Results 141 to 150 of about 926 (186)

Speech Dereverberation Using Variational Autoencoders

ICASSP 2021 - 2021 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP), 2021
This paper presents a statistical method for single-channel speech dereverberation using a variational autoencoder (VAE) for modelling the speech spectra. One popular approach for modelling speech spectra is to use non-negative matrix factorization (NMF) where learned clean speech spectral bases are used as a linear generative model for speech spectra.
Deepak Baby, Hervé Bourlard
openaire   +1 more source

Amplitude Consistent Enhancement for Speech Dereverberation

Proceedings of the 2020 6th International Conference on Computing and Artificial Intelligence, 2020
The mapping and masking methods based on deep learning are both essential methods for speech dereverberation at present, which typically enhance the amplitude of the reverberant speech while letting the reverberant phase unprocessed. The reverberant phase and enhanced amplitude are used to synthesize the target speech.
Chunlei Liu 0005, Longbiao Wang, Jianwu Dang 0001   +2 more
openaire   +1 more source

Cepstrum based deconvolution for speech dereverberation

Proceedings of ICASSP '94. IEEE International Conference on Acoustics, Speech and Signal Processing, 2002
We propose an algorithm for the restoration of speech that has been degraded through addition of multiple echoes. The proposed scheme processes the outputs of two microphones using cepstra operations and the theory of signal reconstruction from phase only.
Athina P. Petropulu, Suresh Subramaniam
openaire   +1 more source

Group sparsity for mimo speech dereverberation

2015 IEEE Workshop on Applications of Signal Processing to Audio and Acoustics (WASPAA), 2015
Reverberation can severely affect the speech signals recorded in a room, possibly leading to a significantly reduced speech quality and intelligibility. In this paper we present a batch algorithm employing a signal model based on multi-channel linear prediction in the short-time Fourier transform domain. Aiming to achieve multiple-input multiple-output
Ante Jukic, Toon van Waterschoot, Timo Gerkmann, Simon Doclo   +3 more
openaire   +1 more source

Calculating Inverse Filters for Speech Dereverberation

IEICE Transactions on Fundamentals of Electronics, Communications and Computer Sciences, 2008
Speech dereverberation is one of the most difficult tasks in acoustic signal processing. Of the various problems involved in this task, this paper highlights “over-whitening,” which flattens the characteristics of recovered speech. This distortion sometimes happens when inverse filters are directly calculated from microphone signals. This paper reviews
Masato Miyoshi, Marc Delcroix, Keisuke Kinoshita   +2 more
openaire   +1 more source

Cepstrum-based deconvolution for speech dereverberation

IEEE Transactions on Speech and Audio Processing, 1996
We present a blind deconvolution-based approach for the restoration of speech degraded by the acoustic environment. The proposed scheme processes the outputs of two microphones using cepstra operations and the theory of signal reconstruction from phase only.
Suresh Subramaniam, Athina P. Petropulu, Christopher Wendt   +2 more
openaire   +1 more source

Multimicrophone Based Speech Dereverberation

2018
Speech signal received by distant microphones in real environments contains reverberation and noise. This deteriorates the quality of received signal. However to improve the speech quality, it is essential to remove reverberation and noise. The process of removing reverberation and reproducing original speech is called dereverberation.
Seema Vitthal Arote, Mangesh Sudhir Deshpande   +1 more
openaire   +1 more source

Towards Speech Recognition Oriented Dereverberation

Proceedings. (ICASSP '05). IEEE International Conference on Acoustics, Speech, and Signal Processing, 2005., 2006
We show the effect of reverberation on the speech recognition performance in a far-field microphone. Given the reference of a close-talk microphone signal, an improvement is shown using an asymmetric non-causal inverse filter both in a synthetic and real room environment.
Pamornpol Jinachitra, Ramon Prieto
openaire   +1 more source

Study on Speech Dereverberation with Autocorrelation Codebook

2007 IEEE International Conference on Acoustics, Speech and Signal Processing - ICASSP '07, 2007
This paper proposes a new speech dereverberation approach based on a statistical speech model. An autocorrelation codebook is introduced as a model that can represent time-varying short-time speech characteristics corresponding to the cepstrum and harmonics.
Tomohiro Nakatani, Biing-Hwang Juang, Takafumi Hikichi, Takuya Yoshioka, Keisuke Kinoshita, Marc Delcroix, Masato Miyoshi   +6 more
openaire   +1 more source

A Novel Approach for Dereverberation of Speech Signals

2019 13th International Conference on Signal Processing and Communication Systems (ICSPCS), 2019
In this paper, we propose a novel higher-order statistics based approach to estimate the room impulse response (RIR) of a reverberating auditorium. The RIR is then used to dereverberate the speech signal. It is assumed that the RIR does not change for a brief period of time called the coherence time.
Sauradeep Dey, Pradip Sircar
openaire   +1 more source