Thippur V. Sreenivas

Orcid: 0000-0002-7878-9609

Affiliations:
  • ERNET, India


According to our database1, Thippur V. Sreenivas authored at least 124 papers between 1979 and 2022.

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

Timeline

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Bibliography

2022
Binaural Spatial Transform for Multi-source Localization determining Angular Extent of Ensemble Source Width.
Proceedings of the IEEE International Conference on Signal Processing and Communications, 2022

2021
Spatiogram: A phase based directional angular measure and perceptual weighting for ensemble source width.
CoRR, 2021

Directional MCLP Analysis and Reconstruction for Spatial Speech Communication.
CoRR, 2021

2020
Binaural Spatial Audiometry Screening Using Android Mobile Device Audio I/O Facility.
Proceedings of the 2020 National Conference on Communications, 2020

2019
Late Reverberation Cancellation Using Bayesian Estimation of Multi-Channel Linear Predictors and Student's t-Source Prior.
IEEE ACM Trans. Audio Speech Lang. Process., 2019

Multi-loudspeaker Rendering of Musical Ensemble: Role of Timbre in Source Width Perception.
Proceedings of the TENCON 2019, 2019

Comparison of low-dimension speech segment embeddings: Application to speaker diarization.
Proceedings of the National Conference on Communications, 2019

Clean speech AE-DNN PSD constraint for MCLP based reverberant speech enhancement.
Proceedings of the 27th European Signal Processing Conference, 2019

Ad-hoc mobile array based audio segmentation using latent variable stochastic model.
Proceedings of the 27th European Signal Processing Conference, 2019

2018
TDOA-Based Multiple Acoustic Source Localization Without Association Ambiguity.
IEEE ACM Trans. Audio Speech Lang. Process., 2018

Time-varying sinusoidal demodulation for non-stationary modeling of speech.
Speech Commun., 2018

LSTM based AE-DNN constraint for better late reverb suppression in multi-channel LP formulation.
CoRR, 2018

Latent variable approach to diarization of audio recordings using ad-hoc randomly placed mobile devices.
CoRR, 2018

Enhanced Directional Sensitivity using Acoustic Dish Reflector.
Proceedings of the 2018 International Conference on Signal Processing and Communications (SPCOM), 2018

Linear Prediction Based Diffuse Signal Estimation for Blind Microphone Geometry Calibration.
Proceedings of the 16th International Workshop on Acoustic Signal Enhancement, 2018

2017
Joint Bayesian Estimation of Time-Varying LP Parameters and Excitation for Speech.
IEEE Signal Process. Lett., 2017

Mel-scale sub-band modelling for perceptually improved time-scale modification of speech and audio signals.
Proceedings of the Twenty-third National Conference on Communications, 2017

Parameter estimation of a moving acoustic source: Linear chirplet transform vs WVD.
Proceedings of the Twenty-third National Conference on Communications, 2017

Perceptual evaluation of simulated auditory source width expansion.
Proceedings of the Twenty-third National Conference on Communications, 2017

Quantized Melodic Contours in Indian Art Music Perception: Application to Transcription.
Proceedings of the 18th International Society for Music Information Retrieval Conference, 2017

2016
Reverberation robust TDOA estimation using convex region prior.
Proceedings of the 2016 International Conference on Signal Processing and Communications (SPCOM), 2016

Who spoke what? A latent variable framework for the joint decoding of multiple speakers and their keywords.
Proceedings of the 2016 International Conference on Signal Processing and Communications (SPCOM), 2016

Reverberation-Robust One-Bit TDOA Based Moving Source Localization for Automatic Camera Steering.
Proceedings of the 17th Annual Conference of the International Speech Communication Association, 2016

Feature selection and model optimization for semi-supervised speaker spotting.
Proceedings of the 24th European Signal Processing Conference, 2016

2015
Event-triggered sampling using signal extrema for instantaneous amplitude and instantaneous frequency estimation.
Signal Process., 2015

Successive approximation algorithm for LPC estimation using sparse residual constraint.
Proceedings of the Twenty First National Conference on Communications, 2015

Moving sound source parameter estimation using a single microphone and signal extrema samples.
Proceedings of the 2015 IEEE International Conference on Acoustics, 2015

Multi-instrument detection in polyphonic music using Gaussian Mixture based factorial HMM.
Proceedings of the 2015 IEEE International Conference on Acoustics, 2015

Influence of time-varying pitch on timbre: "Coherence and incoherence" based on spectral centroid.
Proceedings of the 2015 IEEE International Conference on Acoustics, 2015

TDE sign based homing algorithm for sound source tracking using a Y-shaped microphone array.
Proceedings of the 23rd European Signal Processing Conference, 2015

2014
Time varying linear prediction using sparsity constraints.
Proceedings of the IEEE International Conference on Acoustics, 2014

2013
Fast Likelihood Computation in Speech Recognition using Matrices.
J. Signal Process. Syst., 2013

Identification of Active Sources in Single-Channel Convolutive Mixtures Using Known Source Models.
IEEE Signal Process. Lett., 2013

Hierarchical Classification of Carnatic Music Forms.
Proceedings of the 14th International Society for Music Information Retrieval Conference, 2013

Student's-t mixture model based multi-instrument recognition in polyphonic music.
Proceedings of the IEEE International Conference on Acoustics, 2013

2012
A Mixture Model Approach for Formant Tracking and the Robustness of Student's-t Distribution.
IEEE Trans. Speech Audio Process., 2012

Automatic Speech Segmentation Using Probabilistic Latent Component Modeling.
Proceedings of the 13th Annual Conference of the International Speech Communication Association, 2012

Joint Pitch-Analysis Formant-Synthesis framework for CS recovery of speech.
Proceedings of the 13th Annual Conference of the International Speech Communication Association, 2012

Sparse signal reconstruction based on signal dependent non-uniform samples.
Proceedings of the 2012 IEEE International Conference on Acoustics, 2012

2011
Carnatic music analysis: Shadja, swara identification and rAga verification in AlApana using stochastic models.
Proceedings of the IEEE Workshop on Applications of Signal Processing to Audio and Acoustics, 2011

Fast computation of Gaussian likelihoods using low-rank matrix approximations.
Proceedings of the IEEE Workshop on Signal Processing Systems, 2011

Compressive Sensing for Music Signals: Comparison of Transforms with Coherent Dictionaries.
Proceedings of the AES International Conference Semantic Audio 2011, 2011

Time-Varying Signal Adaptive Transform and IHT Recovery of Compressive Sensed Speech.
Proceedings of the 12th Annual Conference of the International Speech Communication Association, 2011

2010
Multi-Pattern Viterbi Algorithm for joint decoding of multiple speech patterns.
Signal Process., 2010

Joint evaluation of multiple speech patterns for speech recognition and training.
Comput. Speech Lang., 2010

Multi-channel iterative dereverberation based on codebook constrained iterative multi-channel wiener filter.
Proceedings of the 11th Annual Conference of the International Speech Communication Association, 2010

A multimodal density function estimation approach to formant tracking.
Proceedings of the 11th Annual Conference of the International Speech Communication Association, 2010

Robust mixture modeling using t-distribution: application to speaker ID.
Proceedings of the 11th Annual Conference of the International Speech Communication Association, 2010

2009
Linear filtering in DCT IV/DST IV and MDCT/MDST domain.
Signal Process., 2009

Blocking artifacts in speech/audio: Dynamic auditory model-based characterization and optimal time-frequency smoothing.
Signal Process., 2009

Reduced complexity two stage vector quantization.
Digit. Signal Process., 2009

Enhancement of binaural speech using codebook constrained iterative binaural wiener filter.
Proceedings of the 10th Annual Conference of the International Speech Communication Association, 2009

Compressive sensing for sparsely excited speech signals.
Proceedings of the IEEE International Conference on Acoustics, 2009

Analysis-by-synthesis based switched transform domain split VQ using Gaussian mixture model.
Proceedings of the IEEE International Conference on Acoustics, 2009

2008
Block Convolution Using Discrete Trigonometric Transforms and Discrete Fourier Transform.
IEEE Signal Process. Lett., 2008

Optimum Transform Domain Split VQ.
IEEE Signal Process. Lett., 2008

Predicting VQ Performance Bound for LSF Coding.
IEEE Signal Process. Lett., 2008

Switched Conditional PDF-Based Split VQ Using Gaussian Mixture Model.
IEEE Signal Process. Lett., 2008

Optimum switched split vector quantization of LSF parameters.
Signal Process., 2008

Pruned universal symbol sequences for LZW based language identification.
Proceedings of the Odyssey 2008: The Speaker and Language Recognition Workshop, 2008

Comparison of AM-FM based features for robust speech recognition.
Proceedings of the 9th Annual Conference of the International Speech Communication Association, 2008

Subspace based speech enhancement using Gaussian mixture model.
Proceedings of the 9th Annual Conference of the International Speech Communication Association, 2008

Two stage iterative Wiener filtering for speech enhancement.
Proceedings of the 9th Annual Conference of the International Speech Communication Association, 2008

Online unsupervised pattern discovery in speech using parallelization.
Proceedings of the 9th Annual Conference of the International Speech Communication Association, 2008

GMM based Bayesian approach to speech enhancement in signal / transform domain.
Proceedings of the IEEE International Conference on Acoustics, 2008

Forward/Backward Algorithms for joint multi pattern speech recognition.
Proceedings of the 2008 16th European Signal Processing Conference, 2008

Optimal local polynomial regression of noisy time-varying signals.
Proceedings of the 2008 16th European Signal Processing Conference, 2008

Adaptive window for local polynomial regression from noisy nonuniform samples.
Proceedings of the 2008 16th European Signal Processing Conference, 2008

Speech enhancement using intra-frame dependency in DCT domain.
Proceedings of the 2008 16th European Signal Processing Conference, 2008

Low complexity wideband LSF quantization using GMM of uncorrelated Gaussian mixtures.
Proceedings of the 2008 16th European Signal Processing Conference, 2008

2007
Analysis of Conditional PDF-Based Split VQ.
IEEE Signal Process. Lett., 2007

Conditional PDF-Based Split Vector Quantization of Wideband LSF Parameters.
IEEE Signal Process. Lett., 2007

Increased watermark-to-host correlation of uniform random phase watermarks in audio signals.
Signal Process., 2007

Joint inter-frame and intra-frame predictive coding of LSF parameters.
Proceedings of the 9th International Symposium on Signal Processing and Its Applications, 2007

Computationally efficient optimum weighting function for vector quantization of LSF parameters.
Proceedings of the 9th International Symposium on Signal Processing and Its Applications, 2007

Robust and high-resolution voiced/unvoiced classification in noisy speech using a signal smoothness criterion.
Proceedings of the 8th Annual Conference of the International Speech Communication Association, 2007

Normalized two stage SVQ for minimum complexity wide-band LSF quantization.
Proceedings of the 8th Annual Conference of the International Speech Communication Association, 2007

Sequential Split Vector Quantization of LSF Parameters using Conditional Pdf.
Proceedings of the IEEE International Conference on Acoustics, 2007

Joint decoding of multiple speech patterns for robust speech recognition.
Proceedings of the IEEE Workshop on Automatic Speech Recognition & Understanding, 2007

2006
Signal-to-noise ratio estimation using higher-order moments.
Signal Process., 2006

Time-varying filter interpretation of Fourier transform and its variants.
Signal Process., 2006

LZW Based Distance Measures for Spoken Language Identification.
Proceedings of the Odyssey 2006: The Speaker and Language Recognition Workshop, 2006

Comparison of prediction based LSF quantization methods using split VQ.
Proceedings of the Ninth International Conference on Spoken Language Processing, 2006

Two stage transform vector quantization of LSFs for wideband speech coding.
Proceedings of the Ninth International Conference on Spoken Language Processing, 2006

Low complexity LID using pruned pattern tables of LZW.
Proceedings of the Ninth International Conference on Spoken Language Processing, 2006

Dynamic Programming Based Optimum Non-Uniform Samples For Speech Reconstruction and Coding.
Proceedings of the 2006 IEEE International Conference on Acoustics Speech and Signal Processing, 2006

Novel auditory motivated subband temporal envelope based fundamental frequency estimation algorithm.
Proceedings of the 14th European Signal Processing Conference, 2006

2005
Auditory motivated level-crossing approach to instantaneous frequency estimation.
IEEE Trans. Signal Process., 2005

Dynamic programming based segmentation approach to LSF matrix reconstruction.
Proceedings of the 9th European Conference on Speech Communication and Technology, 2005

Stochastic pronunciation modeling by ergodic-HMM of acoustic sub-word units.
Proceedings of the 9th European Conference on Speech Communication and Technology, 2005

Speech enhancement using Markov model of speech segments.
Proceedings of the 9th European Conference on Speech Communication and Technology, 2005

Automatic Speech Segmentation Using Average Level Crossing Rate Information.
Proceedings of the 2005 IEEE International Conference on Acoustics, 2005

Matrix Quantization Based Time-Varying Filter Speech Enhancement.
Proceedings of the 2005 IEEE International Conference on Acoustics, 2005

2004
Effect of interpolation on PWVD computation and instantaneous frequency estimation.
Signal Process., 2004

Adaptive Window Zero-Crossing-Based Instantaneous Frequency Estimation.
EURASIP J. Adv. Signal Process., 2004

Improved iterative wiener filtering for non-stationary noise speech enhancement.
Proceedings of the 8th International Conference on Spoken Language Processing, 2004

Neural "spike rate spectrum" as a noise robust, speaker invariant feature for automatic speech recognition.
Proceedings of the 8th International Conference on Spoken Language Processing, 2004

Mixture Gaussian model training against impostor model parameters: an application to speaker identification.
Proceedings of the 8th International Conference on Spoken Language Processing, 2004

Novel approach to AM-FM decomposition with applications to speech and music analysis.
Proceedings of the 2004 IEEE International Conference on Acoustics, 2004

Automatically derived units for segment vocoders.
Proceedings of the 2004 IEEE International Conference on Acoustics, 2004

Music instrument recognition: from isolated notes to solo phrases.
Proceedings of the 2004 IEEE International Conference on Acoustics, 2004

2003
Adaptive spectrogram vs. adaptive pseudo-Wigner-Ville distribution for instantaneous frequency estimation.
Signal Process., 2003

Language identification using parallel sub-word recognition - an ergodic HMM equivalence.
Proceedings of the 8th European Conference on Speech Communication and Technology, EUROSPEECH 2003, 2003

A novel method of analysing and comparing responses of hearing aid algorithms using auditory time-frequency representation.
Proceedings of the 8th European Conference on Speech Communication and Technology, EUROSPEECH 2003, 2003

Language identification using parallel sub-word recognition.
Proceedings of the 2003 IEEE International Conference on Acoustics, 2003

Instantaneous frequency estimation using level-crossing information.
Proceedings of the 2003 IEEE International Conference on Acoustics, 2003

2002
IF estimation using higher order TFRs.
Signal Process., 2002

Automatic language identification using acoustic sub-word units.
Proceedings of the 7th International Conference on Spoken Language Processing, ICSLP2002, 2002

Teager energy based blood cell segmentation.
Proceedings of the 14th International Conference on Digital Signal Processing, 2002

Robust parameters for automatic segmentation of speech.
Proceedings of the IEEE International Conference on Acoustics, 2002

2001
A switched DPCM/subband coder for pre-echo reduction.
Proceedings of the EUROSPEECH 2001 Scandinavia, 2001

Mixture Gaussian envelope chirp model for speech and audio.
Proceedings of the IEEE International Conference on Acoustics, 2001

1999
Cone-kernel representation versus instantaneous power spectrum.
IEEE Trans. Signal Process., 1999

EARLYZER: perceptualy motivated robust TFR of speech.
Proceedings of the Sixth European Conference on Speech Communication and Technology, 1999

1998
Vector quantization of scale factors in advanced audio coder (AAC).
Proceedings of the 1998 IEEE International Conference on Acoustics, 1998

1996
Codebook constrained Wiener filtering for speech enhancement.
IEEE Trans. Speech Audio Process., 1996

1995
On incorporating phonemic constraints in hidden Markov models for speech recognition.
Proceedings of the Fourth European Conference on Speech Communication and Technology, 1995

1994
Phoneme recognition in continuous speech using large inhomogeneous hidden Markov models.
Proceedings of ICASSP '94: IEEE International Conference on Acoustics, 1994

1993
Vectorized backpropagation and automatic pruning for MLP network optimization.
Proceedings of International Conference on Neural Networks (ICNN'88), San Francisco, CA, USA, March 28, 1993

1992
Zero-crossing based spectral analysis and SVD spectral analysis for formant frequency estimation in noise.
IEEE Trans. Signal Process., 1992

1990
Spectral resolution and noise robustness in auditory modeling.
Proceedings of the 1990 International Conference on Acoustics, 1990

1988
Modelling LPC-residue by components for good quality speech coding.
Proceedings of the IEEE International Conference on Acoustics, 1988

1984
Pitch estimation of aperiodic and noisy speech signals.
Speech Commun., 1984

1979
On sensitivity of vocal tract area functions.
Proceedings of the IEEE International Conference on Acoustics, 1979


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