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Karim Helwani

8 accepted papers

2024

NOLACE: Improving Low-Complexity Speech Codec Enhancement Through Adaptive Temporal Shaping

ICASSP 2024accepted

Speech codec enhancement methods are designed to remove distortions added by speech codecs. While classical methods are very low in complexity and add zero delay, their effectiveness is rather limited. Compared to that, DNN-based methods deliver higher quality but they are typically high in complexi…

Cited by 0SourceScholar
2024

Real-Time Stereo Speech Enhancement with Spatial-Cue Preservation Based on Dual-Path Structure

ICASSP 2024accepted

We introduce a real-time, multichannel speech enhancement algorithm which maintains the spatial cues of stereo recordings including two speech sources. Recognizing that each source has unique spatial information, our method utilizes a dual-path structure, ensuring the spatial cues remain unaffected…

Cited by 0SourceScholar
2023

Generative Modeling Based Manifold Learning for Adaptive Filtering Guidance

ICASSP 2023accepted

In most practical adaptive filtering problems, estimated filters are not arbitrary, but instead lie on a manifold that encapsulates characteristics of the problem at hand. Consequently, it is desirable to steer adaptation towards filters that lie on that manifold. In this paper, we propose a novel a…

Cited by 4SourceScholar
2021

Enhancing Audio Augmentation Methods with Consistency Learning

ICASSP 2021accepted

Data augmentation is an inexpensive way to increase training data diversity, and is commonly achieved via transformations of existing data. For tasks such as classification, there is a good case for learning representations of the data that are invariant to such transformations, yet this is not expl…

Cited by 0SourceScholar
2021

Low-Complexity, Real-Time Joint Neural Echo Control and Speech Enhancement Based On Percepnet

ICASSP 2021accepted

Speech enhancement algorithms based on deep learning have greatly surpassed their traditional counterparts and are now being considered for the task of removing acoustic echo from hands-free communication systems. This is a challenging problem due to both real-world constraints like loudspeaker non-…

Cited by 0SourceScholar
2019

Blind Signal Processing for Time-varying Convolutive Mixing Systems Based on Sequence Estimation on Partly Smooth Manifolds

ICASSP 2019accepted

In this paper we focus on Bayesian blind and semi-blind adaptive signal processing based on a broadband MIMO FIR model (e.g., for blind source separation (BSS) and blind system identification (BSI)). Specifically, we study in this paper a framework allowing us to systematically incorporate various t…

Cited by 0SourceScholar
2017

Efficient adaptive filtering in compressive domains for sparse systems and relation to transform-domain adaptive filtering

ICASSP 2017accepted

In this paper we introduce a novel class of efficient multichannel adaptive filtering algorithms for sparse FIR systems. By suitably integrating ideas from compressed sensing and adaptive filter theory, this class of algorithms allows to significantly reduce the actual number of adaptive coefficient…

Cited by 0SourceScholar
2016

Localization of sound sources with known statistics in the presence of interferers

ICASSP 2016accepted

Methods are available for simultaneous localization of multiple (unknown) audio sources using microphone arrays. Typical algorithms aim at localizing all active sources. They moreover require that the number of sources is known and is less than or equal the number of microphones. This constraint can…

Cited by 0SourceScholar