David Mansour - Academia.edu (original) (raw)

Papers by David Mansour

Research paper thumbnail of Multistream Hybrid in Band on Channel FM Systems for Digital Audio Broadcasting

Abstract- New approaches to hybrid in band on chan-nel (HIBOC) FM systems for digital audio broad... more Abstract- New approaches to hybrid in band on chan-nel (HIBOC) FM systems for digital audio broadcasting based on multistream transmission methodology and mul-tidescriptive audio coding techniques are introduced in this paper. These ideas combined with a lower per side-band audio coding rate and more powerful channel codes result in robust transmission and graceful degradation in variable interference channels. By also using orthogonal frequency division multiplexing techniques with a nonuni-form power profile combined with unequal error protection and sideband time diversity, we arrive at new HIBOC FM schemes with extended coverage and better peak audio quality than previously proposed. The paper provides ap-proximate performance analysis for potential systems in-cluding audio coding quality.

Bookmarks Related papers MentionsView impact

Research paper thumbnail of Antenne compacte pour communication par satellite

L'invention a trait a une antenne receptrice et/ou emettrice compacte, qui comprend un reseau... more L'invention a trait a une antenne receptrice et/ou emettrice compacte, qui comprend un reseau d'elements d'antenne qui recueille et concentre les ondes millimetriques ou un autre rayonnement. Les elements d'antenne sont physiquement adaptes de facon que le rayonnement d'une longueur d'onde de syntonisation qui rencontre l'antenne a un angle d'incidence particulier soit recueilli par les elements et concentre en phase. L'antenne peut comprendre egalement au moins deux rotateurs mecaniques destines a modifier l'angle d'incidence du rayonnement entrant ou sortant, afin qu'il coincide avec l'angle d'incidence particulier.

Bookmarks Related papers MentionsView impact

Research paper thumbnail of Effects of hybrid nonlinearity on a full-duplex telephone network with vocoder

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

In most applications of adaptive filtering algorithms to the telephone echo cancellation problem,... more In most applications of adaptive filtering algorithms to the telephone echo cancellation problem, a linear model is assumed for the hybrid. In this paper, a nonlinear model based on a Volterra series representation is adopted. By using a new and efficient nonlinear frequency domain adaptive algorithm, a nonlinear model for a hybrid in an actual connection has been identified. The

Bookmarks Related papers MentionsView impact

Research paper thumbnail of Study of echo cancelling algorithms for full duplex telephone networks with vocoders

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

In the cancellation of telephone echo an algorithm with fast convergence and relatively simple im... more In the cancellation of telephone echo an algorithm with fast convergence and relatively simple implementation is desired. For narrow band digital voice transmission, it is also of interest to know what effects vocoders have on telephone echo and vice versa [1]. From our experiments, it has been shown that the frequency domain LMS [2] algorithm provides faster convergence than the Widrow LMS algorithm [3], whereas the gradient lattice algorithm [4] produced inferior results. It is also found that when the echo is strong, its effects are more perceptually annoying with an LPC vocoder in the loop. However, when the echo is weak, an LPC vocoder helps to suppress the echo.

Bookmarks Related papers MentionsView impact

Research paper thumbnail of A highly parallel architecture for adaptive multichannel algorithms

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

In this paper an adaptive lattice structured filter for processing wideband spatial data received... more In this paper an adaptive lattice structured filter for processing wideband spatial data received by an array of sensors is discribed. Using a special architecture that implements a modified version of the Gram-Schmidt orthogonalization procedure a very simple derivation of number of algorithms involving multichannel inputs are presented. A wavefront array processing architecture is described that solve the following algorithms: multichannel spatio-temporal exact least square adaptive lattice filter, matrix-inversion algorithm, linear equations solution and the Fadeev problem. The same architecture and processor elements are shared by all four algorithms.

Bookmarks Related papers MentionsView impact

Research paper thumbnail of Unconstrained frequency-domain adaptive filter

IEEE Transactions on Acoustics, Speech, and Signal Processing

Bookmarks Related papers MentionsView impact

Research paper thumbnail of Frequency domain non-linear adaptive filter

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

A new non-linear adaptive filter is presented. The algorithm is based on the Volterra series impl... more A new non-linear adaptive filter is presented. The algorithm is based on the Volterra series implemented in the frequency domain. For a finite memory of length N, the algorithm converges to the equivalent time domain non-linear adaptive filter as proposed by Roy and Sherman [1]. The frequency domain implementation offers a significant reduction in computation. For the second order Volterra

Bookmarks Related papers MentionsView impact

Research paper thumbnail of Efficient nonlinear system identification

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

Bookmarks Related papers MentionsView impact

Research paper thumbnail of The short-time modified coherence representation and noisy speech recognition

IEEE Transactions on Acoustics, Speech, and Signal Processing

A technique for robust spectral representation of all-pole sequences is proposed. It is shown tha... more A technique for robust spectral representation of all-pole sequences is proposed. It is shown that the autocorrelation of an all-pole sequence, obtained by passing white noise through an all-pole filter 1/A(z), is an all-pole sequence of the form 1/ A2(z). A short-time modified coherence (SMC) representation is proposed that is an all-pole modeling of the autocorrelation sequence with a spectral shaper. The spectral shaper, essentially a square root operator in the frequency domain, compensates for the inherent spectral distortion introduced by the autocorrelation operation on the autocorrelation sequence of the signal. The properties of the SMC representation, especially its robustness to additive white noise, are analyzed. Initial implementation of the SMC in a speaker-dependent isolated word recognizer shows an improvement in recognition accuracy equivalent to an increase in input SNR of approximately 13 dB, as compared to the LPC recognizer

Bookmarks Related papers MentionsView impact

Research paper thumbnail of A family of distortion measures base upon projection operation for robust speech recognition

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

The authors aim at the formulation of similarity measures for robust speech recognition. Their co... more The authors aim at the formulation of similarity measures for robust speech recognition. Their consideration focuses on the speech cepstrum derived from linear prediction coefficients (the LPC cepstrum). By using common models for noisy speech, they analytically and empirically show how the ambient noise can affect some important attributes of the LPC cepstrum such as the vector norm, coefficient order,

Bookmarks Related papers MentionsView impact

Research paper thumbnail of The short-time modified coherence representation and its application for noisy speech recognition

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

A technique for robust spectral representation of all-pole sequences is proposed. It is shown tha... more A technique for robust spectral representation of all-pole sequences is proposed. It is shown that the autocorrelation of an all-pole sequence, obtained by passing white noise through an all-pole filter 1/A(z), is an all-pole sequence of the form 1/ A2(z). The short-time modified coherence (SCM) representation, proposed here, is an all-pole modeling of the autocorrelation sequence followed by a spectral

Bookmarks Related papers MentionsView impact

Research paper thumbnail of Adaptive decision directed speech recognition bias equalization method and apparatus

Bookmarks Related papers MentionsView impact

Research paper thumbnail of Low profile antenna

Bookmarks Related papers MentionsView impact

Research paper thumbnail of Multistream-in-band-on-channel transmission system

Bookmarks Related papers MentionsView impact

Research paper thumbnail of Multistream in-band on-channel systems

Bookmarks Related papers MentionsView impact

Research paper thumbnail of Speech recognition bias equalisation method and apparatus

Bookmarks Related papers MentionsView impact

Research paper thumbnail of Low profile antenna for satellite communication

Bookmarks Related papers MentionsView impact

Research paper thumbnail of Use of satellites by terminals with small antennas

Bookmarks Related papers MentionsView impact

Research paper thumbnail of On least square frequency-domain adaptive filters

This paper presents a least square approach to the Dentino et al. [1] frequency-domain adaptive f... more This paper presents a least square approach to the Dentino et al. [1] frequency-domain adaptive filter by minimizing a frequencydomain error criterion. The constantmuthat controls the LMS convergence behavior is replaced by an adaptivemuthat every new iteration achieves this least squared error. The proposed approach can be extended and then applied to modify other frequency-domain adaptive algorithms. For example, a

Bookmarks Related papers MentionsView impact

Research paper thumbnail of Algorithm development of a telephone signal conditioner for the wideband integrated network

During this program a full-scale testbed simulation of the interfacing of a telephone to the wide... more During this program a full-scale testbed simulation of the interfacing of a telephone to the wideband integrated network was completed. This simulation includes the use of different types of echo cancelling algorithms and an LPC vocoder, and allows for other studies to be carried on. Echo cancelling algorithms were studied with a number of interesting conclusions. Because of the non-stationarity of the speech signals, made more so by the vocoder, standard LMS algorithms and lattice techniques are not adequate because of their convergence properties. With the vocoder in the loop, convergence must be faster than without the vocoder, because synthetic speech signals are not so rich in components as are the actual speech signals. An unconstrained frequency domain adaptive filter algorithm was the most effective. The echo cancelling is limited by the nonlinearities of the system. It was found that a nonlinear adaptive filter could reduce the signal-to-noise ratio by a few more dB when us...

Bookmarks Related papers MentionsView impact

Research paper thumbnail of Multistream Hybrid in Band on Channel FM Systems for Digital Audio Broadcasting

Abstract- New approaches to hybrid in band on chan-nel (HIBOC) FM systems for digital audio broad... more Abstract- New approaches to hybrid in band on chan-nel (HIBOC) FM systems for digital audio broadcasting based on multistream transmission methodology and mul-tidescriptive audio coding techniques are introduced in this paper. These ideas combined with a lower per side-band audio coding rate and more powerful channel codes result in robust transmission and graceful degradation in variable interference channels. By also using orthogonal frequency division multiplexing techniques with a nonuni-form power profile combined with unequal error protection and sideband time diversity, we arrive at new HIBOC FM schemes with extended coverage and better peak audio quality than previously proposed. The paper provides ap-proximate performance analysis for potential systems in-cluding audio coding quality.

Bookmarks Related papers MentionsView impact

Research paper thumbnail of Antenne compacte pour communication par satellite

L'invention a trait a une antenne receptrice et/ou emettrice compacte, qui comprend un reseau... more L'invention a trait a une antenne receptrice et/ou emettrice compacte, qui comprend un reseau d'elements d'antenne qui recueille et concentre les ondes millimetriques ou un autre rayonnement. Les elements d'antenne sont physiquement adaptes de facon que le rayonnement d'une longueur d'onde de syntonisation qui rencontre l'antenne a un angle d'incidence particulier soit recueilli par les elements et concentre en phase. L'antenne peut comprendre egalement au moins deux rotateurs mecaniques destines a modifier l'angle d'incidence du rayonnement entrant ou sortant, afin qu'il coincide avec l'angle d'incidence particulier.

Bookmarks Related papers MentionsView impact

Research paper thumbnail of Effects of hybrid nonlinearity on a full-duplex telephone network with vocoder

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

In most applications of adaptive filtering algorithms to the telephone echo cancellation problem,... more In most applications of adaptive filtering algorithms to the telephone echo cancellation problem, a linear model is assumed for the hybrid. In this paper, a nonlinear model based on a Volterra series representation is adopted. By using a new and efficient nonlinear frequency domain adaptive algorithm, a nonlinear model for a hybrid in an actual connection has been identified. The

Bookmarks Related papers MentionsView impact

Research paper thumbnail of Study of echo cancelling algorithms for full duplex telephone networks with vocoders

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

In the cancellation of telephone echo an algorithm with fast convergence and relatively simple im... more In the cancellation of telephone echo an algorithm with fast convergence and relatively simple implementation is desired. For narrow band digital voice transmission, it is also of interest to know what effects vocoders have on telephone echo and vice versa [1]. From our experiments, it has been shown that the frequency domain LMS [2] algorithm provides faster convergence than the Widrow LMS algorithm [3], whereas the gradient lattice algorithm [4] produced inferior results. It is also found that when the echo is strong, its effects are more perceptually annoying with an LPC vocoder in the loop. However, when the echo is weak, an LPC vocoder helps to suppress the echo.

Bookmarks Related papers MentionsView impact

Research paper thumbnail of A highly parallel architecture for adaptive multichannel algorithms

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

In this paper an adaptive lattice structured filter for processing wideband spatial data received... more In this paper an adaptive lattice structured filter for processing wideband spatial data received by an array of sensors is discribed. Using a special architecture that implements a modified version of the Gram-Schmidt orthogonalization procedure a very simple derivation of number of algorithms involving multichannel inputs are presented. A wavefront array processing architecture is described that solve the following algorithms: multichannel spatio-temporal exact least square adaptive lattice filter, matrix-inversion algorithm, linear equations solution and the Fadeev problem. The same architecture and processor elements are shared by all four algorithms.

Bookmarks Related papers MentionsView impact

Research paper thumbnail of Unconstrained frequency-domain adaptive filter

IEEE Transactions on Acoustics, Speech, and Signal Processing

Bookmarks Related papers MentionsView impact

Research paper thumbnail of Frequency domain non-linear adaptive filter

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

A new non-linear adaptive filter is presented. The algorithm is based on the Volterra series impl... more A new non-linear adaptive filter is presented. The algorithm is based on the Volterra series implemented in the frequency domain. For a finite memory of length N, the algorithm converges to the equivalent time domain non-linear adaptive filter as proposed by Roy and Sherman [1]. The frequency domain implementation offers a significant reduction in computation. For the second order Volterra

Bookmarks Related papers MentionsView impact

Research paper thumbnail of Efficient nonlinear system identification

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

Bookmarks Related papers MentionsView impact

Research paper thumbnail of The short-time modified coherence representation and noisy speech recognition

IEEE Transactions on Acoustics, Speech, and Signal Processing

A technique for robust spectral representation of all-pole sequences is proposed. It is shown tha... more A technique for robust spectral representation of all-pole sequences is proposed. It is shown that the autocorrelation of an all-pole sequence, obtained by passing white noise through an all-pole filter 1/A(z), is an all-pole sequence of the form 1/ A2(z). A short-time modified coherence (SMC) representation is proposed that is an all-pole modeling of the autocorrelation sequence with a spectral shaper. The spectral shaper, essentially a square root operator in the frequency domain, compensates for the inherent spectral distortion introduced by the autocorrelation operation on the autocorrelation sequence of the signal. The properties of the SMC representation, especially its robustness to additive white noise, are analyzed. Initial implementation of the SMC in a speaker-dependent isolated word recognizer shows an improvement in recognition accuracy equivalent to an increase in input SNR of approximately 13 dB, as compared to the LPC recognizer

Bookmarks Related papers MentionsView impact

Research paper thumbnail of A family of distortion measures base upon projection operation for robust speech recognition

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

The authors aim at the formulation of similarity measures for robust speech recognition. Their co... more The authors aim at the formulation of similarity measures for robust speech recognition. Their consideration focuses on the speech cepstrum derived from linear prediction coefficients (the LPC cepstrum). By using common models for noisy speech, they analytically and empirically show how the ambient noise can affect some important attributes of the LPC cepstrum such as the vector norm, coefficient order,

Bookmarks Related papers MentionsView impact

Research paper thumbnail of The short-time modified coherence representation and its application for noisy speech recognition

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

A technique for robust spectral representation of all-pole sequences is proposed. It is shown tha... more A technique for robust spectral representation of all-pole sequences is proposed. It is shown that the autocorrelation of an all-pole sequence, obtained by passing white noise through an all-pole filter 1/A(z), is an all-pole sequence of the form 1/ A2(z). The short-time modified coherence (SCM) representation, proposed here, is an all-pole modeling of the autocorrelation sequence followed by a spectral

Bookmarks Related papers MentionsView impact

Research paper thumbnail of Adaptive decision directed speech recognition bias equalization method and apparatus

Bookmarks Related papers MentionsView impact

Research paper thumbnail of Low profile antenna

Bookmarks Related papers MentionsView impact

Research paper thumbnail of Multistream-in-band-on-channel transmission system

Bookmarks Related papers MentionsView impact

Research paper thumbnail of Multistream in-band on-channel systems

Bookmarks Related papers MentionsView impact

Research paper thumbnail of Speech recognition bias equalisation method and apparatus

Bookmarks Related papers MentionsView impact

Research paper thumbnail of Low profile antenna for satellite communication

Bookmarks Related papers MentionsView impact

Research paper thumbnail of Use of satellites by terminals with small antennas

Bookmarks Related papers MentionsView impact

Research paper thumbnail of On least square frequency-domain adaptive filters

This paper presents a least square approach to the Dentino et al. [1] frequency-domain adaptive f... more This paper presents a least square approach to the Dentino et al. [1] frequency-domain adaptive filter by minimizing a frequencydomain error criterion. The constantmuthat controls the LMS convergence behavior is replaced by an adaptivemuthat every new iteration achieves this least squared error. The proposed approach can be extended and then applied to modify other frequency-domain adaptive algorithms. For example, a

Bookmarks Related papers MentionsView impact

Research paper thumbnail of Algorithm development of a telephone signal conditioner for the wideband integrated network

During this program a full-scale testbed simulation of the interfacing of a telephone to the wide... more During this program a full-scale testbed simulation of the interfacing of a telephone to the wideband integrated network was completed. This simulation includes the use of different types of echo cancelling algorithms and an LPC vocoder, and allows for other studies to be carried on. Echo cancelling algorithms were studied with a number of interesting conclusions. Because of the non-stationarity of the speech signals, made more so by the vocoder, standard LMS algorithms and lattice techniques are not adequate because of their convergence properties. With the vocoder in the loop, convergence must be faster than without the vocoder, because synthetic speech signals are not so rich in components as are the actual speech signals. An unconstrained frequency domain adaptive filter algorithm was the most effective. The echo cancelling is limited by the nonlinearities of the system. It was found that a nonlinear adaptive filter could reduce the signal-to-noise ratio by a few more dB when us...

Bookmarks Related papers MentionsView impact