Summary
Researchers at the Centre for Communications Systems Research, University of Surrey, have developed a novel method for isolating multiple sound sources in a noisy environment. Sound sources can be individually separated, emphasized, suppressed, or modified and then recombined in any 3D spatial configuration. All processing is done in real-time and no prior knowledge of the number or location of the sources is required.
Benefits
The intensity vector method supports numerous important advantages over conventional BSS techniques.
Feature
Conventional BSS techniques
Intensity vector analysis
Number of sources
The number of sound sources is limited to the number of microphones.
An infinite number of directivity functions can be calculated to separate more sources than microphones, although the performance would be limited. For large number of sources, it may be more practical to use fixed directivity functions for each window, as their calculation would be computationally demanding.
Moving sources
Independent component analysis requires the sound sources to be stationary.
Real time separation is achievable within 25 msec. Thus the system can lock-on to moving sound sources.
Compactness
The accuracy of time-delay-of-arrival techniques generally increases with the size of the microphone array.
The physical separation of microphones in the array must be small compared to the acoustic wavelength in air. Source separation performance improves with smaller microphone arrays, such as those manufactured using MEMS.
Applications
Availability
Available for licence
IP Status
Patent Pending
Technical
Blind source separation (BSS) is performed using acoustic pressure gradients derived from a small array of condenser microphones, or obtained directly from commercial B-format tetrahedral microphones. Time-frequency representations of the pressure and pressure gradient signals are calculated using a modified discrete cosine transform or fast Fourier transform. These are used to derive intensity vector directions. Beamforming is applied using a directivity function defined for each sound source and time-frequency bin. Finally, individual time-domain signals are obtained using an inverse modified cosine transform or inverse fast Fourier transform.
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