Changchun Bao

Orcid: 0000-0002-5606-5343

According to our database1, Changchun Bao authored at least 159 papers between 2003 and 2024.

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

Timeline

Legend:

Book 
In proceedings 
Article 
PhD thesis 
Dataset
Other 

Links

On csauthors.net:

Bibliography

2024
Exploring the power of pure attention mechanisms in blind room parameter estimation.
EURASIP J. Audio Speech Music. Process., December, 2024

Coarse-to-Fine Target Speaker Extraction Based on Contextual Information Exploitation.
IEEE ACM Trans. Audio Speech Lang. Process., 2024

Joint DOA Estimation and Dereverberation Based on Multi-Channel Linear Prediction Filtering and Azimuth Sparsity.
IEEE ACM Trans. Audio Speech Lang. Process., 2024

Harmonic-Aware Frequency and Time Attention for Automatic Piano Transcription.
IEEE ACM Trans. Audio Speech Lang. Process., 2024

First-Order Relative Harmonic Coefficient-Based Time-Frequency Points Selection for Multi-Source DOA Estimation.
IEEE ACM Trans. Audio Speech Lang. Process., 2024

Three-Dimensional Room Transfer Function Parameterization Based on Multiple Concentric Planar Circular Arrays.
IEEE ACM Trans. Audio Speech Lang. Process., 2024

A distortionless convolution beamformer design method based on the weighted minimum mean square error for joint dereverberation and denoising.
Speech Commun., 2024

SS-BRPE: Self-Supervised Blind Room Parameter Estimation Using Attention Mechanisms.
CoRR, 2024

Target Speaker Extraction by Directly Exploiting Contextual Information in the Time-Frequency Domain.
Proceedings of the IEEE International Conference on Acoustics, 2024

Attention Is All You Need For Blind Room Volume Estimation.
Proceedings of the IEEE International Conference on Acoustics, 2024

2023
Coarse-to-fine speech separation method in the time-frequency domain.
Speech Commun., November, 2023

Study of MVDR Beamforming with Spatially Distributed Source: Theoretical Analysis and Efficient Microphone Array Geometry Optimization Method.
Circuits Syst. Signal Process., August, 2023

Monaural Speech Separation Method Based on Recurrent Attention with Parallel Branches.
Proceedings of the 24th Annual Conference of the International Speech Communication Association, 2023

Predictive Packet Loss Concealment Method Based on Conformer and Temporal Convolution Module.
Proceedings of the IEEE International Conference on Signal Processing, 2023

A Beam-TFDPRNN Based Speech Separation Method in Reverberant Environments.
Proceedings of the IEEE International Conference on Signal Processing, 2023

A Time-domain Packet Loss Concealment Method by Designing Transformer-based Convolutional Recurrent Network.
Proceedings of the IEEE International Conference on Signal Processing, 2023

Multi-Source Localization Method Based on the Log-Mel Spectrum Augmented Noise Subspace.
Proceedings of the IEEE International Conference on Signal Processing, 2023

GAN-Based Time-Domain Packet Loss Concealment Method with Consistent Mapping Approach.
Proceedings of the Asia Pacific Signal and Information Processing Association Annual Summit and Conference, 2023

2022
Speech Enhancement With Robust Beamforming for Spatially Overlapped and Distributed Sources.
IEEE ACM Trans. Audio Speech Lang. Process., 2022

Phase unwrapping based packet loss concealment using deep neural networks.
Speech Commun., 2022

Analysis-by-synthesis based training target extraction of the DNN for noise masking.
Speech Commun., 2022

A Neural Vocoder Based Packet Loss Concealment Algorithm.
CoRR, 2022

Multi-source wideband DOA estimation method by frequency focusing and error weighting.
Proceedings of the 23rd Annual Conference of the International Speech Communication Association, 2022

Embedding Recurrent Layers with Dual-Path Strategy in a Variant of Convolutional Network for Speaker-Independent Speech Separation.
Proceedings of the 23rd Annual Conference of the International Speech Communication Association, 2022

A Neural Vocoder Based Packet Loss Concealment Algorithm.
Proceedings of the IEEE International Conference on Signal Processing, 2022

DPTNet-based Beamforming for Speech Separation.
Proceedings of the IEEE International Conference on Signal Processing, 2022

An Effective Dereverberation Algorithm by Fusing MVDR and MCLP.
Proceedings of the IEEE International Conference on Signal Processing, 2022

Dereverberation and Noise Reduction Based on PSD Estimation with Low Complexity.
Proceedings of the IEEE International Conference on Signal Processing, 2022

Speech Recognition Method based on CTC Multilayer Loss.
Proceedings of the 2022 11th International Conference on Computing and Pattern Recognition, 2022

A New Parametric Coding Method Combined Linear Microphone Array Topology.
Proceedings of the Data Compression Conference, 2022

DNN-based Multi-Channel Speech Coding Employing Sound Localization.
Proceedings of the Data Compression Conference, 2022

2021
Multi-Source DOA Estimation in Reverberant Environments by Jointing Detection and Modeling of Time-Frequency Points.
IEEE ACM Trans. Audio Speech Lang. Process., 2021

Phoneme-Unit-Specific Time-Delay Neural Network for Speaker Verification.
IEEE ACM Trans. Audio Speech Lang. Process., 2021

Power Exponent Based Weighting Criterion for DNN-Based Mask Approximation in Speech Enhancement.
IEEE Signal Process. Lett., 2021

Multi-source localization by using offset residual weight.
EURASIP J. Audio Speech Music. Process., 2021

Design of a Fixed-Wing UAV Controller Based on Adaptive Backstepping Sliding Mode Control Method.
IEEE Access, 2021

GAN-Based Inter-Channel Amplitude Ratio Decoding in Multi-Channel Speech Coding.
Proceedings of the 12th International Symposium on Chinese Spoken Language Processing, 2021

Multi-channel Speech Coding Based on Phase Alignment and Energy Ratio of Internal Channels.
Proceedings of the IEEE International Conference on Signal Processing, 2021

A New Four-Channel Speech Coding Method Based on Recurrent Neural Network.
Proceedings of the IEEE International Conference on Signal Processing, 2021

Kronecker Product Based Linear Prediction Kalman Filter for Dereverberation and Noise Reduction.
Proceedings of the IEEE International Conference on Signal Processing, 2021

An Implementaion of the CNN-Based MVDR Beamforming For Speech Enhancement.
Proceedings of the IEEE International Conference on Signal Processing, 2021

An NMF-based MMSE Approach for Single Channel Speech Enhancement Using Densely Connected Convolutional Network.
Proceedings of the IEEE International Conference on Signal Processing, 2021

A multi-source localization method based on clustering and outlier removal.
Proceedings of the Asia-Pacific Signal and Information Processing Association Annual Summit and Conference, 2021

2020
A Parallel-Data-Free Speech Enhancement Method Using Multi-Objective Learning Cycle-Consistent Generative Adversarial Network.
IEEE ACM Trans. Audio Speech Lang. Process., 2020

Speech Enhancement Based on Beamforming and Post-Filtering by Combining Phase Information.
Proceedings of the 21st Annual Conference of the International Speech Communication Association, 2020

Multi-channel Speech Enhancement with Multiple-target GANs.
Proceedings of the IEEE International Conference on Signal Processing, 2020

A Weekly Supervised Speech Enhancement Strategy using Cycle-GAN.
Proceedings of the IEEE International Conference on Signal Processing, 2020

Multi-channel Speech Enhancement Based on the MVDR Beamformer and Postfilter.
Proceedings of the IEEE International Conference on Signal Processing, 2020

GEV Beamforming with BAN Integrating LPS Estimation and Post-filtering.
Proceedings of the IEEE International Conference on Signal Processing, 2020

Prediction of NMF-based Wiener Filter for Speech Enhancement Using Deep Neural Networks.
Proceedings of the IEEE International Conference on Signal Processing, 2020

Autoregressive Parameter Estimation with Dnn-Based Pre-Processing.
Proceedings of the 2020 IEEE International Conference on Acoustics, 2020

2019
RS-CAE-Based AR-Wiener Filtering and Harmonic Recovery for Speech Enhancement.
IEEE ACM Trans. Audio Speech Lang. Process., 2019

IRM with Phase Parameterization for Speech Enhancement.
Proceedings of the 2019 IEEE Workshop on Applications of Signal Processing to Audio and Acoustics, 2019

Masking Estimation with Phase Restoration of Clean Speech for Monaural Speech Enhancement.
Proceedings of the 20th Annual Conference of the International Speech Communication Association, 2019

Sound Field Reproduction in Reverberant Room Using the Alternating Direction Method of Multipliers Based Lasso and Regularized Least-Square.
Proceedings of the Intelligent Computing Theories and Application, 2019

Linear Prediction-based Part-defined Auto-encoder Used for Speech Enhancement.
Proceedings of the IEEE International Conference on Acoustics, 2019

GSC Based Speech Enhancement with Generative Adversarial Network.
Proceedings of the 2019 Asia-Pacific Signal and Information Processing Association Annual Summit and Conference, 2019

CycleGAN-based speech enhancement for the unpaired training data.
Proceedings of the 2019 Asia-Pacific Signal and Information Processing Association Annual Summit and Conference, 2019

End-to-End Speech Enhancement Using Fully Convolutional Networks with Skip Connections.
Proceedings of the 2019 Asia-Pacific Signal and Information Processing Association Annual Summit and Conference, 2019

Phase Unwrapping Based Speech Enhancement.
Proceedings of the 2019 Asia-Pacific Signal and Information Processing Association Annual Summit and Conference, 2019

2018
Design of a Planar First-Order Loudspeaker Array for Global Active Noise Control.
IEEE ACM Trans. Audio Speech Lang. Process., 2018

IRM estimation based on data field of cochleagram for speech enhancement.
Speech Commun., 2018

Separation of multiple speech sources by recovering sparse and non-sparse components from B-format microphone recordings.
Speech Commun., 2018

Multiple Sound Sources Localization with Frame-by-Frame Component Removal of Statistically Dominant Source.
Sensors, 2018

Multiple Source Localization by Using Improved Single Source Bins Detection.
J. Inf. Hiding Multim. Signal Process., 2018

Weakly supervised CRNN system for sound event detection with large-scale unlabeled in-domain data.
CoRR, 2018

Speech Enhancement via Generative Adversarial LSTM Networks.
Proceedings of the 16th International Workshop on Acoustic Signal Enhancement, 2018

DNN-Based Speech Enhancement Using MBE Model.
Proceedings of the 16th International Workshop on Acoustic Signal Enhancement, 2018

Dnn-Based Ar-Wiener Filtering for Speech Enhancement.
Proceedings of the 2018 IEEE International Conference on Acoustics, 2018

Speech Enhancement with Phase Correction based on Modified DNN Architecture.
Proceedings of the Asia-Pacific Signal and Information Processing Association Annual Summit and Conference, 2018

2017
Multiplicative Update of Auto-Regressive Gains for Codebook-Based Speech Enhancement.
IEEE ACM Trans. Audio Speech Lang. Process., 2017

A data-driven speech enhancement method based on A* longest segment searching technique.
Speech Commun., 2017

Real-time multiple sound source localization and counting using a soundfield microphone.
J. Ambient Intell. Humaniz. Comput., 2017

Simulating the Three-Dimensional Room Transfer Function for a Rotatable Complex Source.
IEICE Trans. Fundam. Electron. Commun. Comput. Sci., 2017

A Mask Estimation Method Integrating Data Field Model for Speech Enhancement.
Proceedings of the 18th Annual Conference of the International Speech Communication Association, 2017

Improved Codebook-Based Speech Enhancement Based on MBE Model.
Proceedings of the 18th Annual Conference of the International Speech Communication Association, 2017

A codebook-driven speech enhancement method by exploiting speech harmonicity.
Proceedings of the 2017 IEEE International Conference on Signal Processing, 2017

Multiple audio source separation by using intra-object-sparsity encoding framework.
Proceedings of the 2017 IEEE International Conference on Signal Processing, 2017

Codebook-driven speech enhancement using DNN and harmonic emphasis.
Proceedings of the 2017 Asia-Pacific Signal and Information Processing Association Annual Summit and Conference, 2017

Multiple source localization by using energy weighted single source zone detection.
Proceedings of the 2017 Asia-Pacific Signal and Information Processing Association Annual Summit and Conference, 2017

Speech enhancement based on binaural cues.
Proceedings of the 2017 Asia-Pacific Signal and Information Processing Association Annual Summit and Conference, 2017

2016
Audio Bandwidth Extension Based on Ensemble Echo State Networks with Temporal Evolution.
IEEE ACM Trans. Audio Speech Lang. Process., 2016

Speech enhancement based on AR model parameters estimation.
Speech Commun., 2016

Audio bandwidth extension using ensemble of recurrent neural networks.
EURASIP J. Audio Speech Music. Process., 2016

HMM-based cue parameters estimation for speech enhancement.
Proceedings of the 10th International Symposium on Chinese Spoken Language Processing, 2016

Speech enhancement with binaural cues derived from a priori codebook.
Proceedings of the 10th International Symposium on Chinese Spoken Language Processing, 2016

Multiplicative update of AR gains in codebook-driven speech enhancement.
Proceedings of the 2016 IEEE International Conference on Acoustics, 2016

Speech enhancement method with geometric phase estimation by incorporating MIXMAX model.
Proceedings of the Asia-Pacific Signal and Information Processing Association Annual Summit and Conference, 2016

Measurement of the acoustic transfer function using compressed sensing techniques.
Proceedings of the Asia-Pacific Signal and Information Processing Association Annual Summit and Conference, 2016

2015
Encoding Multiple Audio Objects Using Intra-Object Sparsity.
IEEE ACM Trans. Audio Speech Lang. Process., 2015

Sparse Hidden Markov Models for Speech Enhancement in Non-Stationary Noise Environments.
IEEE ACM Trans. Audio Speech Lang. Process., 2015

Speech enhancement based on Bayesian decision and spectral amplitude estimation.
EURASIP J. Audio Speech Music. Process., 2015

Codebook-based speech enhancement using Markov process and speech-presence probability.
Proceedings of the 16th Annual Conference of the International Speech Communication Association, 2015

A data-driven speech enhancement method based on modeled long-range temporal dynamics.
Proceedings of the 16th Annual Conference of the International Speech Communication Association, 2015

Sparse HMM-based speech enhancement method for stationary and non-stationary noise environments.
Proceedings of the 2015 IEEE International Conference on Acoustics, 2015

3D multizone soundfield reproduction using spherical harmonic analysis.
Proceedings of the IEEE China Summit and International Conference on Signal and Information Processing, 2015

A novel Bayesian framework for speech enhancement using speech presence uncertainty.
Proceedings of the IEEE China Summit and International Conference on Signal and Information Processing, 2015

Conversion of multichannel sound signals based on spherical harmonics with L1-norm constraint.
Proceedings of the IEEE China Summit and International Conference on Signal and Information Processing, 2015

3D multizone soundfield reproduction in the reverberant room using a spherical loudspeaker array.
Proceedings of the Asia-Pacific Signal and Information Processing Association Annual Summit and Conference, 2015

An analysis-by-synthesis encoding approach for multiple audio objects.
Proceedings of the Asia-Pacific Signal and Information Processing Association Annual Summit and Conference, 2015

Codebook-based speech enhancement with Bayesian LP parameters estimation.
Proceedings of the Asia-Pacific Signal and Information Processing Association Annual Summit and Conference, 2015

A gain-adaptive parallel HMM for speech enhancement.
Proceedings of the Asia-Pacific Signal and Information Processing Association Annual Summit and Conference, 2015

An improved dictionary learning method for speech enhancement.
Proceedings of the Asia-Pacific Signal and Information Processing Association Annual Summit and Conference, 2015

2014
Wiener filtering based speech enhancement with Weighted Denoising Auto-encoder and noise classification.
Speech Commun., 2014

Speech enhancement using generalized weighted β-order spectral amplitude estimator.
Speech Commun., 2014

Audio bandwidth extension based on temporal smoothing cepstral coefficients.
EURASIP J. Audio Speech Music. Process., 2014

A blind bandwidth extension method for audio signals based on phase space reconstruction.
EURASIP J. Audio Speech Music. Process., 2014

The design of Ambisonic reproduction system based on dynamic gain parameters.
Proceedings of the IEEE International Conference on Acoustics, 2014

The design of ambisonics decoders for irregular speaker array conforming to subjective perception.
Proceedings of the International Conference on Audio, 2014

Relative distance estimation in multi-channel spatial audio signal.
Proceedings of the International Conference on Audio, 2014

Multi-source sound field reproduction using cylindrical harmonic analysis.
Proceedings of the IEEE China Summit & International Conference on Signal and Information Processing, 2014

Speech enhancement based on a few shapes of speech spectrum.
Proceedings of the IEEE China Summit & International Conference on Signal and Information Processing, 2014

The design of HOA irregular decoders based on the optimal symmetrical virtual microphone response.
Proceedings of the Asia-Pacific Signal and Information Processing Association Annual Summit and Conference, 2014

Range extrapolation of Head-Related Transfer Function using improved Higher Order Ambisonics.
Proceedings of the Asia-Pacific Signal and Information Processing Association Annual Summit and Conference, 2014

Speech enhancement based on a novel weighting spectral distortion measure.
Proceedings of the Asia-Pacific Signal and Information Processing Association Annual Summit and Conference, 2014

A novel speech enhancement method using power spectra smooth in Wiener filtering.
Proceedings of the Asia-Pacific Signal and Information Processing Association Annual Summit and Conference, 2014

2013
Compressed domain speech enhancement method based on ITU-T G.722.2.
Speech Commun., 2013

Blind bandwidth extension of audio signals based on harmonic mapping in phase space.
Proceedings of the 36th International Conference on Telecommunications and Signal Processing, 2013

Speech enhancement with weighted denoising auto-encoder.
Proceedings of the 14th Annual Conference of the International Speech Communication Association, 2013

A speech enhancement method by coupling speech detection and spectral amplitude estimation.
Proceedings of the 14th Annual Conference of the International Speech Communication Association, 2013

Speech Intelligibility Improvement Using the Constraints on Speech Distortion and Noise Over-estimation.
Proceedings of the Ninth International Conference on Intelligent Information Hiding and Multimedia Signal Processing, 2013

Spectral envelope estimation used for audio bandwidth extension based on RBF neural network.
Proceedings of the IEEE International Conference on Acoustics, 2013

Speech coding based on pitch synchrony and two-stage transformation.
Proceedings of the IEEE International Conference on Acoustics, 2013

A speech enhancement algorithm based on β-order GARCH model.
Proceedings of the 2013 IEEE China Summit and International Conference on Signal and Information Processing, 2013

Clipping detection of audio signals based on kernel Fisher discriminant.
Proceedings of the 2013 IEEE China Summit and International Conference on Signal and Information Processing, 2013

An adaptive β-order MMSE estimator for speech enhancement using super-Gaussian speech model.
Proceedings of the 2013 IEEE China Summit and International Conference on Signal and Information Processing, 2013

A prior knowledge-based noise reduction method with dual microphones.
Proceedings of the Asia-Pacific Signal and Information Processing Association Annual Summit and Conference, 2013

Audio bandwidth extension based on Grey model.
Proceedings of the Asia-Pacific Signal and Information Processing Association Annual Summit and Conference, 2013

2012
A novel supervised learning algorithm for musical instrument classification.
Proceedings of the 35th International Conference on Telecommunications and Signal Processing, 2012

An LP spectrum modification method for noisy speech based on linear extrapolation.
Proceedings of the 35th International Conference on Telecommunications and Signal Processing, 2012

Compressed domain automatic level control based on ITU-T G.722.2.
Proceedings of the IEEE International Symposium on Signal Processing and Information Technology, 2012

A blind bandwidth extension method of audio signals based on Volterra series.
Proceedings of the Asia-Pacific Signal and Information Processing Association Annual Summit and Conference, 2012

2011
A novel hiss noise reduction method for audio signals based on MDCT.
Proceedings of the 2011 International Conference on Wireless Communications & Signal Processing, 2011

A MDCT-based click noise reduction method for MPEG-4 AAC codec.
Proceedings of the 2011 International Conference on Wireless Communications & Signal Processing, 2011

A sinusoidal audio and speech analysis/synthesis model based on improved EMD by adding pure tone.
Proceedings of the 2011 IEEE International Workshop on Machine Learning for Signal Processing, 2011

A restoration method of the clipped audio signals based on MDCT.
Proceedings of the 2011 IEEE International Symposium on Signal Processing and Information Technology, 2011

Compressed domain speech enhancement based on the joint modification of codebook gains.
Proceedings of the 2011 IEEE International Symposium on Signal Processing and Information Technology, 2011

Nonlinear bandwidth extension of audio signals based on hidden Markov model.
Proceedings of the 2011 IEEE International Symposium on Signal Processing and Information Technology, 2011

Audio bandwidth extension based on RBF neural network.
Proceedings of the 2011 IEEE International Symposium on Signal Processing and Information Technology, 2011

An improved β-order WEDM spectral amplitude estimator for speech enhancement.
Proceedings of the 2011 IEEE International Symposium on Signal Processing and Information Technology, 2011

An embedded stereo speech and audio coding method based on principal component analysis.
Proceedings of the 2011 IEEE International Symposium on Signal Processing and Information Technology, 2011

Robust understanding of spoken Chinese through character-based tagging and prior knowledge exploitation.
Proceedings of the 2011 IEEE Workshop on Automatic Speech Recognition & Understanding, 2011

2010
An efficient transcoding algorithm between AMR-NB and G.729ab.
Speech Commun., 2010

Robust character based tagging with domain lexical features for Chinese spoken language understanding.
Proceedings of the Sixth International Conference on Natural Computation, 2010

High frequency reconstruction of audio signal based on chaotic prediction theory.
Proceedings of the IEEE International Conference on Acoustics, 2010

2009
Improving Voice Search Using Forward-Backward LVCSR System Combination.
Proceedings of the Sixth International Symposium on Neural Networks, 2009

2008
A novel transcoding algorithm between 3GPP AMR-NB (7.95kbit/s) and ITU-t g.729a (8kbit/s).
Proceedings of the 9th Annual Conference of the International Speech Communication Association, 2008

Recognizing named entities in spoken Chinese dialogues with a character-level maximum entropy tagger.
Proceedings of the 9th Annual Conference of the International Speech Communication Association, 2008

A 8.32 kb/s embedded wideband speech coding candidate for ITU-t EV-VBR standardization.
Proceedings of the 9th Annual Conference of the International Speech Communication Association, 2008

2007
A novel 2kb/s waveform interpolation speech coder based on non-negative matrix factorization.
Proceedings of the 8th Annual Conference of the International Speech Communication Association, 2007

2006
A Low-complexity Improved WI Speech Coding at 2kbps.
Proceedings of the 5th International Symposium on Chinese Spoken Language Processing, 2006

A Fast Search Approach for LSF Parameters Codebook.
Proceedings of the 2006 IEEE International Conference on Acoustics Speech and Signal Processing, 2006

2005
A novel voicing cut-off determination for low bit-rate harmonic speech coding.
Proceedings of the 9th European Conference on Speech Communication and Technology, 2005

2004
Low complexity decomposition for the characteristic waveform of speech signal.
Proceedings of the 2004 International Symposium on Chinese Spoken Language Processing, 2004

A novel two-step SVM classifier for voiced/unvoiced/silence classification of speech.
Proceedings of the 2004 International Symposium on Chinese Spoken Language Processing, 2004

Quantization of SEW and REW magnitude for 2 kb/s waveform interpolation speech coding.
Proceedings of the 2004 International Symposium on Chinese Spoken Language Processing, 2004

High quality harmonic excitation linear predictive speech coding at 2 kb/s.
Proceedings of the 2004 International Symposium on Chinese Spoken Language Processing, 2004

An improved 4 kbit/s CELP speech coding algorithm.
Proceedings of the 2004 International Symposium on Chinese Spoken Language Processing, 2004

2003
Harmonic excitation LPC (HE-LPC) speech coding at 2.3 kb/s.
Proceedings of the 2003 IEEE International Conference on Acoustics, 2003


  Loading...