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Hybrid ARQ with Incremental Redundant Index Assignments for Iterative Source-Channel Decoding

Hybrid ARQ with Incremental Redundant Index Assignments for ...

引用

IEEE International Conference on communications (ICC)

作者： T. Clevorn L. Schmalen P. Vary M. Adrat Design Center NRW Infineon Technologies Duisburg Germany Institute of Communication Systems and Data Processing RWTH Aachen University Aachen Germany FKIE Communication Systems Fraunhofer-Gesellschaft Wachtberg Germany

Iterative source-channel decoding (ISCD) exploits the residual redundancy of source codec parameters by using the Turbo principle. In this paper we extend the excellent capabilities of ISCD to a hybrid automatic repeat request (HARQ) scheme by novel incremental redundant index assignments. The incremental redundancy for HARQ is supplied by the source encoder and not the channel encoder. Simulation results show an excellent performance over a wide range of channel conditions, with inherently adapted bandwidth and complexity.

关键词： Automatic repeat request Iterative decoding Speech Multiaccess communication Cyclic redundancy check Convolutional codes Computational complexity communications Society Paper technology data processing

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On the Overshooting Effect in EXIT Charts of Iterative Source-Channel Decoding

On the Overshooting Effect in EXIT Charts of Iterative Sourc...

引用

IEEE International Conference on communications (ICC)

作者： M. Adrat M. Antweiler L. Schmalen P. Vary T. Clevorn FKIE Communication Systems Fraunhofer-Gesellschaft Wachtberg Germany Institute of Communication Systems and Data Processing RWTH University Aachen Germany Design Center NRW Infineon Technologies Duisburg Germany

Source codec parameters determined by modern source encoders are usually very sensitive to noise on the transmission link. Natural residual redundancy of these parameters can be utilized by Soft Decision Source Decoding (SDSD)to increase the error resistance. Serially concatenating Forward Error Correction (FEC) and SDSD leads to a Iterative Source-Channel Decoding (ISCD) scheme which offers further improvements in robustness. ISCD exploits natural residual source redundancy by SDSD and artificial channel coding redundancy due to FEC in an iterative Turbo-like process. The convergence behavior of ISCD schemes can be analyzed by EXtrinsic Information Transfer (EXIT) charts. However, if an advanced ISCD system design is applied, offering incremental quality improvements for many iterations, the EXIT curves for SDSD do not specify a tight bound for the decoding trajectory anymore. In the initial iterations, the decoding trajectory "overshoots" the EXIT curves of SDSD, e.g.. In this paper, we give reasons for this overshooting effect. In addition, we propose a novel solution for determining the EXIT curve of SDSD which allows a more precise convergence analysis.

关键词： Iterative decoding Codecs Redundancy Forward error correction Virtual manufacturing communications Society data processing Noise robustness Channel coding Convergence

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Turbo Source Compression with Jointly Optimized Inner Irregular and Outer Irregular Codes

Turbo Source Compression with Jointly Optimized Inner Irregu...

引用

IEEE Conference on Vehicular Technology (VTC)

作者： Laurent Schmalen Peter Vary Thorsten Clevorn Marc Adrat Inst. of Comm. Systems and Data Processing RWTH Aachen University Germany Infineon Technologies Duisburg Germany Fraunhofer Gesellschaft FKIE Communication Systems Germany

In this paper, we present a near-lossless compression scheme for scalar-quantized source codec parameters based on iterative source-channel decoding (ISCD). The scheme is comparable to a Turbo source encoder and can inherently incorporate protection against transmission errors. In order to realize a close-to-capacity transmission/compression scheme, we employ irregular redundant bit mappings and irregular inner channel codes. The irregular inner code is composed of (pseudo) randomly punctured convolutional codes of rate > 1, i.e., the compression is performed by strong puncturing of the inner code. The optimization goal is a minimum number of transmitted bits given certain source and channel conditions. We show how the inner and outer component can be jointly optimized resulting in a constrained nonlinear programming problem. An additional successive approximation based on linear programming is also presented. Simulation examples show the advantage over systems employing only a single irregular component.

关键词： Optimization Iterative decoding Decoding Convolutional codes Encoding Joints Speech

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VOICE SOURCE ESTIMATION FOR ARTIFICIAL BANDWIDTH EXTENSION OF TELEPHONE SPEECH

VOICE SOURCE ESTIMATION FOR ARTIFICIAL BANDWIDTH EXTENSION O...

引用

IEEE International Conference on Acoustics, Speech, and Signal processing

作者： Mark R. P. Thomas Jon Gudnason Patrick A. Naylor Bernd Geiser Peter Vary Communications and Signal Processing Group Imperial College London Exhibition Road London SW7 2AZ UK Institute of Communication Systems and Data Processing (iNd) RWTH Aachen University Germany

ISBN: (纸本)9781424442959

Artificial bandwidth extension (ABWE) of speech signals aims to estimate wideband speech (50 Hz - 7 kHz) from narrowband signals (300 Hz - 3.4 kHz). Applying the source-filter model of speech, many existing algorithms estimate vocal tract filter parameters independently of the source signal. However, many current methods for extending the narrowband voice source signal are limited to straightforward signal processing techniques which are only effective for high-band estimation. This paper presents a method for ABWE that employs novel data-driven modelling and an existing spectral mirroring technique to estimate the wideband source signal in both the high and low extension bands. A state-of-the-art Hidden Markov Model-based estimator evaluates the temporal and spectral envelopes in the missing frequency bands, with which the ABWE speech signal is synthesized. Informal listening tests comparing two existing source estimation techniques and two permutations of the proposed approach show an improvement in the perceived bandwidth of speech signals, in particular towards low frequencies. Subjective tests on the same data show a preference for the proposed techniques over the existing methods under test.

关键词： Speech enhancement artificial bandwidth extension voice source modelling Speech enhancement Speech Bandwidth Voice Narrowband signals Voice Signal Frequency bands

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Improving UMTS LTE performance by UEP in high order modulation

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7th International Workshop on Multi-Carrier systems and Solutions, MC-SS 2009

作者： Lüders, Helge Minwegen, Andreas Vary, Peter Institute of Communication Systems and Data Processing RWTH Aachen University Germany

ISBN: (纸本)9789048125296

In this contribution we investigate the performance of the UMTS Long Term Evolution (LTE) physical layer using turbo coding and 64QAM with Gray mapping. We show how the mapping of systematic and parity bits to the six different bit positions defining one complex 64QAM symbol influences the convergence of the turbo decoder and thereby the bit error rate (BER) performance as well as number of necessary decoding iterations. Exploiting the unequal error protection (UEP) property of Gray mapped 64QAM results in an SNR performance gain of approximately 2 dB for the non-iterative system and in addition leads to a significant reduction of the necessary decoding iterations when iterative decoding is performed. © 2009 Springer-Verlag.

关键词： Bit error rate

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Joint pre-echo control and frame erasure concealment for voip audio codecs

Joint pre-echo control and frame erasure concealment for voi...

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17th European Signal processing Conference, EUSIPCO 2009

作者： Geiser, Bernd Vary, Peter Institute of Communication Systems and Data Processing RWTH Aachen University Germany

Pre-echo artifacts are a common problem of transform audio codecs owing to their comparatively large block lengths. If such codecs are used in Voice-over-IP (VoIP) applications, the concealment of lost audio packets poses another major issue. These two problems are usually solved by two separate algorithmic components: A mechanism for pre-echo control (PEC) and a frame erasure concealment (FEC) algorithm. In this contribution we propose a combined PEC/FEC approach which elegantly solves both problems based on a single side information stream. In particular, our proposal is suitable for (subband) transform audio codecs that are based on a frequency transform with half-overlapped windowing (such as the popular MDCT). The described method is part of the Huawei/ETRI candidate for the future ITU-T super-wideband extensions of Rec. G.729.1 and G.718 where it has been successfully applied in the context of high frequency (8-14 kHz) encoding. © EURASIP, 2009.

关键词： Voice/data communication systems

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Properties and performance bounds of linear analog block codes

Properties and performance bounds of linear analog block cod...

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43rd Asilomar Conference on Signals, systems and Computers

作者： Rüngeler, Matthias Schotsch, Birgit Vary, Peter Institute of Communication Systems and Data Processing RWTH Aachen University Germany

ISBN: (纸本)9781424458271

Linear analog block codes have been considered for transmission of discrete-time and continuous-amplitude data. In this paper, the fidelity measure parameter SNR (pSNR) at the receiver is derived for an arbitrary generator matrix P using an additive white Gaussian noise (AWGN) channel. In contrast to [1], it is shown that the performance of linear analog bock codes is dependent on the eigenvalues of the matrix PTP and not only on the dimensions of the matrix P. Surprisingly, the quality of the received values is independent of the code rate r, and e.g. a simple identity matrix has the optimal eigenvalues. Furthermore, the theoretical fidelity bound OPTA (Optimum Performance Theoretically Attainable) is used to assess the performance of a transmission system of continuous-amplitude data. © 2009 IEEE.

关键词： Block codes

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A new method for snr-estimation in impulse response measurements

A new method for snr-estimation in impulse response measurem...

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17th European Signal processing Conference, EUSIPCO 2009

作者： Telle, Aulis Vary, Peter Institute of Communication Systems and Data Processing RWTH Aachen University 52056 Aachen Germany

Methods for measuring the impulse response of a linear transmission system and system identification algorithms in general must be robust against noise in the measured system response. To handle the noise it is of great advantage to know the instantaneous signal-to-noise ratio (SNR), especially in situations with changing noise conditions. In this paper we present a new approach for estimating the SNR during an impulse response measurement by means of the so-called sliding window correlation (SWiC) as introduced in this paper. The performance of the proposed method is evaluated by means of simulation results. © EURASIP, 2009.

关键词： Signal to noise ratio

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Near end listening enhancement optimized with respect to speech intelligibility index

Near end listening enhancement optimized with respect to spe...

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17th European Signal processing Conference, EUSIPCO 2009

作者： Sauert, Bastian Vary, Peter Institute of Communication Systems and Data Processing RWTH Aachen University 52056 Aachen Germany

Signal processing algorithms for near end listening enhancement allow to improve the intelligibility of clean (far end) speech for the near end listener who perceives not only the far end speech but also ambient background noise. A typical scenario is mobile communication conducted in the presence of acoustical background noise such as traffic or babble noise. In this contribution we analyze the calculation rules of the Speech Intelligibility Index (SII) and derive a simple condition for the speech spectrum level of every subband that maximizes the SII for a given noise spectrum level. This rule is used to derive a theoretical bound for a maximum achievable SII as well as a new SII optimized algorithm for near end listening enhancement. The impact of ignoring masking effects in the algorithm is also investigated and seconds our SNR recovery algorithm proposed earlier. Instrumental evaluation shows that the new algorithm performs close to the established theoretical bound. © EURASIP, 2009.

关键词： Signal to noise ratio

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General least-squares design of allpass transformed DFT filter-banks

General least-squares design of allpass transformed DFT filt...

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17th European Signal processing Conference, EUSIPCO 2009

作者： Löllmann, Heinrich W. Dartmann, Guido Vary, Peter Institute of Communication Systems and Data Processing RWTH Aachen University 52056 Aachen Germany Institute for Integrated Signal Processing Systems RWTH Aachen University 52056 Aachen Germany

The allpass transformation is a common approach to design a frequency warped analysis-synthesis filter-bank (AS FB) with non-uniform time-frequency resolution. Such filter-banks are beneficial, e.g., for speech and audio subband processing, but their synthesis filter-bank design is more complex as for uniform filter-banks. This problem is tackled by a new least-squares error (LS) synthesis filter-bank design. Magnitude and phase distortions are minimized with the constraints for complete aliasing cancellation and a restricted signal delay. The coefficie nts of the synthesis filters are determined by a linear set of equations with linear constraints which can be easily solved. The obtained FIR synthesis subband filters exhibit a frequency selectivity similar to that of the IIR analysis filters. The new concept is very general and can be applied in case of an allpass transformation of first or higher order as well as for polyphase network (PPN) filter-banks where the prototype filter length exceeds the number of channels. These properties make the proposed design of interest for various subband processing systems requiring non-uniform frequency bands. © EURASIP, 2009.

关键词： Filter banks

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