacific Signal and Information Processing Association Annual Summit and Conference, 2021

2020
Perfect Match: Self-Supervised Embeddings for Cross-Modal Retrieval.
IEEE J. Sel. Top. Signal Process., 2020

Effective Emotion Transplantation in an End-to-End Text-to-Speech System.
IEEE Access, 2020

Speaker-Adaptive Neural Vocoders for Parametric Speech Synthesis Systems.
Proceedings of the 22nd IEEE International Workshop on Multimedia Signal Processing, 2020

Intra-Class Variation Reduction of Speaker Representation in Disentanglement Framework.
Proceedings of the Interspeech 2020, 2020

A Cross-Channel Attention-Based Wave-U-Net for Multi-Channel Speech Enhancement.
Proceedings of the Interspeech 2020, 2020

MIRNet: Learning Multiple Identities Representations in Overlapped Speech.
Proceedings of the Interspeech 2020, 2020

Seeing Voices and Hearing Voices: Learning Discriminative Embeddings Using Cross-Modal Self-Supervision.
Proceedings of the Interspeech 2020, 2020

FaceFilter: Audio-Visual Speech Separation Using Still Images.
Proceedings of the Interspeech 2020, 2020

Emotional Speech Synthesis with Rich and Granularized Control.
Proceedings of the 2020 IEEE International Conference on Acoustics, 2020

Improving LPCNET-Based Text-to-Speech with Linear Prediction-Structured Mixture Density Network.
Proceedings of the 2020 IEEE International Conference on Acoustics, 2020

Speaker-invariant Psychological Stress Detection Using Attention-based Network.
Proceedings of the Asia-Pacific Signal and Information Processing Association Annual Summit and Conference, 2020

ExcitGlow: Improving a WaveGlow-based Neural Vocoder with Linear Prediction Analysis.
Proceedings of the Asia-Pacific Signal and Information Processing Association Annual Summit and Conference, 2020

LP-WaveNet: Linear Prediction-based WaveNet Speech Synthesis.
Proceedings of the Asia-Pacific Signal and Information Processing Association Annual Summit and Conference, 2020

A Study on Conditional Features for a Flow-based Neural Vocoder.
Proceedings of the 54th Asilomar Conference on Signals, Systems, and Computers, 2020

2019
A Joint Learning Algorithm for Complex-Valued T-F Masks in Deep Learning-Based Single-Channel Speech Enhancement Systems.
IEEE ACM Trans. Audio Speech Lang. Process., 2019

An Effective Style Token Weight Control Technique for End-to-End Emotional Speech Synthesis.
IEEE Signal Process. Lett., 2019

Dry Electrode-Based Body Fat Estimation System with Anthropometric Data for Use in a Wearable Device.
Sensors, 2019

Effective parameter estimation methods for an ExcitNet model in generative text-to-speech systems.
CoRR, 2019

Orthonormal Embedding-based Deep Clustering for Single-channel Speech Separation.
CoRR, 2019

Parameter Enhancement for MELP Speech Codec in Noisy Communication Environment.
Proceedings of the Interspeech 2019, 2019

Gradient-based Active Learning Query Strategy for End-to-end Speech Recognition.
Proceedings of the IEEE International Conference on Acoustics, 2019

Perfect Match: Improved Cross-modal Embeddings for Audio-visual Synchronisation.
Proceedings of the IEEE International Conference on Acoustics, 2019

ExcitNet Vocoder: A Neural Excitation Model for Parametric Speech Synthesis Systems.
Proceedings of the 27th European Signal Processing Conference, 2019

A Study on Acoustic Parameter Selection Strategies to Improve Deep Learning-Based Speech Synthesis.
Proceedings of the 2019 Asia-Pacific Signal and Information Processing Association Annual Summit and Conference, 2019

2018
SVD-Based Adaptive QIM Watermarking on Stereo Audio Signals.
IEEE Trans. Multim., 2018

Phase-Sensitive Joint Learning Algorithms for Deep Learning-Based Speech Enhancement.
IEEE Signal Process. Lett., 2018

LP-WaveNet: Linear Prediction-based WaveNet Speech Synthesis.
CoRR, 2018

Speaker-adaptive neural vocoders for statistical parametric speech synthesis systems.
CoRR, 2018

Session details: Oral Session 2.
Proceedings of the 2018 Workshop on Audio-Visual Scene Understanding for Immersive Multimedia, 2018

AVSU: Workshop on Audio-Visual Scene Understanding for Immersive Multimedia.
Proceedings of the 2018 ACM Multimedia Conference on Multimedia Conference, 2018

A Deep Learning-based Stress Detection Algorithm with Speech Signal.
Proceedings of the 2018 Workshop on Audio-Visual Scene Understanding for Immersive Multimedia, 2018

A Unified Framework for the Generation of Glottal Signals in Deep Learning-based Parametric Speech Synthesis Systems.
Proceedings of the Interspeech 2018, 2018

Dnn-Based Wireless Positioning in an Outdoor Environment.
Proceedings of the 2018 IEEE International Conference on Acoustics, 2018

Modeling-By-Generation-Structured Noise Compensation Algorithm for Glottal Vocoding Speech Synthesis System.
Proceedings of the 2018 IEEE International Conference on Acoustics, 2018

Two electrode based healthcare device for continuously monitoring ECG and BIA signals.
Proceedings of the 2018 IEEE EMBS International Conference on Biomedical & Health Informatics, 2018

2017
Effective Spectral and Excitation Modeling Techniques for LSTM-RNN-Based Speech Synthesis Systems.
IEEE ACM Trans. Audio Speech Lang. Process., 2017

Deep bi-directional long short-term memory based speech enhancement for wind noise reduction.
Proceedings of the Hands-free Speech Communications and Microphone Arrays, 2017

A study on search grid points for data-driven 3-D beamsteering.
Proceedings of the Hands-free Speech Communications and Microphone Arrays, 2017

Continuous bladder volume monitoring system for wearable applications.
Proceedings of the 2017 39th Annual International Conference of the IEEE Engineering in Medicine and Biology Society (EMBC), 2017

Perceptual quality and modeling accuracy of excitation parameters in DLSTM-based speech synthesis systems.
Proceedings of the 2017 IEEE Automatic Speech Recognition and Understanding Workshop, 2017

2016
On pre-filtering strategies for the GCC-PHAT algorithm.
Proceedings of the IEEE International Workshop on Acoustic Signal Enhancement, 2016

Improved Time-Frequency Trajectory Excitation Vocoder for DNN-Based Speech Synthesis.
Proceedings of the Interspeech 2016, 2016

Parametric-based non-intrusive speech quality assessment by deep neural network.
Proceedings of the 2016 IEEE International Conference on Digital Signal Processing, 2016

A pitch-synchronous speech analysis and synthesis method for DNN-SPSS system.
Proceedings of the 2016 IEEE International Conference on Digital Signal Processing, 2016

Multi-class learning algorithm for deep neural network-based statistical parametric speech synthesis.
Proceedings of the 24th European Signal Processing Conference, 2016

Efficient deep neural networks for speech synthesis using bottleneck features.
Proceedings of the Asia-Pacific Signal and Information Processing Association Annual Summit and Conference, 2016

2015
A Priori SNR Estimation Using Air- and Bone-Conduction Microphones.
IEEE ACM Trans. Audio Speech Lang. Process., 2015

Deep neural network-based statistical parametric speech synthesis system using improved time-frequency trajectory excitation model.
Proceedings of the INTERSPEECH 2015, 2015

Systematic integration of acoustic echo canceller and noise reduction modules for voice communication systems.
Proceedings of the INTERSPEECH 2015, 2015

Improved time-frequency trajectory excitation modeling for a statistical parametric speech synthesis system.
Proceedings of the 2015 IEEE International Conference on Acoustics, 2015

Coherent channel based subband multichannel dereverberation.
Proceedings of the 2015 IEEE International Conference on Acoustics, 2015

Detection of fiducial points in ECG waves using iteration based adaptive thresholds.
Proceedings of the 37th Annual International Conference of the IEEE Engineering in Medicine and Biology Society, 2015

A constrained two-layer compression technique for ECG waves.
Proceedings of the 37th Annual International Conference of the IEEE Engineering in Medicine and Biology Society, 2015

Robust formant features for speaker verification in the lombard effect.
Proceedings of the Asia-Pacific Signal and Information Processing Association Annual Summit and Conference, 2015

2014
Online Speech Dereverberation Algorithm Based on Adaptive Multichannel Linear Prediction.
IEEE ACM Trans. Audio Speech Lang. Process., 2014

An Efficient Multichannel Linear Prediction-Based Blind Equalization Algorithm in Near Common Zeros Condition.
IEEE Signal Process. Lett., 2014

Knowledge-Based Manner Class Segmentation Based on the Acoustic Event and Landmark Detection Algorithm.
IEICE Trans. Inf. Syst., 2014

New aliasing cancelation algorithm for the transition between non-aliased and TDAC-based coding modes.
EURASIP J. Audio Speech Music. Process., 2014

Fixed-point implementation of MPEG-D unified speech and audio coding decoder.
Proceedings of the 19th International Conference on Digital Signal Processing, 2014

A maximum a Posterior-based reconstruction approach to speech bandwidth expansion in noise.
Proceedings of the IEEE International Conference on Acoustics, 2014

Factored adaptation of speaker and environment using orthogonal subspace transforms.
Proceedings of the IEEE International Conference on Acoustics, 2014

Detecting pathological speech using contour modeling of harmonic-to-noise ratio.
Proceedings of the IEEE International Conference on Acoustics, 2014

Mean normalization of power function based cepstral coefficients for robust speech recognition in noisy environment.
Proceedings of the IEEE International Conference on Acoustics, 2014

2013
Refinement of Landmark Detection and Extraction of Articulator-Free Features for Knowledge-Based Speech Recognition.
IEICE Trans. Inf. Syst., 2013

A source-filter based adaptive harmonic model and its application to speech prosody modification.
Proceedings of the INTERSPEECH 2013, 2013

Enhancement of spectral clarity for HMM-based text-to-speech systems.
Proceedings of the IEEE International Conference on Acoustics, 2013

Adaptive multichannel linear prediction based dereverberation in time-varying room environments.
Proceedings of the 21st European Signal Processing Conference, 2013

Adaptation of HMM dynamic parameters in reverberant environment.
Proceedings of the 21st European Signal Processing Conference, 2013

Vector Taylor series based HMM adaptation for generalized cepstrum in noisy environment.
Proceedings of the 2013 IEEE Workshop on Automatic Speech Recognition and Understanding, 2013

Speech enhancement for pathological voice using time-frequency trajectory excitation modeling.
Proceedings of the Asia-Pacific Signal and Information Processing Association Annual Summit and Conference, 2013

Detecting pathological speech using local and global characteristics of harmonic-to-noise ratio.
Proceedings of the Asia-Pacific Signal and Information Processing Association Annual Summit and Conference, 2013

2012
Waveform Interpolation-Based Speech Analysis/Synthesis for HMM-Based TTS Systems.
IEEE Signal Process. Lett., 2012

Two-stage source tracking method using a multiple linear regression model in the expanded phase domain.
EURASIP J. Adv. Signal Process., 2012

Survey of Speech Enhancement Supported by a Bone Conduction Microphone.
Proceedings of the 10th ITG Conference on Speech Communication, 2012

2011
An Interactive 3-D Audio System With Loudspeakers.
IEEE Trans. Multim., 2011

Robust Session Variability Compensation for SVM Speaker Verification.
IEEE Trans. Speech Audio Process., 2011

A Two-Channel Noise Estimator for Speech Enhancement in a Highly Nonstationary Environment.
IEEE ACM Trans. Audio Speech Lang. Process., 2011

Estimating redundancy information of selected features in multi-dimensional pattern classification.
Pattern Recognit. Lett., 2011

Transient noise reduction in speech signal with a modified long-term predictor.
EURASIP J. Adv. Signal Process., 2011

Classification of Fricatives Using Feature Extrapolation of Acoustic-Phonetic Features in Telephone Speech.
Proceedings of the INTERSPEECH 2011, 2011

Enhanced long-term predictor for Unified Speech and Audio Coding.
Proceedings of the IEEE International Conference on Acoustics, 2011

Signal and feature domain enhancement approaches for robust speech recognition.
Proceedings of the 8th International Conference on Information, 2011

2010
Selecting Feature Frames for Automatic Speaker Recognition Using Mutual Information.
IEEE Trans. Speech Audio Process., 2010

On the Importance of Transition Regions for Automatic Speaker Recognition.
IEICE Trans. Inf. Syst., 2010

Harmonic Enhancement in Low Bitrate Audio Coding Using an Efficient Long-Term Predictor.
EURASIP J. Adv. Signal Process., 2010

Enhancing loudspeaker-based 3D audio with room modeling.
Proceedings of the 2010 IEEE International Workshop on Multimedia Signal Processing, 2010

A variable frame length and rate algorithm based on the spectral kurtosis measure for speaker verification.
Proceedings of the INTERSPEECH 2010, 2010

Personal 3D audio system with loudspeakers.
Proceedings of the 2010 IEEE International Conference on Multimedia and Expo, 2010

Binaural loudness based speech reinforcement with a closed-form solution.
Proceedings of the IEEE International Conference on Acoustics, 2010

2009
On the Study of Noise Allocation for Speech Signal in Low Bit-Rate Audio Coding.
IEEE Signal Process. Lett., 2009

Normalized minimum-redundancy and maximum-relevancy based feature selection for speaker verification systems.
Proceedings of the IEEE International Conference on Acoustics, 2009

A robust time difference of arrival estimator in reverberant environments.
Proceedings of the 17th European Signal Processing Conference, 2009

2008
Speech Bandwidth Extension Using Temporal Envelope Modeling.
IEEE Signal Process. Lett., 2008

Designing a unified speech/audio codec by adopting a single channel harmonic source separation module.
Proceedings of the IEEE International Conference on Acoustics, 2008

2007
Applying a Speaker-Dependent Speech Compression Technique to Concatenative TTS Synthesizers.
IEEE Trans. Speech Audio Process., 2007

Speech quality estimation using packet loss effects in CELP-type speech coders.
Proceedings of the INTERSPEECH 2007, 2007

A Soft-Decision Adaptation Mode Controller for an Efficient Frequency-Domain Generalized Sidelobe Canceller.
Proceedings of the IEEE International Conference on Acoustics, 2007

2006
Performance analysis of various single channel speech enhancement algorithms for automatic speech recognition.
Proceedings of the INTERSPEECH 2006, 2006

An efficient segment-based speech compression technique for hand-held TTS systems.
Proceedings of the INTERSPEECH 2006, 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

2005
A fast adaptive-codebook search algorithm for G.723.1 speech coder.
IEEE Signal Process. Lett., 2005

An information-theoretic perspective on feature selection in speaker recognition.
IEEE Signal Process. Lett., 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 INTERSPEECH 2005, 2005

An improved estimation of a priori speech absence probability for speech enhancement : in perspective of speech perception.
Proceedings of the 2005 IEEE International Conference on Acoustics, 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 INTERSPEECH 2004, 2004

On the time variability of vocal tract for speaker recognition.
Proceedings of the INTERSPEECH 2004, 2004

Performance analysis of transcoding algorithms in packet-loss environments.
Proceedings of the INTERSPEECH 2004, 2004

Theory for speaker recognition over IP.
Proceedings of the INTERSPEECH 2004, 2004

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

A bit-rate/bandwidth scalable speech coder based on ITU-T G.723.1 standard.
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
Improving the transcoding capability of speech coders.
IEEE Trans. Multim., 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 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
An adaptive short-term postfilter based on pseudo-cepstral representation of line spectral frequencies.
Speech Commun., 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 frame erasure concealment algorithm based on gain parameter re-estimation for CELP coders.
IEEE Signal Process. Lett., 2001

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

A candidate for the ITU-T 4 kbit/s speech coding standard.
Proceedings of the IEEE International Conference on Acoustics, 2001

2000
Low-rate quantization of spectrum parameters.
Proceedings of the IEEE International Conference on Acoustics, 2000

1999
Low delay analysis/synthesis schemes for joint speech enhancement and low bit rate speech coding.
Proceedings of the Sixth European Conference on Speech Communication and Technology, 1999

Phase adjustment in waveform interpolation.
Proceedings of the 1999 IEEE International Conference on Acoustics, 1999

Pitch quantization in low bit-rate speech coding.
Proceedings of the 1999 IEEE International Conference on Acoustics, 1999

1998
Quantization of the spectral envelope for sinusoidal coders.
Proceedings of the 1998 IEEE International Conference on Acoustics, 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

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 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


  Loading...