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作者： Tim Claussen Viet Duc Nguyen Ulrich Heute Gerhard Schmidt Digital Signal Processing and System Theory University of Kiel Kiel Germany

ISBN: (纸本)9781467393744

Cognitive signal processing has been intensively studied over the past years. Nevertheless, cognitive sonar systems only draw little attention in the research community. In this paper the signal processing of a cognitive sonar system for diver detection is shown in detail. It is inspired by the target-detection system of a bat and thus able to adapt its signal processing units to measured parameters of particular targets. This is done by adjusting the transmitted signal as well as the direction of the transmitted energy of the active sonar system by transmit beamforming. Thereby, it is possible to improve on the one hand the probability of detection of a target for a given signal-to-noise ratio (SNR) as well as signal-to-reverberation ratio (SRR) and, on the other hand, to extend the spatial awareness by adapting the transmit beamformer. This technique leads to a combination of two spatial filters: the receive beampattern and the transmit beampattern. To obtain target-specific parameters, the data form an internal tracking system is used. After showing a general system overview, the core parts of the signal processing are shown in detail. At this point we focus on the Doppler processing with its real-time interpolation and correlation algorithms. MonteCarlo simulations illustrate the potential of the proposed system by comparing the results with conventional active-sonar signal processing.

关键词： Sonar Target tracking Array signal processing Real-time systems signal to noise ratio Doppler effect

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Voice activity detection based on modulation-phase differences

Voice activity detection based on modulation-phase differenc...

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12. ITG-Fachtagung Sprachkommunikation - 12th ITG Conference on Speech Communication

作者： Graf, Simon Herbig, Tobias Buck, Markus Schmidt, Gerhard Acoustic Speech Enhancement Research Nuance Communications Deutschland GmbH Ulm Germany Digital Signal Processing and System Theory Christian-Albrechts-Universität zu Kiel Kiel Germany

ISBN: (纸本)9783800742752

Many speech processing algorithms rely on voice activity detection (VAD) that separates speech from noise. For this task, several features have been introduced that employ different characteristic properties of speech. In this contribution, we introduce a new feature that is robust against various types of noise. By considering an alternating excitation structure of low and high frequencies, speech is detected with a high confidence. The computationally low complex feature can cope even with the limited spectral resolution that is typical for in-car-communication systems. By combining the feature with a conventional modulation feature, the performance can be improved. Our simulations confirm the robustness of the feature and show the increasing performance compared to established VAD features. © 2016 VDE VERLAG GMBH.

关键词： Speech processing

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REAL-TIME EMPIRICAL MODE DECOMPOSITION FOR EEG signal ENHANCEMENT

REAL-TIME EMPIRICAL MODE DECOMPOSITION FOR EEG SIGNAL ENHANC...

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European signal processing Conference

作者： Alina Santillan-Guzman Martin Fischer Ulrich Heute Gerhard Schmidt Kiel University Faculty of Engineering Digital Signal Processing and System Theory

Electroencephalography (EEG) recordings are used for brain research. However, in most cases, the recordings not only contain brain waves, but also artifacts of physiological or technical origins. A recent approach used for signal enhancement is Empirical Mode Decomposition (EMD), an adaptive data-driven technique which decomposes non-stationary data into so-called Intrinsic Mode Functions (IMFs). Once the IMFs are obtained, they can be used for denoising and detrending purposes. This paper presents a real-time implementation of an EMD-based signal enhancement scheme. The proposed implementation is used for removing noise, for suppressing muscle artifacts, and for detrending EEG signals in an automatic manner and in real-time. The proposed algorithm is demonstrated by application to a simulated and a real EEG data set from an epilepsy patient. Moreover, by visual inspection and in a quantitative manner, it is shown that after the EMD in real-time, the EEG signals are enhanced.

关键词： EMD Real-time Denoising Detrending EEG

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Kurtosis-controlled babble noise suppression

Kurtosis-controlled babble noise suppression

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12. ITG-Fachtagung Sprachkommunikation - 12th ITG Conference on Speech Communication

ISBN: (纸本)9783800742752

When a speech application is employed in a crowded environment, the user's voice superposes with many interfering voices. This babble noise is a challenge for many speech processing algorithms since assumptions like stationarity of the noise or a good SNR may not be valid. In this contribution, characteristics of babble noise are discussed and distinctive features are summarized that allow for distinguishing the desired foreground speech from background noise. In particular, the kurtosis of a signal is identified as a good measure to detect the presence of speech in babble noise. Furthermore, a babble noise suppression system is introduced that employs a speech detector to dynamically control the aggressiveness of noise suppression. It is desirable that the residual noise after processing the signal is perceived as more pleasant by human listeners. To evaluate the improvements, a subjective listening test is conducted. Speech distortions are quantified using an objective measure. © 2016 VDE VERLAG GMBH.

关键词： Speech processing

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Automatic Localization of an Ultrasound Probe with the Help of Magnetic Sensors

Automatic Localization of an Ultrasound Probe with the Help ...

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作者： Bald, Christin Bergholz, Robert Schmidt, Gerhard Digital Signal Processing and System Theory Institute of Electrical Engineering and Information Technology Faculty of Engineering Christian-Albrechts-Universität zu Kiel Kaiserstraße 2 Kiel Germany Digital Signal Processing and System Theory Institute of Electrical Engineering and Information Technology Faculty of Engineering Christian-Albrechts-Universität zu Kiel Kiel Germany Pediatric Surgery Faculty of Medicine Christian-Albrechts-Universität zu Kiel Kiel Germany

Ultrasound measurements are a widely used instrument in clinical practice. For later traceability of the images, the position (and orientation) of the ultrasound probe must be recorded during the measurement. Until now this has to be done manually by the physician. An easier and more accurate approach would be the automatic tracking of the ultrasound probe. This contribution shows a first approach for automatically localizing the ultrasonic head during measurement. The proposed method is based on coils surrounding the patient bed and a 3D magnetic sensor placed on the ultrasound head. Besides some pre- and postprocessing steps, the proposed localization algorithm is based on trilateration followed by a least mean squares approach for refinement of the estimation. In a first proof-of-concept measurement with fixed positions and orientations of the ultrasound head a mean accuracy of 2.85 cm and 8.94◦ was achieved. Additionally, a measurement with a moving ultrasound head is presented to demonstrate the real-time capability of the system. Finally, future steps for improving the automatic measurement are discussed, including a graphical user interface for the physician and the use of magnetoelectric sensors for measurement. © 2022 The Author(s), published by De Gruyter.

关键词： Ultrasonics

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Magnetoneurograhy of an electrically stimulated arm nerve: Usability of magnetoelectric (ME) sensors for magnetic measurements of peripheral arm nerves

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Current Directions in Biomedical Engineering 2018年第1期4卷 363-366页

作者： Elzenheimer, Eric Laufs, Helmut Sander, Tilmann Schmidt, Gerhard Kiel University Digital Signal Processing and System Theory Kaiserstr. 2 Kiel 24143 Germany UKSH Department of Neurology Arnold-Heller-Str. 3 Kiel 24105 Germany Physikalisch-Technische Bundesanstalt (PTB) Abbestr. 2-12 Berlin 10587 Germany

For further development and optimization of novel uncooled magnetoelectric (ME) sensors, a better understanding of spectral power distribution and signal strength of nerve signals is of high interest. For obtaining information on these signal properties, Superconducting Quantum Interference Device (SQUID) measurements were performed at the Physikalisch-Technische Bundesanstalt (PTB) in Berlin. signal amplitudes were subject-dependent and ranged from 17 fT to 60 fT in a frequency range from 100 Hz to 1 kHz. The required SQUID averaging time was in the range of minutes, while for current ME sensors significantly longer averaging times are expected to be necessary. © 2018 Eric Elzenheimer et al.

关键词： SQUIDs

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On the use of vocal-tract approximations for instrumental quality assessment 10

On the use of vocal-tract approximations for instrumental qu...

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10th ITG Symposium on Speech Communication, ITGspeech 2012

作者： Norrenbrock, Christoph R. Heute, Ulrich Hinterleitner, Florian Digital Signal Processing and System Theory Christian-Albrechts-Universität zu Kiel Kiel24143 Germany Quality and Usability Lab. Telekom Innovation Laboratories TU Berlin Berlin10587 Germany

ISBN: (纸本)9783800734559

The nature of linear prediction becomes most vivid through the relation between predictor and reflection coefficients in the context of the source-filter model of speech production. In particular, an approximation of the vocal-tract shape can be recursively calculated using the reflection coefficients only. To date, it is not fully clear to what extent such an approximation can be exploited for a perception-related quality analysis of connected speech signals. In this paper, we will review known approaches from literature and elaborate on a novel extension towards instrumental quality assessment of synthesized speech signals. © VDE VERLAG GMBH • Berlin • Offenbach.

关键词： Reflection

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Improved performance measures for voice activity detection 11

Improved performance measures for voice activity detection

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11th ITG Symposium on Speech Communication

作者： Graf, Simon Herbig, Tobias Buck, Markus Schmidt, Gerhard Acoustic Speech Enhancement Research Nuance Communications Deutschland GmbH Ulm89077 Germany Department of Digital Signal Processing and System Theory Christian-Albrechts-Universität Zu Kiel Kiel24143 Germany

ISBN: (纸本)9783800736409

Voice activity detection is an essential part of many speech processing algorithms. The requirements of the speech application determine the design of voice activity detection. Some applications need low-latency results whereas the accuracy of speech detection is more important for other applications. The performance is generally evaluated by Receiver Operating Characteristic (ROC) curves, which perform a static analysis averaged over speech and nonspeech segments, respectively. We adopt the ROC curves but evaluate them for specific speech classes, e.g., voiced or unvoiced speech, to describe the overall accuracy of speech detection. In addition, we present a new measure for the dynamic behavior that considers the delay and latency of speech on- And offset detection. Finally, we present a unified measure for both aspects. This measure may be used to find appropriate voice activity detection features for a given application. An automotive noise scenario is employed to demonstrate the measures as it contains stationary and non-stationary noise. © VDE VERLAG GMBH Berlin Offenbach.

关键词： Speech processing

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Real-time cognitive sonar system with target-optimized adaptive signal processing through multi-layer data fusion

Real-time cognitive sonar system with target-optimized adapt...

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International Conference on Multisensor Fusion and Integration for Intelligent systems (MFI)

作者： Tim Claussen Viet Duc Nguyen Faculty of Electrical and Information Engineering Digital Signal Processing and System Theory Kiel University Kiel Germany Faculty of Electrical and Information Engineering Digital Signal Processing and System Theory Kiel University Kiel Schleswig-Holstein DE

In this contribution a proof-of-concept of a real-time cognitive sonar system is shown and tested in simulations. In the field of radar applications the principle of a cognitive system has been intensively studied over the past years. However, cognitive sonar systems draw only little attention in the reseach community. In this paper we are showing a new approach for a cognitive sonar system, that is able to adapt its signal processing to measured parameters of particular targets. This is possible by the fusion of the information of different data layers. Data resulting from an internal tracking module is used to adapt the signal processing at sensor layer. This is done by adjusting the transmit signal and the direction of the transmitted energy via transmit beamforming. These adjustments improve, on the one hand, the signal-to-noise ratio (SNR) and the signal-to-reverberation ratio (SRR) of the target through the adaption of the transmitted signal, and, on the other hand, extend the spatial awareness by adapting the transmit beamformer, which results in a combination of two spatial filters: the receive beampattern and the transmit beampattern.

关键词： Target tracking Radar tracking Sonar detection Correlation

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Estimation of time-variant acoustic feedback paths in in-car communication systems

Estimation of time-variant acoustic feedback paths in in-car...

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International Workshop on Acoustic signal Enhancement (IWAENC)

作者： Jochen Withopf Gerhard Schmidt Digital Signal Processing and System Theory Christian-Albrechts-Universität zu Kiel Kiel

In-car communication systems aim at improving the speech communication inside a vehicle by picking up the talking passenger's voice by means of microphones and playing it back over loudspeakers that are located close to the listening passengers. Because of this closed-loop operation, system instability caused by acoustical feedback is a limiting factor for the maximum possible system support. In this contribution, we present a feedback cancellation algorithm based on the NLMS algorithm for estimating the frequency responses of the multiple, time-variant feedback paths. The step-size parameter is calculated time- and frequency dependently and signal decorrelation, which is necessary for convergence of the filters, is performed by a phase modulation filter for frequency shifting. Measurements in a test car prove the effectiveness of the approach in the in-car communication application, even in the presence of background noise.

关键词： Loudspeakers Decorrelation Acoustics Speech Microphones Correlation Frequency modulation

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