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AN EXTENSION OF THE PEAQ MEASURE BY A BINAURAL HEARING MODEL

AN EXTENSION OF THE PEAQ MEASURE BY A BINAURAL HEARING MODEL

引用

IEEE International Conference on Acoustics, Speech and Signal processing

作者： Magnus Schafer Mohammad Bahram Peter Vary Institute of Communication Systems and Data Processing (iNd) RWTH Aachen University

ISBN: (纸本)9781479903573

Instrumental evaluation of the perceived audio signal quality is an important tool for the development of audio signal enhancement and transmission systems. There are various single channel measures which can be used for different application scenarios. Binaural signals have not received much focus so far and no sophisticated model of spatial perception is utilized in the available measures. In this contribution, an extension to Perceptual Evaluation of Audio Quality (PEAQ) is presented which makes use of a recently proposed binaural hearing model. It is shown that the inclusion of spatial information into the instrumental quality measurement leads to a strongly increased correlation between the instrumental measure and a listening test.

关键词： PEAQ Space Perception instrument design and development Quality management Transmission systems audio signals signal enhancement

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Wideband extension of telephone speech using a hidden Markov model

Wideband extension of telephone speech using a hidden Markov...

引用

IEEE Workshop on Speech Coding Proceedings

作者： P. Jax P. Vary Institute of Communication Systems and Data Processing RWTH Aachen Aachen Germany

In this paper we propose an algorithm to recover wideband speech from lowpass-bandlimited speech. The narrowband input signal is classified into a limited number of speech sounds for which the information about the wideband spectral envelope is taken from a pre-trained codebook. For the codebook search algorithm a statistical approach based on a hidden Markov model is used, which takes different features of the bandlimited speech into account, and minimizes a mean squared error criterion. The new algorithm needs only one single wideband codebook and inherently guarantees the transparency of the system in the base-band. The enhanced speech exhibits a significantly larger bandwidth than the input speech without introducing objectionable artifacts.

关键词： Wideband Telephony Hidden Markov models Bandwidth Speech processing Frequency Finite impulse response filter Signal processing Narrowband Speech enhancement

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A binaural room impulse response database for the evaluation of dereverberation algorithms

A binaural room impulse response database for the evaluation...

引用

International Conference on Digital Signal processing (DSP)

作者： Marco Jeub Magnus Schafer Peter Vary Institute of Communication Systems and Data Processing RWTH Aachen University Germany

This paper describes a new database of binaural room impulse responses (BRIR), referred to as the aachen impulse response (AIR) database. The main field of application of this database is the evaluation of speech enhancement algorithms dealing with room reverberation. The measurements with a dummy head took place in a low-reverberant studio booth, an office room, a meeting room and a lecture room. Due to the different dimensions and acoustic properties, it covers a wide range of situations where digital hearing aids or other hands-free devices can be used. Besides the description of the database, a motivation for using binaural instead of monaural measurements is given. Furthermore an example using a coherence-based dereverberation technique is provided to show the advantage of this database for algorithm evaluation. The AIR database is being made available online.

关键词： databases Acoustic measurements Hearing aids Speech enhancement Auditory system Microphones Reverberation Magnetic heads data processing Acoustic devices

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A Bit-Mapping Strategy for Joint Iterative Channel Estimation and Turbo-Decoding

A Bit-Mapping Strategy for Joint Iterative Channel Estimatio...

引用

IEEE Conference on Vehicular Technology (VTC)

作者： Susanne Godtmann Helge Luders Gerd Ascheid Peter Vary Institute for Integrated Signal Processing Systems RWTH Aachen University Aachen Germany Institute of Communication Systems and Data Processing RWTH Aachen University Aachen Germany

In this paper, we investigate turbo-coded transmission over a temporally correlated flat Rayleigh fading channel. Conventionally, channel estimation is performed prior to decoding and is solely based on periodically inserted pilot symbols. We here consider the case that the pilot spacing is not small enough to satisfy the Nyquist criterion for the bandlimited channel process. It is demonstrated that iterative code-aided channel estimation can then significantly improve the bit error rate (BER). Furthermore, we show that the convergence speed of the iterative channel estimation can be significantly accelerated without additional computational power by just carefully choosing the mapping strategy of systematic bits to data symbols.

关键词： Channel estimation Convergence Decoding Iterative decoding Synchronization Fading Bit error rate

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Performance Evaluation of the Markov Chain Monte Carlo MIMO Detector based on Mutual Information

Performance Evaluation of the Markov Chain Monte Carlo MIMO ...

引用

IEEE International Conference on communications (ICC)

作者： M. Senst G. Ascheid H. Lüders Institute for Integrated Signal Processing Systems RWTH Aachen University Aachen Germany Institute of Communication Systems and Data Processing RWTH Aachen University Aachen Germany

Recently, MIMO detectors which are based on Markov Chain Monte Carlo (MCMC) simulation techniques have been proposed as alternatives to, e.g., the well-known sphere detector. In this paper, we present a systematic analysis of the performance of MCMC detectors. We study the impact of several parameters such as the list size and the number of independently running chains. As a performance criterion, our analysis is based on the mutual information over an equivalent modulation channel, rather than on coded bit error rates, because this metric is independent of the outer channel code and provides valuable insights over the whole SNR range of interest. Furthermore, we show that combining the MCMC detector with a hard-output sphere detector removes the error floor at high SNR, which is a well-known problem of the MCMC principle.

关键词： Monte Carlo methods MIMO Detectors Mutual information Performance analysis Modulation coding Bit error rate Wireless communication Stochastic processes Niobium

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MODEL-BASED SPEECH ENHANCEMENT USING SNR DEPENDENT MMSE ESTIMATION

MODEL-BASED SPEECH ENHANCEMENT USING SNR DEPENDENT MMSE ESTI...

引用

IEEE International Conference on Acoustics, Speech and Signal processing

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

This contribution presents a modified Kalman filter approach for single channel speech enhancement which is operating in the frequency domain. In the first step, temporal correlation of successive frames is exploited yielding estimates of the current speech and noise DFT coefficients. This first prediction is updated in the second step applying an SNR dependent MMSE estimator which is adapted to the (measured) statistics of the speech prediction error signal. Objective measurements show consistent improvements compared to estimators which do not take into account the temporal correlation or the influence of the input SNR on the statistics of the prediction error signal.

关键词： Signal to noise ratio Speech Kalman filters Speech enhancement Discrete Fourier transforms Attenuation

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An information theoretic view on Artificial Bandwidth Extension in noisy environments

An information theoretic view on Artificial Bandwidth Extens...

引用

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

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

Artificial Bandwidth Extension (ABWE) exploits spectral dependencies of speech signals and aims at recovering missing high frequency components if only the narrowband speech signal is available. This contribution provides an information theoretic view on ABWE when used in noisy conditions. Based on the results of [1], a performance bound of ABWE is formulated if the narrowband signal is disturbed by additive noise. The performance bound is evaluated using real entropy measurements and the influence of noise suppression prior to ABWE is investigated.

关键词： Vectors Narrowband Speech Noise measurement Mutual information Entropy Signal to noise ratio

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Efficient musical noise suppression for speech enhancement system

Efficient musical noise suppression for speech enhancement s...

引用

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

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

Noise reduction techniques that are relying on spectral weighting rules often generate annoying musical noise artifacts in the processed signal. In this paper, we present a postfilter (PF) for the spectral weighting gains that is capable of reducing musical noise in a simple but efficient way. It includes a robust detector for speech pauses and low SNR conditions and adaptively smoothes the weighting gains over frequency based on soft-decisions. Objective and subjective measurements show consistent improvements if the postfilter is applied to conventional noise reduction techniques.

关键词： Speech enhancement Noise reduction Signal generators Noise generators Signal processing Noise robustness Detectors Signal to noise ratio Frequency Noise measurement

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NEAR END LISTENING ENHANCEMENT OPTIMIZED WITH RESPECT TO SPEECH INTELLIGIBILITY INDEX AND AUDIO POWER LIMITATIONS

NEAR END LISTENING ENHANCEMENT OPTIMIZED WITH RESPECT TO SPE...

引用

European Signal processing Conference

作者： Bastian Sauert Peter Vary Institute of Communication Systems and Data Processing (iNd) RWTH Aachen University

In speech communications, 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 telephony in acoustical background noise such as traffic or babble noise. In these situations, it is often not acceptable/possible to increase the audio power amplification. In this contribution we use a theoretical analysis of the Speech Intelligibility Index (SII) to develop an algorithm which numerically maximizes the SII under the constraint of an unchanged average power of the audio signal.

关键词： NATIONAL EMERGENCY ALARM REPEATER intelligibility Speech BACKGROUND NOISE Audio TCEA1 gene power amplification audio signals Mobile phones mobile communication Theoretical analysis Traffic

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BPSK modulation with unequal power allocation considering a priori knowledge

BPSK modulation with unequal power allocation considering a ...

引用

IEEE International Symposium on Personal, Indoor and Mobile Radio communications (PIMRC)

作者： T. Bruggen M. Adrat P. Vary Institute of Communication Systems and Data Processing RWTH Aachen University Germany

A well known technique to protect bits with different sensitivities against channel errors is unequal error protection (UEP) by selective channel coding. However, some transmission systems such as Bluetooth and DECT include only weak or no channel coding. For those systems modulation with unequal power allocation (MUPA) is an appropriate technique to realize UEP without channel coding, MUPA allocates different transmission power to the modulation symbols according to the individual bit error sensitivities. In this paper we apply the MUPA concept to systems with softbit source decoding and soft demodulation, and enhance the MUPA approach by taking a 1 st order Markov model in terms of 1 st order a priori knowledge (AK1) into account (MUPA-AK1)

关键词： Binary phase shift keying Channel coding Error correction codes Decoding Power system protection Bluetooth Demodulation Proposals Autocorrelation data processing

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