Signal Processing for Communications II

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Full List of Titles
1: Speech Processing
CELP Coding
Large Vocabulary Recognition
Speech Analysis and Enhancement
Acoustic Modeling I
ASR Systems and Applications
Topics in Speech Coding
Speech Analysis
Low Bit Rate Speech Coding I
Robust Speech Recognition in Noisy Environments
Speaker Recognition
Acoustic Modeling II
Speech Production and Synthesis
Feature Extraction
Robust Speech Recognition and Adaptation
Low Bit Rate Speech Coding II
Speech Understanding
Language Modeling I
2: Speech Processing, Audio and Electroacoustics, and Neural Networks
Acoustic Modeling III
Lexical Issues/Search
Speech Understanding and Systems
Speech Analysis and Quantization
Utterance Verification/Acoustic Modeling
Language Modeling II
Adaptation /Normalization
Speech Enhancement
Topics in Speaker and Language Recognition
Echo Cancellation and Noise Control
Coding
Auditory Modeling, Hearing Aids and Applications of Signal Processing to Audio and Acoustics
Spatial Audio
Music Applications
Application - Pattern Recognition & Speech Processing
Theory & Neural Architecture
Signal Separation
Application - Image & Nonlinear Signal Processing
3: Signal Processing Theory & Methods I
Filter Design and Structures
Detection
Wavelets
Adaptive Filtering: Applications and Implementation
Nonlinear Signals and Systems
Time/Frequency and Time/Scale Analysis
Signal Modeling and Representation
Filterbank and Wavelet Applications
Source and Signal Separation
Filterbanks
Emerging Applications and Fast Algorithms
Frequency and Phase Estimation
Spectral Analysis and Higher Order Statistics
Signal Reconstruction
Adaptive Filter Analysis
Transforms and Statistical Estimation
Markov and Bayesian Estimation and Classification
4: Signal Processing Theory & Methods II, Design and Implementation of Signal Processing Systems, Special Sessions, and Industry Technology Tracks
System Identification, Equalization, and Noise Suppression
Parameter Estimation
Adaptive Filters: Algorithms and Performance
DSP Development Tools
VLSI Building Blocks
DSP Architectures
DSP System Design
Education
Recent Advances in Sampling Theory and Applications
Steganography: Information Embedding, Digital Watermarking, and Data Hiding
Speech Under Stress
Physics-Based Signal Processing
DSP Chips, Architectures and Implementations
DSP Tools and Rapid Prototyping
Communication Technologies
Image and Video Technologies
Automotive Applications / Industrial Signal Processing
Speech and Audio Technologies
Defense and Security Applications
Biomedical Applications
Voice and Media Processing
Adaptive Interference Cancellation
5: Communications, Sensor Array and Multichannel
Source Coding and Compression
Compression and Modulation
Channel Estimation and Equalization
Blind Multiuser Communications
Signal Processing for Communications I
CDMA and Space-Time Processing
Time-Varying Channels and Self-Recovering Receivers
Signal Processing for Communications II
Blind CDMA and Multi-Channel Equalization
Multicarrier Communications
Detection, Classification, Localization, and Tracking
Radar and Sonar Signal Processing
Array Processing: Direction Finding
Array Processing Applications I
Blind Identification, Separation, and Equalization
Antenna Arrays for Communications
Array Processing Applications II
6: Multimedia Signal Processing, Image and Multidimensional Signal Processing, Digital Signal Processing Education
Multimedia Analysis and Retrieval
Audio and Video Processing for Multimedia Applications
Advanced Techniques in Multimedia
Video Compression and Processing
Image Coding
Transform Techniques
Restoration and Estimation
Image Analysis
Object Identification and Tracking
Motion Estimation
Medical Imaging
Image and Multidimensional Signal Processing Applications I
Segmentation
Image and Multidimensional Signal Processing Applications II
Facial Recognition and Analysis
Digital Signal Processing Education

Author Index
A B C D E F G H I
J K L M N O P Q R
S T U V W X Y Z

Orthogonal Pulse Shape Design via Semidefinite Programming

Authors:

Timothy N Davidson, Communications Research Laboratory, McMaster University, Hamilton, Ontario, Canada (Canada)
Zhi-Quan Luo, Communications Research Laboratory, McMaster University, Hamilton, Ontario, Canada (Canada)
Kon Max Wong,

Page (NA) Paper number 2014

Abstract:

In digital communications, orthogonal pulse shapes are often used to represent message symbols for transmission through a channel. The design of such pulse shapes is formulated as a convex semidefinite programming problem, from which a globally optimal pulse shape can be efficiently found using interior point methods. The formulation is used to design filters which achieve the minimal bandwidth for a given filter length, and the minimal filter length for a given bandwidth. The effectiveness of the method is demonstrated by the design of waveforms with substantially improved performance over the `chip' waveforms specified in recent standards for digital mobile telecommunications.

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Reed-Solomon Codes and Their Performance for FCMA Systems in Fading Satellite Channel

Authors:

Mahmoud Ahmed Attia Ali,
Atef Abou-El-Azm,
M.F. Marie,

Page (NA) Paper number 5069

Abstract:

Fading in mobile satellite communications severely degrades the performance of data transmission. The emphasis in this paper is on the performance of uncoded and coded FCMA in the fading environment. The channel is modeled with nonfrequency selective Rice and Rayleigh fading using noncoherent demodulation with Reed-Solomon (RS) codes and hard decision decoding. Berlekamp-Massey decoding of RS codes is used to compensate for the fading. The bit error rate (BER) after decoding is calculated for specific codes and for different values of Rician channel parameters. The results of this paper are obtained by simulation techniques. They show that substantial coding gains are obtained compared to the uncoded reference system. They are also useful as reference for validating the results of simulation studies.

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Highly Efficient Signal Processing for Frequency Agile Power Line Communications

Authors:

Maja Sliskovic, Faculty of Electrical Engineering and Computing, Zagreb, Croatia (Croatia)

Page (NA) Paper number 1005

Abstract:

A modified frequency hopping signaling scheme has recently received a considerable attention by the designers of power line communication systems because of its insensitivity to frequency-selective and time-variant attenuation and high level of interference. In frequency hopping system, the increase of the data rate without increase of the hop rate is possible only if the receiver can detect all information-bearing tones simultaneously. This paper presents the structure of the noncoherent optimum receiver that minimizes number of required multiplications and memory locations. The receiver structure is based on computation of equidistant DFT coefficients. The proposed demodulator can be used in systems with variable bit rate. With slightly modified algorithm, rough synchronization can also be achieved. Comparison of computational complexity confirms superiority of presented algorithm over direct DFT calculation and FFT algorithm.

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Synchronization by Pilot Signal

Authors:

Jyrki J Joutsensalo,
Tapani - Ristaniemi,

Page (NA) Paper number 1146

Abstract:

In this paper, we propose a novel approach for code acquisition in the code-division multiple access (CDMA) communication system. The essential assumption is that the pilot signal of the desired user is available. If the codes are exactly orthogonal, the method can be derived from an optimization criterion. Using the pilot signal, the performance can be greatly improved without increasing computational complexity. Simulations show that our method clearly outperforms minimum variance method, eigenvector-based MUSIC, and matched filter.

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Nonlinear Prediction of Mobile Radio Channels: Measurements and MARS Model Designs

Authors:

Torbjörn Ekman,
Gernot Kubin,

Page (NA) Paper number 1305

Abstract:

The rapid time variation of mobile radio channels is often modeled as a random process with second order moments reflecting vehicle speed, bandwidth and the scattering environment. These statistics typically show that there is little room for prediction of channel properties such as received power or complex taps of the impulse response coefficients, at least when linear predictor structures are considered. We use mutual information estimation to measure statistical dependencies in sequences of wideband mobile radio channel data and find significant nonlinear dependencies, far exceeding the linear component. Based on these upper limits for the predictability of channel evolution over time intervals up to 30 ms ahead, we develop practical nonlinear predictor systems using Multivariate Adaptive Regression Splines (MARS). We demonstrate computationally efficient schemes that increase the prediction horizon beyond 10 ms, compared to less than 4 ms with linear predictors at comparable prediction gains.

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Multi-Stage Adaptive Predistortion of HPA Saturation Effects for Digital Television Transmission

Authors:

John T Stonick,
Virginia L Stonick,
José M.F. Moura,

Page (NA) Paper number 1480

Abstract:

This paper presents a new structure for adaptive predistortion of memoryless, nonlinear saturation effects caused by High Power Amplifiers (HPA). Timely compensation of HPA distortions is critical for cost-effective prevention of the cliff effect during terrestrial transmission of digital broadcast television. The new structure provides a two-stage approach: First, the forward model is identified using measured data, and then its inverse is computed. Replacing the analog system by the HPA forward model eliminates measurement noise and analog system delays during inverse modeling, yielding faster adaptation and reduced solution bias. Additionally, block processing reduces noise in the forward modeling. In the inverse modeling stage, the use of synthetic data and a closed form expression for the gradient result in more efficient convergence and more accurate solutions. Simulation results using measured HPA data demonstrate about a 5 dB improvement in SINAD over standard SNR operating conditions.

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Co-Channel Interference Cancellation For HDTV Receivers

Authors:

Monisha Ghosh,

Page (NA) Paper number 1564

Abstract:

A new method of co-channel interference rejection for digital television receivers is presented that uses a different rejection filter from the comb filter that was used in the prototype built by Zenith. This filter is optimized for rejection of co-channel NTSC interference in the presence of white noise and hence suffers a penalty of only 0.4 dB in AWGN as compared to 3.5 dB with the comb. The receiver structure with this filter, including required equalizer and trellis decoder modifications is presented, along with simulation results showing the improvement in performance in co-channel-plus-AWGN interference and hence in coverage area.

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Optimum Codec Companding for High-Speed PCM Data Transmission in Telephone Networks

Authors:

Nirmal Warke,
Murtaza Ali,

Page (NA) Paper number 2083

Abstract:

Codecs used in telephone networks in the United States have a mu-law companding characteristic solely designed for the purpose of transmitting voice signals. These mu-law codecs are not well suited for the latest generation of high-speed digital voiceband modems requiring PCM data transmission at the quantization levels of the codec. In this paper, we have shown that using a linear codec in the telephone network is optimal for PCM transmission in terms of a minimum symbol error rate performance of these PCM modems without any increase in the complexity of the modem. In practice, it would be necessary and relatively simple to implement an additional feature in the mu-law codec to detect a voiceband modem during the initial phase of training and switch to a linear companding characteristic.

IC992083.PDF (From Author) IC992083.PDF (Rasterized)

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Information Measure Based Stochastic System Identification Of ATM Network Traffic

Authors:

Baibing Li,
Bart De Moor,

Page (NA) Paper number 1545

Abstract:

For ATM network traffic, a new approach based on the Kullback-Leibler information measure is proposed for stochastic system identification of packet traffic. This approach, equivalent to the maximum marginal likehood estimate, can overcome the over-modeling problem in [1] such that much more parsimonious model order N can be obtained, and then lead significant reduction in the latter queueing analysis involving in O(N3) computational complexity. A practical case study is provided for a set of Internet traffic data.

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