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IEEE Workshop on Signal processing Advances in Wireless comm.nications (SPAWC)

作者： Christian Hofbauer Werner Haselmayr Mario Huemer Wireless Communications Silicon Austria Labs GmbH Linz Austria Inst. for Comm. Engineering and RF-Systems Johannes Kepler University Linz Linz Austria Institute of Signal Processing Johannes Kepler University Linz Linz Austria

ISBN: (数字)9781728154787

ISBN: (纸本)9781728154794

Unique word-orthogonal frequency division multiplexing (UW-OFDM) provides several performance benefits over conventional cyclic prefix (CP) based OFDM, such as excellent spectral sidelobe suppression properties and an outstanding bit error ratio performance. Aside from idealized assumptions, each comm.nication system faces challenges in conducting various system parameter estimation tasks. This work extends the current UW-OFDM signaling scheme by optimally including pilots tones in the frequency domain. Pilot tone based estimation of a carrier frequency offset (CFO) is conducted as a practical estimation example. Simulation results in a multipath environment demonstrate a significantly increased estimation accuracy in UW-OFDM over CP-OFDM, a result predicated on the inherent redundancy present in an UW-OFDM signal. The difference in estimation quality gets even more dominant with increasing CFO.

关键词： OFDM Estimation Frequency-domain analysis Time-domain analysis Optimization Generators Discrete Fourier transforms

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Performance comparison of algorithms for blind reverberation time estimation from speech

Performance comparison of algorithms for blind reverberation...

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International Workshop on Acoustic Signal Enhancement, IWAENC 2012

作者： Gaubitch, N.D. Löllmann, H.W. Jeub, M. Falk, T.H. Naylor, P.A. Vary, P. Brookes, M. Imperial College London United Kingdom Inst. of Comm. Systems and Data Proc. RWTH Aachen University Germany Institute National de La Recherche Scientifique MontrealQC Canada

ISBN: (纸本)9783800734511

The reverberation time, T60, is one of the key parameters used to quantify room acoustics. It can provide information about the quality and intelligibility of speech recorded in a reverberant environment, and it can be used to increase robustness to reverberation of speech processing algorithms. T60 can be determined directly from a measurement of the acoustic impulse response, but in situations where this is unavailable it must be estimated blindly from reverberant speech. In this contribution, we provide a study of three state-of-the-art methods for blind T60 estimation. Experimental results with a large number of talkers, simulated and measured acoustic impulse responses, and various levels of additive white Gaussian noise are presented. The relative merits of the three methods in terms of computational time, estimation accuracy, noise sensitivity and inter-talker variance are discussed. In general, all three methods are able to estimate the reverberation time to within 0.2 s for T60 ≤ 0.8 s and SNR ≥ 30 dB, while increasing the noise level causes overestimation. The relative computational speed of the three methods is also assessed. © 2012, inst.tute of Electrical and Electronics Engineers Inc. All rights reserved.

关键词： Reverberation

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Performance Comparison of Algorithms for Blind Reverberation Time Estimation from Speech

Performance Comparison of Algorithms for Blind Reverberation...

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IWAENC 2012;International Workshop on Acoustic Signal Enhancement

作者： N. D. Gaubitch H. W. Loellmann M. Jeub T. H. Falk P. A. Naylor P. Vary M. Brookes Centre for Law Enforcement Audio Research (CLEAR) Imperial College London Inst. of Comm. Systems and Data Proc. RWTH Aachen University Institute National de la Recherche Scientifique

The reverberation time, T60, is one of the key parameters used to quantify room acoustics. It can provide information about the quality and intelligibility of speech recorded in a reverberant environment, and it can be used to increase robustness to reverberation of speech processing algorithms. T60 can be determined directly from a measurement of the acoustic impulse response, but in situations where this is unavailable it must be estimated blindly from reverberant speech. In this contribution, we provide a study of three stateof- the-art methods for blind T60 estimation. Experimental results with a large number of talkers, simulated and measured acoustic impulse responses, and various levels of additive white Gaussian noise are presented. The relative merits of the three methods in terms of computational time, estimation accuracy, noise sensitivity and intertalker variance are discussed. In general, all three methods are able to estimate the reverberation time to within 0.2 s for T60 = 0.8 s and SNR = 30 dB, while increasing the noise level causes overestimation. The relative computational speed of the three methods is also assessed. Index Terms - Reverberation time, blind estimation

关键词： Speech Reverberation Signal to noise ratio Modulation Estimation error

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Hybrid ARQ with incremental redundant index assignments for iterative source-channel decoding

Hybrid ARQ with incremental redundant index assignments for ...

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2010 IEEE International Conference on comm.nications, ICC 2010

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

ISBN: (纸本)9781424464043

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. ©2010 IEEE.

关键词： Signal encoding

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

Turbo Source Compression with Jointly Optimized Inner Irregu...

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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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Turbo DeCodulation using Highly Redundant Index Assignments and Multi-Dimensional Mappings

Turbo DeCodulation using Highly Redundant Index Assignments ...

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4th International Symposium on Turbo Codes & Related Topics;6th International ITG-Conference on Source and Channel Coding

作者： Thorsten Clevorn Marc Adrat Peter Vary Inst. of Comm. Systems & Data Proc. (IND) RWTH Aachen University Germany Research Establishment for Applied Sciences (FGAN) Dept. FKIE/KOM Wachtberg Germany

We propose a block coded Turbo DeCodulation scheme, combining iterative demodulation, channel decoding, and source decoding at the receiver. Turbo DeCodulation resembles a serial concatenation of bit-interleaved coded modulation (BICM) and iterative source-channel decoding (ISCD) where the two iterative loops are connected by a single convolutional code. In our contribution, explicit channel (de)coding by a convolutional code, used in most Turbo systems, is omitted. inst.ad, highly redundant index assignments and multi-dimensional mappings, both based on short block codes, introduce the artificial redundancy. Thus, the proposed receiver possesses only a single iterative loop. The simulations results show a competitive or even superior performance compared to previously known systems. Furthermore, we present an extension to a flexible multi-mode system and a novel, simple design method for the index assignments.

关键词： Demodulation Robustness

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A 6.1 to 13.3-kb/s variable rate CELP codec (VR-CELP) for AMR speech coding

A 6.1 to 13.3-kb/s variable rate CELP codec (VR-CELP) for AM...

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

作者： S. Heinen M. Adratm O. Steil P. Vary W. Xu Inst. of Comm. Systems and Data Processing RWTH Aachen University of Technology Germany Rheinisch-Westfalische Technische Hochschule Aachen Aachen Nordrhein-Westfalen DE Department of Mobile Phone Development Siemens AG Munich Germany

We propose a new 6.1 to 13.3-kb/s speech codec called variable rate code-excited linear prediction (VR-CELP) for adaptive multi-rate (AMR) transmission over mobile radio channels such as GSM or UMTS. The AMR concept allows to operate with almost wireline speech quality for poor channel conditions and better quality for good channel conditions. This is achieved by dynamically splitting the gross bit rate of the transmission system between source and channel coding according to the current channel conditions. Thus the source coding scheme must be designed for seamless switching between rates without annoying artifacts. To enhance the transmission quality under very poor channel conditions, a new powerful error concealment strategy based on estimation theory is applied.

关键词： Speech coding Bit rate Speech codecs Speech analysis Vector quantization Linear predictive coding Decoding Testing Iterative algorithms Encoding

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Trade-Off Analysis of a Low-Power Image Coding Algorithm

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Journal of VLSI Signal processing systems for Signal, Image, and Video Technology 1998年第1期18.0卷 65-80页

作者： Masselos, K. Merakos, P. Stouraitis, T. Goutis, C.E. Dept. of Elec. and Comp. Engineering University of Patras Rio 26500 Greece University of Patras Greece VLSI Systems Engineering Univ. Manchester Inst. Sci. Technol. United Kingdom Inter-Univ. Micro Electronics Centre Belgium IEEE Technical Chamber of Greece Greece Dept. of Elec. and Comp. Engineering University of Patras Greece Technical Chamber of Greece Greece Ohio State University United States University of Florida United States University of Cincinnati United States University of Athens Greece Athena Group Inc. Greece Natl. Elec. Manufacturers Assoc. L. Cincinnati OH United States VLSI Syst. and Applic. Tech. Comm. IEEE Circuits and Systems Society United States DSP Technical Committee IEEE Signal Processing Society United States School of Physics and Mathematics University of Athens Greece Greek P.T.T. Greece Dept. of Elec. and Electron. Eng. University of Strathclyde United Kingdom Dept. of Elec. and Electron. Eng. University of Newcastle upon Tyne United Kingdom Dept. of Elec. and Comp. Engineering University of Patras Greece

In this paper, a novel algorithm for low-power image coding and decoding is presented and the various inherent trade-offs are described and investigated in detail. The algorithm reduces the memory requirements of vector quantization, i.e., the size of memory required for the codebook and the number of memory accesses by using small codebooks. This significantly reduces the memory-related power consumption, which is an important part of the total power budget. To compensate for the loss of quality introduced by the small codebook size, simple transformations are applied on the codewords during coding. Thus, small codebooks are extended through computations and the main coding task becomes computation-based rather than memory-based. Each image block is encoded by a codeword index and a set of transformation parameters. The algorithm leads to power savings of a factor of 10 in coding and of a factor of 3 in decoding, at least in comparison to classical full-search vector quantization. In terms of SNR, the image quality is better than or comparable to that corresponding to full-search vector quantization, depending on the size of the codebook that is used. The main disadvantage of the proposed algorithm is the decrease of the compression ratio in comparison to vector quantization. The trade-off between image quality and power consumption is dominant in this algorithm and is mainly determined by the size of the codebook.

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