NDLI: A DSP core for speech coding applications

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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

Volume v - 1994

Volume i - 1994

Volume iii - 1994

Volume vi - 1994

Volume iv - 1994

Volume ii - 1994

Large vocabulary continuous speech recognition using HTK

Noise immunization using neural net for speech recognition

Combining stochastic trajectory model and discriminative feature in speech recognizer

A new model-discriminant training algorithm for hybrid NN-HMM systems

Incorporating linguistic features in a hybrid HMM/MLP speech recognizer

"Eigenlips" for robust speech recognition

Vowel classification using a neural predictive HMM: a discriminative training approach

A neural architecture for hierarchical clustering

Hybrid model decomposition of speech and noise in a radial basis function neural model framework

Application of a generalized probabilistic descent method to recurrent neural network based speech recognition

Isolated word recognition using a hybrid neural network

Mutual information neural networks: a new connectionist approach for dynamic speech recognition tasks

Performance of radar target recognition schemes using neural networks-a comparative study

On-line cursive script recognition using time delay neural networks and hidden Markov models

A connectionist recognizer for on-line cursive handwriting recognition

Incremental learning using the time delay neural network

Transformation of optimized prototypes for handwritten digit recognition

Using artificial neural networks to improve the mechanical signature analysis test

On two-pattern classification and feature selection using neural networks

A new composite feature vector for Arabic handwritten signature recognition

Invariant property of spatio-temporal feature maps using gated neuronal architecture

On-line handwritten character recognition using parallel neural networks

Multiple neural networks using the reduced input dimension

Optimal, matching-score network for pattern classification

Application of neural networks to detecting misfire in automotive engines

Neural receiver for CPM signals

A new cascaded projection pursuit network for nonlinear regression

Approach of using a density equalizing function to self-organizing learning for solving travelling salesman problem

A simulation environment for very large neural networks

Echocardiogram structure and tissue classification using hierarchical fuzzy neural networks

A multiresolution probabilistic neural network for image segmentation

M-lattice: a novel non-linear dynamical system and its application to halftoning

The design of neural network configuration for object recognition

A new competitive learning algorithm for vector quantization

Bimodal recognition experiments with recurrent neural networks

The conditional expectation via a general class of nonlinear networks

A constrained neural network with complex activation function: application to time-frequency analysis

An invariance property of neural networks

Design of neural estimators for multisensors: second order backpropagation, initialization and generalization

Noise reduction in state space using the focused gamma neural network

On the design of nonlinear speech predictors with recurrent nets

A self-structuring algorithm for artificial neural networks

Square-root, reciprocal, sine/cosine, arctangent cell for signal and image processing

A multi-channel decimator for a tri-level delta-sigma modulator

An analysis of FPGA-based custom computers for DSP applications

Optimal VLSI architecture for distributed arithmetic-based algorithms

Using preliminary decisions to reduce complexity in adaptive equalizers

A novel VLSI array design for the discrete Hartley transform using cyclic convolution

Design of high throughput, low latency and low cost structures for linear systems

A 'Jacobi' signal processing unit for time-adaptive SVD

RECALS II: a new list scheduling algorithm

Word-length determination and scaling software for a signal flow block diagram

Dynamic data flow and control flow in high level DSP code synthesis

Behavioral synthesis of low-cost partial scan designs for DSP applications

Geolocalization by combined range difference and range rate difference measurements

Automatic model parameter generation for the speech training of deaf children

Algorithm and architecture: for non-linear noise filtering via piecewise linear compression

A novel efficient parallel algorithm for RNS to binary conversion for arbitrary moduli set

Computation of the real Schur decomposition of nonsymmetric matrices and its hardware implementation

On the formal derivation of a systolic array for recursive least squares estimation

Specification and support for multidimensional DSP in the SILAGE language

Generating efficient loop code for programmable DSPs

Retiming of DSP programs for optimum vectorization

Minimizing memory requirements for chain-structured synchronous dataflow programs

Automatic code generation for heterogeneous multiprocessors

Application-driven design of DSP architectures and compilers

Concurrency characteristics in DSP programs

A DSP core for speech coding applications

Secure speech and data communication over the public switching telephone network

Arithmetic codec from behavioral description based LSI-CAD for fully programmable image coding system

Video DSP architecture for MPEG2 codec

A hierarchical multiprocessor architecture based on heterogeneous processors for video coding applications

A low power subband video decoder architecture

Digit pipelined discrete wavelet transform

A VLSI architecture for real-time hierarchical encoding/decoding of video using the wavelet transform

Time delay estimation using integrated polyspectrum

An efficient closed-form localization solution from time difference of arrival measurements

An adaptive linear multiuser receiver for deep water acoustic local area networks

Estimation of the causal impulse response of underwater target

Detection, identification and tracking of active scatterers

Weak radar signal detection based on wavelet transform

Radar target recognition using range profiles

Correction of I/Q errors in homodyne step frequency radar refocuses range profiles

Deconvolution of sensor array signals using time-scaling

Extraction of multiple periodic waveforms from noisy data

High-resolution bathymetric simulations based on Kirchhoff scattering theory and anisotropic seafloor modeling

A deconvolution approach to moving sources localization

Resolving ambiguities in estimating spatial frequencies in sparse linear array

Bayesian time delay estimation for ocean acoustic tomography

Matched field processing in shallow ocean: signal arrival identification using EM algorithm

Extraction of pertinent parameters in oceanographic data

Statistics of the phase delays between array receivers estimated from the eigendecomposition of the signal correlation matrix

Transient sonar signal classification using hidden Markov model and neural net

Optimum Costas-like decompositions of Costas arrays for channel characterization and communications

Maneuvering target motion analysis using hidden Markov model

An integrated approach to passive target classification

Quantisation error modelling of narrowband adaptive arrays using projected perturbation sequences

Probabilistic ray identification: a new tool for ocean acoustic tomography

GASTOM 90 tomography experiment data inversion

Adaptive detection of moving signal using shallow sea hydroacoustic data

The use of non-supervised neural networks to detect lines in lofargram

A comparison between spectral and bispectral analysis for ship detection from acoustical time series

Adaptive waveform selection for active sonar

Far field array processing with neural networks

Parametric separation applied to underwater acoustic multipath propagation

Acoustic event localization using a crosspower-spectrum phase based technique

Colouration in assisted reverberation systems

New adaptive-filtering techniques applied to speech echo cancellation

Effects of room reverberation on time-delay estimation performance

Characterization of talker radiation pattern using a microphone array

Fully adaptive generalized recursive control system for active acoustic attenuation

A time varying step size normalized LMS echo canceler algorithm

DSP implementation and performance evaluation of a compact stereo echo canceller

Active control using IIR filters-a second look

Detection of human speech in structured noise

Recursive restoration of pitch variation defects in musical recordings

Real time speech rate converting system for elderly people

Real-time, loose-harmonic matching fundamental frequency estimation for musical signals

Improved musical pitch tracking using principal decomposition analysis

A two-channel approach to the removal of impulsive noise from archived recordings

Low rate audio coder with hierarchical filterbanks and lattice vector quantization

Recursive tracking of formants in speech signals

A new paradigm for reliable automatic formant tracking

Optimization of time-frequency masking filters using the minimum classification error criterion

Subband-Autocorrelation analysis and its application for speech recognition

Fine pitch contour extraction by voice fundamental wave filtering method

Prosodic phrase segmentation by pitch pattern clustering

An automatic intonation tone contour labelling and classification algorithm

Suprasegmental features and continuous speech recognition

Automatic classification of prosodically marked phrase boundaries in German

A generalised model for utilising prosodic information in continuous speech recognition

Hybrid system combining expert-TDNNs and HMMs for continuous speech recognition

A channel-bank-based phone detection strategy

Fast match acoustic models in large vocabulary continuous speech recognition

An algorithm of high resolution and efficient multiple string hypothesization for continuous speech recognition using inter-word models

Non-uniform unit parsing for SSS-LR continuous speech recognition

Context-free large-vocabulary connected speech recognition with evolutional grammars

Search algorithm that merges candidates in meaning level for very large vocabulary spontaneous speech recognition

Large vocabulary word recognition based on tree-trellis search

An initial study on a segmental probability model approach to large-vocabulary continuous Mandarin speech recognition

Large vocabulary continuous speech recognition of Wall Street Journal data

A real-time wideband CELP coder for a videophone application

Multi-band residual coding of CELP codecs at 8 kb/s

A pitch synchronous innovation CELP (PSI-CELP) coder for 2-4 kbit/s

CELP speech coding with almost no codebook search

Perceptual enhancement of CELP speech coders

A computationally efficient wavelet transform CELP coder

8 kbit/s ACELP coding of speech with 10 ms speech-frame: a candidate for CCITT standardization

Implementation and performance of an 8-kbit/s conjugate structure CELP speech coder

A hierarchical LPNN network for noise reduction and noise degraded speech recognition

Non-linear regression based feature extraction for connected-word recognition in noise

Speech parameter extraction in noisy environment using a masking model

Auditory model inversion for sound separation

Steady-state performance limitations of full-band acoustic echo cancellers

Speech recognition in noisy car environment based on OSALPC representation and robust similarity measuring techniques

Noisy speech recognition using cepstral-time features and spectral-time filters

Environment normalization for robust speech recognition using direct cepstral comparison

Markov model based noise modeling and its application to noisy speech recognition using dynamical features of speech

A family of MLP based nonlinear spectral estimators for noise reduction

On the limitations of cepstral features in noise

Minimum cost based phoneme class detection for improved iterative speech enhancement

Parsing word graphs using a linguistic grammar and a statistical language model

Learning consistent semantics from training data

A multimodal, keyword-based spoken dialogue system-MultiksDial

On jointly learning the parameters in a character-synchronous integrated speech and language model

Heuristic search integrating syntactic, semantic and dialog-level constraints

Detecting misrecognitions and out-of-vocabulary words

Integrating semantic constraints into the Sphinx-II recognition search

Efficient chart parsing of speech recognition candidates

Improving sentence recognition in stochastic context-free grammars

A stochastic language model for speech recognition integrating local and global constraints

Phonological parsing for reversible letter-to-sound/sound-to-letter generation

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

A DSP core for speech coding applications

Content Provider	IEEE Xplore Digital Library
Author	Nurmi, J. Eerola, V. Ofner, E. Gierlinger, A. Jernej, J. Karema, T. Raita-aho, T.
Copyright Year	1994
Description	Author affiliation: Signal Process. Lab., Tampere Univ. of Technol., Finland (Nurmi, J.)
Abstract	An application specific processor core for mobile speech coding applications has been designed and implemented. Since the architecture is tailored to the application, it has a very low power consumption, making it attractive for handheld devices. The low power consumption, flexible design and high performance have been achieved by a standby mode, optimized full custom design, and a low clock frequency, together with a highly parallel architecture. All the parallelism is accessible to the user on the assembler level. Engineering samples of the processor have been fabricated and tested. The silicon area required for the core is approximately 25 mm/sup 2/ using 1.0 /spl mu/m CMOS. A typical average power consumption for a GSM full rate speech codec implementation using this core is less than 50 mW at 5 V operating voltage, and the complex algorithm is executed in less than 5 ms for each 20 ms speech frame (including encode, decode, VAD and DTX operations).<>
File Size	394323
File Format	PDF
ISBN	0780317750
ISSN	15206149
DOI	10.1109/ICASSP.1994.389629
Language	English
Publisher	Institute of Electrical and Electronics Engineers, Inc. (IEEE)
Publisher Date	1994-04-19
Access Restriction	Subscribed
Rights Holder	Institute of Electrical and Electronics Engineers, Inc. (IEEE)
Subject Keyword	Digital signal processing Speech coding Energy consumption Speech codecs Application specific processors Handheld computers Design optimization Clocks Frequency Parallel architectures
Content Type	Text
Resource Type	Article

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