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Communication Technology Colloquium at IKS

von Irina Esser

Dear subscribers of the colloquium newsletter, we are happy to inform you about the next date of our Communication Technology Colloquium. *Monday, April 25, 2022* *Speaker*: Alexander Sobolew *Time*: 03:00 p.m. *Location*: https://rwth.zoom.us/j/97904157921?pwd=SWpsbDl0MWhrWjY1ZkZaeFRoYmErZz09 Meeting-ID: 979 0415 7921 Passwort: 481650 *Master-Lecture*: Investigation of Specialized Recurrent Units for Acoustic Echo Cancellation In today's communication, hands-free devices e.g. remote communication are widely used. Without further action, these would suffer from an acoustic echo that arises from the coupling between speaker and microphone. To minimize these disturbances, acoustic echo cancellation is indispensable. Model-based adaptive algorithms exist to solve this issue. However, they require careful tuning of parameters whose optimum differs between devices and acoustic situations. In this thesis, a new data-driven approach for acoustic echo cancellation is developed and investigated. In contrast to the purely model-based approach, the algorithm is supposed to learn the optimal performance from data without the necessity to be tuned. In new situations, the unknown parameters should be estimated. At its core, the novel structure is similar to a frequency adaptive filter. However, it is extended by the gating mechanism known from recurrent neural networks. The development also includes the determination of optimal training paradigms. When choosing the model structure, attention is paid to a reasonable training complexity. Major challenges in this thesis include the investigation of the gating mechanism, which is represented by a learn gate and a reset gate. The prior is used to estimate a time-varying step size of the iterative algorithm. Gated Recurrent Units provide an internal memory to accommodate the sequential information in speech, while skip connections optimize the gradient flow during training. Independent use of a reset gate to reset the impulse response estimation in case of situation change is outperformed by exploitation of weight sharing. Using weight sharing, the learn and reset gates have direct information about each other's behavior due to a shared partial network. It was shown that when using backpropagation through time, the truncation order can be reduced to a certain extent, which reduced the training complexity but did not decrease the performance. The developed model outperforms the tuned Fast Block Normalized Least-Mean-Square algorithm in reconvergence speed and steady-state performance in far-end single talk and double talk. Furthermore, our model repeatedly outperforms the tuned diagonalized Kalman filter in certain scenarios and offers significantly improved overall performance in single talk. and *Monday, April 25, 2022* *Speaker*: Alexej Sobolew *Time*: 04:00 p.m. *Location*: https://rwth.zoom.us/j/97904157921?pwd=SWpsbDl0MWhrWjY1ZkZaeFRoYmErZz09 Meeting-ID: 979 0415 7921 Passwort: 481650 *Master-Lecture*: Investigation of Generative Neural Networks for Speech Enhancement Speech enhancement aims to reduce noise in speech signals and is widely used in hearing aids and mobile speech communication. Speech synthesis, on the other hand, aims to generate high-quality human speech and is used, e. g., in text-to-speech generation. Noise reduction and speech synthesis can be combined since conventional noise reduction methods often only improve the magnitude spectrum and keep the noisy phase. However, the phase has an important influence on speech quality and intelligibility. In addition, training neural networks with complex spectrograms is more difficult, so it is reasonable to first denoise the magnitude spectrum and subsequently synthesize the waveform of the speech based on it. Mentioned applications often require low execution times and low computational overhead. This is achievable by exploiting parallel processors using the non-autoregressive property and by reducing the number of parameters in the neural network. Hence, this thesis aims to investigate noise reduction, speech synthesis, and joint interaction. In this thesis, the first use case considered is phase reconstruction and speech synthesis based on clean data. A non-autoregressive three-stage speech enhancement system is developed for the second use case of combined noise reduction and speech synthesis based on magnitude spectra. For speech synthesis on clean magnitude spectra such as mel-spectrograms, the neural network called WaveGlow from Nvidia is taken as a basis. WaveGlow achieves similar subjective performance as the Griffin-Lim algorithm but is better suited for fast applications. For the reduction of parameters in WaveGlow, the SqueezeWave is used, resulting in a decrease in the number of parameters and the inference time by up to 70%. During the switch to additional noise reduction besides speech synthesis, it is shown that WaveGlow alone is not suitable for performing both tasks simultaneously. Consequently, the problem is divided into three stages: masking, inpainting, and synthesis. The models for masking and inpainting are adapted to mel-spectrograms and studied in detail for noise reduction. As result, they are able to reduce the noise significantly. It is worth noting that in this thesis the mel-filter is used as downsampling adapted to human perception due to its non-linear property, resulting in a reduction of the number of computations in the first two models. Subsequently, the performance of the entire three-stage speech enhancement system is investigated. It improves the speech quality and intelligibility of noisy data while exploiting parallel processors and can compete with existing state-of-the-art methods. The system achieves better noise reduction than the Convolutional Recurrent Network (CRN) and additionally does not rely on the noisy phase. All interested parties are cordially invited, registration is not required. General information on the colloquium, as well as a current list of dates of the Communication Technology Colloquium can be found at: http://www.iks.rwth-aachen.de/aktuelles/kolloquium -- Irina Esser Institute of Communication Systems (IKS) RWTH Aachen University Muffeter Weg 3a, 52074 Aachen, Germany +49 241 80 26958 (phone) ronkartz(a)iks.rwth-aachen.de http://www.iks.rwth-aachen.de/

2 Jahre

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Communication Technology Colloquium at IKS

von Irina Esser

Dear subscribers of the colloquium newsletter, we are happy to inform you about the next date of our Communication Technology Colloquium. *Wednesday, March, 30, 2022* *Speaker*: Nora Pöhlau *Time*: 10:00 a.m. *Location*: https://rwth.zoom.us/j/97904157921?pwd=SWpsbDl0MWhrWjY1ZkZaeFRoYmErZz09 Meeting-ID: 979 0415 7921 Passwort: 481650 *Master Lecture*: Performance Evaluation of Sound Field Translation Methods for Recorded Virtual Reality The Higher-Order Ambisonics (HOA) format allows directional recording and playback of sound, making it an attractive tool for spatial audio or immersive sound applications. Because Higher-Order Ambisonics are mathematically based on Spherical Harmonics (SHs), they offer full rotational freedom for the listener (3DoF). However, the sound field can only be correctly reconstructed in a small area around the original recording position due to physical constraints. Three algorithms developed at the Institute for Communications Systems (IKS) make it possible to allow an additional translational movement of the user, even beyond the sweet spot. These algorithms deviate from the physically correct reconstruction in favour of an acoustically plausible playback. In this thesis, the three algorithms of Space Warping (SW), Adaptive Space Warping (ASW) and Adaptive Beamforming (ABF) are perceptually compared by conducting multiple listening tests. ABF and ASW split the sound signal into a primary and an ambient part and apply the translation operation only to the primary part. In two web-based listening tests, it was found that this separation is an acoustically valid approach. It was not distinguishable for the listeners if the primary part contained only direct sound or additional early reflexions. In a second step, a listening test in the laboratory was conducted. Here, the algorithms were compared for different translation distances. For small distances, ABF showed the best performance of all algorithms. ABF introduced fluctuating residual noise for higher distances but still obtained the highest source position ratings. Besides that, a newly proposed variant of SW has proven to perform surprisingly well and scored second best in all ratings behind ABF. All interested parties are cordially invited, registration is not required. General information on the colloquium, as well as a current list of dates of the Communication Technology Colloquium can be found at: http://www.iks.rwth-aachen.de/aktuelles/kolloquium -- Irina Esser Institute of Communication Systems (IKS) RWTH Aachen University Muffeter Weg 3a, 52074 Aachen, Germany +49 241 80 26958 (phone) ronkartz(a)iks.rwth-aachen.de http://www.iks.rwth-aachen.de/

2 Jahre, 1 Monat

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Kommunikationstechnisches Kolloquium am IKS

von Irina Esser

Sehr geehrte Abonnenten des Kolloquium-Newsletters, gerne informieren wir Sie über den nächsten Termin unseres Kommunikationstechnischen Kolloquiums. *Dienstag, 22. März 2022* *Vortrage**nder*: Anatolii Skovitin *Zeit*: 14:00 Uhr *Ort*: https://rwth.zoom.us/j/97904157921?pwd=SWpsbDl0MWhrWjY1ZkZaeFRoYmErZz09 Meeting-ID: 979 0415 7921 Passwort: 481650 *Bachelor-Vortrag:* Sprachsynthese von gestörten Betragsspektren mit Methoden des maschinellen Lernens Sprachsignale sind ein wichtiges Mittel der Kommunikation zwischen Menschen. In der digitalen Welt werden Sprachsignale über Telefone oder das Internet übertragen. Dazu müssen sie zunächst in den Zeit-Frequenzbereich transformiert werden. Ein resultierendes Zeit-Frequenzspektrum setzt sich zusammen aus dem Magnituden- und dem Phasenspektrum. Sprachsignale sind häufig Störungen ausgesetzt, bei denen das Nutzsignal verzerrt wird. Es gibt Methoden die diese Verzerrungen abschwächen oder entfernen können. Häufig wird jedoch nur das Magnitudenspektrum betrachtet und das Phasenspektrum bleibt aufgrund seiner vergleichsweise geringen Bedeutung unverändert. Allerdings werden sich im Normalfall die verarbeiteten Magnitudenspektren den perfekten Magnitudenspektren annähern. Andere Methoden liefern überhaupt kein Phasenspektrum, sondern nur eine Schätzung des Magnitudenspektrums. In dieser Arbeit wird die Methode untersucht, die das Phasenspektrum von Sprachsignalen aus den geschätzten oder gestörten Magnitudenspektren rekonstruiert. Zu diesem Zweck werden Ansätze aus dem Bereich des maschinellen Lernens verwendet. Um die Methoden der Phasenrekonstruktion möglichst unabhängig von den spezifischen Störungsarten eines bestimmten Systems zu untersuchen, wird eine künstliche Störung verwendet. Die vorbereiteten Daten werden für das Training der neuronalen Netze verwendet. Die besten Modelle des neuronalen Netzes werden dann ausgewählt. Sie werden auf unterschiedlich gestörten Daten angewendet, um herauszufinden, wie gut die neuronalen Netze für verschiedene Arten von Störungen geeignet sind. Schließlich werden die verschiedenen Phasenrekonstruktionsmethoden angewendet und die resultierenden Sprachsignale bewertet. Außerdem wird ein Vergleich mit dem Griffin-Lim-Algorithmus durchgeführt. Alle Interessierten sind herzlich eingeladen, eine Anmeldung ist nicht erforderlich. Allgemeine Informationen zum Kolloquium sowie eine aktuelle Liste der Termine des Kommunikationstechnischen Kolloquiums finden Sie unter: https://www.iks.rwth-aachen.de/aktuelles/kolloquium/ -- Irina Esser Institute of Communication Systems (IKS) RWTH Aachen University Muffeter Weg 3a, 52074 Aachen, Germany +49 241 80 26958 (phone) esser(a)iks.rwth-aachen.de http://www.iks.rwth-aachen.de/

2 Jahre, 1 Monat

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Einladung Promotionsvortrag

von Irina Esser

Sehr geehrte Abonnenten des Kolloquium-Newsletters, gerne laden wir Sie zu einem Promotionsvortrag ein. Vortragender: Herr Matthias Schrammen, M. Sc. Thema: *Front-End Signal Processing for Far-Field Speech Communication* Zeit: Freitag, 18. März 2022, 10:00 Uhr Zoom-Meeting: https://rwth.zoom.us/j/97904157921?pwd=SWpsbDl0MWhrWjY1ZkZaeFRoYmErZz09 <https://rwth.zoom.us/j/97904157921?pwd=SWpsbDl0MWhrWjY1ZkZaeFRoYmErZz09>Meeting-ID: 979 0415 7921 Passwort: 481650 Devices for speech communication operated in hands-free mode offer a very natural way of human communication. The capturing device, e.g., a smartphone, smart speaker or tablet, is often located up to several meters away from the human speaker. Furthermore, detrimental effects like noise and reverberation are present in everyday acoustic environments. Therefore, the signal-to-noise ratio at the microphones mounted on the device is typically too low to offer sufficient speech quality for the listener at the other end of the communication link. In addition, the loudspeaker of the device is located much closer to the microphones than the human speaker. Therefore, a strong echo signal from the loudspeaker couples into the microphones degrading the conversation quality for the remote listener even further. State-of-the-art approaches that tackle the above-mentioned problems usually rely on multiple microphones to improve the signal-to-noise ratio with methods like beamforming. Beamforming combines the digitally filtered signals of several microphones to obtain an enhanced speech signal at the output. In addition, acoustic echo cancellation is employed to attenuate the echo signal more specifically. This is achieved by adaptive estimation of a digital model of the acoustic echo path and subsequent subtraction of the synthesized echo signal from the microphone signal. However, the solutions are usually optimized for one specific device and are only applicable when the positions of the microphones are fixed and known to the algorithm. Furthermore, the combination of multi-microphone enhancement and echo cancellation is not trivial and low complexity solutions are lacking performance in terms of tracking dynamic acoustic scenarios. Finally, low costs, small form factors, and high desired sound pressure levels result in loudspeakers that operate at their physical limits. This adds significant nonlinear components to the sound emitted by the loudspeaker. Therefore, conventional linear acoustic echo cancellation cannot compensate for the nonlinear parts of the echo and the conversational quality is not satisfactory. The task of the dissertation is to alleviate these shortcomings. The developed signal processing algorithms should be more flexible with respect to desired features in real devices. Among these are microphone positions that are unknown or change during operation and the use of beamforming and acoustic echo cancellation at the same time. Furthermore, the developed solutions should be able to handle nonlinear echo paths and should introduce a low computational complexity to be attractive for battery-powered devices, too. Alle Interessierten sind herzlich eingeladen, eine Anmeldung ist nicht erforderlich. -- Irina Esser Institute of Communication Systems (IKS) RWTH Aachen University Muffeter Weg 3a, 52074 Aachen, Germany +49 241 80 26958 (phone) esser(a)iks.rwth-aachen.de http://www.iks.rwth-aachen.de/

2 Jahre, 1 Monat

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Kommunikationstechnisches Kolloquium am IKS

von Irina Esser

Sehr geehrte Abonnenten des Kolloquium-Newsletters, gerne informieren wir Sie über den nächsten Termin unseres Kommunikationstechnischen Kolloquiums. *Mittwoch, 9. März 2022* *Vortragender*: Marcel Kohn *Zeit*: 14:00 Uhr *Ort*: https://rwth.zoom.us/j/97904157921?pwd=SWpsbDl0MWhrWjY1ZkZaeFRoYmErZz09 Meeting-ID: 979 0415 7921 Passwort: 481650 *Master-Vortrag*: Robuste Schätzung der Sprecherstimme bei Hearables und Hörgeräten mit Mehr-Sensor-Systemen Durch den Verschluss des Gehörgangs durch Hörgeräte oder Hearables wird die Eigenwahrnehmung der eigenen Stimme gestört. Wenn keine Gegenmaßnahmen ergriffen werden, wird die eigene Stimme als dröhnend empfunden, was als Okklusionseffekt bezeichnet wird. Er kann durch die so genannte aktive Okklusionsunterdrückung (AOC) verhindert werden. Dieser Algorithmus nutzt jedoch in der Regel einen akustischen Hear-Through aus, so dass er in lauten Umgebungen eher nachteilig ist. Bei einem alternativen Ansatz arbeitet das Gerät im aktiven Geräuschunterdrückungsmodus (ANC), so dass alle Umgebungsgeräusche blockiert werden. Dann kann eine verbesserte Wahrnehmung der eigenen Stimme erreicht werden, wenn eine Schätzung des Luftschalls der eigenen Stimme durch das Gerät wiedergegeben wird, um das Gefühl eines unverschlossenen Ohres zu erzeugen. Da ANC-Geräte jedoch in der Regel in lauten Umgebungen getragen werden, ist die Schätzung dieses Signals eine Herausforderung. Eine Möglichkeit, die natürliche Wahrnehmung wiederherzustellen, besteht darin, den gedämpften Luftschall der Stimme zu rekonstruieren und so eine natürliche Wahrnehmung zu erzeugen. In dieser Arbeit wrid ein neuronales Netzwerk in ein ANC-System integriert, um die Sprachkomponenten der Stimme des Sprechers von den Umgebungsgeräuschen zu trennen. Nach Anwendung eines Equalizers zur Berücksichtigung weiterer akustischer Einflüsse wird das entrauschte Sprachsignal über einen Kopfhörerlautsprecher im Gehörgang wiedergegeben. Im Vergleich zu bestehenden Sprachverbesserungssystemen wird das Signal eines zusätzlichen Mikrofons an der Innenseite des Kopfhörers als Nebeninformation berücksichtigt. Innerhalb einer Messreihe werden die für das Training benötigten Daten mit Testpersonen aufgenommen. Darüber hinaus werden gerätebezogene Übertragungsfunktionen gemessen, die zusammen mit Ambisonics-Aufnahmen höherer Ordnung (HOA) zur Vergrößerung der Trainingsdatenmenge verwendet werden können, was zu 1736 Stunden Audiodaten für 21 Testpersonen führt. Eine Untersuchung verschiedener rekurrenter neuronaler Faltungsnetzwerke zeigt insbesondere, dass die Verwendung des inneren Mikrofons sowohl zu der gewünschten Störgeräuschreduktion als auch zu einer Verschlechterung der Sprachqualität führt. Weitere Änderungen an der Netzarchitektur des untersuchten Netzes führen zu erhöhten Werten bei wahrnehmungsmotivierten Metriken. Darüber hinaus wird eine auf Multimasking basierende Netzwerkerweiterung getestet, die in der Lage ist, die Dämpfung von Störsignalen dynamisch durch einen einzigen Parameter zu reduzieren, was zu vergleichbar hohen Ergebnissen führt. Alle Interessierten sind herzlich eingeladen, eine Anmeldung ist nicht erforderlich. Allgemeine Informationen zum Kolloquium, sowie eine aktuelle Liste der Termine des Kommunikationstechnischen Kolloquiums finden Sie unter: http://www.iks.rwth-aachen.de/aktuelles/kolloquium -- Irina Esser Institute of Communication Systems (IKS) RWTH Aachen University Muffeter Weg 3a, 52074 Aachen, Germany +49 241 80 26958 (phone) ronkartz(a)iks.rwth-aachen.de http://www.iks.rwth-aachen.de/

2 Jahre, 2 Monate

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Einladung Promotionsvortrag

von Irina Esser

Sehr geehrte Abonnenten des Kolloquium-Newsletters, gerne laden wir Sie zu einem Promotionsvortrag ein. Vortragender: Herr Stefan Kühl, M. Sc. Thema: *Adaptive Algorithms for the Identification of Time-Variant Acoustic Systems* Zeit: Montag, 7. März 2022, 11:00 Uhr Zoom-Meeting: https://rwth.zoom.us/j/97904157921?pwd=SWpsbDl0MWhrWjY1ZkZaeFRoYmErZz09 <https://rwth.zoom.us/j/97904157921?pwd=SWpsbDl0MWhrWjY1ZkZaeFRoYmErZz09>Meeting-ID: 979 0415 7921 Passwort: 481650 Many digital speech and audio communication systems incorporate models of acoustic systems during signal processing. Often, the impulse responses of these acoustic systems have to be identified during or before using a communication system by means of adaptive algorithms. Acoustic systems describe how sound is affected during transmission between a source and a receiver. Examples of acoustic systems comprise rooms with reflections from boundaries and scattering from objects, communication devices such as smartphones or smart home devices, or even a human head where shadowing effects occur. In many situations, the impulse responses of these systems need to be identified. Possible scenarios are acoustic measurements or system identification in speech communication applications, e.g., for accoustic echo cancellation (AEC). Depending on the specific use case, vertain aspects have to be taken into account. For a measurement the excitation signal can be designed, whereas for speech communication applications the system's excitation is the speech signal that cannot be altered. Hence, for the latter case, correlation has to be considered during the system identification. In addition, the systems to be identified may vary over time when the acoustic environment changes, e.g., due to moving objects. Therefore, adaptive algorithms must be used to track the system's state. Additional challenges arise when considering the identification of multiple channels simultaneously. This thesis considers different aspects of systems identification of time-variant acoustic systems in a joint framework for diverse scenarios. Alle Interessierten sind herzlich eingeladen, eine Anmeldung ist nicht erforderlich. -- Irina Esser Institute of Communication Systems (IKS) RWTH Aachen University Muffeter Weg 3a, 52074 Aachen, Germany +49 241 80 26958 (phone) ronkartz(a)iks.rwth-aachen.de http://www.iks.rwth-aachen.de/

2 Jahre, 2 Monate

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Communication Technology Colloquium at IKS

von Irina Esser

Dear subscribers of the colloquium newsletter, we are happy to inform you about the next date of our Communication Technology Colloquium. *Friday, January 21, 2022* *Speaker*: Ali Yilmaz Yildirim *Time*: 11:00 a.m. *Location*: https://rwth.zoom.us/j/97904157921?pwd=SWpsbDl0MWhrWjY1ZkZaeFRoYmErZz09 Meeting-ID: 979 0415 7921 Passwort: 481650 *Master Lecture*: Dynamic Spectrum Access in Highly Mobile Environments Dynamic Spectrum Access (DSA) has been introduced to enable the opportunistic utilization of licensed frequency bands by secondary users (SUs). DSA was introduced for ground-based users, and algorithms were developed by considering those users. With the increase in air data traffic, DSA has started to be considered for airborne communication. Nevertheless developed algorithms for ground-based users can not be reused for airborne communication. The main reason for that is the high mobility of aerial vehicles. Therefore, current DSA algorithms should be examined and adapted for highly mobile environments. With this need, this thesis aims to develop algorithms for DSA in highly mobile environments by focusing on the aspects of DSA that require to be adapted. As a result of SUs being considered as temporary users, SUs should vacate the channel with the transmission of the primary users (PUs) and find a new common idle channel to continue their communication. With the highly varying spectral environment as a result of high mobility, communication between SUs will be interrupted with PU transmission more frequently compared to stationary environments. That is why SUs should detect PU transmission and react by finding a new common idle channel in the shortest time. By considering the need for an adaptation in this aspect of DSA, this thesis proposes algorithms and a frame structure to identify the PU presence and to find a new common channel in highly mobile environments. All interested parties are cordially invited, registration is not required. General information on the colloquium, as well as a current list of dates of the Communication Technology Colloquium can be found at: http://www.iks.rwth-aachen.de/aktuelles/kolloquium -- Irina Esser Institute of Communication Systems (IKS) RWTH Aachen University Muffeter Weg 3a, 52074 Aachen, Germany +49 241 80 26958 (phone) ronkartz(a)iks.rwth-aachen.de http://www.iks.rwth-aachen.de/

2 Jahre, 3 Monate

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Coomunication Technology Colloquium at IKS

von Irina Esser

Dear subscribers of the colloquium newsletter, we are happy to inform you about the next date of our Communication Technology Colloquium. *Wednesday, November 24, 2021* *Speaker*: Georg Krekel *Time*: 2:00 p.m. *Location*: https://rwth.zoom.us/j/97904157921?pwd=SWpsbDl0MWhrWjY1ZkZaeFRoYmErZz09 Meeting-ID: 979 0415 7921 Passwort: 481650 *Bachelor Lecture*: Comparison of Sound Field Translation Methods Higher Order Ambisonics (HOA) signals can be used to allow directional playback of sound signals, where the listener can rotate their head around all three rotational axis. HOA signals can also be adapted to allow translational movement of the listener using sound field translation. Sound field translation is the process of evaluating a sound field at a different position than it was recorded at. At the IKS (Institute of Communication Systems, RWTH Aachen) multiple algorithms have been proposed, that try to achieve psychoacoustically plausible sound field translation, however the inner workings of these algorithms are not well understood. This thesis compares different aspects of two algorithms, Adaptive Beamforming and Adaptive Spacewarping, mainly by deriving analytical expressions that allow deeper understanding of the underlying effects. The algorithms were split into two main stages, the translation stage and the filtering stage. There are significant differences between ASW and ABF. In the translation stage ASW is distorted compared to ABF. However, differences shrink with the order of the HOA input signal. The filtering stage for ABF and ASW is very similar, especially if oracle information of covariance matrices is not available. All interested parties are cordially invited, registration is not required. General information on the colloquium, as well as a current list of dates of the Communication Technology Colloquium can be found at: http://www.iks.rwth-aachen.de/aktuelles/kolloquium -- Irina Esser Institute of Communication Systems (IKS) RWTH Aachen University Muffeter Weg 3a, 52074 Aachen, Germany +49 241 80 26958 (phone) ronkartz(a)iks.rwth-aachen.de http://www.iks.rwth-aachen.de/

2 Jahre, 5 Monate

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Kommunikationstechnisches Kolloquium am IKS

von Irina Ronkartz

Sehr geehrte Abonnenten des Kolloquium-Newsletters, gerne informieren wir Sie über den nächsten Termin unserer Kommunikationstechnischen Kolloquiums. *Dienstag, 12.10.2021* *Vortragender:* Henning Konermann *Zeit: *11:00 Uhr *Ort*: https://rwth.zoom.us/j/97904157921?pwd=SWpsbDl0MWhrWjY1ZkZaeFRoYmErZz09 Meeting-ID: 979 0415 7921 Passwort: 481650 *Bachelor-Vortrag*: Modellbasiertes 3D Headtracking zur Anwendung in adaptiven Übersprechkompensationssystemen In adaptiven Übersprechkompensationssystemen werden verlässliche 3D Positions- und Orientierungsdaten eines Zuhörers in Echtzeit benötigt. Für die Bestimmung dieser Daten wird meist ein zusätzliches physisches Messsystem installiert. Eine Vermeidung dieses Aufbaus hat erhebliche Kosten- und Aufwandsvorteile. Durch das Schätzen der Laufzeit aus den in adaptiven Übersprechkompensationssystemen verwendeten kopfbezogenen Impulsantworten kann anhand eines geometrischen Ansatzes Headtracking betrieben werden. Die Literatur zeigt, dass mit zusätzlichen physischen Messsystemen geometrisch basiertes 2D und 3D Headtracking möglich ist. Außerdem wurde gezeigt, dass für den 2D Fall ohne zusätzliches Messsystem ein modellbasierter Ansatz einem geometrischen Ansatz überlegen ist. Ziel dieser Arbeit ist die Erstellung und Realisierung eines Modells, welches den modellbasierten Ansatz auf den 3D Fall erweitert. Bei der Erstellung eines durchschnittlichen Modells sowie individualisierter Modelle werden kopfbezogene Übertragungsfunktionen verwendet. Das Tracking erfolgt durch die Minimierung eines aus der Literatur adaptierten Optimierungsproblems. Im Anschluss werden Experimente simuliert. Die Trackinggenauigkeit verschieden stark personalisierter Modelle wird mit einem geometrischen Ansatz verglichen. In den meisten Anwendungsfällen lässt sich die Überlegenheit des Modells feststellen. Weiterhin wird erkannt, dass das Modell durch Orientierungslimitierungen einzuschränken ist, um sinnvolle Ergebnisse zu erzielen. Alle Interessierten sind herzlich eingeladen, eine Anmeldung ist nicht erforderlich. Allgemeine Informationen zum Kolloquium sowie eine aktuelle Liste der Termine des Kommunikationstechnischen Kolloquiums finden Sie unter: http://www.iks.rwth-aachen.de/aktuelles/kolloquium -- Irina Ronkartz Institute of Communication Systems (IKS) RWTH Aachen University Muffeter Weg 3a, 52074 Aachen, Germany +49 241 80 26958 (phone) ronkartz(a)iks.rwth-aachen.de http://www.iks.rwth-aachen.de/

2 Jahre, 6 Monate

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Communication Technology Colloquium at IKS

von Irina Ronkartz

Dear subscribers of the colloquium newsletter, we are happy to inform you about the next date of our Communication Technology Colloquium. *Friday, October 8, 2021* *Speaker:* Frederick Pietschmann *Time:* 10:00 a.m. *Location*: https://rwth.zoom.us/j/97904157921?pwd=SWpsbDl0MWhrWjY1ZkZaeFRoYmErZz09 Meeting-ID: 979 0415 7921 Passwort: 481650 *Bachelor Lecture*: Model-Based Algorithms for Retrospective Individualization of Binaural Signals With binaural signals, it is possible to capture the sensation of spatial hearing that humans are used to from real-life auditory scenes. However, because binaural recordings are always individualized for a speciﬁc target person, the authenticity of a binaural recording is perceived differently by different listeners. It is therefore desirable that a binaural recording is always individualized for the respective listener. In this thesis, the existing Binaural Cue Adaptation (BCA) system from the Institute of Communication Systems is adjusted and reinterpreted for it to be able to retrospectively individualize an existing binaural signal for a new target person. Existing approaches to the main components of the BCA system, the Localization block and the Modiﬁcation block, are reviewed and analyzed and multiple new concepts and ideas are introduced and algebraically described: New optimization target functions, a concept for joint optimization over multiple frequencies and multiple approaches that attempt to lessen the impact of artifacts which may occur due to model mismatches and / or erroneous location estimates. A theoretical evaluation is conducted in which the localization performance of both existing and new approaches is benchmarked. The results conﬁrm that some of the newly proposed ideas and concepts improve the overall performance of the BCA system. All interested parties are cordially invited, registration is not required. General information on the colloquium, as well as a current list of dates of the Communication Technology Colloquium can be found at: http://www.iks.rwth-aachen.de/aktuelles/kolloquium -- Irina Ronkartz Institute of Communication Systems (IKS) RWTH Aachen University Muffeter Weg 3a, 52074 Aachen, Germany +49 241 80 26958 (phone) ronkartz(a)iks.rwth-aachen.de http://www.iks.rwth-aachen.de/

2 Jahre, 7 Monate

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