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System identification with perfect sequence excitation – Efficient NLMS vs. Inverse cyclic convolution 11

System identification with perfect sequence excitation – Ef...

引用

11th ITG Symposium on Speech communication

作者： Antweiler, Christiane Kühl, Stefan Sauert, Bastian Vary, Peter Institute of Communication Systems and Data Processing RWTH Aachen University Aachen52056 Germany

ISBN: (纸本)9783800736409

Linear transmission systems are often characterized by their impulse responses. A simple and fast approach to acquire these impulse responses is the normalized least-mean-square (NLMS) algorithm in combination with a perfect sequence excitation. It is not only applicable to static impulse response measurements, but has been optimized especially for the tracking of time varying linear systems. In this paper, different implementation strategies of the perfect sequence excited NLMS algorithm, namely the efficient NLMS and the inverse cyclic convolution algorithm, are discussed and compared in terms of performance, complexity, and applicability. As a main result it is shown that for certain conditions all algorithmic variants can be transferred to each other. © VDE VERLAG GMBH.

关键词： Impulse response

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Energy efficiency of network-based acoustic echo control in mobile radio 10

Energy efficiency of network-based acoustic echo control in ...

引用

10th ITG Symposium on Speech communication, ITGspeech 2012

作者： Pawig, Matthias Vary, Peter Institute of Communication Systems and Data Processing RWTH Aachen University Aachen52056 Germany

ISBN: (纸本)9783800734559

In this contribution we investigate the energy efficiency of network-based acoustic echo control (AEC). Usually, AEC is implemented in the mobile terminal. Due to the limited resources of the mobile terminal in terms of the battery, computational capacity and memory, the complexity of the AEC algorithm has to be limited. In this paper we will investigate the idea to move the AEC processing from the mobile terminal to a network-based processing unit which allows to use more sophisticated algorithms. As this might require the use of improved speech codecs with higher bit rate, we will evaluate the trade-off between increased power for the radio transmission with additional bit rate and the reduced power for signal processing at the mobile terminal. This implies to take into consideration the attenuation of the radio channel according to, e.g., the Okumura-Hata path loss model. © VDE VERLAG GMBH • Berlin • Offenbach.

关键词： Energy efficiency

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Selflearning codebook speech enhancement 11

Selflearning codebook speech enhancement

引用

11th ITG Symposium on Speech communication

作者： Heese, Florian Nelke, Christoph Matthias Niermann, Markus Vary, Peter Institute of Communication Systems and Data Processing RWTH Aachen University Aachen52056 Germany

ISBN: (纸本)9783800736409

A novel speech enhancement system is presented which exploits a codebook for noise estimation. In contrast to state-of-the-art noise estimators which usually rely on the assumption that the noise signal is only slightly time-varying, codebook approaches allow also non-stationary environments. The basic concept of the proposed codebook noise estimation is a superposition of a scaled speech and noise codebook entry. In order to be independent of a priori noise knowledge, the new estimator is able to learn new noise types online. Training vectors for codebook updates are identified using a speech activity detector (VAD) and a codebook mismatch measure. The VAD is realized as part of the codebook matching. A Wiener filter or any state-of-the-art weighting rule can be applied subsequently for speech enhancement. Experiments confirmed that the new system is able to learn new noise types and provides improved performance compared to state-of-the-art algorithms. © VDE VERLAG GMBH Berlin Offenbach.

关键词： Speech enhancement

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Steganographic packet loss concealment for wireless VoIP

Steganographic packet loss concealment for wireless VoIP

引用

8. ITG-Fachtagung Sprachkommunikation 2008 - 8th ITG Conference on Speech communication 2008

作者： Geiser, Bernd Mertz, Frank Vary, Peter Institute of Communication Systems and Data Processing RWTH Aachen University Aachen52056 Germany

ISBN: (纸本)9783800731206

We propose a method for the concealment of frame erasures in packet switched speech communication systems such as Voice over IP that is particularly tailored to the case of wireless transmission. At the receiver, the concealment algorithm is assisted by specific side information that is communicated via a steganographic channel within the bitstream of the employed speech codec. An exemplary implementation for the AMR codec is described and the obtained results are discussed. © Sprachkommunikation 2008 - 8. ITG-Fachtagung.

关键词： Packet loss

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Exploiting temporal correlation of speech and noise magnitudes using a modified kalman filter for speech enhancement

Exploiting temporal correlation of speech and noise magnitud...

引用

8. ITG-Fachtagung Sprachkommunikation 2008 - 8th ITG Conference on Speech communication 2008

作者： Esch, Thomas Vary, Peter Institute of Communication Systems and Data Processing RWTH Aachen University Aachen52056 Germany

ISBN: (纸本)9783800731206

A new speech enhancement algorithm using a modified Kalman filter in the frequency domain is proposed. The new approach consists of two steps. In the first step, the temporal trajectories of the speech and noise magnitudes are modeled by low order autoregressive (AR) processes, i.e., the current coefficients are propagated in time based on information taken from previous, enhanced coefficients, followed by a subsequent phase estimation. In the second step, the first estimation is updated. Therefore, two statistical estimators are utilized. The performance of the proposed method is shown to be considerably better than purely statistical estimators. © Sprachkommunikation 2008 - 8. ITG-Fachtagung.

关键词： Kalman filters

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Near end listening enhancement by means of warped low delay filter-banks

Near end listening enhancement by means of warped low delay ...

引用

8. ITG-Fachtagung Sprachkommunikation 2008 - 8th ITG Conference on Speech communication 2008

作者： Sauert, Bastian Löllmann, Heinrich W. Vary, Peter Institute of Communication Systems and Data Processing RWTH Aachen University Aachen52056 Germany

ISBN: (纸本)9783800731206

The concept of near end listening enhancement allows to improve the speech intelligibility of telecommunication devices in the presence of ambient background noise. It raises adaptively the average speech spectrum of the received signal from the far end speaker over the average noise spectrum of the near end background noise, which leads to reduced listening efforts for the near end listener. In this paper, we will propose an improved algorithm for near end listening enhancement which uses a non-uniform filter-bank with Bark-scaled frequency bands. The employed filter-bank has a significantly lower signal delay than commonly used non-uniform analysis-synthesis filter-banks. This allows to achieve a comparable speech intelligibility with low latency, which is of interest, e. g., for applications in mobile phones. © Sprachkommunikation 2008 - 8. ITG-Fachtagung.

关键词： Filter banks

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Efficient speech dereverberation for binaural hearing AIDS 10

Efficient speech dereverberation for binaural hearing AIDS

引用

10th ITG Symposium on Speech communication, ITGspeech 2012

作者： Löllmann, Heinrich W. Vary, Peter Institute of Communication Systems and Data Processing RWTH Aachen University Aachen52056 Germany

ISBN: (纸本)9783800734559

An efficient two-stage algorithm for binaural cue preserving speech dereverberation by spectral weighting is presented. The first stage reduces late reverberant speech components where the second stage aims at a suppression of non-coherent early and remaining late reverberant speech components. The proposed system is based on the binaural speech dereverberation algorithm of Jeub et al. [3]. In contrast to that contribution, the realization of the needed delay-and-sum beamformer and the blind estimation of reverberation time and time-difference-of-arrival with low complexity and low delay is tackled. In addition, the problem of a decreased signal energy for the enhanced signal due to spectral weighting is addressed by an efficient approach. © VDE VERLAG GMBH • Berlin • Offenbach.

关键词： Reverberation

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

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

引用

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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Iterative source-channel decoding with cross-layer support for wireless VoIP

Iterative source-channel decoding with cross-layer support f...

引用

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

作者： Breddermann, Tobias Lüders, Helge Vary, Peter Aktas, Ismet Schmidt, Florian Institute of Communication Systems and Data Processing RWTH Aachen University Germany Distributed Systems Group RWTH Aachen University Germany

ISBN: (纸本)9781424468720

This paper presents a cross-layer approach for iterative source-channel decoding (ISCD) in wireless VoIP networks. The novelty of the proposed method is the incorporation of both, speech bits as well as protocol header bits, into the ISCD process. The header bits take the role of pilot bits having perfect reliability. These bits are distributed over the frame as strong supporting points for the MAP decoder which results in a significant enhancement of the output speech quality compared to the benchmark scheme using ISCD for speech only. For this approach, we exploit cross-layer concepts that support the direct communication between non-adjacent layers. These concepts enable the iterative exchange of extrinsic information between the source decoder located on the application layer and the channel decoder located on the physical layer. This technique can also be applied to audio and video transmission. © VDE Verlag -GMBH.

关键词： Network layers

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Improved design of oversampled allpass transformed DFT filter-banks with near-perfect reconstruction

Improved design of oversampled allpass transformed DFT filte...

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15th European Signal processing Conference, EUSIPCO 2007

作者： Löllmann, Heinrich W. Vary, Peter Institute of Communication Systems and Data Processing RWTH Aachen University D-52056 Aachen Germany

ISBN: (纸本)9788392134022

In this paper, the design of oversampled allpass transformed analysis-synthesis filter-banks is treated, which can be employed for non-uniform subband processing. Near-perfect reconstruction can be achieved by a synthesis filter-bank design which uses phase equalizers to compensate the phase modifications due to the allpass transformation of the analysis filter-bank. It is shown that an FIR phase equalizer, which is obtained by a least-squares approximation of the desired non-causal IIR phase equalizer, leads to almost perfect signal reconstruction and causes only a comparatively low signal delay. The devised synthesis filter-bank design can be described by a closed-form expression and allows to control the trade-off between reconstruction errors on the one hand and signal delay and computational complexity on the other hand. © 2007 EURASIP.

关键词： Least squares approximations

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