NDLI: Minimum phoneme error based heteroscedastic linear discriminant analysis for speech recognition

Please wait, while we are loading the content...

2015 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)

2014 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)

2013 IEEE International Conference on Acoustics, Speech and Signal Processing

2012 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)

2011 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)

2010 IEEE International Conference on Acoustics, Speech and Signal Processing

2009 IEEE International Conference on Acoustics, Speech and Signal Processing

2008 IEEE International Conference on Acoustics, Speech and Signal Processing

2007 IEEE International Conference on Acoustics, Speech and Signal Processing - ICASSP '07

2006 IEEE International Conference on Acoustics Speech and Signal Processing Proceedings

Proceedings. (ICASSP '05). IEEE International Conference on Acoustics, Speech, and Signal Processing, 2005.

Volume 5 - 2005

Volume 4 - 2005

Volume 3 - 2005

Volume 2 - 2005

Volume 1 - 2005

2005 IEEE International Conference on Acoustics, Speech, and Signal Processing

2005 IEEE International Conference on Acoustics, Speech, and Signal Processing

IEEE Signal Processing Society 2004 Board of Governors

ICASSP 2005 Conference Committee

Technical Program Committee

Future ICASSP conferences

Table of contents

ICASSP 2005 Proceedings

Polyglot synthesis using a mixture of monolingual corpora

Introducing roughness in individuality transformation through jitter modeling and modification

Spectral conversion based on maximum likelihood estimation considering global variance of converted parameter

A study on residual prediction techniques for voice conversion

Voice forgery using ALISP: indexation in a client memory

An improved spectral and prosodic transformation method in STRAIGHT-based voice conversion

Incorporating discourse features into confidence scoring of intention recognition results in spoken dialogue systems

Semantic interpretation with error correction

Dialog act tagging using graphical models

A clarification algorithm for spoken dialogue systems

Model adaptation for spoken language understanding

Unsupervised semantic intent discovery from call log acoustics

Proposal on objective speech quality assessment for wideband IP telephony

Neural cell type recognition between globus pallidus externus and globus pallidus internus by Gaussian mixture modeling

Analysis of relationship between overall quality and psychological factors affecting high-quality speech communication services

Can you understand him? Let's look at his word accuracy-automatic evaluation of tracheoesophageal speech

A warped bandwidth expansion filter

Relative energy and intelligibility of transient speech information

Rejection using rank statistics based on HMM state shortlists

Speaker adaptive confidence scoring using Bayesian combining

Improving utterance verification using additional confidence measures in isolated speech recognition interfaces

Generalized posterior probability for minimum error verification of recognized sentences

Robust speech recognition by integrating speech separation and hypothesis testing

Combination of multiple predictors to improve confidence measure based on local posterior probabilities

Adaptation of precision matrix models on large vocabulary continuous speech recognition

Discriminative training of CDHMMs for maximum relative separation margin

Statistical performance analysis of MCE/GPD learning in Gaussian classifiers and hidden Markov models [speech recognition example]

Discriminative training of acoustic models applied to domains with unreliable transcripts [speech recognition applications]

Minimum classification error for large scale speech recognition tasks using weighted finite state transducers

Discriminative training based on the criterion of least phone competing tokens for large vocabulary speech recognition

Multi-frame GMM-based block quantisation of line spectral frequencies for wideband speech coding

Non-intrusive GMM-based speech quality measurement

A multiple-description PCM speech coder using structured dual vector quantizers

A new segment quantizer for line spectral frequencies using Lempel-Ziv algorithm [speech coding applications]

Predictive VQ for bandwidth scalable LSP quantization [speech coding applications]

Coding with side information techniques for LSF reconstruction in voice over IP

Signal subspace speech enhancement for audible noise reduction

A wavelet Kalman filter with perceptual masking for speech enhancement in colored noise

Adaptive time segmentation of noisy speech for improved speech enhancement

Speech enhancement using harmonic regeneration

Instant noise estimation using Fourier transform of AMDF and variable start minima search

Speech enhancement based on speech spectral complex Gaussian mixture model

Improved phonetic speaker recognition using lattice decoding

SRI's 2004 NIST speaker recognition evaluation system

The 2004 MIT Lincoln Laboratory speaker recognition system

Speaker verification using adapted articulatory feature-based conditional pronunciation modeling

Prosody modeling and eigen-prosody analysis for robust speaker recognition

Prosodic modeling for speaker recognition based on sub-band energy temporal trajectories

Sub-phonetic polynomial segment model for large vocabulary continuous speech recognition

Constructing ensembles of ASR systems using randomized decision trees

Efficient generation of high-order context dependent weighted finite state transducers for speech recognition

The IBM 2004 conversational telephony system for rich transcription

Training LVCSR systems on thousands of hours of data

Landmark-based speech recognition: report of the 2004 Johns Hopkins summer workshop

Speech analysis by estimating perceptually relevant pole locations

Coherent envelope detection for modulation filtering of speech

Speech signal analysis with exponential autoregressive model

Comparison of autoregressive parameter estimation algorithms for speech processing and recognition

An algorithm for locating fundamental frequency markers in speech signals

An auto-regressive, non-stationary excited signal parameter estimation method and an evaluation of a singing-voice recognition

Static and dynamic spectral features: their noise robustness and optimal weights for ASR

Log-energy dynamic range normalization for robust speech recognition

A companding front end for noise-robust automatic speech recognition

Multi-resolution spectral entropy feature for robust ASR

Particle filter based non-stationary noise tracking for robust speech recognition

Online cepstral filtering using a sequential EM approach with Polyak averaging and feedback [speech recognition applications]

Improving the understandability of speech synthesis by modeling speech in noise

An automatic prosody recognizer using a coupled multi-stream acoustic model and a syntactic-prosodic language model

F0 control characterization by perceptual impressions on speaking attitudes using multiple dimensional scaling analysis

Additive modeling of English F0 contour for speech synthesis

Prosody analysis and modeling for emotional speech synthesis

Sliding window smoothing for maximum entropy based intonational phrase prediction in Chinese

Identification and synthesis of Cantonese tones based on the command-response model for F/sub 0/ contour generation

Compression of exception lexicons for small footprint grapheme-to-phoneme conversion

Prediction of pronunciation variations for speech synthesis: a data-driven approach

Recording script design for corpus-based TTS system based on coverage of various phonetic elements

Optimal subset selection from text databases

Comparative study of automatic phone segmentation methods for TTS

Increased robustness against bit errors for distributed speech recognition in wireless environments

"Of all things the measure is man" automatic classification of emotions and inter-labeler consistency [speech-based emotion recognition]

Disordered speech evaluation using objective quality measures

Meta-classifiers in acoustic and linguistic feature fusion-based affect recognition

Packet loss concealment based on VQ replicas and MMSE estimation applied to distributed speech recognition

A comparison of soft-feature distributed speech recognition with candidate codecs for speech enabled mobile services

A hidden trajectory model with bi-directional target filtering: cascaded vs. integrated implementation for phonetic recognition

A comparison of classifiers for detecting emotion from speech

Soft decoding of temporal derivatives for robust distributed speech recognition in packet loss

DBN-based multi-stream models for Mandarin toneme recognition

Sparse KPCA for feature extraction in speech recognition

Effects of phoneme characteristics on TEO feature-based automatic stress detection in speech

Scalable concatenative speech synthesis based on the plural unit selection and fusion method

Adaptive training for hidden semi-Markov model [speech synthesis applications]

Perceptually weighted long term modeling of sinusoidal speech amplitude trajectories

Speech recognition in the blind condition based on multiple directivity patterns using a microphone array

An unsupervised quantitative measure for word prominence in spontaneous speech

Speech modelling based on generalized Gaussian probability density functions

Bayesian model based non-intrusive speech quality evaluation

Robust pitch estimation at very low SNR exploiting time and frequency domain cues

Fundamental frequency estimation and vocal tremor analysis by means of Morlet wavelet transforms

Automatic speech segmentation using average level crossing rate information

DWT-based phonetic groups classification using neural networks

A novel KLT algorithm optimized for small signal sets [speech processing applications]

Voicing-state classification of co-channel speech using nonlinear state-space reconstruction

Speech rate estimation via temporal correlation and selected sub-band correlation

Closely coupled array processing and model-based compensation for microphone array speech recognition

Context dependent duration modeling [speech recognition applications]

Recognising speech in the presence of a competing speaker using a 'speech fragment decoder'

An environment compensated maximum likelihood training approach based on stochastic vector mapping [speech recognition applications]

Effect of phase-sensitive environment model and higher order VTS on noisy speech feature enhancement [speech recognition applications]

Towards speech recognition oriented dereverberation

Noisy speech recognition based on robust end-point detection and model adaptation

Analysis of a large in-car speech corpus and its application to the multimodel ASR

Building an effective corpus by using acoustic space visualization (COSMOS) method [speech recognition applications]

HMM/ANN based spectral peak location estimation for noise robust speech recognition

Acoustic feature combination for robust speech recognition

Acoustic training from heterogeneous data sources: experiments in Mandarin conversational telephone speech transcription

Dynamic match phone-lattice searches for very fast and accurate unrestricted vocabulary keyword spotting

A stream-weight optimization method for multi-stream HMMs based on likelihood value normalization

Lip reading for robust speech recognition on embedded devices

Novel techniques for time-compressing speech: an exploratory study

Fast two-stage vocabulary independent search in spontaneous speech

An HMM-based text segmentation method using variational Bayes approach and its application to LVCSR for broadcast news

Detecting group interest-level in meetings

Semantic data mining of short utterances

Automatic processing of audio lectures for information retrieval: vocabulary selection and language modeling

Blind change detection for audio segmentation

Combining multiple subword representations for open-vocabulary spoken document retrieval

Robust lip-motion features for speaker identification

Variational Bayesian feature saliency for audio type classification

Pitch-synchronous ZCPA (PS-ZCPA)-based feature extraction with auditory masking

MFCC compensation for improved recognition of filtered and bandlimited speech

Speech feature smoothing for robust ASR

On desensitizing the Mel-cepstrum to spurious spectral components for robust speech recognition

Two-stage noise spectra estimation and regression based in-car speech recognition using single distant microphone

Mask estimation based on sound localisation for missing data speech recognition

Speech processing using joint features derived from the modified group delay function

Influence of autocorrelation lag ranges on robust speech recognition

Subspace-based speaker-independent vowel recognition

Robust speech recognition based on spectral adjusting and warping

Robust speech activity detection using LDA applied to FF parameters

Joint discriminative language modeling and utterance classification

Language model estimation for optimizing end-to-end performance of a natural language call routing system

Language identification using phonetic and prosodic HMMs with feature normalization

Rapid language model development using external resources for new spoken dialog domains

Using local & global phonotactic features in Chinese dialect identification

Random clusterings for language modeling

Dialect/accent classification via boosted word modeling

Web-data augmented language models for Mandarin conversational speech recognition

An efficient algorithm for clustering short spoken utterances

Maximum entropy based generic filter for language model adaptation

Language identification using pitch contour information

Integrating multiple layers of concept information into n-gram modeling for spoken language understanding

Automatic language identification using ergodic-HMM

Discriminative power of transient frames in speaker recognition

Speaker identification in unknown noisy conditions - a universal compensation approach

Extracting additional information from Gaussian mixture model probabilities for improved text independent speaker identification

Combining selection tree with observation reordering pruning for efficient speaker identification using GMM-UBM

Advances in channel compensation for SVM speaker recognition

Improved speaker model migration via stochastic synthesis [speaker recognition applications]

Factor analysis simplified [speaker verification applications]

Minimum classification error interactive training for speaker identification [interactive robot applications]

A new common component GMM-based speaker recognition method

GMM-based Bhattacharyya kernel Fisher discriminant analysis for speaker recognition

A study of the relative importance of temporal characteristics in text dependent and text constrained speaker verification

Noise robust speaker verification using mel-frequency discrete wavelet coefficients and parallel model compensation

Initializing subspace constrained Gaussian mixture models

Multi-rate and variable-rate modeling of speech at phone and syllable time scales [speech recognition applications]

Optimal clustering and non-uniform allocation of Gaussian kernels in scalar dimension for HMM compression [speech recognition applications]

Hierarchical correlation compensation for hidden Markov models [speech recognition applications]

Cluster-dependent acoustic modeling [speech recognition applications]

Fuzzy parameter clustering method in speech recognition

Automatic training set segmentation for multi-pass speech recognition

Generalized statistical modeling of pronunciation variations using variable-length phone context

On initialization of Gaussian mixtures: a hybrid genetic EM algorithm

Acoustic model training using greedy EM

Modeling successive frame dependencies with hybrid HMM/BN acoustic model

Improved covariance modeling for maximum likelihood multiple subspace transformations [speech recognition applications]

A probabilistic measure of modality reliability in speaker verification

A correlation metric for speaker tracking using anchor models

Estimating and evaluating confidence for forensic speaker recognition

F-ratio client dependent normalisation for biometric authentication tasks

Clustering speech utterances by speaker using Eigenvoice-motivated vector space models

T-Norm for text-dependent commercial speaker verification applications: effect of lexical mismatch

A session-GMM generative model using test utterance Gaussian mixture modeling for speaker verification

ALIZE, a free toolkit for speaker recognition

Speaker adaptive cohort selection for Tnorm in text-independent speaker verification

Hybrid speaker-based segmentation system using model-level clustering

Robustness of bit-stream based features for speaker verification

Two-way cluster voting to improve speaker diarisation performance

Speaker detection without models

Improving the 2.4 kb/s military standard-MELP (MS-MELP) coder using pitch-synchronous analysis and synthesis techniques [speech coding]

Ultra low bit rate speech coding using an ergodic hidden Markov model

Towards iLBC speech coding at lower rates through a new formulation of the start state search

A missing-data approach to noise-robust LPC extraction for voiced speech using auxiliary sensors

A technique of multi-tap long term predictor (LTP) filter using sub-sample resolution delay [speech coding applications]

Voice activity detection based on generalized gamma distribution

Increasing the robustness of CELP-based coders by constrained optimization

Joint optimization of excitation parameters in analysis-by-synthesis speech coders having multi-tap long term predictor

Block-based bandwidth extension of narrowband speech signal by using CDHMM

Segmentation-based speech enhancement for intelligibility improvement in MELP coders using auxiliary sensors

Stochastic integration and long-term predictor estimation under noisy conditions for speech enhancement

A robust narrowband to wideband extension system featuring enhanced codebook mapping

Artificial bandwidth expansion method to improve intelligibility and quality of AMR-coded narrowband speech

A soft decision based noise cross power spectral density estimation for two-microphone speech enhancement systems

Lattice segmentation and support vector machines for large vocabulary continuous speech recognition

First steps in fast acoustic modeling for a new target language: application to Vietnamese

Cross domain automatic transcription on the TC-STAR EPPS corpus

Using rule-based knowledge to improve LVCSR

Adaptation strategies for the acoustic and language models in bilingual speech transcription

A study on knowledge source integration for candidate rescoring in automatic speech recognition

Development of the CUHTK 2004 Mandarin conversational telephone speech transcription system

Bayesian model combination (BAYCOM) for improved recognition

Investigation of acoustic modeling techniques for LVCSR systems

Improved confusion network algorithm and shortest path search from word lattice

Thai automatic speech recognition

Development of the CU-HTK 2004 broadcast news transcription systems

Cross-language acoustic model refinement for the Indonesian language

Analysis of spectral measures for voiced speech with varying noise and perturbation levels

Automatic dysphonia recognition using biologically-inspired amplitude-modulation features

Voiced/unvoiced determination of speech signal in noisy environment using harmonicity measure based on instantaneous frequency

SNR and local noise power estimations based on Gaussian mixture modeling on the log-power domain

Detection of symbolic gestural events in articulatory data for use in structural representations of continuous speech

Mathematical evidence of the acoustic universal structure in speech

Modeling of the front cavity and sublingual space in American English rhotic sounds

Objective quality measures for glottal inverse filtering of speech pressure signals

Effects of glottal and lip boundary conditions on vocal-tract area function estimates from speech signals

Adaptive filterbanks inspired by the auditory system for speech feature extraction

Multi-speaker articulatory reconstruction based on an eigen articulatory HMM

A graphical model for formant tracking

Dysphonic speech analysis using generalized variogram

Training wideband acoustic models using mixed-bandwidth training data via feature bandwidth extension

Minimum phoneme error based heteroscedastic linear discriminant analysis for speech recognition

A study of auditory modeling and processing for speech signals

A wavelet and filter bank framework for phonetic classification

Automatic syllable stress detection using prosodic features for pronunciation evaluation of language learners

Predicting formant frequencies from MFCC vectors [speech recognition applications]

Tonotopic multi-layered perceptron: a neural network for learning long-term temporal features for speech recognition

Towards an intelligent acoustic front-end for automatic speech recognition: built-in speaker normalization (BISN)

Quasi-continuous local codebook features for multilingual acoustic phonetic modelling

GARCH coefficients as feature for speech recognition in Persian isolated digit

fMPE: discriminatively trained features for speech recognition

Variational Bayesian adaptation for speaker clustering

Automatic disfluency removal on recognized spontaneous speech - rapid adaptation to speaker-dependent disfluencies

Aggregate a posteriori linear regression for speaker adaptation

Two-stage speaker adaptation of hybrid tied-posterior acoustic models

Various reference speakers determination methods for embedded kernel eigenvoice speaker adaptation

Kernel eigenspace-based MLLR adaptation using multiple regression classes [speech recognition applications]

Automatically transcribing meetings using distant microphones

A novel method for rapid speaker adaptation based on support speaker weighting

Adaptive training using simple target models [speech recognition applications]

Learning pronunciation and formulation variants in continuous speech applications

Alternate phone models for conversational speech

Whispery speech recognition using adapted articulatory features

Open vocabulary ASR for audiovisual document indexation

Constrained phrase-based translation using weighted finite-state transducers

Unsupervised vocabulary expansion for automatic transcription of broadcast news

Classification of structured descriptions

Maximum entropy segmentation of broadcast news

The AT&T WATSON speech recognizer

Open vocabulary Chinese name recognition with the help of character description and syllable spelling recognition

Error prediction in spoken dialog: from signal-to-noise ratio to semantic confidence scores

Incorporating dialogue context and topic clustering in out-of-domain detection

Structuring baseball live games based on speech recognition using task dependent knowledge and emotion state recognition

A new ASR evaluation measure and minimum Bayes-risk decoding for open-domain speech understanding

Speech recognition of a named entity

Automatic dialog act segmentation and classification in multiparty meetings

Sentence extraction-based presentation summarization techniques and evaluation metrics

Blind dereverberation based on estimates of signal transmission channels without precise information on channel order [speech processing applications]

Fast estimation of a precise dereverberation filter based on speech harmonicity

Codebook-based Bayesian speech enhancement

Overcoming the statistical independence assumption w.r.t. frequency in speech enhancement

A two-stage algorithm for enhancement of reverberant speech

Matrix quantization based time-varying filter speech enhancement

Leakage model and teeth clack removal for air- and bone-conductive integrated microphones

Speech enhancement using a MMSE short time spectral amplitude estimator with Laplacian speech modeling

Separation of fricatives and affricates

Speech enhancement based on filtering the spectrotemporal modulations

Improved Kalman filtering for speech enhancement

Adaptive decorrelation filtering algorithm for speech source separation in uncorrelated noises

An improved estimation of a priori speech absence probability for speech enhancement: in perspective of speech perception

Speech enhancement using a switching Kalman filter with a perceptual post-filter

2004 IEEE International Conference on Acoustics, Speech, and Signal Processing

2003 IEEE International Conference on Acoustics, Speech, and Signal Processing, 2003. Proceedings. (ICASSP '03).

2002 IEEE International Conference on Acoustics, Speech, and Signal Processing

2001 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.01CH37221)

2000 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.00CH37100)

1999 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings. ICASSP99 (Cat. No.99CH36258)

Proceedings of the 1998 IEEE International Conference on Acoustics, Speech and Signal Processing, ICASSP '98 (Cat. No.98CH36181)

1997 IEEE International Conference on Acoustics, Speech, and Signal Processing

1996 IEEE International Conference on Acoustics, Speech, and Signal Processing Conference Proceedings

1995 International Conference on Acoustics, Speech, and Signal Processing

Proceedings of ICASSP '94. IEEE International Conference on Acoustics, Speech and Signal Processing

1993 IEEE International Conference on Acoustics, Speech, and Signal Processing

[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing

[Proceedings] ICASSP 91: 1991 International Conference on Acoustics, Speech, and Signal Processing

International Conference on Acoustics, Speech, and Signal Processing

International Conference on Acoustics, Speech, and Signal Processing,

ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing

ICASSP '87. IEEE International Conference on Acoustics, Speech, and Signal Processing

ICASSP '86. IEEE International Conference on Acoustics, Speech, and Signal Processing

ICASSP '85. IEEE International Conference on Acoustics, Speech, and Signal Processing

ICASSP '84. IEEE International Conference on Acoustics, Speech, and Signal Processing

ICASSP '83. IEEE International Conference on Acoustics, Speech, and Signal Processing

ICASSP '82. IEEE International Conference on Acoustics, Speech, and Signal Processing

ICASSP '81. IEEE International Conference on Acoustics, Speech, and Signal Processing

ICASSP '80. IEEE International Conference on Acoustics, Speech, and Signal Processing

ICASSP '79. IEEE International Conference on Acoustics, Speech, and Signal Processing

ICASSP '78. IEEE International Conference on Acoustics, Speech, and Signal Processing

ICASSP '77. IEEE International Conference on Acoustics, Speech, and Signal Processing

ICASSP '76. IEEE International Conference on Acoustics, Speech, and Signal Processing

Minimum phoneme error based heteroscedastic linear discriminant analysis for speech recognition

Content Provider	IEEE Xplore Digital Library
Author	Bing Zhang Matsoukas, S.
Copyright Year	2005
Description	Author affiliation: BBN Technol., Cambridge, MA, USA (Bing Zhang)
Abstract	We introduce a discriminative feature analysis method that seeks to minimize phoneme errors in lattice-based training frameworks. This technique, referred to as minimum phoneme error heteroscedastic linear discriminant analysis (MPE-HLDA), is shown to be more robust than traditional LDA methods in high dimensional spaces, and easy to incorporate with existing training procedures, such as HLDA-SAT and discriminative training of hidden Markov models (HMMs). Results on conversational telephone speech and broadcast news corpora also show that the recognition accuracy is improved using features selected by MPE-HLDA.
Sponsorship	IEEE Signal Processing Soc
File Size	288548
File Format	PDF
ISBN	0780388747
ISSN	15206149
DOI	10.1109/ICASSP.2005.1415266
Language	English
Publisher	Institute of Electrical and Electronics Engineers, Inc. (IEEE)
Publisher Date	2005-03-23
Publisher Place	USA
Access Restriction	Subscribed
Rights Holder	Institute of Electrical and Electronics Engineers, Inc. (IEEE)
Subject Keyword	Linear discriminant analysis Speech recognition Speech analysis Hidden Markov models Maximum likelihood estimation Robustness Telephony Broadcasting Control systems Analysis of variance
Content Type	Text
Resource Type	Article

Central Library (ISO-9001:2015 Certified)
Indian Institute of Technology Kharagpur
Kharagpur, West Bengal, India | PIN - 721302

See location in the Map
03222 282435
Mail: support@ndl.gov.in

Sl.	Authority	Responsibilities	Communication Details
1	Ministry of Education (GoI), Department of Higher Education	Sanctioning Authority	https://www.education.gov.in/ict-initiatives
2	Indian Institute of Technology Kharagpur	Host Institute of the Project: The host institute of the project is responsible for providing infrastructure support and hosting the project	https://www.iitkgp.ac.in
3	National Digital Library of India Office, Indian Institute of Technology Kharagpur	The administrative and infrastructural headquarters of the project	Dr. B. Sutradhar bsutra@ndl.gov.in
4	Project PI / Joint PI	Principal Investigator and Joint Principal Investigators of the project	Dr. B. Sutradhar bsutra@ndl.gov.in Prof. Saswat Chakrabarti will be added soon
5	Website/Portal (Helpdesk)	Queries regarding NDLI and its services	support@ndl.gov.in
6	Contents and Copyright Issues	Queries related to content curation and copyright issues	content@ndl.gov.in
7	National Digital Library of India Club (NDLI Club)	Queries related to NDLI Club formation, support, user awareness program, seminar/symposium, collaboration, social media, promotion, and outreach	clubsupport@ndl.gov.in
8	Digital Preservation Centre (DPC)	Assistance with digitizing and archiving copyright-free printed books	dpc@ndl.gov.in
9	IDR Setup or Support	Queries related to establishment and support of Institutional Digital Repository (IDR) and IDR workshops	idr@ndl.gov.in