Phase Inversion Filter for Correcting Low Frequency Phase Distortion in a Loudspeaker System

Abstract

A filter for correcting phase distortion produced at low frequencies in a loudspeaker system is created by inverting the phase response of the determined complex-valued frequency response of a loudspeaker system. The inverted phase response is obtained by taking the complex conjugate of the phase response. The impulse response for the inverted phase response is obtained by means of an inverse Fourier transform of the inverted phase response. The impulse response provides a linear phase FIR filter having a long filter length. The linear phase FIR filter is applied to the audio signal input to the loudspeaker system. Prior to inverting the phase response, the determined complex-valued frequency response of a loudspeaker system can be subjected to high frequency blanking and polynomial smoothing. Also, the linear phase FIR filter can be subjected to a window function prior to applying the filter to the audio signal.

What we claim is: 
     
         1 . A method of creating a filter for correcting phase distortion produced at low frequencies in a loudspeaker system having a transducer driven by a piston with mass, comprising:
 a. obtaining the complex-valued frequency response of the loudspeaker system, said frequency response including a magnitude component (the magnitude response) and a phase component (the phase response) and having a number of data points for producing a relatively high resolution representation of the frequency response of the loudspeaker system,   b. inverting the phase response obtained in step (a) by taking the complex conjugate of the phase response to produce an inverted phase response,   c. obtaining the impulse response for the inverted phase response by means of an inverse Fourier transform of the inverted phase response, wherein said impulse response is a FIR filter having a long filter length characterized by a series of RR coefficients, and   d. applying the FIR filter o the audio signal input to the loudspeaker system for which the frequency response was obtained in step (a).   
     
     
         2 . The method of  claim 1  wherein the complex-valued frequency response of the loudspeaker system is measured under free-field conditions. 
     
     
         3 . The method of  claim 1  wherein the phase of the complex-valued frequency response of the loudspeaker system is set to zero above a high frequency cut-off point to create a high frequency blanked phase trace, and wherein said cut-off point is located at a frequency below which the obtained phase response of the loudspeaker system begins to continuously move away from zero degrees. 
     
     
         4 . The method of  claim 3  wherein a second stage smoothing function is applied to the phase response prior to inverting the phase response, thereby creating a smooth polynomial approximation of the high frequency blanked phase response. 
     
     
         5 . The method of  claim 4  wherein said second stage smoothing function is a least-square smoothing spline algorithm applied to the high frequency blanked phase response. 
     
     
         6 . The method of  claim 1  wherein a second stage smoothing function is applied to the phase response prior to inverting the phase response, thereby creating a smooth polynomial approximation of the phase response. 
     
     
         7 . The method of  claim 6  wherein said second stage smoothing function is a least-square smoothing spline algorithm applied to the phase response. 
     
     
         8 . The method of  claim 1  wherein a window function is applied to the FIR filter obtained in step (c) of  claim 1  to remove audible distortions produced by the FIR filter. 
     
     
         9 . The method of  claim 8  wherein a Kaiser window is applied to said FIR filter. 
     
     
         10 . The method of  claim 8  wherein pre-correction is iteratively applied to said window function to remove side effects, wherein the side effects are a low frequency attenuation in the magnitude response of the FIR filter. 
     
     
         11 . The method of  claim 1  wherein said FIR filter is modified such that it is a symmetric filter. 
     
     
         12 . The method of  claim 1  wherein the length of said FIR filter is between 5,000 and 50,000 points. 
     
     
         13 . A method of creating digital filter for correcting phase distortion produced at low frequencies in a loudspeaker system having a transducer driven by a piston with mass, comprising:
 a, obtaining the complex-valued frequency response of the loudspeaker system, said frequency response including a magnitude component (the magnitude response) and a phase component (the phase response), and having a number of data points for producing a relatively high resolution representation of the frequency response of the loudspeaker system,   b. inverting the phase response obtained in step (a) by taking the complex conjugate of the phase response to produce an inverted phase response,   c. obtaining the impulse response for the inverted phase response by means of an inverse Fourier transform of the inverted phase response, wherein said impulse response is a FIR filter having a long filter length that depends on a love frequency cut-off that is selected and is characterized by a series of FIR coefficients,   d. applying a symmetric window function to the symmetric linear phase FIR filter to force the FIR coefficients to decay to zero by the end of the FIR filter length,   e. adding pre-correction to the windowed symmetric linear phase FIR filter to correct for magnitude attenuation introduced by step (d) at low frequencies, and   f. applying the pre-corrected windowed FIR filter to the audio signal input to the loudspeaker system for which the frequency response was obtained in step (a).   
     
     
         14 . The method of  claim 13  wherein in step (a) the complex-valued frequency response of the loudspeaker system is measured under free-field conditions. 
     
     
         15 . The method of  claim 13  wherein a second stage smoothing function is applied to the phase response prior to inverting the phase response, thereby creating a smooth polynomial approximation of the phase response. 
     
     
         16 . The method of  claim 15  wherein said second stage smoothing function is a least-square smoothing spline algorithm applied to the phase response. 
     
     
         17 . The method of  claim 16  wherein, prior to applying a smoothing function, the phase response is interpolated on a logarithmic frequency scale so that the information per octave is constant across the operating range of the loudspeaker system. 
     
     
         18 . A filter for correcting phase distortion produced at low frequencies in a loudspeaker system, said filter being created in accordance with the method of  claim 1 , wherein the filter is applied to the audio signal in real time. 
     
     
         19 . A filter for correcting phase distortion produced at low frequencies in a loudspeaker system said filter being created in accordance with the method of  claim 1 , wherein the filter is applied to the audio signal off-line and placed on a storage medium prior to playback through the loudspeaker system. 
     
     
         20 . A filter for correcting phase distortion produced at low frequencies in a loudspeaker system, said filter being created in accordance with the method of  claim 13 , wherein the filter is applied to the audio signal off-line and placed on a storage medium prior to playback through the loud speaker system.