NDLI: Arithmetic performane of floating point formarts available in VLSI

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.

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

Volume 11 - 1986

Digital audio magnetic recording: Progress in packing density and key technology

Perpendicular magnetic recording and its application to digital recording

Error correction method for R-DAT and its evaluation

Efficient data reduction for digital audio using a digital filter array

Digital data equalizing in multitrack digital audio recording

A comparative study of the proposed high quality coding schemes for digital music

Permissible value of group delay distortion on tone quality due to low-pass filters

A study on the digitization of audio signals for video tape recorder

An audio sampling frequency conversion using digital signal processors

An approach to high resolution D/A converter utilizing a linear predictive coding

Mixture autoregressive hidden Markov models for speaker independent isolated word recognition

Markov modeling of continuous parameters in speech recognition

Maximum mutual information estimation of hidden Markov model parameters for speech recognition

An IBM PC based large-vocabulary isolated-utterance speech recognizer

A syllable-based isolated word recognition experiment

Quantitative knowledge on word structure, from a phonetic corpus, with application to large vocabularies recognition systems

Word preselection for large vocabulary speech recognition

Compact isolated word recognition system for large vocabulary

Coarse classification using a hierarchical decision tree and top down parsing

Incremental network generation in word recognition

A computational model for separating two simultaneous talkers

Evaluation of multisensor speech input for speech recognition in high ambient noise

Enhancement of noisy speech by forward/Backward adaptive digital filtering

Performance characteristics of a hardware implementation of the cross-talk resistant adaptive noise canceller

Acoustic noise reduction by two dimensional spectral smoothing and spectral amplitude transformation

Speech enhancement using multi-pulse excited linear prediction system

Waveform substitution techniques for recovering missing speech segments in packet voice communications

Fast and accurate pitch detection using pattern recognition and adaptive time-domain analysis

Pitch detection using the short-term phase spectrum

Frequency identification by complex spectrum

An adaptive non-uniform sign clipping preprocessor (ANUSC) for real-time autocorrelative pitch detection

A high quality 9.6 kbps speech coding system

Adaptive discrete cosine transform image coding using gain/Shape vector quantizers

Vector quantization of color images

Low-rate image coding with finite-state vector quantization

A gain-shape vector quantizer for image coding

Vector quantization in transformed image coding

A technique using a one-dimensional mapping for vector quantisation of images

A hybrid coding involving ADM and vector quantization for digital video image compression

The performance of a hybrid videoconferencing coder using displacement estimation in the transform domain

Contour based representation of the displacement field for motion compensated image coding

Hybrid interframe coding of video signals with backward-acting motion detection

High-resolution spectral estimation: Rethinking the Fourier transform

Single vector approaches to eigenstructure analysis for harmonic retrieval

Estimation of frequencies of sinusoids in colored noise

High resolution spectral estimation of sinusoids in colored noise using a modified Pisarenko decomposition

Spectral and parameter estimation problems arising in the metrology of high performance mirror surfaces

On the use of linear programming for spectral estimation

A Toeplitz approximation approach to coherent source direction finding

Performance of multichannel autoregressive spectral estimators

Linear predictive spectral analysis via the Lpnorm

Refined ARMA digital lattice filter

ARMA parameter estimation based on sample covariances, for missing data

A new FFT approach to the interpolation of discrete-time signals

Fast algorithms for the discrete Fourier transform and for other transforms

Performance of fixed-point FFT's: Rounding and scaling considerations

A fast Fourier transform algorithm using Hadamard transform

Cyclic convolution of real sequences: Hartley versus Fourier and new schemes

Computation of complex number theoretic transforms using quadratic residue number systems

A hardware efficient realisation of number theoretic convolvers

Half-Fourier transform and application to radar signals

Optimal design of multiplierless DFTS and FFTS

An efficient vector implementation of the FFT algorithm on IBM 3090VF

Immitance-domain Levinson algorithms

Efficient multi-processor implementation of recursive digital filters

A unified derivation of the fast RLS algorithms

A fast codebook search algorithm for nearest-neighbor pattern matching

The continuous-time limit of the discrete-time stability theory

Maximum entropy deconvolution

L1-norm noisy Tauberian deconvolution

A new homomorphic deconvolution system

Generalized Lanczos method for signal smoothing

FFT and convolution algorithms on DSP microprocessors

A unified transform architecture

The array matrix generalization for signal processing

A fast B-spline transform and its applications

A generalization of Levinson algorithm for solving Toeplitz systems

A demiphoneme network representation of speech and automatic labeling techniques for speech data base construction

A speech data base at the united states air force academy

A PCM/VCR speech database exchange format

A Japanese language speech database

B.D.L.E.X. : A data and cognition base of spoken French

The development of the MIT Lisp-machine based speech research workstation

A microprocessor-based word recognition system for large vocabularies

Signal processor application to voice dialing equipment

The analog voice privacy system

Development of a 16-kb/s speech codec for telephone applications

24-channel 32 kb/s ADPCM transcoder using the CCITT recommendation G.721

A special architecture for dynamic programming

A modular architecture for dynamic programming and maximum likelihood sequence estimation

Speech recognition on the DADO/DSP multiprocessor

A new systolic decomposition for the dynamic time warping algorithm

VDP : A versatile high performance vector distance processor

Design of an efficient dynamic time warping LSI

A single-chip speaker independent voice recognition system

VLSI Architecture for a real-time LPC-based feature extractor

On the IC architecture and design of a 2 µm CMOS 8 MIPS digital signal processor with parallel processing capability: The PCB5010/5011

VLSI Digital signal processor (PSI)

Architecture and applications of a 100-ns CMOS VLSI digital signal processor

A high performance microprocessor for DSP applications

A 50ns floating-point signal processor VLSI

Architecture of high-speed 22-bit floating-point digital signal processor

Advanced single-chip signal processor

A next-generation 32-bit VLSI signal processor

Architectural considerations for a sub 10 nanosecond DSP building block family

The architecture, instruction set and development support for the $WE^{®}DSP32$ digital signal processor

Digital signal processing in a 16kbps APC-AB codec by fixed point digital signal processor (FDSP-3)

A high performance digital voice echo canceller on a single TMS32020

Hidden Markov models applied to very low bit rate speech coding

Design and performance of trellis vector quantizers for speech signals

Joint time-spectral vector quantization and inverse filter set

Fully vector-quantized multipulse LPC at 4800 bps

Low-rate speech encoding using vector quantization and subband coding

The self excited vocoder - an alternate approach to toll quality at 4800 bps

Low bit rate multi-pulse speech coder with natural speech quality

A novel LPC synthesis model using a binary pulse source excitation

High quality glottal LPC-vocoding

On the behaviour of reduced complexity code-excited linear prediction (CELP)

Voiced/Unvoiced classification of speech with applications to the U.S. government LPC-10E algorithm

Optical circuitry and architectures for digital optical computing

Optical parallel array logic system

Use of optical signal processing techniques to spectrum analysis of speech

Optical signal processors consisting of collinear acousto-optic channel waveguides

Reproduction of the sounds from old wax phonographic cylinders using the laser-beam reflection method

New cascaded lattice structures for FIR filters having extremely low coefficent sensitivity

Synthesis of minimum sensitivity structures in linear systems using controllability and observability measures

Embedded max quantization

Nonlinear quantization and data communication

Floating point error analysis of recursive least squares and least means squares adaptive filters

Design of limit-cycle-free digital biquad filters

Predictive coding of multi-viewpoint image sets

An architecture for the universal video codec

A fractal based approach to image compression

Bit-sliced progressive transmission and reconstruction of transformed images

A binary representation of mixed documents (text/Graphic/Image) that compresses

A representation method of color pictures by arranging fixed size dots of the primary colors

Design of 3-D IIR filters via transformations of 2-D circularly symmetric rotated filters

A computer program for designing optimum 2-D FIR linear phase digital filters

An algorithm for the design of optimal finite wordlength 2-D FIR digital filters

Reduction of the number of parameters in the design of 2-D digital filters

On the stability test for 2-D digital recursive filters

A fast image filtering processor-FIFP

Lower bounds in parameter estimation - summary of results

New forms of least squares lattice algorithms and a comparison of their round-off error characteristics

Effective adaptive Pisarenko spectrum estimate

Determining MA models as salvos of pulses

On the limiting behavior of estimates based on sample covariances

Performance of narrowband signal-subspace processing

Phase-strip sequence estimation: A phase unwrapping/Tracking algorithm

A new functional method of parameter estimation for the energy acoustic system based on an Lxevaluation criterion

Estimation of anti-resonance frequencies by using an over-determined high-order Yule-Walker equation

A new methodological trial on state estimation of linear structure vibration model with noisy power observation mechanism of non-Gaussian type

Asymptotic properties of high-order Yule-Walker estimates of frequencies of multiple sinusoids

Assessment of speech discrimination ability of hearing impaired subjects using FIR digital filter

A speech processor with lateral inhibition for an eight channel cochlea implant and its evaluation by subject testing

Sound processing for cochlear implant

A wearable, pocket-sized processor for digital hearing aid and other hearing prostheses applications

Assessment of electronic aids for the hearing impaired which transmit visible or tactile speech

Speech training devices for profoundly deaf children

Speech training systems for handicapped children using vocal tract lateral shapes

Speech-analysis-based devices for diagnosis and education of speech and hearing impaired people

Refutation based recognition to help vowel articulation

Memory-intensive recognition for word articulation training

Speech control of assistive devices for the physically disabled

Integrated communication aids for the blind

Japanese speech synthesis system in a book reader for the blind

A lexical prediction system

An adaptive comb filtering method as applied to acoustic analyses of pathological voice

Multidimensional analysis of phonological degeneration in pathological voices

Acoustic measurement of pathological voice qualities for medical purposes

A microcomputer based system for acoustic analysis of voice characteristics

Wave-flow index: A measure of vocal efficiency

Electroglottography: Assessing articulatory function to aid the handicapped

Analysis and classification of snoring signals

Speaker independent isolated Arabic word recognition system

Isolated-word recognition of the complete vocabulary of spoken Chinese

On hidden Markov models in isolated word recognition

Large-vocabulary isolated word recognition with fast coarse time alignment

A proposal of a knowledge based isolated word recognition

Isolated word recognition based on finite-state vector quantization

Speech processing with a Boltzmann machine

Simulating an acoustic recognizer

Experiments in speech recognition over the telephone network

Recognition of speech under stress and in noise

Word recognition using multisensor speech input in high ambient noise

Noise compensation for speech recognition using probabilistic models

Optimal and suboptimal training strategies for automatic speech recognition in noise, and the effects of adaptation on performance

Improved speech recognition in noise

Clustering acoustic prototypes with self organizing distortion measures

Spectral slope based distortion measures for all-pole models of speech

A weighted cepstral distance measure for speech recognition

On the use of bandpass liftering in speech recognition

Comparative study of various spectrum matching measures on noise robustness

An adaptive filtering approach for speaker independent vowel recognition

Systolic image processor for automatic detection and recognition of traffic violations

Image processing on linear transputer arrays

Data flow chip ImPP and its system for image processing

Construction methods and performance evaluation for image processing systems with ISP LSIs

Architecture of a real-time video signal processor

Video signal processor configuration by multiprocessor approach

A custom chip set for real-time image processing

A single chip video rate 16×16 discrete cosine transform

Video signal processing LSI and its application to TV CODEC

A study of VLSI design for DPCM coding

CCITT Standardizing activities on speech coding

An advanced 32 kbit/s ADPCM coding to transmit speech and high-speed voiceband data

A real-time ADPCM encoder using variable order prediction

Adaptive postfiltering of 16 kb/s-ADPCM speech

Amplitude normalization and its application to speech coding

Range extended speech decoder for Modulo-PCM

Considerations on quantization and dynamic bit-allocation in subband coders

16 kbit/s Split-band APC coder using vector quantization and dynamic bit allocation

Adaptive predictive coding with dynamic quantization adjustment (APC-DQA) at 16 kbits/s

Performance of 16 kbit/s APC codec with maximum likelihood quantization for low-speed voiceband data

A robust 16 kbits/s vector adaptive predictive coder for mobile communications

Language identification using noisy speech

Methods and experiments for text-independent speaker recognition over telephone channels

A new method of text-independent speaker recognition

Evaluation of a vector quantization talker recognition system in text independent and text dependent modes

On the use of instantaneous and transitional spectral information in speaker recognition

High performance speaker verification using principal spectral components

The effects of voice disguise upon formant transition

Contributions of pitch, formant frequency and bandwidth to the perception of voice-personality

The effect of formant trajectory and spectral shape on the tense/Lax distinction in American vowels

A pitch perception model

Comparative evaluation of the speech quality of speech coders and text-to-speech synthesizers

Acoustical transformation technique as a practical tool

Grasp and development of spatial informations in a room by closely located four-point microphone method

Acoustic noise cancellation

Inverse control of room acoustics using multiple loudspeakers and/Or microphones

Modelling of reverberations and audioconference rooms

Extraction of fundamental physical parameters in the sound field by multidimensional analysis

Impulse response measurement using Golay codes

A new method to locate sound sources by searching the minimum value of error function

Headphone responses on real ears and a head and torso simulator

Pre-howling howlback detection method

Octave and fractional octave band digital filtering based on the proposed ANSI standard

Data equalization based on the constant modulus adaptive filter

The least-squares CMA: A new technique for rapid correction of constant modulus signals

An adaptive filter convergence method for echo cancellation and decision feedback equalization

Convergence of an adaptive echo cancellation system with an augmented predictor

An adaptive echo canceller using parallel Kalman filters

Adaptive rate conversion filter

Interference cancellation for enhanced detection of frequency hopped signals

Noise cancellation for hearing aids

Adaptive optimization of microphone arrays under a nonlinear constraint

Adaptive discrete cosine transform coding with vector quantization for color images

Motion compensated vector quantization

A novel method for constructing a set of Hadamard matrices

A nearly optimal transformation

Two-dimensional transform domain decimation techniques

Sub-band coding of images

Contour coding of images

An adaptive state-prediction digital facsimile coding technique

Proposal of line element chain coding

Two alternatives for motion compensated coding of composite NTSC video signal

2-D Multilevel FIR-median hybrid filters

High speed delayed multipath 2-D digital filter structures

Optimized shift-varying filters for image interpolation

New interpolatory approximation method with application to the design of multi-dimensional FIR filters

2D Spectral factorization and stability test for 2D matrix polynomials based on the radon projection

A unified study on the roundoff noise in 2-D state space digital filters

A generalized state-space model for noncausal 2-D systems

On compression of multi-level document images

3-D Lattice predictive modeling of random fields

Theory and realization of M-D nonlinear digital filters

A continuous training procedure for connected digit recognition

An efficient algorithm for combining vector quantization and stochastic modeling for speaker-independent speech recognition

Connected digit speech recognition by using demi-word pair reference pattern (DWPR)

A speaker independent recognition algorithm for connected word using word boundary hypothesizer

Global connected digit recognition using Baum-Welch algorithm

Fast feature-based preclassification of segments in continuous digit recognition

Lexical access and verification in a broad phonetic approach to continuous digit recognition

Speaker-independent recognition of connected Swedish digits

Network-based connected digit recognition using explicit acoustic-phonetic modeling

Connected word recognition by overlap and split of reference patterns and its performance evaluation tests

Two novel algorithms for variable frame analysis and word matching for connected word recognition

A new network-based speaker-independent connected-word recognition system

New DP matching algorithms for connected word recognition

Syllable-based connected spoken word recognition by two pass O(n) DP matching and hidden Markov models

Voice-activated word processor with automatic learning for dynamic optimization of syllable-templates

Japanese text input system based on continuous speech recognition

A "Bunsetsu" speech recognition system using a top-down processor controlled by bottom-up information

Speech understanding system using knowledge engineering techniques

A linguistic processor for Japanese continuous speech recognition

A grammatical approach to reducing the statistical sparsity of language models in natural domains

N-gram driven search for sentences in a syntactic network

On the analysis of systolic arrays

Implementations of arbitrarily fast adaptive lattice filters with multiple slow processing elements

A systolic discrete Fourier transform using residue number systems over the ring of Gaussian integers

A high speed CMOS/SOS implementation of a bit level systolic correlator

Systolic implementation for deconvolution iterative algorithm

Doubly pipelined Cordic array for digital signal processing algorithms

High speed signal processing, pipelining, and VLSI

A dataflow algorithm for digital filtering

A round-robin architecture for relaxation operation

A parallel architecture for recursive least square identification

Fault-tolerant digital filtering structures for wafer scale VLSI

An expert system for speech spectrogram reading

An expert spectrogram reader: A knowledge-based approach to speech recognition

APHODEX, design and implementation of an acoustic-phonetic decoding expert system

A representational approach to knowledge-based acoustic-phonetic processing in speech recognition

Producing and organizing phonetic knowledge from acoustics facts in multi-level data-information

A hybrid recogniser for speech patterns

Plan refinement in a knowledge-based system for automatic speech recognition

A continuous speech recognition system based on knowledge engineering techniques

A family of formant trackers based on hidden Markov models

A new algorithm for estimation of formant trajectories directly from the speech signal based on an extended Kalman-filter

A background for sinusoid based representation of voiced speech

A speech waveform analysis and reconstruction process based on non-euclidean error minimization and matrix array procesing techniques

Continuously variable duration hidden Markov models for speech analysis

Analysis of speech signals of short pitch period by the sample-selective linear prediction

Efficient implementation of the multipulse LPC method for speech

A speech analysis-synthesis system based on the ARMA model and its evaluation

Analysis of time fluctuating characteristics of linear predictive coefficients

Duality theory of composite sinusoidal modeling and linear prediction

Optimal speech data reduction using a stochastic realization algorithm

Musical sound models for digital synthesis

A signal source in a digital musical synthesizer

Transcription of sung song

A study on annoyance of musical signal using LAeq measurement and digital signal processing

Model analysis of a hammer-string interaction

Source separation and note identification in polyphonic music

Piano tone synthesis using digital filters by computer simulation

Digital and analog bows: Hybrid mechanical-electrical systems

Hardware implementation of FM multipath distortion canceller

Multipath echo cancellation in FM transmission using adaptive FIR filters

Echo return loss required for acoustic echo canceller based on subjective assessment

Dynamically-reduced complexity implementation of echo cancelers

An acoustic echo canceller for teleconference systems

Stereophonic speech teleconferencing

A new adaptive summing technique for audio teleconferencing

Stability and insertion loss required for audio conference bridges

An integrated voice codec and echo canceller implemented in a single DSP processor

Decision-directed echo cancellation for full-duplex data transmission at 4800bps

A system design for real time signal processing

Spectral analysis using the QD algorithm

A new series of digital filter windows B-spline windows

Spectral analysis of order statistic filters

Spectral analysis for speech signals with bounded disturbance

Analysis of a moving sound source - compensation of the Doppler effect

Parallel AR computation with a reconfigurable signal processor

A noise-compensated long correlation matching method for AR spectral estimation of noisy signals

Subsets of reflection coefficients

Maximum-entropy spectrum from a non-extendable autocorrelation function

Accurate estimation of transfer function of all-pole model system from the signal buried in noise

New relationships between ARMA and AR processes

Maximum entropy pole-zero estimation

Prony's method based on eigenanalysis and overdetermined system approach

Improved frequency estimation using total least square approach

Adaptive algorithms for estimating the complete covariance eigenstructure

Some results on constrained maximum likelihood estimation

"Fast maximum likelihood joint estimation of frequency and frequency rate"

Estimation of partitioned set of parameters

Multidimensional spectral analysis and the decomposition of ionospheric mode structures

Complexity of waveform formulated from Fourier spectrum -- Theory and experiment

On estimation with bilinear time series

Adaptive spectrum estimation by including prior information into the window method

Describing greylevel textures through curvature primal sketching

Partitioning of texture image using two-dimensional linear prediction model

Automatic classification and recognition of defects in an eddy current non destructive testing

Feature extraction by uniform structure threshold logic networks

Non-stationary 2D estimation: An application to turbulence field analysis

An automatic visual inspection method of pattern using a circular feature extraction filter

Silhouette understanding system

Image recognition with inexact processing

Edge detection using zero crossings of directional derivatives of a random field model

Fourier analysis of object boundaries from two dimensional digitized images

The use of context in image restoration

Automatic classification of zooplancton by image analysis

An iterative method for restoring noisy images

Regularized iterative image restoration in a weighted Hilbert space

A parallel identification procedure for images with noncausal symmetric blurs

Karhunen-Loeve transform-based fast algorithms for image restoration

A fast local PPF restoration filter

Two-dimensional target recognition using normalization and recorrelation to eliminate noise

Restoration theory for digital images under additive noises by a constrained generalized inverse matrix

A projection operator for reconstruction images from sparse, ungridded data

Least square method for enhancement of laser radar images based on piecewise linear transformations of gray scales

A real-time digital mixer IC

A single-chip SC line equalizer system for full duplex multi-bit rate digital transmission

VLSI Architecture for an adaptive equalizer in ISDN line termination

An adaptive transversal filter VLSI

FIR Lowpass filter for signal decimation with 15 MHz clock frequency

Application specific integrated filters for HIFI digital audio signal processing

Experiences with automatic generation of audio band digital signal processing circuits

VLSI- A to D and D to A converters with multi-stage noise shaping modulators

A dynamic time warp VLSI processors for continuous speech recognition

System design for large-vocabulary speech recognition LSI

A wafer scale DTW multiprocessor

Syllable-based acoustic-phonetic decoding and wordhypotheses generation in fluently spoken speech

Generating word hypotheses in continuous speech

An efficient word lattice parsing algorithm for continuous speech recognition

A distributed system architecture for speech recognition

A speech recognition system of continuously spoken Japanese sentences and an application to a speech input device

Representation of a continuous speech understanding and dialog system in a homogeneous semantic net achitecture

A new architecture of a speech understanding system - A hybrid of a hierarchical and a network models

A network model dealing with focus of conversation for speech understanding system

The role of word-dependent coarticulatory effects in a phoneme-based speech recognition system

A continuous parameter and frequency domain based Markov model

Analysis, synthesis and perception of the French nasal vowels

Proposal and evaluation of models for the glottal source waveform

On the source-filter model of the vocal tract

Area estimation from ARMA analysis based on vocal-tract model

A model for nonstationary analysis of speech

Measurements on the effects of glottal opening and flow on the glottal impedance

Measurement and analysis of speech sound radiated from vocal tract wall

Effects of tempo and context on jaw openings for vowel in vowel sequence words

Simultaneous high-speed digital recording of vocal fold vibration and speech signal

A hierarchical classification of signals and corresponding approximation method based on minimum norm criterion

Effects of representative distortion and noise on wavelet recovery via the complex cepstrum

Performance and limitation of discrete band-limited signal extrapolation

Signal recovery from nonuniform samples and spectral analysis on random nonuniform samples

Spline fitting in presense of uniform noise

A unified approach to generalized sampling theorems

Integer programming approach to optimal smoothing of two-state Markov sequences

A time-domain signal resolution problem

Discrete signal reconstruction from its spectral magnitude and some samples

Ignorance-based signal estimation given multiple noisy realizations

An algorithm for bandlimited signal interpolation

High-quality speech at low bit rates: Multi-pulse and stochastically excited linear predictive coders

CELP Coding for high-quality speech at 8 kbit/s

High quality multi-pulse speech coder with pitch prediction

Vector adaptive predictive coding of speech at 9.6 kb/s

8 kbits/s Speech coder with pitch adaptive vector quantizer

Speech coder using phase equalization and vector quantization

Time-scale modification in medium to low rate speech coding

A study on the realtionships between stochastic and harmonic coding

Phase modelling and its application to sinusoidal transform coding

Single DSP 8kbps speech CODEC

A high-efficiency speech coding algorithm based on ADPCM with multi-quantizer

Rapid magnetic resonance imaging using the Hankel tranform approach

Consistency in tomographic reconstruction by iterative methods

Image reconstruction from truncated projections: A linear prediction approach

High speed image reconstruction using direct two-dimensional Fourier transform method

3-Dimensional reconstructing algorithm for digital tomo-synthesis

Detector width artifact correction in computer assisted tomography

Maximum entropy diffraction tomography

Constrained resolution enhancement in optical microscopic tomography

Blood flow phase detection on digital subtraction angiography

Computation of synthetical radiographs application to the analysis of high velocity flows

An automatic analyzing system of left ventricular cineangiograms

Signal restoration of distorted kinesiographic measurement

Ultrasonic beamwidth effect on evaluation of attenuation by sampling medical B-mode imaging data

Measurement methods of ultrasonic velocity by dip point analysis from echo spectrum and ultrasonic attenuation considering a dispersion

One approach to color image processing

Compensation of motion related blur in CCD color endoscope images

A semi-automatic system for schistosome eggs detection

Patterns in Hilbert transforms and Wigner-Ville distributions of electrocardiogram data

Novel method for real-time detection of QRS complex in the presence of strong noise

New signal processing methods applied in ECG microcomputer automatic diagnosis system recently designed

Estimation of intramuscular EMG signals from surface EMG signal analysis

A nonlinear digital filter for separating nonstationary waves from stationary waves of the EEG

Design and construction of a multiplexed electrode for monitoring evoked responses in cerebral cortex

A precise bearing estimation method of digital split-beam array system

A simple fast adaptive array based on a null steering beamformer

Noise influence on the accuracy of source localising, comparison of various receivers with an imposed achitecture

Multichannel lattice filter for an adaptive array processor with linear constraints

Eigenstructure approach to direction-of-arrival estimation in IR detector arrays

Processing technique for a linear FM pulse array

Bearing estimation with a linear array

An adaptive recursive algorithm for array processing of coherent signal

A novel compact beamformer for a high resolution sonar

A partitionable distributed beamformer and signal conditioner architecture

A digital signal processing approach to data handling for ultrasound beam-forming

The integrator as a time delay estimator

Frequency domain compensation for failed elements in linear sonar arrays

Eigenstructure based broadband source location estimation

Toeplitzization of correlation matrix in multipath environment

High frequency acoustic propagation in a range dependent guiding-to-antiguiding ocean channel transition

The application of linear constraints to an adaptive beamformer

Source localization with two dimensional array subject to uncertainty in sensor location

The effect of the character of the noise field upon the error in the optimum gain of an antenna array

Time delay estimation via generalized correlation with adaptive spatial prefiltering

Analysis of a moving sound source-Orbit estimation using linear sensor array

Characterization and identification of trajectories using a leg by leg approach of bearings only target motion analysis

On smoothed rank profile tests in eigenstructure approach to directions-of-arrival estimation

Performance analysis of the MUSIC algorithm

A fast and accurate algorithm for SAR Doppler center frequency estimation

Eigenstructure approach to Doppler estimation for wideband signals

Complex deconvolution in non-coherent radar system

Constraint optimization filter design for radar target enhancement

Iterative enhancement of noncoherent radar data

Group delay processing of radar signals

Resolution of coherent radar targets in the near field of an array antenna

A fast algorithm for evaluating the false alarm probability of a CFAR detector

Fundamental limits on multiple PRF radar performance

A microprocessor controlled CCD programmable transversal filter for radar applications

An improvement of phase correction for spotlight mapping radar

A model of peripheral auditory transduction using a phase vocoder with modified channel signals

A new approach to continuous speech recognition based on considerations on human processes of speech perception

How to make more efficient use of the fact that the speech signal is dynamic and redundant

Toward a massively parallel system for word recognition

Perceptually based processing in automatic speech recognition

Experiments with a computational model of the cochlea

Speech recognition using a cochlear model

A computational model for the peripheral auditory system: Application of speech recognition research

Representation of vowels in connected speech based on speech perception experiments

Speaker-independent isolated word recognition based on emphasized spectral dynamics

Speech analysis/Synthesis based on matching the synthesized and the original representations in the auditory nerve level

A nonlinear articulatory speech synthesizer using both time- and frequency-domain elements

Dynamically controlled excitation source for a time-varying format synthesizer

A modified-superposition speech synthesizer and its applications

Improved vocal tract models for speech synthesis

Diphone synthesis using an overlap-add technique for speech waveforms concatenation

The speech synthesis system for an unlimited Japanese vocabulary

Acoustic parameters and phonological rules of a text-to-speech system for Chinese

The effects of speaking rate on formant transitions in French synthesis-by-rule

The Japanese CV-Syllable positioning rule for speech synthesis

Microprosodic timing rules for Italian consonant clusters

Acoustic characteristics and the underlying rules of intonation of the common Japanese used by radio and television announcers

Pitch contour control in Japanese conversational speech

Multi-scale edge detection with Gaussian filters

A unified method for segmentation and edge detection using graph theory

Detection of object boundaries in synthetic aperture radar imagery using a human visual model

Contour recognition: A syntax-directed approach

Fast Hough transform for multidimensional signal processing

Applications of morphological filtering to image analysis and processing

Script recognition using hidden Markov models

Dynamic programming for region based pattern recognition

Constraint and consistency in stereo matching

A multi-resolution representation of 3D imagery for efficient image enhancement and processing

Iterative local parallel operations for structural analysis of line patterns using distance transformation

A new IIR type adaptive algorithm of parallel type structure

Derivation of the fast pole-zero (ARMA) recursive least squares algorithm using geometric projections

An adaptive IIR channel equaliser: A Kalman filter approach

Adaptive filters realized with second order sections

Bandpass adaptive pole-zero filtering

An adaptive stochastic filter with no positive real condition

A covariance RLS lattice for adaptive filtering

Rounding error analysis of the triangular lattice and escalator algorithms

An adaptive delay filter with varibale gains and variable taps

A flexible, numerically robust array processing algorithm and its relationship to the givens transformation

Numerical design of nonlinear adaptive filters

A high-speed programmable architecture using barrel shifters for FFT and digital filtering

Kalman filtering and systolic processors

A systolic vector quantization processor for real-time speech coding

A VLSI array for computing the DFT based on RNS

A single chip VLSI Reed-Solomon decoder

A WSI oriented two dimensional systolic array for FFT

Mapping system level functions on to bit level systolic arrays

A fast adder-based multiplication unit for customised digital signal processors

Two chip DSP implementation on gate array

A cascadable multimode VLSI signal processor

A high performance digital signal processor VLSI: MN 1900 series

A practical study on the architecture of VLSI DSP

An open architecture CMOS digital signal processor

A 32kbps ADPCM with improvement in coding characteristics for 9600bps modem signal

Wideband speech codec at 64kb/s using an advanced digital signal processor "DSSP-1"

Implementation of a new type DSP PLL using high performance DSP DSSP-1

A dynamic programming-based DUR recognizer implemented on analog devices' DSP

Efficient bit-serial complex multiplication and sum-of-products computation using distributed arithmetic

A comparison of micro-DSP and Silicon compiler implementations of a polyphase network filter bank

A bit-serial VLSI vector quantizer

Design and implementation of real time video processor

Frequency domain noise canceller: Frequency bin adaptive filtering (FBAF)

VLSI Implementation of real-time Kalman filter

Deciphering speech waveforms

A speech processing based on syllable identification by using phonological patterns

Definition and evaluation of phonetic units for speech recognition by hidden Markov models

Using hidden Markov models to define linguistic units

Phoneme recognition in connected speech using both static and dynamic properties of spectrum described by vector quantization

Estimation of articulatory parameters by table look-up method and its application for speaker independent phoneme recognition

Phoneme recognition in contiunous speech using phoneme discriminant filters

The C-MU phonetic classification system

A broad phonetic classifier

Steady part recognition of continuous speech for acoustic-phonetic decoding

On the determination of speech boundaries: A tool for providing anchor time points in speech recognition

Voiced/Unvoiced segmentation

Automatic segmentation of continuous speech signals

Interactive extraction of phonemic variation rules in continuous speech

Prosodic aids to structural analysis of conversational speech

Recognition of lexical stress in a continuous speech understanding system - A pattern recognition approach

Phonemic recognition of each frame with vector field feature using continuous dynamic programming

Syllable recognition for continuous Japanese speech recognition

VCV Segmentation and phoneme recognition in continuous speech

Top-down controlled and optimal syllable-template chaining algorithm for large vocabulary speech recognition

Word spotting method taking account of duration change characteristics for stable and transient parts of speech

On quick word spotting techniques

A stochastic process model of sound program signals

Envelope and instantaneous phase methods for the propagation of transients through oscillatory nonlinear systems

Wigner-Ville analysis of non-stationary random signals. (with application to turbulent microstructure signals)

The relationship between Wigner-Ville and evolutionary spectra for frequency modulated random processes

On detection-estimation procedures in the time-frequency plane

ARMA Modeling of time varying systems with lattice filters

Nonstationary ARMA models via simultaneous AR and MA estimation

Estimation of time-varying coherence functions

Spectral line tracking for nonstationary random processes

Spectrum enhancement using linear programming

Time series modeling for acoustic signals by a generalized adaptive function and its application to actual random noise data

Composite permutation coding of speech waveforms

Reducing computation in optimal amplitude multipulse coders

Implementation of a multi-pulse coder on a single chip floating-point signal processor

Low cost voice compression for mobile digital radios

Efficient procedures for finding the optimum innovation in stochastic coders

Magnitude-phase coding of base-band speech signals

Comparison between vector and scalar quantization in variable rate subband coders

A high quality subband speech coder with backward adaptive predictor and optimal time-frequency bit assignment

Linear transformation for low bit rate speech coding

An improved modulo-PCM speech encoder with an imbedded vector quantizer

Dynamic time-scale compression for waveform speech coding

Linguistic processing in the Kth multi-lingual text-to-speech system

A grammatical component for a text-to-speech system

Word identification method for Japanese text-to-speech conversion system

Generation of prosodic symbols for rule-synthesis of connected speech of Japanese

Multiparametric generation of French prosody from unrestricted text

Stress assignment in letter to sound rules for speech synthesis

English text to speech conversion with delta

Japanese text-to-speech synthesizer based on residual excited speech synthesis

Phoneme based text-to-speech synthesis system

A Chinese text-to-speech system based upon a syllable concatenation model

A diphone speech synthesis approach applicable to different languages

A new model of motion blurred images and estimation of its parameter

An application of median filters to digital television

Image deconvolution using 2-D non-causal fast Kalman filters

A binary image preprocessor for document quality improvement and data reduction

Image enhancement via transform amplitude sharpening

A multi-threshold adaptive filtering for image enhancement

FIR Anisotropic filters for image enhancement

A new recursive estimation method for digital image restoration filter and its applications

Lognormal random field models and their applications to radar image synthesis

Support-limited extrapolation of offset Fourier data

Iterative reconstruction of multidimensional signals from nonuniformly spaced samples

Proper orthogonal projection - multiple signal classification (POP-MUSIC)

Direction-of-arrival estimation by subspace rotation methods - ESPRIT

Adaptive interference and multipath rejection in linear direction-finding arrays

Multipath and multiple source array processing via the EM algorithm

Detection of acoustic signal source by use of cross spectra between every pair of sensors distributed in space

Hyperbolic householder transformations, definition and application

Multiple target detection - optimum resolution

Linear prediction for aperture extrapolation in line array beamforming

The CM array: An adaptive beamformer for constant modulus signals

A new approach to the design of robust narrow-band processors

Fresnel region imaging with a linear array

On the oscillatory roots of one and two dimensional median type filters

On the optimality of rank order operations

Theory of uniform DFT, parallel, quadrature mirror filter banks

Design of doubly-complementary IIR digital filters, using a single complex allpass filter

Bandwidth compression with periodically time-varying FIR filters

A new class of uniform filter banks based on recursive Nth-band filters

FIR Filters with minimum coefficient sensitivity

Spline filter--A time-varying digital filter and its application

Perfect transmultiplexers

An adaptive tracking filter controlling sampling frequency

Design of computationally efficient FIR filters for sampling rate alteration and multiband filtering with arbitrary passbands and time response

A unified design method of state-space digital filters using system balancing concept

The implementation of the generalized Lagrange FIR filter structure defined over finite fields or rings

Design of IIR group delay digital filters using the iterative near least-squares criteria

Efficient FIR, IIR, and hybrid nyquist filters with zero intersymbol interference

Antialiasing filters for continuously varying sampling rate conversion

A robust adaptive filter for noise reduction problems

Complex wave digital filters for the analysis of complex signals

Analysis of a periodically time-varying digital filter

Noncausal digital filters with antisymmetrical impulse response

Design of quadrature mirror filters by linear programming

A simple phase-reversal tone disabler

Techniques for realization of high-speed recursive digital filters using residue number system arithmetic

On the dimensionality of steady-state vowel normalization

Nonlinear frequency warp for speech recognition

Speech recognition using dual-stage hierarchical structure: Gross and fine phonetic features

Unsupervised speaker adaptation methods for vowel templates

Speaker adaptation through vector quantization

Speaker-adpatation methods using selective linear prediction

Synthesis of speaker-adaptive word templates by concatenation of the monosyllabic sounds

Speaker-adaptive connected syllable recognition based on the multiple similarity method

Spectral transformations through canonical correlation analysis for speaker adptation in ASR

New classification method of place of articulation of consonants in connected speech using formants

Speaker adaptation for a hidden Markov model

Multi-speaker validation of coarticulation models of syllabic nuclei

A relaxation technique for seeking optimal vowel candidate sequence

Speaker-independent isolated word recognition using label histograms

Speaker-independent French digits recognition using word-based vector quantization and hidden Markov models

Speaker-independent isolated word recognition for telephone voice using phoneme-like templates

New clustering algorithms applied to speaker independent isolated word recognition

Speaker-independent word recognition for large vocabulary using pre-selection and non-linear spectral matching

Speaker independent digit recognition with reference frame-specific distance measures

Discrimination of isolated vowels and stop consonants using features extracted from onset spectrum

A unified theory of deterministic system parameter identification

Structural identification algorithms of nonlinear systems using volterra functional series

Results of least-squares identification algorithm for unstable systems

A dynamical identification method based on conditioned observations for a sound insulation system with the order determined by a new criterion index

On-line identification of time variant parameters using covariance information

Digital estimation of linear/Quadratic transfer functions with a general random input

Some properties of the lattice algorithm for the direct computation of the matricial spectral factor

The locally quasi-stationary processing by the sigular value decomposition

Modeling of a nonstationary stochastic process produced by a linear time-varying system

A semi-Markovian parametric model for slowly time-varying binary switchings

ARMA Covariance realization from noisy data

Visual characterization of speech spectrograms

Automatic generation of consonant discrimination rules

Phoneme classification using Markov models

Phonetic properties of the basic vocabulary of five European languages: Implications for speech recognition

Detection and recognition of nasal consonants in American English

Recognition of word-final unstressed syllables

A phonetically based semivowel recognition system

Fine phonetic labeling methodology for speech recognition research

Automatic labeling system using speaker-dependent phonetic unit references

Compensating for vowel coarticulation in continuous speech recognition

Characterization and modeling of speech-segment durations

Detection in underwater noises modeled as a Gaussian-Gaussian mixture

Time delay estimation with nonstationary signals

Determining time delay via frequency estimation

Detection of unknown-frequency sinusoids in noise via autoregressive modeling

A new method for signal detection and estimation using the eigenstructure of the covariance difference

Application of the block Kalman filter to multisensor estimation with uncertain measurements

Separated estimation of wave parameters and spectral parameters by maximum likelihood

Range and speed estimation from frequency measurements alone

Direct estimation of multipath signals

The estimation of wide-angle bottom loss by a normal incident acoustic pulse

Cepstral methods for the determination of reflection coefficients for dispersive systems

A new approach to three-dimensional underwater tracking that considers acoustic medium effects

Three-dimensional imaging with a multifrequency holographic method

Fourier domain reconstruction of ultrasonic cross-sectional images

Three dimensional reconstruction system from TV camera acquired projections for the dynamic evaluation of scoliosis

FFT Processor as a digital lens for radio patrol camera in astrophysics

Accelerated skew coordinate transformation for high resolution SAR image formation

High speed SAR data processing on data-flow computer NEDIPS

Fast SAR signal processing on the DAP

Three-dimensional underwater imaging method - synthetic aperture image holography using an acoustic lens

Reconstruction of propagating complex wave fields using the Hilbert-Hankel transform

An underground object imaging system with computerized reconstruction

Signal processing of holographic under-snow radar with a displaying systems of three-dimensional information

A parallel signal processor system

The optimal synchronous cyclo-static array: A multiprocessor supercomputer for digital signal processing

Using warp as a supercomputer in signal processing

Arithmetic performane of floating point formarts available in VLSI

On the difficulties of utilizing current technologies to perform 100MHz DSP

A very fast FFT spectrum analyzer for radio astronomy

Conceptual diagnosis of signal processing systems

Evaluation of a high level language oriented program development system for high performance DSP DSSP1

The DSP workbench: Modeling parallel architectures as concurrent processes

Iteration independent subroutine form of Durbin's recursion for programmable signal processors

A topological sorting and loop cleansing algorithm for a constrained MIMD compiler of shift-invariant flow graphs

A highly parallel architecture for adaptive multichannel algorithms

A variable length lattice filter for adaptive noise cancellation

Adaptive lattice noise canceller and optimal step size

A family of pseudo-least squares estimation algorithms without division

Continuous-time discrete-order lattice filters

Extended Kalman frequency domain adaptive filtering with data-aided state initialization

On the rectangular transform approach for BLMS adaptive filtering

Adaptive noise cancelling in the spectrum domain

Suboptimal frequency domain adaptive array processing in a broadband environment

Chandrasekhar adaptive regularizer for adaptive filtering

Adaptive filter for concurrent echo cancellation and equalization using fast Kalman estimation

Adaptive signal processing using a modified gradient estimation technique

A TAP selection algorithm for adaptive filters

Performance analysis of the minimal parameter adaptive notch filter with constrained poles and zeros

Recursive equlaizers using a multistage adaptive algorithm

A microprocessor-based memory-tap echo canceler

Estimation of random spectral misadjustment of an adaptive filter

A stability problem in sign-sign adaptive algorithms

Stability margins of linear prediction polynomials

Long term stability in fractionally-space adaptive equalizers

Coefficient wordlength limitation in FLS adaptive filters

Trade-off studies on adaptive filtering algorithms used for adaptive moving target indicators

Adaptive phase locked loop employing frequency and phase processing technique

A new structure for adaptive signal processing with combined FIR & IIR filtering algorithms

A continuously adaptive vector predictive coder (AVPC) for speech encoding

A frequency domain waveform speech compression system based on product vector quantizers

A reduced search vector quantizer for speech

Performance of a real time low rate voice codec

Reducing signal delay in multipulse coding at 16kb/s

Multi-pulse LPC using linear programming in the frequency domain

Envelope constrained multipulse speech coding

Complexity reduction methods for vector excitation coding

An improved CELP by the separate coding of pulsive and random residuals

Multi-dimensional quantization applied to predictive coding of speech

A novel approach to estimating excitation code in code-excited linear prediction coding

A variable rate coding by APC with maximum likelihood quantization from 4.8 kbits/s to 16 kbits/s

Adaptive subbands excited transform (ASET) coding

Variable rate coding for speech storage

A low complexity regular pulse coding scheme with a reduced transmission delay

Selective modeling of the LPC residual during unvoiced frames: White noise or pulse excitation

A robust ADPCM system using an error-correcting code

Hardware realization of a multistage speech waveform vector quantizater

Subband coding with vector quantization

Two-dimensional spectrum analysis in sonic logging

On VLSI array architectures for digital image processing

New applications of digital signal processing in communications

Speech enhancement

Report of the 1985 IEEE ASSPS workshop on speech recognition

Use of statistical models for time series analysis

Predictive coding of speech: Historical review and directions for future research

Signal processing for radio astronomy at the Nobeyama radio observatory

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

Arithmetic performane of floating point formarts available in VLSI

Content Provider	IEEE Xplore Digital Library
Author	Williams, F.
Copyright Year	1986
Description	Author affiliation: TRW LSI Products Division, La Jolla, California (Williams, F.)
Abstract	This paper reports on the results obtained by simulating 1024, 256, and 128 point complex-to-complex Fast Fourier Transforms with four different floating point formats: the IEEE P754 Rev.10 standard, the MIL-STD-1750A standard, the TRW 22-bit DSP format, and the Hitachi 16/20-bit format. Since many existing systems use a sixteen-bit format, performance of a sixteen bit format (with block floating point where appropriate) was used as a baseline for comparison. The results were evaluated for bias and variance (in the form of signal-to-noise ratio) for all functions. The performance of FIR and IIR filters was analyzed for the same numerical representations.
Starting Page	2899
Ending Page	2902
File Size	133183
Page Count	4
File Format	PDF
DOI	10.1109/ICASSP.1986.1168605
Language	English
Publisher	Institute of Electrical and Electronics Engineers, Inc. (IEEE)
Publisher Date	1986-04-07
Publisher Place	Japan
Access Restriction	Subscribed
Rights Holder	Institute of Electrical and Electronics Engineers, Inc. (IEEE)
Subject Keyword	Floating-point arithmetic Very large scale integration Dynamic range IIR filters FETs Finite impulse response filter Digital signal processing Signal to noise ratio Intermodulation distortion Passband
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