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Noise PSD estimation by logarithmic baseline tracing 40

Noise PSD estimation by logarithmic baseline tracing

引用

40th IEEE International Conference on Acoustics, Speech, and Signal processing, ICASSP 2015

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

ISBN: (纸本)9781467369978

A novel noise power spectral density (PSD) estimator for disturbed speech signals which operates in the short-time Fourier domain is presented. A noise PSD estimate is provided by constrained tracing with time of the noisy observation separately for each frequency bin. The constraint is a limitation of the logarithmic magnitude change between successive time frames. Since speech onset is assumed as sudden rises in the noisy observation, a fixed and adaptive tracing parameter β has been derived to track the contained noise while preventing speech leakage to the noise PSD estimate. The experimental evaluation and comparison with state-of-the-art algorithms, SPP and Minimum Statistics, confirms a lower logarithmic noise estimation error and superior speech enhancement rated in a standard noise reduction system. The proposed concept has extremely low computational complexity and memory usage. Thus, it is well suited for applications where processing power and memory is limited. © 2015 IEEE.

关键词： low complexity low memory Noise power estimation noise reduction speech enhancement

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Quality control for UMTS-AMR speech channels

Quality control for UMTS-AMR speech channels

引用

9th European Conference on Speech communication and Technology

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

In UMTS speech transmissions, the Adaptive Multi-Rate (AMR) speech codec is employed to allow for a dynamic assignment of data rates to individual users. The control of AMR modes is based on quality measurements of the transmission channel and aims at the maximization of speech quality by selecting the mode which is best suited to the current interference situation. In contrast to GSM, a reduction of AMR modes also leads to an increased cell capacity in UMTS. In this paper, a decentralized method for AMR mode switching is presented which is based on individual softbit measurements at the Viterbi channel decoder and optimizes the speech quality for a wide range of channel conditions. This method was developed by optimizing the AMR mode switching thresholds with respect to PESQ speech quality scores. The second part of the paper describes a new instrumental non-intrusive method for speech quality measurement by the evaluation of UMTS transmission parameters for application to UMTS-AMR speech transmissions. High correlations with the reference PESQ scores were observed.

关键词： Speech recognition

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A new method for snr-estimation in impulse response measurements

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

引用

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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Inverse cyclic convolution 11

Inverse cyclic convolution

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 leastmean- 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 Berlin Offenbach.

关键词： Impulse response

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Efficient voice communication in wireless packet networks

Efficient voice communication in wireless packet networks

引用

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

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

ISBN: (纸本)9783800731206

We present a methodology to optimize the system parameterization for speech and audio transmission in heterogeneous packet networks with wireless access. To this end, we determine the theoretical performance of forward error correction (FEC) schemes on packet level, i.e., the resulting frame loss distribution at the receiver after error correction. The calculation utilizes a statistical model for packet loss distribution which is adaptable for different packet transmission time intervals and packet sizes. The optimal parameterization of the transmission scheme, i.e., the encoding rate, frame length, and amount of redundancy per packet, are then found by optimizing the resulting conversational quality. We utilize the ITU-T E-model which provides a quality prediction in dependence on codec, end-to-end delay, and frame loss distribution. The procedure will be illustrated for realistic transmission scenarios of wireless packet network access technologies. © Sprachkommunikation 2008 - 8. ITG-Fachtagung.

关键词： Packet networks

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Near end listening enhancement optimized with respect to speech intelligibility index

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

引用

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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Audio coding for beamforming with distributed microphones 11

Audio coding for beamforming with distributed microphones

引用

11th ITG Symposium on Speech communication

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

ISBN: (纸本)9783800736409

In this contribution we investigate the influence of audio coding on beamforming algorithms. Usually, a beamforming system would be implemented in the same device as the microphones. However, for some devices such as wireless distributed microphone arrays, or digital hearing aids, it may be necessary to transmit the microphone signals prior to combining. Since any digital signal transmission will introduce quantization noise, the subsequent beamformer processing will be influenced. In this paper, we investigate the behavior of a differential beamformer and a generalized sidelobe canceler after encoding and decoding of microphone signals. Furthermore, an example of a suitable audio coding scheme is given. © VDE VERLAG GMBH Berlin Offenbach.

关键词： Beamforming

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On reverse waterfilling in closed-loop LPC with noise shaping 11

On reverse waterfilling in closed-loop LPC with noise shapin...

引用

11th ITG Symposium on Speech communication

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

ISBN: (纸本)9783800736409

A new criterion is derived for the design of the noise weighting filter in linear predictive coding (LPC) that accounts for the issue of noise propagation. It will be shown that the new filter design constitutes an approximation of the reverse waterfilling procedure known from rate-distortion theory and audio transform coding. It improves the overall coding SNR of closed-loop LPC with noise shaping. A practical application within the 3GPP AMRWB codec is presented. Objective test results indicate notable quality improvements. © VDE VERLAG GMBH Berlin Offenbach.

关键词： Electric distortion

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Logarithmic cubic vector quantization 10

Logarithmic cubic vector quantization

引用

10th ITG Symposium on Speech communication, ITGspeech 2012

作者： Rohlfing, C. Krüger, H. Vary, P. Institute of Communication Systems and Data Processing RWTH Aachen University Aachen52074 Germany

ISBN: (纸本)9783800734559

In this paper a novel type of gain-shape vector quantization (GSVQ) is presented, denoted as Logarithmic Cubic Vector Quantization (LCVQ). LCVQ is based on a decomposition of the vector to be quantized into a gain factor and a shape vector which is a normalized version of the input vector. Both components are quantized independently and transmitted to the decoder. Compared to other GSVQ approaches, in LCVQ the input vectors are normalized such that all shape vectors are located on the surface of the unit hypercube. As a conclusion, the shape vector quantizer can be realized based on uniform scalar quantizers. This yields low computational complexity as well as high memory efficiency even in case of very high vector dimensions. In order to demonstrate the coding efficiency of the proposed quantization scheme, LCVQ is compared to existing quantization schemes, in particular the recently proposed Logarithmic Spherical Vector Quantization (LSVQ) [1]. © VDE VERLAG GMBH • Berlin • Offenbach.

关键词： Vector quantization

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A blind algorithm for joint noise suppression and dereverberation

A blind algorithm for joint noise suppression and dereverber...

引用

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

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

ISBN: (纸本)9783800731206

This contribution presents a novel speech enhancement algorithm for the suppression of background noise and late reverberation without a priori knowledge. The speech enhancement is performed by a generalized spectral weighting rule based on spectral power estimations for the late reverberant speech and background noise. By this, speech distortions due to late room reflections can be suppressed without knowledge about the complete room impulse response. In contrast to existing methods, all needed quantities are estimated entire blindly from the reverberant and noisy speech signal. A possible application of the proposed algorithm are speech enhancement systems for hands-free devices, mobile phones or hearing aids. © Sprachkommunikation 2008 - 8. ITG-Fachtagung.

关键词： Hearing aids

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