Vladimir Tourbabin

Ben-Gurion University of the Negev | bgu · Department of Electrical and Computer Engineering

A two-stage multi-channel speech enhancement method is proposed which consists of a novel adaptive beamformer, Hybrid Minimum Variance Distortionless Response (MVDR), Isotropic-MVDR (Iso), and a novel multi-channel spectral Principal Components Analysis (PCA) denoising. In the first stage, the Hybrid-MVDR performs multiple MVDRs using a dictionary...

Background As well known to any photographer, controlling the “field of view” offers an extremely powerful mechanism by which to adjust target acquisition. Only a few natural sensory systems can actively control their field of view (e.g., dolphins, whales, and bats). Bats are known for their active sensing abilities and modify their echolocation si...

Spatial audio has been studied for several decades, but has seen much renewed interest recently due to advances in both software and hardware for capture and playback, and the emergence of applications such as virtual reality and augmented reality. This renewed interest has led to the investment of increasing efforts in developing signal processing...

In many applications, such as hearing aids and virtual reality, spatial audio is used to provide a more natural experience to the users. However, when captured in the real world, the audio signals may suffer from noise and interference. In this case, the challenge is to attenuate the undesired signals, while preserving the desired signals with thei...

Acoustic rake filters perform coherent summation of the early room reflections using beamforming, with the aim of improving beamforming performance. This concept has been investigated for speech enhancement applications, improving noise reduction and late reverberation attenuation. Current studies typically assume that the parameters of the early r...

Augmented Reality (AR) as a platform has the potential to facilitate the reduction of the cocktail party effect. Future AR headsets could potentially leverage information from an array of sensors spanning many different modalities. Training and testing signal processing and machine learning algorithms on tasks such as beam-forming and speech enhanc...

Non-interactive and linear experiences like cinema film offer high quality surround sound audio to enhance immersion, however the perspective is usually fixed to the recording microphone position. With the rise of virtual reality, there is a demand for recording and recreating real-world experiences that allow users to move throughout the reproduct...

Non-interactive and linear experiences like cinema film offer high quality surround sound audio to enhance immersion, however the listener's experience is usually fixed to a single acoustic perspective. With the rise of virtual reality, there is a demand for recording and recreating real-world experiences in a way that allows for the user to intera...

A recent approach to improving the robustness of sound localization in reverberant environments is based on pre-selection of time- frequency pixels that are dominated by direct sound. This approach is equivalent to applying a binary time-frequency mask prior to the localization stage. Although the binary mask approach was shown to be effective, it...

Signals recorded by microphones form the basis for a wide range of audio signal processing systems. In some applications, such as humanoid robots, the microphones may be moving while recording the audio signals. A common practice is to assume that the microphone is stationary within a short time frame. Although this assumption may be reasonable und...

The auditory system of humanoid robots has gained increased attention in recent years. This system typically acquires the surrounding sound field by means of a microphone array. Signals acquired by the array are then processed using various methods. One of the widely applied methods is direction of arrival estimation. The conventional direction of...

Processing of microphone arrays of various configurations involves mathematical models of the surrounding sound fields. These models are based on different space and time conventions found throughout the literature. In traditional open array processing, interchanging different space and time conventions can lead to confusion between the arrival and...

Auditory systems of humanoid robots usually acquire the surrounding sound field by means of microphone arrays. These arrays can undergo motion related to the robot’s activity. The conventional approach to dealing with this motion is to stop the robot during sound acquisition. This approach avoids changing the positions of the microphones during the...

One of the important tasks of a humanoid-robot auditory system is speaker localization. It is used for the construction of the surrounding acoustic scene and as an input for additional processing methods. Localization is usually required to operate indoors under high reverberation levels. Recently, an algorithm for speaker localization under these...

An important aspect of a humanoid robot is audition. Previous work has presented robot systems capable of sound localization and source segregation based on microphone arrays with various configurations. However, no theoretical framework for the design of these arrays has been presented. In the current paper, a design framework is proposed based on...

A recent and fast evolving application for microphone arrays is the auditory systems of humanoid robots. These arrays, in contrast to conventional arrays, are not fixed in a given position, but move together with the robot. While imposing a challenge to most conventional array processing algorithms, this movement offers an opportunity to enhance pe...

An important part of a human-like robot is robot audition. Previous work presented systems capable of sound localization and source segregation based on microphone arrays of various configurations. However, no theoretical framework for assessing the quality of these array configurations has been presented. In the current paper such a measure is pro...

The human auditory system is capable of performing various tasks related to spatial hearing including sound localization and source segregation. The performance of these tasks depends on many factors, including the complexity of the sound field; the way in which the information from the sound field is transferred to the ears; and the ability of the...

One of the uses of sensor arrays is for spatial filtering or beamforming. Current digital signal processing methods facilitate complex-weighted beamforming, providing flexibility in array design. Previous studies proposed the use of real-valued beamforming weights, which although reduce flexibility in design, may provide a range of benefits, e.g.,...

Head-related transfer functions (HRTFs) are widely used in the study of the human auditory system. It is known that HRTFs contain spatial cues allowing individuals to localize sounds. Inter-aural phase and level differences, and high-frequency magnitude variations, are among the most common cues. While these cues require detailed analysis of specif...

Microphone arrays are commonly used for spatial filtering or beamforming. A variety of beamforming techniques offering high spatial resolution have been developed and are described in the literature. However, improved resolution may require an increase in the number of sensors and channels, resulting in complicated and expensive systems. In the cur...

Directional microphones are widely used in spatial filtering applications. The directivity and steering capabilities are usually realized using arrays of sensors connected to a digital processor. In general, these techniques are very complicated and expensive. In this work a continuous-time electronic steering technique for microphone array based o...

Intreduction and Background Experimental ProceduresResults and DiscussionSummary and Conclusions Acknowledgment