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Speech enhancement under non-stationary environments is still a challenging problem. This contribution presents a noise reduction system that is capable of tracking and suppressing both time varying harmonic noise and stationary noise. In a first stage, the harmonic noise power is estimated and attenuated using a modified Minimum Statistics approach that performs frequency warping according to the...
This contribution presents a wideband (50 Hz-7 kHz) speech enhancement system that is operating in the frequency domain. As a novel feature, techniques known from artificial bandwidth extension (BWE) are used to improve the spectral estimation process by exploiting the statistical dependencies between the low band (50Hz - 4kHz) and the high band (4-7kHz). Conventional noise suppression is used in...
The ability of the human auditory system for sound localization mainly depends on the binaural cues, especially interaural time and level differences (ITD and ILD). In the context of digital hearing aids and binaural audio transmission systems, these cues can be severely degraded by independent bilateral signal processing such as dereverberation or noise reduction. This contribution presents a novel...
Noise reduction techniques that are relying on spectral weighting rules often generate annoying musical noise artifacts in the processed signal. In this paper, we present a postfilter (PF) for the spectral weighting gains that is capable of reducing musical noise in a simple but efficient way. It includes a robust detector for speech pauses and low SNR conditions and adaptively smoothes the weighting...
In contrast to common noise reduction systems, this contribution presents a digital signal processing algorithm to improve intelligibility of clean far end speech for the near end listener who is located in an environment with background noise. Since the noise reaches the ears of the near end listener directly and therefore can hardly be influenced, a sensible option is to manipulate the far end speech...
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