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Estimation of Rapidly Time-Varying Harmonic Noise for Speech Enhancement

IEEE Signal processing Letters 2012年第10期19卷 659-662页

作者： Esch, Thomas Rüngeler, Matthias Heese, Florian Vary, Peter Institute of Communication Systems and Data Processing RWTH Aachen University Aachen 52074 Germany

Robust speech enhancement relies on the estimation of stationary as well as non-stationary background noise. This contribution presents a novel approach for estimating the short-term power spectral densities (ST-PSDs) of rapidly time-varying harmonic noise as produced, e.g., by cars or motorcycles. The well-known Minimum Statistics algorithm is modified by frequency warping controlled by the fundamental frequency of the harmonic noise which is assumed to be known a priori. The resulting noise estimates are used for the enhancement of the noisy signal. A detailed description of the algorithm is given and it is shown by a thorough analysis that the new solution considerably outperforms two conventional noise ST-PSD estimation techniques. © 2012 IEEE

关键词： Adverse environments noise reduction non-stationary noise speech enhancement

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A semi-analytical model for the binaural coherence of noise fields

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IEEE Signal processing Letters 2011年第3期18卷 197-200页

作者： Jeub, Marco Dörbecker, Matthias Vary, Peter Institute of Communication Systems and Data Processing RWTH Aachen University Aachen 52074 Germany Harman/Becker Automotive Systems GmbH Karlsbad Germany

A novel semi-analytical signal processing model for the binaural coherence of homogeneous isotropic noise fields is presented in this contribution. This is derived from a simplified geometrical model of the human head, where the shadowing between the left and right ear is modeled by two nonreflecting circular plates. Based on Kirchhoff's diffraction theory, it is shown how the corresponding coherence is calculated. This model can be used as part of various binaural signal processing algorithms, such as speech enhancement for digital hearing aids or binaural speech transmission systems. In experiments using an artificial head in a highly reverberant environment, it is confirmed that the proposed theoretical model shows a good match with the coherence obtained from measurements. © 2011 IEEE.

关键词： Speech enhancement

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A novel approach for impulse response measurements in environments with time-varying noise

A novel approach for impulse response measurements in enviro...

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20th International Congress on Acoustics 2010, ICA 2010 - Incorporating the 2010 Annual Conference of the Australian Acoustical Society

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

ISBN: (纸本)9781617827457

A typical measure taken against unavoidable noise during the measurement of an impulse response of an acoustical system is to repeat the measurement a couple of times and coherently average the distinct measurements. With stationary noise this is the optimal solution which gives an improvement of the signal-to-noise ratio of about 3 dB per doubling of the number of measurements averaged. If transient or time varying noise components are present the resulting accuracy suffers significantly. In this contribution, the authors propose a weighted averaging approach where the distinct measurements are weighted according to the present noise power. The noise power is estimated with the so-called SWiC method (sliding window correlation) recently published by the authors. Copyright © (2010) by the International Congress on Acoustics.

关键词： Impulse response

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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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Dual microphone wind noise reduction by exploiting the complex coherence 11

Dual microphone wind noise reduction by exploiting the compl...

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11th ITG Symposium on Speech communication

作者： Nelke, Christoph Matthias Vary, Peter Institute of Communication Systems and Data Processing RWTH Aachen University Germany

ISBN: (纸本)9783800736409

This contribution presents methods for detecting, estimating and reducing the effect of wind noise picked up by a dual microphone communication device. To discriminate the sound induced by the wind stream the complex coherence properties of the two microphone signals are exploited. Therefore, a coherence analysis of recorded signals is given. Based on the phase variance of the complex coherence function a wind noise detection mechanism is presented. Furthermore, an approach for the enhancement of distorted speech is developed, which uses the phase variance and the magnitude of the complex cross power spectral density (PSD). This stage is realized by a noise PSD estimation and a dedicated spectral subtraction weighting. The reduction of wind noise by means of spectral weighting often leads to severe degradations to the speech signal. However, the evaluation of the proposed algorithm demonstrates a better performance compared to related approaches. © VDE VERLAG GMBH Berlin Offenbach.

关键词： Speech communication

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Semi-analytical solution for rate distortion function and OPTA for sources with arbitrary distribution

Semi-analytical solution for rate distortion function and OP...

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2010 International ITG Conference on Source and Channel Coding, SCC 2010

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

ISBN: (纸本)9781424468720

The rate distortion function is a widely used theoretical bound which describes the minimum mean square error (MMSE) distortion for a given number of quantization bits when quantizing a scalar random variable. An analytical solution for this function is only available for a small number of probability density functions (pdf), such as the Gaussian pdf. For arbitrary pdfs, the Blahut-Arimoto algorithm [1], [2] needs to be applied to iteratively estimate the rate distortion function. We propose a novel (semi-)analytical and non-iterative method to calculate the rate distortion function for sources with arbitrary pdfs. This method is based on the Guo-Shamai-Verdú (GSV) theorem [3]. Furthermore, it is possible to apply the proposed method for calculating the Optimum Performance Theoretically Attainable (OPTA) for arbitrarily distributed input symbols observed through an AWGN channel. © VDE Verlag -GMBH.

关键词： Mean square error

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Error resilient turbo compression of source codec parameters using inner irregular codes

Error resilient turbo compression of source codec parameters...

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2010 International ITG Conference on Source and Channel Coding, SCC 2010

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

ISBN: (纸本)9781424468720

In this paper, we present a novel near-lossless compression scheme for scalar quantized source codec parameters. The scheme is comparable to a Turbo source coding approach and can inherently incorporate protection against transmission errors. We show that using the concept of EXIT charts and inner irregular convolutional codes of rate > 1, a linear programming optimization problem can be formulated resulting in the minimization of the transmitted number of bits given a certain channel quality. The solution of this problem leads to an irregular inner code offering the best possible compression. The advantage of this method in comparison to a previous proposal is the easy adaptability to varying source conditions. © VDE Verlag -GMBH.

关键词： Convolutional codes

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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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The FlexCode source-channel coding approach for audio and speech transmission

The FlexCode source-channel coding approach for audio and sp...

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2010 International ITG Conference on Source and Channel Coding, SCC 2010

作者： Schmalen, Laurent Schlien, Thomas Vary, Peter Institute of Communication Systems and Data Processing RWTH Aachen University Germany

ISBN: (纸本)9781424468720

The FlexCode project is a joint research project of KTH Stockholm, rwth aachen university, Ericsson AB, Nokia, and Orange/France Telecom under the umbrella of the sixth framework programme of the European Commission (http://***). In this paper we present the generic channel coding platform developed within the FlexCode project. This channel encoder provides iterative source-channel decoding at the receiver in order to achieve near-capacity transmission of the source coder parameters. The structure of the encoder allows to flexibly adapt instantaneously the source and channel coding rates according to varying conditions. The approach handles both main operating modes of the FlexCode project: constrained resolution scalar quantization and constrained entropy scalar quantization which lead to either constant or variable bit rates. © VDE Verlag -GMBH.

关键词： Channel coding

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