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18th IEEE International Conference on Machine Learning and Applications, ICMLA 2019

作者： Apfeld, Sabine Charlish, Alexander Ascheid, Gerd Dept. Sensor Data and Information Fusion Fraunhofer Fkie Wachtberg Germany Institute for Communication Technologies and Embedded Systems Rwth Aachen University Germany

ISBN: (纸本)9781728145495

This paper adapts and extends the previously published hierarchical modelling approach of multifunction radars as systems that speak a language. We propose Long Short-Term Memory neural networks for learning the emitters' grammar and predicting their emissions. The approach is demonstrated using simulations of an airborne multifunction radar with three different resource management techniques of varying complexity. A comparison with simple prediction strategies shows that a huge improvement in accuracy can be achieved by using Long Short-Term Memory networks for predicting complex radar emitter behaviour. © 2019 IEEE.

关键词： Long short-term memory

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Noise-shaping for closed-loop Multi-Channel Linear Prediction

Noise-shaping for closed-loop Multi-Channel Linear Predictio...

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International Conference on Acoustics, Speech, and Signal processing (ICASSP)

作者： Niklas Koep Magnus Schäfer Peter Vary Institute for Theoretical Information Technology RWTH Aachen University Institute of Communication Systems and Data Processing (iNd) RWTH Aachen University Germany

In this paper, a novel noise-shaping method for Multi-Channel Linear Prediction (MCLP) is presented. Without special consideration, the quantization noise of the prediction error poses a serious problem in multi-chann... 详细信息

关键词： Quantization (signal) Signal to noise ratio Noise shaping Gain Decoding Psychoacoustics

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Convergence behavior of iterative source-channel decoding

Convergence behavior of iterative source-channel decoding

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International Conference on Acoustics, Speech, and Signal processing (ICASSP)

作者： M. Adrat U. von Agris P. Vary Institute of Communication Systems and Data Processing Aachen University of Technology Institute of Communication Systems and Data Processing Aachen University of Technology Germany

Whenever digital communication is subject to transmission errors, the utilization of reliability information at the receiver is beneficial. Soft-channel decoding techniques improve the error correcting capabilities and in addition, softbit source decoding performs error concealment. While the reliability gain of channel decoding is based on artificial diversities explicitly introduced at the transmitter, softbit source decoding exploits implicit redundancies remaining in the bitstream after state-of-the-art source encoding. In an iterative source-channel decoding scheme the reliability gains of both are exchanged iteratively in order to enhance the common error resistance. However, first investigations have shown that the number of profitable iterations is limited to small values. In this paper, the convergence behavior is analyzed using a modern tool called EXIT charts. EXIT charts permit us to predict the convergence behavior by studying the individual components. As a novelty, EXIT characteristics are specified for softbit source decoding. The convergence analysis is confirmed by simulation.

关键词： Convergence Iterative decoding Computer errors Redundancy Codecs Algebra AWGN data processing Digital communication Analytical models

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Transactions Papers source-optimized channel coding for digital transmission channels

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IEEE Transactions on communications 2005年第4期53卷 592-600页

作者： S. Heinen P. Vary Infineon Technologies AG Development Center Düsseldorf Dusseldorf Germany Institute of Communication Systems and Data Processing RWTH Aachen University Aachen Germany

We present a new class of nonlinear block codes called source-optimized channel codes (SOCCs), which are particularly designed for parametric source encoding of speech, audio, and video. In contrast to conventional channel codes, the new codes are not optimized for minimizing residual bit-error rate, but maximizing the signal-to-noise ratio of transmitted source codec parameters. The decoding of SOCCs is not based on bit-error correction, but on parameter estimation. We compare SOCCs with other approaches to joint source/channel coding such as channel-optimized vector quantization, channel-constrained vector quantization, unequal error protection, and source-controlled channel decoding. In terms of performance, SOCCs show better robustness if under channel mismatch conditions. For real-world applications, SOCCs are attractive, since the separation of source and channel codec is preserved.

关键词： Channel coding Codecs Decoding Vector quantization Block codes Speech coding Bit error rate Signal to noise ratio Parameter estimation Error correction codes

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NOVEL ITERATIVE MULTIPLE DESCRIPTION CODING FOR CORRELATED SOURCES

NOVEL ITERATIVE MULTIPLE DESCRIPTION CODING FOR CORRELATED S...

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European Signal processing Conference

作者： Laurent Schmalen Matthias Tschauner Tobias Breddermann Peter Vary Alcatel-Lucent Bell Labs Institute of Communication Systems and Data Processing RWTH Aachen University

In this paper, we study a novel multiple description coding approach which uses a convolutional code to generate the individual descriptions. Contrary to most conventional multiple description schemes, which attempt to partially reconstruct the signal in the presence of a packet loss, we are interested in the question, whether the signal can be completely reconstructed at the receiver if one description is missing by exploiting the residual source correlation. Usually, audio-visual source encoders for digital mobile communications extract parameters that - due to delay and complexity constraints - exhibit some residual redundancy. This residual redundancy is exploited in a multiple description receiver by performing iterative source-channel decoding (ISCD). The source correlation required for near perfect reconstruction in case of a loss of one description is analyzed by means of EXIT charts and simulation results show the superior performance of the new approach.

关键词： Decoding Iterative decoding Correlation Receivers Encoding Indexes Vectors

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Flexible nonuniform filter banks using allpass transformation of multiple order

Flexible nonuniform filter banks using allpass transformatio...

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European Signal processing Conference (EUSIPCO)

作者： M. Kappelan B. Strauß P. Vary Institute of Communication Systems and Data Processing (IND) RWTH Aachen University of Technology D-52056 Aachen Germany

This paper deals with allpass frequency transformations of uniform filter banks to achieve nonuniform band-widths. The known transformation with an allpass of first order [1] [4] [5] [6] is extended to an allpass tran... 详细信息

关键词： Prototypes Delays Filter banks Finite impulse response filters Transfer functions Frequency modulation

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Perfect sequence lms for rapid acquisition of continuous-azimuth head related impulse responses

Perfect sequence lms for rapid acquisition of continuous-azi...

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IEEE Workshop on Applications of Signal processing to Audio and Acoustics

作者： Christiane Antweiler Gerald Enzner Institute of Communication Systems and Data Processing RWTH Aachen University Aachen Germany Institute of Communication Acoustics Ruhr University of Bochum Bochum Germany

In recent publications, continuous-azimuth inference of head related impulse responses (HRIRs) was treated as a time-varying system identification problem on the basis of dynamical measurements. The system identification thus can be handled by LMS-type adaptive filters for which we have the freedom to choose the excitation signal in this application. In order to provide the perspective of reducing the measurement time to a minimum, we now suggest the optimal excitation signal in terms of the rate of convergence. This excitation signal is given by perfect sequences (PSEQs) out of the larger family of periodic pseudo-noise signals. After the discussion of specific implications of perfect sequences, we compare the performances of our perfect-sequence LMS algorithm (PSEQ-LMS) to the results of white noise processing. We demonstrate a uniform improvement by PSEQ-LMS in terms of instrumental mean-square error analysis as well as subjective listening to dynamic HRIRs. Both measures turn out to be consistent.

关键词： System identification Time varying systems Adaptive filters Signal processing Time measurement Convergence Least squares approximation White noise Instruments Error analysis

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Perfect-sweep NLMS for time-variant acoustic system identification

Perfect-sweep NLMS for time-variant acoustic system identifi...

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International Conference on Acoustics, Speech, and Signal processing (ICASSP)

作者： C. Antweiler A. Telle P. Vary G. Enzner Institute of Communication Systems and Data Processing RWTH Aachen University Aachen Germany Institute of Communication Acoustics Ruhr University of Bochum Bochum Germany

Fast and robust acoustic system identification is still a research topic of interest, because of the typically time-variant nature of acoustic systems and the natural performance limitation of electroacoustic measurement equipment. In this paper, we propose NLMS-type adaptive identification with perfect-sweep excitation. The perfect-sweep is derived from the more general class of perfect sequences and, thus, it inherits periodicity and especially the desired decorrelation property known from perfect sequences. Moreover, the perfect-sweep shows the desirable characteristics of swept sine signals regarding the immunity against non-linear loudspeaker distortions. On this basis, we first demonstrate the fast tracking ability of the perfect-sweep NLMS algorithm via computer generated simulation of a time-variant acoustic system. Then, the robustness of the perfect-sweep NLMS algorithm against non-linear characteristics of real measurements in a time-invariant case is presented. By finally addressing the measurement of quasi-continuous head-related impulse responses, we face the combined challenge of time-variant and possibly non-linear distorted acoustic system identification in a real application scenario and we can demonstrate the superiority of the perfect-sweep NLMS algorithm.

关键词： Acoustics Acoustic measurements Loudspeakers Distortion measurement Acoustic distortion Signal to noise ratio Convergence

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On joint source-channel decoding for correlated sources

On joint source-channel decoding for correlated sources

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International Conference on Acoustics, Speech, and Signal processing (ICASSP)

作者： M. Adrat R. Haenel P. Vary Institte of Communication Systems and Data Processing RWTH Aachen University of Technology Germany

Whenever digital communications is subject to constraints in delay and complexity, some residual redundancy remains after source coding. Shannon's separation theorem does not hold strictly and hence joint source-channel decoding is beneficial. In this paper we discuss two competing approaches exploiting residual redundancy in terms of (auto)-correlation. Both will be analyzed with respect to their impact on quantization noise and channel distortion. The first approach exploits redundancy at the receiver for minimum mean squared error estimation of source codec parameters. The second approach exploits correlation at the transmitter by predictive encoding. The prediction gain allows to reduce quantization noise or bitrate. The spare bitrate can be used for specific error protection and hence reduction of channel distortion. But otherwise, less redundancy is utilizable at the receiver for parameter estimation. Simulations will show that the predictive encoding scheme improves quality in a wide range of channel conditions, but in contrast to Shannon's separation theorem the utilization of correlation at the receiver side is beneficial in very bad channel conditions.

关键词： Signal to noise ratio Decoding Noise measurement

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Dependency of the Power and Delay Domain Parameters on Antenna Height and Distance in Urban Macro Cell

Dependency of the Power and Delay Domain Parameters on Anten...

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European Conference on Antennas and Propagation

作者： Annika Bottcher Christian Schneider Peter Vary Reiner S. Thoma Institute of Communication Systems and Data Processing RWTH Aachen University Institute of Information Technology Ilmenau University of Technology

ISBN: (纸本)9781457702501

Large scale parameters (LSP) mainly describe the distribution of the transmitted power over different dimensions of the channel. The required statistic is obtained from MIMO channel measurements with an appropriate post processing procedure. In this contribution, we perform the characterization of large scale parameters within an urban macro cell scenario. We focus on investigating the influence of the base station antenna height and the distance between the mobile terminal and the base station. The parameter analysis is conform to the 3GPP SCM and WINNER channel models. We found that parameters as delay spread, K-factor and cross polarization ratio show significant dependency. Because of their potential impact on the system performance these results should be considered and validated in current and future channel models.

关键词： Large scale parameter Measurement data Channel sounding Spatial channel modeling

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