relative transfer function
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Author(s):  
Amit Sofer ◽  
Tomas Kounovsky ◽  
Jaroslav Cmejla ◽  
Zbynek Koldovsky ◽  
Sharon Gannot

Information ◽  
2020 ◽  
Vol 11 (11) ◽  
pp. 505
Author(s):  
Jounghoon Beh

In this paper, a sequential approach is proposed to estimate the relative transfer functions (RTF) used in developing a generalized sidelobe canceller (GSC). The latency in calibrating microphone arrays for GSC, often suffered by conventional approaches involving batch operations, is significantly reduced in the proposed sequential method. This is accomplished by an immediate generation of the RTF from initial input segments and subsequent updates of the RTF as the input stream continues. From the experimental results via the mean square error (MSE) criterion, it has been shown that the proposed method exhibits improved performance over the conventional batch approach as well as over recently introduced least mean squares approaches.


2019 ◽  
Vol 9 (12) ◽  
pp. 2520 ◽  
Author(s):  
Juan M. Martín-Doñas ◽  
Antonio M. Peinado ◽  
Iván López-Espejo ◽  
Angel Gomez

This paper deals with speech enhancement in dual-microphone smartphones using beamforming along with postfiltering techniques. The performance of these algorithms relies on a good estimation of the acoustic channel and speech and noise statistics. In this work we present a speech enhancement system that combines the estimation of the relative transfer function (RTF) between microphones using an extended Kalman filter framework with a novel speech presence probability estimator intended to track the noise statistics’ variability. The available dual-channel information is exploited to obtain more reliable estimates of clean speech statistics. Noise reduction is further improved by means of postfiltering techniques that take advantage of the speech presence estimation. Our proposal is evaluated in different reverberant and noisy environments when the smartphone is used in both close-talk and far-talk positions. The experimental results show that our system achieves improvements in terms of noise reduction, low speech distortion and better speech intelligibility compared to other state-of-the-art approaches.


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