In listening room compensation, the aim is to compensate for the degradations that are rendered to an audio signal by transmission in a closed room. Due to multiple reflections of the soundwaves, the listener receives a superposition of delayed and attenuated versions of the source signal. A filter is designed so that the convolution of the room impulse response and the equalizer contains better acoustic properties than the original acoustic channel. For dereverberation, the filter is usually designed by optimizing a solely time-domain based cost function and, therefore, may introduce spectral distortions. Recent methods also consider the frequency-domain representation of the overall system and aim at yielding a flat overall frequency response. However, in some cases, it may be desirable to follow a predefined frequency response rather than obtaining a flat one. Hearing impaired persons, for example, may prefer or even need an amplification of certain frequency ranges. In this work, we propose a new method to jointly shape the time- and the frequency-domain representations of the overall acoustic channel according to prescribed curves. Furthermore, we integrate the concept of auditory scales into the filter design.
|Titel||2015 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)|
|Publikationsstatus||Veröffentlicht - 01.04.2015|
|Veranstaltung||40th IEEE International Conference on Acoustics, Speech, and Signal Processing - Brisbane, Australien|
Dauer: 19.04.2015 → 24.04.2015