Analysis of two-sensors forward BSS structure with post-filters in the presence of coherent and incoherent noise

Djendi, Mohamed; Scalart, Pascal; Gilloire, André

doi:10.1016/j.specom.2013.06.001

Cited by 42 publications

(16 citation statements)

References 22 publications

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“…This mixing model can be simplified by taking h 11 (n) = h 22 (n) = δ(n) (the Kronecker impulse response).Then the output equations become as follows 11 : h 12 (n) and h 21 (n) represent the crosscoupling effects between the 2 channels, and K is the impulse response length.…”

Section: Multimicrophone and Simplified Two-microphone Mixing Modelsmentioning

confidence: 99%

“…1,2,[11][12][13] We recall here that we have proposed in the previous studies [11][12][13] and in 2 studies 44,45 several techniques that estimate this postfilters. The effect of this postfilter is important when the 2 microphones are closely spaced.…”

Section: Multimicrophone and Simplified Two-microphone Mixing Modelsmentioning

confidence: 99%

“…11,56,57 To assess the performance of the proposed algorithm in objective criteria with a real-life database, we have evaluated the SegSNR, the CD, and the time convergence criteria of each of the adaptive and nonadaptive algorithms of Table 5. 11,56,57 To assess the performance of the proposed algorithm in objective criteria with a real-life database, we have evaluated the SegSNR, the CD, and the time convergence criteria of each of the adaptive and nonadaptive algorithms of Table 5.…”

Section: Assessment Of the Proposed Tm-gspap Algorithm With Real-limentioning

confidence: 99%

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A new two‐microphone Gauss‐Seidel pseudo affine projection algorithm for speech quality enhancement

Djendi

2017

Adaptive Control & Signal

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This study addresses the problem of speech quality enhancement by adaptive and nonadaptive filtering algorithms. The well-known two-microphone forward blind source separation (TM-FBSS) structure has been largely studied in the literature. Several two-microphone algorithms combined with TM-FBSS have been recently proposed. In this study, we propose 2 contributions: In the first, a new two-microphone Gauss-Seidel pseudo affine projection (TM-GSPAP) algorithm is combined with TM-FBSS. In the second, we propose to use the new TM-GSPAP algorithm in speech enhancement. Furthermore, we show the efficiency of the proposed TM-GSPAP algorithm in speech enhancement when highly noisy observations are available. To validate the good performances of our algorithm, we have evaluated the adaptive filtering properties in computational complexity and convergence speed performance by system mismatch criteria. A fair comparison with adaptive and nonadaptive noise reduction algorithms are also presented. The adaptive algorithms are the well-known two-microphone normalized least mean square algorithm, and the recently published two-microphone pseudo affine projection algorithm. The nonadaptive algorithms are the one-microphone spectral subtraction and the two-microphone Wiener filter algorithm. We evalute the quality of the output speech signal in each algorithm by several objective and subjective criteria as the segmental signalto-noise ratio, cepstral distance, perceptual evaluation of speech quality, and the mean opinion score. Finally, we validate the superior performances of the proposed algorithm with physically measured signals.

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Section: Multimicrophone and Simplified Two-microphone Mixing Modelsmentioning

confidence: 99%

Section: Multimicrophone and Simplified Two-microphone Mixing Modelsmentioning

confidence: 99%

Section: Assessment Of the Proposed Tm-gspap Algorithm With Real-limentioning

confidence: 99%

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A new two‐microphone Gauss‐Seidel pseudo affine projection algorithm for speech quality enhancement

Djendi

2017

Adaptive Control & Signal

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“…The method proposes two models of non-speech and speech instead of existed two states, formally, each model could include several states, also, they analyze other features, such as pitch index, pitch magnitude and fractal dimension of speech and non-speech. In a car environment, where the driver makes a hands-free communication with a farend talker, this distant-talking involves a decrease of the desired speech signal quality [18][19][20][21]. This distortion is originally caused by the high number of quasi-infinite noise sources since vibrations, road, fan and wind noise from open windows generate a continuous and distributed noise background.…”

Section: Introductionmentioning

confidence: 99%

“…FBSS and BBSS) structures applied to enhance corrupted speech signals and to cancel the acoustic noise components. Several works have dealt with these two structures [18][19][20][21][22][23]. However, all of these techniques used a MVAD system that makes them inoperative in practice.…”

Section: Introductionmentioning

confidence: 99%

A new regularized forward blind source separation algorithm for automatic speech quality enhancement

Zoulikha

Djendi

2016

Applied Acoustics

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New efficient adaptive fast transversal filtering (FTF)‐type algorithms for mono and stereophonic acoustic echo cancelation

Djendi

2014

Adaptive Control & Signal

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This paper addresses the field of stereophonic acoustic echo cancelation (SAEC) by adaptive filtering algorithms. Recently, simplified versions of the fast transversal filter (SFTF)-type algorithm has been proposed. In this paper, we propose two major contributions. In the first contribution, we propose two new FTF-type algorithms with low complexity and good convergence speed characteristics. These two proposed algorithms are mainly on the basis of a forward prediction scheme to estimate the so called dual Kalman gain, which is inherent in the filtering part update. This computation complexity is achieved by the introduction of new relations for the computation of the likelihood variables that are simple and lead to further simplifications on the prediction part of the two proposed algorithms. In the second contribution, we propose to adapt then apply these four new SFTF-type algorithms, (the two proposed algorithms in this paper and their original versions) in the SAEC applications. A fair comparison of the proposed algorithms with the original SFTF and the normalized least mean square algorithms, in mono and SAEC applications, is presented. Copyright 274 M. DJENDI Recently, a different approach for the SAEC problems (inter-channel coherence, adaptive filter lengths, etc.) was proposed in [18,19] by using the widely linear model, which is assigned to the processing of noncircular complex valued signals. A new hybrid solution to the same problem that uses frequency shifts to improve the performance of the state-of-art solution based on addition of half-wave rectified signals is presented in [20]. The authors in [21] proposed a windowing frequency domain adaptive filter, and an up-sampling block transforms preprocessing, to solve these constraints. The proposed adaptive filter uses windowing functions with smooth cutoff property to reduce the spectral leakage during filter updating, so the utilization of the independent noise introduced by preprocessing in SAEC can be increased. The proposed preprocessing is operated in short blocks with low processing delay, and it uses frequency-domain up-sampling to meet the minimal block length requirement given by the band limit of simultaneous masking. Another very recent and novel approach to SAEC is proposed in [22] and based on a combination of psychoacoustic effects. The principle of the proposed structure is to use an approach based on the 'missing-fundamental' phenomenon at low frequencies (below 500 Hz) and a modified phase modulation approach at higher frequencies.A basic scheme for SAEC is given in Figure 1, where we illustrate the concept with a transmission room on the left and a receiving room on the right [16,17,23,24]. In Figure 1, when we generate a signal in the transmission room, the two microphones mic 1 and mic 2 receive two sorts of signals: (i) the direct source signal modified by the path g 1 (then received by mic 1 / and g 2 (then received by mic 2 /, respectively, and (ii) the existing diffuse noises n 1 and n 2 in the transmission room and received ...

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Analysis of two-sensors forward BSS structure with post-filters in the presence of coherent and incoherent noise

Cited by 42 publications

References 22 publications

A new two‐microphone Gauss‐Seidel pseudo affine projection algorithm for speech quality enhancement

A new two‐microphone Gauss‐Seidel pseudo affine projection algorithm for speech quality enhancement

A new regularized forward blind source separation algorithm for automatic speech quality enhancement

New efficient adaptive fast transversal filtering (FTF)‐type algorithms for mono and stereophonic acoustic echo cancelation

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