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Audio Signal Processing [clear filter]
Thursday, October 28
 

9:00pm EDT

A Neural Beamforming Front-end for Distributed Microphone Arrays
Robust real-time audio signal enhancement increasingly relies on multichannel microphone arrays for signal acquisition. Sophisticated beamforming algorithms have been developed to maximize the benefit of multiple microphones. With the recent success of deep learning models created for audio signal processing, the task of Neural Beamforming remains an open research topic. This paper presents a Neural Beamformer architecture capable of performing spatial beamforming with microphones randomly distributed over very large areas, even in negative signal-to-noise ratio environments with multiple noise sources and reverberation. The proposed method combines adaptive, nonlinear filtering and the computation of spatial relations with state-of-the-art mask estimation networks. The resulting End-to-End network architecture is fully differentiable and provides excellent signal separation performance. Combining a small number of principal building blocks, the method is capable of low-latency, domain-specific signal enhancement even in challenging environments.

Speakers
JZ

Jonathan Ziegler

Stuttgart Media University
LS

Leon Schröder

Stuttgart Media University
AK

Andreas Koch

HdM Stuttgart
AS

Andreas Schilling

Eberhard Karls University Tuebingen


Thursday October 28, 2021 9:00pm - Friday December 3, 2021 6:00pm EST
On-Demand

9:00pm EDT

Audio-Source Rendering on Flat-Panel Loudspeakers with Non-Uniform Boundary Conditions
Devices from smartphones to televisions are beginning to employ dual purpose displays, where the display serves as both a video screen and a loudspeaker. In this paper we demonstrate a method to generate localized sound-radiating regions on a flat-panel display. An array of force actuators affixed to the back of the panel is driven by appropriately filtered audio signals so the total response of the panel due to the actuator array approximates a target spatial acceleration profile. The response of the panel to each actuator individually is initially measured via a laser vibrometer, and the required actuator filters for each source position are determined by an optimization procedure that minimizes the mean squared error between the reconstructed and targeted acceleration profiles. Since the single-actuator panel responses are determined empirically, the method does not require analytical or numerical models of the system’s modal response, and thus is well-suited to panels having the complex boundary conditions typical of television screens, mobile devices, and tablets. The method is demonstrated on two panels with differing boundary conditions. When integrated with display technology, the localized audio source rendering method may transform traditional displays into multimodal audio-visual interfaces by colocating localized audio sources and objects in the video stream.

Speakers
MH

Michael Heilemann

Assistant Professor, University of Rochester
avatar for Tre DiPassio

Tre DiPassio

PhD Student, University of Rochester
Hello! My name is Tre, and I am in my final semester as a PhD student studying musical acoustics and signal processing under the supervision of Dr. Mark Bocko and Dr. Michael Heilemann. The research lab I am a part of has been developing an emerging type of speaker, called a flat... Read More →
MB

Mark Bocko

University of Rochester


Thursday October 28, 2021 9:00pm - Friday December 3, 2021 6:00pm EST
On-Demand

9:00pm EDT

Informed postprocessing for auditory roughness removal for low-bitrate audio coders
In perceptual audio coding using very low bitrates, modulation artifacts can be introduced onto tonal signal components, which are often perceived as auditory roughness. These artifacts may occur for instance due to quantization errors or may be added when using an audio bandwidth extension, which sometimes causes an irregular harmonic structure at the borders of replicated bands. Especially, the roughness artifacts due to quantization errors are difficult to mitigate without investing considerably more bits in encoding of tonal components. We propose a novel technique to remove these roughness artifacts at the decoder side controlled by a small amount of guidance information transmitted by the encoder.

Speakers
SV

Steven Van De Par

Carl von Ossietzky University, Department of Medical Physics and Acoustics
SD

Sascha Disch

Fraunhofer IIS, Erlangen
AN

Andreas Niedermeier

Fraunhofer IIS, Erlangen
BE

Bernd Edler

Audiolabs Erlangen


Thursday October 28, 2021 9:00pm - Friday December 3, 2021 6:00pm EST
On-Demand

9:00pm EDT

Phoneme Mappings for Online Vocal Percussion Transcription
Vocal Percussion Transcription (VPT) aims at detecting vocal percussion sound events in a beatboxing performance and classifying them into the correct drum instrument class (kick, snare, or hi-hat). To do this in an online (real-time) setting, however, algorithms are forced to classify these events within just a few milliseconds after they are detected. The purpose of this study was to investigate which phoneme-to-instrument mappings are the most robust for online transcription purposes. We used three different evaluation criteria to base our decision upon: frequency of use of phonemes among different performers, spectral similarity to reference drum sounds, and classification separability. With these criteria applied, the recommended mappings would potentially feel natural for performers to articulate while enabling the classification algorithms to achieve the best performance possible. Given the final results, we provided a detailed discussion on which phonemes to choose given different contexts and applications.

Speakers
AL

Alejandro Luezas

Roli / Queen Mary University of London
CS

Charalampos Saitis

Queen Mary University of London
MS

Mark Sandler

Queen Mary University of London


Thursday October 28, 2021 9:00pm - Friday December 3, 2021 6:00pm EST
On-Demand
 
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