Device and a method for determining a component signal with high accuracy

A method of determining a component signal that is suitable for a wave field synthesis system comprising an array of loudspeakers, the wave field synthesis system being configured to exploit an audio signal that is associated with a virtual source and that exists as a discrete signal sampled at an audio sampling frequency, and a source position associated with the virtual source, so as to calculate component signals for the loudspeakers on the basis of the virtual source while taking into account loudspeaker positions of loudspeakers of the array of loudspeakers, the method comprising:

• providing wave field synthesis parameters, which comprise delay values, for the component signal to a loudspeaker of the array of loudspeakers while using the source position and while using a loudspeaker position of the loudspeaker of the array of loudspeakers at a parameter sampling frequency smaller than the audio sampling frequency, the wave field synthesis parameters being delay values;
• interpolating the wave field synthesis parameters so as to produce interpolated wave field synthesis parameters which are present at a parameter interpolation frequency that is higher than the parameter sampling frequency, the interpolated wave field synthesis parameters comprising integer portions of delay values for the component signal and interpolated fractions of delay values for the component signal, said interpolated fractions constituting delays which define fractions of sample intervals of the audio signal;
• and
• processing the audio signal so as to apply the interpolated fractions to the audio signal such that the component signal is calculated with fraction delays which correspond to the interpolated fractions,
• wherein the processing the audio signal comprises:
• processing the audio signal in subfilters, so that each subfilter produces an output signal;
• storing the output signals of the subfilters within a buffer;
• reading out the output signals from a position of the buffer which corresponds to the integer portions of the delay values;
• determining an interpolated value by calculating a polynomial in the interpolated fractions so that a component signal is acquired from the interpolated fractions of the delay values and of the read out output signals of the subfilters.

A non-transitory computer readable medium including a computer program comprising program code for performing, when the program is executed by a computer, the method of determining a component signal that is suitable for a wave field synthesis system comprising an array of loudspeakers, the wave field synthesis system being configured to exploit an audio signal that is associated with a virtual source and that exists as a discrete signal sampled at an audio sampling frequency, and a source position associated with the virtual source, so as to calculate component signals for the loudspeakers on the basis of the virtual source while taking into account loudspeaker positions of loudspeakers of the array of loudspeakers, the method comprising:

• providing wave field synthesis parameters, which comprise delay values, for the component signal to a loudspeaker of the array of loudspeakers while using the source position and while using a loudspeaker position of the loudspeaker of the array of loudspeakers at a parameter sampling frequency smaller than the audio sampling frequency, the wave field synthesis parameters being delay values;
• interpolating the wave field synthesis parameters so as to produce interpolated wave field synthesis parameters which are present at a parameter interpolation frequency that is higher than the parameter sampling frequency, the interpolated wave field synthesis parameters comprising integer portions of delay values for the component signal and interpolated fractions of delay values for the component signal, said interpolated fractions constituting delays which define fractions of sample intervals of the audio signal;
• and
• processing the audio signal so as to apply the interpolated fractions to the audio signal such that the component signal is calculated with fraction delays which correspond to the interpolated fractions,
• wherein the processing the audio signal comprises:
• processing the audio signal in subfilters, so that each subfilter produces an output signal;
• storing the output signals of the subfilters within a buffer;
• reading out the output signals from a position in the buffer which corresponds to the integer portions of the delay values;
• determining an interpolated value by calculating a polynomial in the interpolated fractions so that a component signal is acquired from the interpolated fractions of the delay values and of the read out output signals of the subfilters.