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Filter Bank Design based on Minimization of Individual Aliasing Terms for Minimum Mutual Information Subband Adaptive Beamforming

Related publications (34)

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Spectro-Temporal Features for Automatic Speech Recognition using Linear Prediction in Spectral Domain

Hynek Hermansky, Sriram Ganapathy, Samuel Thomas

Frequency Domain Linear Prediction (FDLP) provides an efficient way to represent temporal envelopes of a signal using auto-regressive models. For the input speech signal, we use FDLP to estimate temporal trajectories of sub-band energy by applying linear p ...

IDIAP2008

Adaptive Beamforming with a Maximum Negentropy Criterion

Philip Neil Garner, John David Scott Dines, Weifeng Li

This paper presents an adaptive beamforming application based on the capture of far-field speech data from a real single speaker in a real meeting room. After the position of a speaker is estimated by a speaker tracking system, we construct a subband-domai ...

IDIAP2008

Effective post-processing for single-channel frequency-domain speech enhancement

Weifeng Li

Conventional frequency-domain speech enhancement filters improve signal-to-noise ratio (SNR), but also produce speech distortions. This paper describes a novel post-processing algorithm devised for the improvement of the quality of the speech processed by ...

IDIAP2007

Minimum Mutual Information Beamforming for Simultaneous Active Speakers

In this work, we consider an acoustic beamforming application where two speakers are simultaneously active. We construct one subband-domain beamformer in \emph{generalized sidelobe canceller} (GSC) configuration for each source. In contrast to normal pract ...

IDIAP2007

Adaptive Beamforming with a Minimum Mutual Information Criterion

IDIAP2007

Novel speech processing techniques for robust automatic speech recognition

Vivek Tyagi

The goal of this thesis is to develop and design new feature representations that can improve the automatic speech recognition (ASR) performance in clean as well noisy conditions. One of the main shortcomings of the fixed scale (typically 20-30 ms long ana ...

EPFL2006

Towards using slide information to enhance speech transcription of meetings

Hervé Bourlard, Artem Peregoudov, Alessandro Vinciarelli

In this paper we investigate the possibility of improving the speech recognition performance of meeting recordings by using slides captured during the recording process. The key hypothesis exploited in this work is that both slides and speech carry correla ...

IDIAP2006

Silicon implementation of an MMSE-based soft demapper for MIMO-BICM

Andreas Peter Burg

The performance of systems employing bit-interleaved coded modulation (BICM) critically depends on the availability of soft information. In the multi-antenna case, the extraction of optimum bit-metrics becomes prohibitively complex, so that suboptimal solu ...

Ieee Service Center, 445 Hoes Lane, Po Box 1331, Piscataway, Nj 08855-1331 Usa2006

Fractal additive synthesis

Pietro Polotti

Musical and audio signals in general form a major part of the large amount of data exchange taking place in our information-based society. Transmission of high quality audio signals through narrow-band channels, such as the Internet, requires refined metho ...

EPFL2003

Wavelet footprints and frames for signal processing and communication

Pier Luigi Dragotti

Linear transforms and expansions are fundamental mathematical tools of signal processing. In particular, the wavelet transform has played an important role in several signal processing tasks, compression being a prime example. A signal can be represented i ...

EPFL2002