Time domain compensation for transducer mismatch

• US 5,224,170 A
• Filed: 04/15/1991
• Issued: 06/29/1993
• Est. Priority Date: 04/15/1991
• Status: Expired due to Term

• Current Assignee: Agilent Technologies, Inc.
• Original Assignee: Hewlett-Packard Company (HP Inc.)
• Inventors: Waite, James W. Jr.
• Primary Examiner(s): Ng, Jin F.
• Assistant Examiner(s): Lefkowitz, Edward
• Application Number:
  US07/685,370
• Time in Patent Office: 806 Days
• Field of Search: 381/92, 381/96, 381/97, 364/553, 364/724.19
• US Class Current: 381/92
• CPC Class Codes: H04R 29/006 Microphone matching

Abstract

A two channel real time octave analyzer is equipped with an adaptive time domain phase compensation filter whose poles and zeros are programmably selected to counteract phase error associated with any given pair of microphones. Precise matching of microphone phase characteristics is thereby achieved. In the preferred embodiment, the transfer function for the adaptive filter is determined using a pole/zero synthesis technique based on cross spectrum data acquired in an FFT spectral analysis of the probe microphones.

Claims

1. In a method of time domain spectral analysis, an improvement comprising the steps:

determining an error characteristic sought to be compensated;

synthesizing a pole/zero transfer function approximating said error characteristic; and

programming a time domain digital filter in accordance with said synthesized transfer function to compensate for the determined error characteristic.

2. A method of time domain spectral analysis according to claim 1 in which the transfer characteristic sought to be compensated is a phase error function associated with first and second measurement channels of a spectral analysis instrument, and in which the determining step comprises determining a phase error between said first and second channels.

3. The method of claim 2 in which the determining step includes switching the measurement channels to determine a phase error associated therewith.

4. The method of claim 2 in which the determining step further includes performing an FFT analysis of signals on the two channels.

5. The method of claim 1 in which the programming step comprises programming an infinite impulse response filter.

6. The method of claim 1 which further includes constraining a pole of the transfer characteristic to be located at DC.

7. The method of claim 1 in which the transfer characteristic sought to be compensated is a phase error function, and in which the time domain digital filter is programmed to correct for said phase error function.

8. A method of determining sound intensity comprising the steps:

determining a phase error function associated with signals from first and second microphones;

processing an output signal from the first microphone through a first series of bandpass filter stages to determine the spectral composition thereof, said processing including digital filtering with a first filter stage at a first sample rate and digital filtering with a second, subsequent filter stage at a second sample rate lower than the first;

processing an output signal from the second microphone through a second series of bandpass filters to determine the spectral composition thereof; and

analyzing output signals from said first and second series of bandpass filters to determine sound intensity;

wherein at least the first of said processing steps further includes time domain digital filtering at a sample rate less than the first sample rate to correct for said determined phase error function so that the sound intensity determination resulting from the analyzing step is compensated for phase error without the need for post-analysis operation and at reduced computational expense due to the lower sampling rate.

9. The method of claim 8 in which the time domain filtering corrects for phase error function below 50 hertz.

10. The method of claim 8 in which the time domain filtering comprises filtering with an infinite impulse response filter.

11. The method of claim 8 which further includes:

synthesizing a transfer function approximating the conjugate of the determined error function; and

programming a digital filter to implement said synthesized transfer function.

12. The method of claim 11 which further includes constraining a pole of the synthesized transfer function to near DC when synthesizing said transfer function.

13. The method of claim 12 in which the time domain filtering corrects for phase error function below 50 hertz.

14. A method of determining sound intensity comprising the steps:

determining a phase error function associated with signals from first and second microphones;

processing an output signal from the first microphone through a first series of bandpass filters to determine the spectral composition thereof;

processing an output signal from the second microphone through a second series of bandpass filters to determine the spectral composition thereof; and

analyzing output signals from said first and second series of bandpass filters to determine sound intensity;

wherein at least one of said processing steps further includes time domain filtering to correct for said determined phase error function so that the sound intensity determination resulting from the analyzing step is compensated for phase error without the need for post-analysis operation, the method further including;

using a single measurement instrument;

perform FFT analyses of signals from the first and second microphones;

determine from said FFT analyses phase error function associated with said microphone signals;

synthesize a transfer function approximating the determined error function;

program a time domain digital filter to implement said synthesized transfer function; and

compensate a digital data stream associated with the first microphone using said time domain digital filter.

15. The method of claim 14 which further includes using said same measurement instrument to perform FFT analyses of cross spectra using the microphones in reference and switched states, and determining from said cross spectra a phase error function associated with the microphone signals.

16. The method of claim 14 in which the time domain filtering corrects for phase error function below 50 hertz.

17. A real time octave analyzer having at least two measurement channels, each measurement channel comprising:

a source of input signals;

an anti-alias filter having an input coupled to the source of input signals;

an analog-to-digital converter having an input coupled to an output of the anti-alias filter;

a plurality of successive analysis stages, each having an input, an output, and at least one digital bandpass filter associated therewith, the digital bandpass filter producing an output signal responsive to sampled data provided thereto;

the input of a first analysis stage being coupled to an output of the analog-to-digital filter, the first analysis stage including a digital bandpass filter operating at a first sample rate;

the input of each successive analysis stage being coupled to an output of a previous analysis stage;

one of said measurement channels further including means for reducing the sample rate of data applied to subsequent analysis stages in said channel;

said one of said measurement channels further including a time domain digital filter operating on data at a sample rate less than the first sample rate, said time domain digital filter being adapted to compensate for a phase error associated with the source of input signals.

18. The analyzer of claim 17 in which the time domain digital filter is an infinite impulse response filter.

19. The analyzer of claim 17 in which the time domain digital filter is programmable so that it may be adapted to compensate for phase errors associated with a variety of input signal sources.

20. The analyzer of claim 17 which further includes FFT analysis means for characterizing the phase error associated with the source of input signals.

21. A real time octave analyzer having at least two measurement channels, each measurement channel comprising:

a source of input signals;

an anti-alias filter having an input coupled to the source of input signals;

an analog-to-digital converter having an input coupled to an output of the anti-alias filter;

a plurality of successive analysis stages, each having an input, an output, and at least one bandpass filter associated therewith;

the input of a first analysis stage being coupled to an output of the analog-to-digital filter;

the input of each successive analysis stage being coupled to an output of a previous analysis stage;

one of said measurement channels further including a time domain digital filter adapted to compensate for a phase error associated with the source of input signals;

wherein the time domain digital filter has a pole near DC.

22. A real time octave analyzer having at least two measurement channels, each measurement channel comprising:

a source of input signals;

an anti-alias filter having an input coupled to the source of input signals;

an analog-to-digital converter having an input coupled to an output of the anti-alias filter;

a plurality of successive analysis stages, each having an input, an output, and at least one bandpass filter associated therewith;

the input of a first analysis stage being coupled to an output of the analog-to-digital converter;

the input of each successive analysis stage being coupled to an output of a previous analysis stage;

one of said measurement channels further including a time domain infinite impulse response filter adapted to compensate for a phase error associated with the source of input signals.

23. A method of determining sound intensity comprising the steps:

determining a phase error function associated with signals from first and second microphones;

approximating the determined phase error function with a curve fitter;

determining pole and zero locations of a real time digital filter in accordance with said approximated phase error function;

processing an output signal from the first microphone through a first series of bandpass filters to determine the spectral composition thereof;

processing an output signal from the second microphone through a second series of bandpass filters to determine the spectral composition thereof; and

analyzing output signals from said first and second series of bandpass filters to determine sound intensity;

wherein at least one of said processing steps further includes time domain filtering with a filter implemented in accordance with the determined pole and zero locations to correct for said determined phase error function.

24. A real time octave analyzer comprising a plurality of cascaded analysis stages, each of which includes a decimation stage, the analyzer further including a programmable time domain phase compensation filter interposed amidst said analysis stages.