scholarly journals Interaural Coherence Estimation for Speech Processing in Reverberant Environment

2020 ◽  
Vol 10 (3) ◽  
pp. 769
Author(s):  
Seong-Hu Kim ◽  
Yong-Hwa Park

Interaural coherence is used to quantify the effects of reverberation on speech, and previous studies applied the conventional method using all previous time data in the form of an infinite impulse response filter to estimate interaural coherence. To consider a characteristic of speech that continuously changes over time, this paper proposes a new method of estimating interaural coherence using time data within a finite length of speech, which is called the quasi-steady interval. The length of the quasi-steady interval is determined with various frequency bands, reverberation times, and short-time Fourier transform (STFT) variables through numerical experiment, and it decreased as reverberation time decreased and the frequency increased. In this interval, a diffuse speech, which is an infinite sum of reflected speeches of different propagating paths, is uncorrelated between two microphones apart from each other; thus, the coherence is close to zero. However, a direct speech measured at the two microphones has steady amplitude and phase difference in this internal; thus, the coherence is close to one. Moreover, the new method is the form of a finite impulse response filter that has a linear phase delay or zero phase delay with respect to speech to frequency; thus, the same or zero time delay for each frequency is applied to the power spectral density. Therefore, the coherence estimation of the new method is closer to the ideal value than the conventional one, and the coherence is accurately estimated at the time–frequency bins of direct speech, which is time-varying according to speech variation.

2015 ◽  
Vol 719-720 ◽  
pp. 826-832 ◽  
Author(s):  
Kai Wu ◽  
Tao Su ◽  
Jin Hao Pang

To solve the space dispersion (SD) problem which makes the beam formed deviate from its expected direction in wideband digital array beamforming, a new wideband digital array beamformer (WDBF) is designed by introducing an identical variable fractional delay (VFD) filter on each sensor. The VFD filter is an all-pass filter with an infinite impulse response, thus it requires less filter order compared with finite impulse response filter. Since the VFD designed is suitable within the whole fractional delay range of-0.5 to 0.5 in sampling interval, the new beamformer designed is suitable for any direction without redesigning the fractional delay filter. In this way, the real-time property of the new beamformer is guaranteed. Simulation results show that the VFD deigned in this paper is high in precision. Applying it in the new WDBF, the SD problem is solved and the beam formed approximates the ideal beam very well.


Author(s):  
Andrzej Handkiewicz ◽  
Mariusz Naumowicz

AbstractThe paper presents a method of optimizing frequency characteristics of filter banks in terms of their implementation in digital CMOS technologies in nanoscale. Usability of such filters is demonstrated by frequency-interleaved (FI) analog-to-digital converters (ADC). An analysis filter present in these converters was designed in switched-current technique. However, due to huge technological pitch of standard digital CMOS process in nanoscale, its characteristics substantially deviate from the required ones. NANO-studio environment presented in the paper allows adjustment, with transistor channel sizes as optimization parameters. The same environment is used at designing a digital synthesis filter, whereas optimization parameters are input and output conductances, gyration transconductances and capacitances of a prototype circuit. Transition between analog s and digital z domains is done by means of bilinear transformation. Assuming a lossless gyrator-capacitor (gC) multiport network as a prototype circuit, both for analysis and synthesis filter banks in FI ADC, is an implementation of the strategy to design filters with low sensitivity to parameter changes. An additional advantage is designing the synthesis filter as stable infinite impulse response (IIR) instead of commonly used finite impulse response (FIR) filters. It provides several dozen-fold saving in the number of applied multipliers.. The analysis and synthesis filters in FI ADC are implemented as filter pairs. An additional example of three-filter bank demonstrates versatility of NANO-studio software.


Author(s):  
David Rivas-Lalaleo ◽  
Sergio Muñoz-Romero ◽  
Monica Huerta ◽  
Víctor Bautista-Naranjo ◽  
Jorge García-Quintanilla ◽  
...  

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