System and method for compensating memoryless non-linear distortion of an audio transducer

1. A method of compensating digital audio samples d(n) of a digital audio signal for an audio transducer, comprising:

storing a lookup table (LUT) for the audio transducer in memory, said LUT including scale factors of the transducer's memoryless nonlinear distortion over a phase plane indexed by sample amplitude, velocity pairs,

measuring an amplitude a(n) of the digital audio signal for each digital audio sample d(n);

estimating a velocity v(n) of the digital audio signal for each digital audio sample d(n);

for each digital audio sample d(n), using the amplitude, velocity pair (a(n),v(n)) to extract a scale factor from the LUT; and

scaling the amplitude a(n) of each digital audio sample d(n) by the extracted scale factor.

2. The method of claim 1 , wherein the amplitude a(n) of each said sample d(n) is scaled by one said extracted scale factor.

3. The method of claim 1 , further comprising extracting a plurality of scale factors closest to the (a(n),v(n)) and performing an interpolation on said plurality to produce the scale factor for the measured (a(n),v(n)) pair.

4. The method of claim 1 , wherein each scale factor is determined by ratio of the amplitude of a test signal s(n) applied to the audio transducer and the amplitude of a recorded signal r(n) reproduced by the audio transducer.

5. The method of claim 4 , wherein said LUT is indexed by the amplitude, velocity pair of the test signal, said digital audio signal being scaled by the scale factor to pre-compensate the digital audio signal.

6. The method of claim 5 , wherein the audio transducer is an earphone, further comprising:

playback of the pre-compensated digital audio signal on the earphone.

7. The method of claim 4 , wherein said LUT is indexed by the amplitude, velocity pair of the recorded signal, said digital audio signal being scaled by the scale factor to post-compensate the audio signal.

8. The method of claim 1 , wherein the digital audio samples d(n) are downsampled to a low-frequency band where the scale factor is extracted and the samples scaled and then upsampled to the full frequency band.

9. A method of compensating digital audio samples d(n) of a digital audio signal for an audio transducer, comprising:

measuring an amplitude a(n) of the digital audio signal for each digital audio sample d(n);

estimating a velocity v(n) of the digital audio signal for each digital audio sample d(n);

using the amplitude, velocity pair (a(n),v(n)) to extract a scale factor from a phase plane representation of the audio transducer, said phase plane representation embodying scale factors of the transducer's memoryless nonlinear distortion over the phase plane as a function of amplitude and velocity, wherein the phase plane representation is a polynomial equation whose only independent variables are the measured signal amplitude a(n) and signal velocity v(n); and

scaling the amplitude a(n) of digital audio signal by the scale factor.

10. A system for compensating digital audio samples d(n) of a digital audio signal for an audio transducer, comprising:

memory for storing a lookup table (LUT) for the audio transducer, said LUT including scale factors of the transducer's memoryless nonlinear distortion over the phase plane indexed by sample amplitude, velocity pairs; and

a processor that measures an amplitude a(n) of the digital audio signal each digital audio sample d(n), estimates a velocity v(n) of the digital signal for each digital audio sample d(n), extracts a scale factor from the LUT using the measured a(n), v(n) pair, and scales the amplitude a(n) of the digital audio sample d(n) by the scale factor.

11. The system of claim 10 , wherein the processor scales the amplitude a(n) of each said sample d(n) by one said scale factor.

12. The system of claim 10 , wherein the processor extracts a plurality of scale factors closest to the measured (a(n),v(n)) pair and performs an interpolation on said plurality to produce the scale factor for the measured (a(n),v(n)) pair.

13. The system of claim 10 , wherein each scale factor is determined by a ratio of the amplitude of a test signal s(n) applied to the audio transducer and the amplitude of a recorded signal r(n) reproduced by the audio transducer.

14. The system of claim 13 , wherein said LUT is indexed by the amplitude, velocity pair of the test signal, said digital audio signal being scaled by the scale factor to pre-compensate the audio signal.

15. The system of claim 14 , wherein the audio transducer is an earphone, said processor directing the pre-compensated digital audio signal for playback on the earphone.

16. The system of claim 13 , wherein said LUT is indexed by the amplitude, velocity pair of the recorded signal, said digital audio signal being scaled by the scale factor to post-compensate the audio signal.

17. The system of claim 10 , wherein the processor downsamples the digital audio samples d(n) to a low-frequency band where the scale factor is extracted and the samples scaled and then up samples the scaled samples to the full frequency band.

18. A system for compensating digital audio samples d(n) of a digital audio signal for an audio transducer, comprising:

memory for storing a phase plane representation of the audio transducer, said phase plane representation embodying scale factors of the transducer's memoryless nonlinear distortion over the phase plane as a function of amplitude and velocity, wherein the phase plane representation is a polynomial equation whose only independent variables are the measured signal amplitude and signal velocity; and

a processor that measures an amplitude a(n) of the digital audio signal for each digital audio sample d(n), estimates a velocity v(n) of the digital audio signal for each digital audio sample d(n), extracts a scale factor from the phase plane representation using the measured a(n), v(n) pair, and scales the amplitude a(n) of the digital audio signal by the scale factor.

19. A method of determining a phase plane representation of scale factors for compensating memoryless nonlinear distortion of an audio transducer, comprising:

synchronized playback and recording of a test signal through the audio transducer; and

storing a ratio of the test signal amplitude s(n) to the recorded signal amplitude r(n) as a scale factor in a lookup table (LUT) indexed by a signal amplitude, signal velocity pair.

20. The method of claim 19 , wherein the amplitude and velocity of the test signal spans at least a desired range of the phase plane.

21. The method of claim 20 , wherein the test signal comprises first and second sine waves with changing frequency and amplitude.

22. The method of claim 19 , further comprising extrapolating the scale factors in the LUT to cover the entire phase plane.

23. The method of claim of claim 19 , further comprising interpolating and resembling the scale factor in the LUT to a desired amplitude, velocity indexing.

24. The method of claim 19 , wherein the LUT is indexed by the amplitude, velocity pair of the test signal for use in pre-compensating an audio signal for playback on an audio transducer.

25. The method of claim 19 , wherein the LUT is indexed by the amplitude, velocity pair of the recorded signal for use in post-compensating an audio signal reconstructed from an audio transducer.

26. The method of claim 19 , further comprising:

approximating the LUT with a polynomial equation whose only independent variables are the signal amplitude and signal velocity.