The Experts below are selected from a list of 159 Experts worldwide ranked by ideXlab platform

Khalid Sayood - One of the best experts on this subject based on the ideXlab platform.

  • Joint Source Channel Coding Using Arithmetic Codes
    2010
    Co-Authors: B.d. Pettijohn, Khalid Sayood, Michael W. Hoffman
    Abstract:

    Reserving space for a symbol that is not in the source alphabet has been shown to provide excellent error detection. In this paper we show how to use this capability using a sequential deCoder structure to provide powerful error correction capability. This joint source/Channel Coder design provides significant packet loss recovery with minimal overhead.

  • Joint source/Channel coding using arithmetic codes
    IEEE Transactions on Communications, 2001
    Co-Authors: B.d. Pettijohn, Michael W. Hoffman, Khalid Sayood
    Abstract:

    Reserving space fur a symbol that is not in the source alphabet has been shown to provide excellent error detection. In this paper, we show how to exploit this capability using two sequential deCoder structures to provide powerful error correction capability. This joint source/Channel Coder design provides significant packet loss recovery with minimal rate overhead, and compares favorably with conventional schemes.

  • joint source Channel coding using arithmetic codes
    IEEE Transactions on Communications, 2001
    Co-Authors: B.d. Pettijohn, Michael W. Hoffman, Khalid Sayood
    Abstract:

    Reserving space fur a symbol that is not in the source alphabet has been shown to provide excellent error detection. In this paper, we show how to exploit this capability using two sequential deCoder structures to provide powerful error correction capability. This joint source/Channel Coder design provides significant packet loss recovery with minimal rate overhead, and compares favorably with conventional schemes.

  • a joint source Channel Coder with block constraints
    IEEE Transactions on Communications, 1999
    Co-Authors: Hasan H. Otu, Khalid Sayood
    Abstract:

    The assumptions made about the source during source Coder design result in a residual redundancy at the output of the source Coder. This redundancy can be utilized for error protection without any additional Channel coding. Joint source/Channel Coders obtained using this idea via maximum a posteriori probability deCoders tend to fail at low probability of error. In this paper, we propose a modification of the standard approach which provides protection at low error rates as well.

  • a joint source Channel Coder with block constraints
    International Conference on Acoustics Speech and Signal Processing, 1998
    Co-Authors: Hasan H. Otu, Khalid Sayood
    Abstract:

    Joint source/Channel Coders obtained using MAP deCoders tend to fail at low probability of error. We propose a modification of the standard approach which provides protection at low error rates as well.

Nguyen, Duy Huy - One of the best experts on this subject based on the ideXlab platform.

  • Renforcer et améliorer la transmission de la qualité de la voix sur le réseau LTE : défis et solutions
    2017
    Co-Authors: Nguyen, Duy Huy
    Abstract:

    LTE (Long Term Evolution) a été développé et normalisé par le 3GPP (3rd Generation Partnership Project). C’est un réseau à commutation de paquets. Cela signifie que la voix sur LTE (VoLTE) est un service de VoIP avec les exigences de qualité de service garantis au lieu de transmettre dans un réseau à commutation de circuits tels que les systèmes existants (2G/3G). VoLTE est déployé dans un réseau entièrement IP combinée avec IMS (IP Sous-système Multimédia). De ce fait, le déploiement de VoLTE est assez complexe et comment assurer la qualité de transmission de la voix sur les réseaux LTE est un très grand défi. Ainsi, il faut plusieurs solutions différentes pour renforcer et améliorer la qualité de transmission de la voix sur les réseaux LTE. Dans cette thèse, nous présentons des solutions en vue d’améliorer la qualité de transmission de la voix sur les réseaux LTE pour les services audio à bandes étroites et larges. Pour cela, il nous faudra différents facteurs complets en solutions. L’un d’eux est QoE (Qualité de l’Expérience) qui est une nouvelle tendance. Et afin de déterminer la perception des utilisateurs pour le service en temps réel tel que VoLTE, nous utilisons le E-model étendu et le WB (large bande) E-model pour des services audio à bandes étroites et larges respectivement. Les solutions proposées ici portent principalement sur des éléments clés dans les réseaux LTE, tels que le codage par chaine, MAC (Contrôle d’Accès Moyen) des systèmes de planification et la qualité de voix du moniteur décrits comme suit. Tout d’abord, des algorithmes améliorés pour renforcer le codec de la chaine LTE (codeur et décodeur) ont été proposés. Pour améliorer le codeur de chaine LTE, un algorithme d’adaptation conjointe a été déployé. Le but de cet algorithme est de minimiser la redondance générée par codage en chaine avec une légère réduction de la perception de l’utilisateur. Ensuite, afin d’améliorer le décodeur par chaine LTE, un algorithme amélioré Log-MAP a été présenté. Cet algorithme vise à obtenir la performance BER (Bit Error Rate) qui est le plus proche du Log-MAP avec une complexité de calcul réduite par rapport à l’état de l’art. Deuxièmement, la chaine et les systèmes QoS de planification améliorés de la perception de l’utilisateur et du mode de priorité VoIP ont été proposés. Ces planificateurs sont déployés à la fois pour les utilisateurs d’audio à larges et à étroites bandes. Les résultats numériques montrent qu’ils surpassent plusieurs planificateurs en vedette tels que FLS, M-LWDF et EXP/PF en termes de retard, de taux de perte de paquets, de débit cellulaire, d’indice et de l’équité et d’efficacité spectrale dans presque tous les cas. Enfin, pour assurer la qualité vocale de transmission sur le réseau LTE, la prédiction de la satisfaction des utilisateurs est essentielle. Pour cette raison, nous présentons deux modèles non intrusifs pour mesurer la qualité de la voix sur les réseaux LTE. Ces modèles sont utilisés pour les utilisateurs d’audio à bandes étroites et larges bandes. Les modèles proposés ne se réfèrent pas au signal original. Par conséquent, ils sont très appropriés pour prédire la qualité de l’appel vocal sur les réseaux LTELTE (Long Term Evolution) has been developed and standardized by 3GPP (3rd Generation Partnership Project). It is a packet-switched network. This means voice over LTE (VoLTE) is a VoIP service with the guaranteed QoS requirements instead of transmitted in a circuit-switched network such as the legacy system (2G/3G). Since VoLTE is deployed in an All-IP network combined with IMS (IP Multimedia Subsytem), thus, the VoLTE deployment is quite complex and how to ensure voice transmission quality over LTE networks is a very big challenge. Therefore, there needs to be many different solutions to enhance and improve voice transmission quality over LTE networks. In this dissertation, we present solutions to enhance and improve voice transmission quality over LTE networks for both narrowband and wideband audio services. In order to do that, there needs to be many different factors complemented in solutions. One of them is QoE (Quality of Experience) which is a new trend. And in order to determine user perception for real-time service such as VoLTE, we use extended E-model and WB (Wideband) E-model for narrowband and wideband audio services, respectively. The proposed solutions in this thesis mainly focus on key elements in LTE networks such as Channel coding, MAC (Medium Access Control) scheduling schemes and monitor voice quality described as follows. First, enhanced/improved algorithms for enhancing LTE Channel codec (Coder and deCoder) have been proposed. In order to enhance LTE Channel Coder, a joint source-Channel code rate adaption algorithm has been deployed. The goal of this algorithm is to minimize redundancy generated by Channel coding with a slight reduction of user perception. Next, in order to enhance LTE Channel deCoder, an improved Log-MAP algorithm has been presented. This algorithm aims at obtaining BER performance that is closest to the LOP-MAP with the computational complexity reduced in comparison with state-of-the-art. Second, Channel- and QoS-Aware scheduling schemes with the enhancement of user perception and VoIP priority mode have been proposed. These schedulers are deployed for both narrowband and wideband audio users. The numerical results show that they outperform several featured schedulers such as FLS, M-LWDF, and EXP/PF in terms of delay, packet loss rate, cell throughput, fairness index, and spectral efficiency in almost cases. Last, in order to ensure voice transmission quality over LTE network, prediction of user satisfaction is essential. For this reason, we present two object non-intrusive models for measuring voice quality in LTE networks. These models are used for narrowband and wideband audio users. The proposed models do not refer to the original signal, thus, they are very suitable for predicting voice call quality in LTE network

  • Renforcer et améliorer la transmission de la qualité de la voix sur le réseau LTE : défis et solutions
    HAL CCSD, 2017
    Co-Authors: Nguyen, Duy Huy
    Abstract:

    LTE (Long Term Evolution) has been developed and standardized by 3GPP (3rd Generation Partnership Project). It is a packet-switched network. This means voice over LTE (VoLTE) is a VoIP service with the guaranteed QoS requirements instead of transmitted in a circuit-switched network such as the legacy system (2G/3G). Since VoLTE is deployed in an All-IP network combined with IMS (IP Multimedia Subsytem), thus, the VoLTE deployment is quite complex and how to ensure voice transmission quality over LTE networks is a very big challenge. Therefore, there needs to be many different solutions to enhance and improve voice transmission quality over LTE networks. In this dissertation, we present solutions to enhance and improve voice transmission quality over LTE networks for both narrowband and wideband audio services. In order to do that, there needs to be many different factors complemented in solutions. One of them is QoE (Quality of Experience) which is a new trend. And in order to determine user perception for real-time service such as VoLTE, we use extended E-model and WB (Wideband) E-model for narrowband and wideband audio services, respectively. The proposed solutions in this thesis mainly focus on key elements in LTE networks such as Channel coding, MAC (Medium Access Control) scheduling schemes and monitor voice quality described as follows. First, enhanced/improved algorithms for enhancing LTE Channel codec (Coder and deCoder) have been proposed. In order to enhance LTE Channel Coder, a joint source-Channel code rate adaption algorithm has been deployed. The goal of this algorithm is to minimize redundancy generated by Channel coding with a slight reduction of user perception. Next, in order to enhance LTE Channel deCoder, an improved Log-MAP algorithm has been presented. This algorithm aims at obtaining BER performance that is closest to the LOP-MAP with the computational complexity reduced in comparison with state-of-the-art. Second, Channel- and QoS-Aware scheduling schemes with the enhancement of user perception and VoIP priority mode have been proposed. These schedulers are deployed for both narrowband and wideband audio users. The numerical results show that they outperform several featured schedulers such as FLS, M-LWDF, and EXP/PF in terms of delay, packet loss rate, cell throughput, fairness index, and spectral efficiency in almost cases. Last, in order to ensure voice transmission quality over LTE network, prediction of user satisfaction is essential. For this reason, we present two object non-intrusive models for measuring voice quality in LTE networks. These models are used for narrowband and wideband audio users. The proposed models do not refer to the original signal, thus, they are very suitable for predicting voice call quality in LTE networksLTE (Long Term Evolution) a été développé et normalisé par le 3GPP (3rd Generation Partnership Project). C’est un réseau à commutation de paquets. Cela signifie que la voix sur LTE (VoLTE) est un service de VoIP avec les exigences de qualité de service garantis au lieu de transmettre dans un réseau à commutation de circuits tels que les systèmes existants (2G/3G). VoLTE est déployé dans un réseau entièrement IP combinée avec IMS (IP Sous-système Multimédia). De ce fait, le déploiement de VoLTE est assez complexe et comment assurer la qualité de transmission de la voix sur les réseaux LTE est un très grand défi. Ainsi, il faut plusieurs solutions différentes pour renforcer et améliorer la qualité de transmission de la voix sur les réseaux LTE. Dans cette thèse, nous présentons des solutions en vue d’améliorer la qualité de transmission de la voix sur les réseaux LTE pour les services audio à bandes étroites et larges. Pour cela, il nous faudra différents facteurs complets en solutions. L’un d’eux est QoE (Qualité de l’Expérience) qui est une nouvelle tendance. Et afin de déterminer la perception des utilisateurs pour le service en temps réel tel que VoLTE, nous utilisons le E-model étendu et le WB (large bande) E-model pour des services audio à bandes étroites et larges respectivement. Les solutions proposées ici portent principalement sur des éléments clés dans les réseaux LTE, tels que le codage par chaine, MAC (Contrôle d’Accès Moyen) des systèmes de planification et la qualité de voix du moniteur décrits comme suit. Tout d’abord, des algorithmes améliorés pour renforcer le codec de la chaine LTE (codeur et décodeur) ont été proposés. Pour améliorer le codeur de chaine LTE, un algorithme d’adaptation conjointe a été déployé. Le but de cet algorithme est de minimiser la redondance générée par codage en chaine avec une légère réduction de la perception de l’utilisateur. Ensuite, afin d’améliorer le décodeur par chaine LTE, un algorithme amélioré Log-MAP a été présenté. Cet algorithme vise à obtenir la performance BER (Bit Error Rate) qui est le plus proche du Log-MAP avec une complexité de calcul réduite par rapport à l’état de l’art. Deuxièmement, la chaine et les systèmes QoS de planification améliorés de la perception de l’utilisateur et du mode de priorité VoIP ont été proposés. Ces planificateurs sont déployés à la fois pour les utilisateurs d’audio à larges et à étroites bandes. Les résultats numériques montrent qu’ils surpassent plusieurs planificateurs en vedette tels que FLS, M-LWDF et EXP/PF en termes de retard, de taux de perte de paquets, de débit cellulaire, d’indice et de l’équité et d’efficacité spectrale dans presque tous les cas. Enfin, pour assurer la qualité vocale de transmission sur le réseau LTE, la prédiction de la satisfaction des utilisateurs est essentielle. Pour cette raison, nous présentons deux modèles non intrusifs pour mesurer la qualité de la voix sur les réseaux LTE. Ces modèles sont utilisés pour les utilisateurs d’audio à bandes étroites et larges bandes. Les modèles proposés ne se réfèrent pas au signal original. Par conséquent, ils sont très appropriés pour prédire la qualité de l’appel vocal sur les réseaux LT

Che Ho Wei - One of the best experts on this subject based on the ideXlab platform.

  • robust transform image Coder over noisy Channel
    Electronics Letters, 2000
    Co-Authors: Hsueh-ming Hang, Che Ho Wei
    Abstract:

    A robust quantiser design for image coding is presented. The proposed quantiser can be viewed as the combination of compound of a quantiser, a variable length code (VLC) Coder, and a Channel Coder. Simulation results show that our proposed scheme has a graceful distortion behaviour within the designed noise range.

  • Robust transform image Coder over noisy Channel
    Visual Communications and Image Processing '99, 1998
    Co-Authors: Hsueh-ming Hang, Che Ho Wei
    Abstract:

    In this paper, we propose a robust quantizer design for image coding. Because the bits representing the reconstruction levels are transmitted directly to the Channel, the proposed quantizer can be viewed as a compound of a quantizer, a VLC Coder, and a Channel Coder. The conventional combined source/Channel design produces a source Coder designed for a Channel with a specific Channel noise. Our proposed quantizer is designed within a noise range. In comparison with the ordinary JPEG Coder, simulation results show that our proposed scheme has a much more graceful distortion behavior within the designed noise range.

J D Gibson - One of the best experts on this subject based on the ideXlab platform.

  • a constrained joint source Channel Coder design
    IEEE Journal on Selected Areas in Communications, 1994
    Co-Authors: Khalid Sayood, Fuling Liu, J D Gibson
    Abstract:

    The design of joint source/Channel Coders in situations where there is residual redundancy at the output of the source Coder is examined. It has previously been shown that this residual redundancy can be used to provide error protection without a Channel Coder. In this paper, this approach is extended to conventional source Coder/convolutional Coder combinations. A family of nonbinary enCoders is developed which more efficiently use the residual redundancy in the source Coder output. It is shown through simulation results that the proposed systems outperform conventional source-Channel Coder pairs with gains of greater than 9 dB in the reconstruction SNR at high probability of error. >

  • a joint source Channel Coder design
    International Conference on Communications, 1993
    Co-Authors: Khalid Sayood, Fuling Liu, J D Gibson
    Abstract:

    The situation in which there is residual redundancy at the source Coder output is examined. It has previously been shown that this residual redundancy can be used to provide error correction without a Channel enCoder. This approach is extended to conventional source Coder/convolutional Coder combinations. A design for nonbinary enCoders for this situation is developed. It is shown through simulation results that the proposed systems consistently outperform conventional source-Channel Coder pairs with gains of greater than 10 dB at high probability of error. >

  • a joint source Channel Coder design
    Asilomar Conference on Signals Systems and Computers, 1991
    Co-Authors: Fuling Liu, Khalid Sayood, J D Gibson
    Abstract:

    Source Coders and Channel Coders are generally designed separately without reference to each other. This approach is justified by a famous result of C.E. Shannons (1948). However, there are many situations in practice in which the assumptions upon which this result is based are violated. Specifically, the authors examine the situation where there is residual redundancy at the source Coder output. They have previously shown that this residual redundancy can be used to provide error correction using a Viterbi deCoder. In this work, they present the second half of the design: the design of enCoders for this situation. They show through simulation results that the proposed Coders consistently outperform conventional source-Channel Coder pairs with gains of up to 12 dB at a high probability of error. >

B.m. Hochwald - One of the best experts on this subject based on the ideXlab platform.

  • Source and Channel coding tradeoff on a Gaussian Channel
    Proceedings. 1998 IEEE International Symposium on Information Theory (Cat. No.98CH36252), 1998
    Co-Authors: B.m. Hochwald
    Abstract:

    Consider a system that quantizes and encodes analog data for transmission across an additive noise Gaussian Channel. To minimize distortion, the Channel code rate must be chosen to optimally allocate the available transmission rate between lossy source coding and block Channel coding. We establish approximations of the Channel code rate that minimizes the average distortion of a vector quantizer cascaded with a Channel Coder and a Gaussian Channel, thus complementing some recently-obtained results for the binary symmetric Channel. Analytic expressions are derived for large signal-to-noise ratios and for large source vector dimension. As in the binary symmetric Channel, the distortion decays exponentially with the number of Channel uses.

  • Tradeoff between source and Channel coding on a Gaussian Channel
    IEEE Transactions on Information Theory, 1998
    Co-Authors: B.m. Hochwald
    Abstract:

    Consider a system that quantizes and encodes analog data for transmission across an additive noise Gaussian Channel. To minimize distortion, the Channel code rate must be chosen to optimally allocate the available transmission rate between lossy source coding and block Channel coding. We establish tight upper and lower bounds on the Channel code rate that minimizes the average distortion of a vector quantizer cascaded with a Channel Coder and a Gaussian Channel, thus extending some recently obtained results for the binary-symmetric Channel. The upper hounds are obtained by averaging, whereas the lower bounds are uniform, over all possible index assignments. Analytic expressions are derived for large and small signal-to-noise ratios, and also for large source vector dimension. As in the binary-symmetric Channel, the optimal Channel code rate is often substantially smaller than the Channel capacity and the distortion decays exponentially with the number of Channel uses. Exact exponents are derived.

  • Tradeoff between source and Channel coding
    IEEE Transactions on Information Theory, 1997
    Co-Authors: B.m. Hochwald, Kenneth Zeger
    Abstract:

    A fundamental problem in the transmission of analog information across a noisy discrete Channel is the choice of Channel code rate that optimally allocates the available transmission rate between lossy source coding and block Channel coding. We establish tight bounds on the Channel code rate that minimizes the average distortion of a vector quantizer cascaded with a Channel Coder and a binary-symmetric Channel. Analytic expressions are derived in two cases of interest: small bit-error probability and arbitrary source vector dimension; arbitrary bit-error probability and large source vector dimension. We demonstrate that the optimal Channel code rate is often substantially smaller than the Channel capacity, and obtain a noisy-Channel version of the Zador (1982) high-resolution distortion formula.