Compensating filters

• US 6,760,451 B1
• Filed: 05/22/1998
• Issued: 07/06/2004
• Est. Priority Date: 08/03/1993
• Status: Expired due to Fees

• Current Assignee: Michael Anthony Gerzon, Peter Graham Craven
• Original Assignee: Michael Anthony Gerzon, Peter Graham Craven
• Inventors: Craven, Peter Graham, Gerzon, Michael Anthony
• Primary Examiner(s): Mei, Xu
• Application Number:
  US09/083,276
• Time in Patent Office: 2,237 Days
• Field of Search: 381/58, 381/59, 381/97, 381/98, 381/101, 381/102, 381/103, 381/56
• US Class Current: 381/98
• CPC Class Codes: H03G 5/005 of digital signals
• CPC Class Codes: H03G 5/165 Equalizers; Volume or gain control in limited frequency bands
• CPC Class Codes: H03H 21/0012 Digital adaptive filters
• CPC Class Codes: H04R 29/001 for loudspeakers H04R29/007 takes precedence
• CPC Class Codes: H04R 3/04 for correcting frequency response
• CPC Class Codes: H04S 7/301 Automatic calibration of stereophonic sound system, e.g. with test microphone
• CPC Class Codes: H04S 7/305 Electronic adaptation of stereophonic audio signals to reverberation of the listening space H04S7/301 takes precedence
• CPC Class Codes: H04S 7/307 Frequency adjustment, e.g. tone control H04S7/301 takes precedence
• CPC Class Codes: H04S 7/40 Visual indication of stereophonic sound image

Abstract

A prefilter (5) for an audio system comprising a loudspeaker (1) in a room (2), which corrects both amplitude and phase errors due to the loudspeaker (1) by a linear phase correction filter response and corrects the amplitude response of the room (2) whilst introducing the minimum possible amount of extra phase distortion by employing a minimum phase correction filter stage. A test signal generator (8) generates a signal comprising a periodic frequency sweep with a greater phase repetition period than the frequency repetition period. A microphone (7) positioned at various points in the room (2) measures the audio signal processed by the room (2) and loudspeaker (1), and a coefficient calculator (6) (e.g. a digital signal processor device) derives the signal response of the room and thereby a requisite minimum phase correction to be cascaded with the linear phase correction already calculated for the loudspeaker (1). Filter (5) may comprise the same digital signal processor as the coefficient calculator (6). Applications in high fidelity audio reproduction, and in car stereo reproduction.

Claims

1. A method of processing an audio signal, said method comprising the steps of:

executing at least one spectral transform operation to derive a signal spectrum,

processing the spectrum to effect smoothing thereof, said smoothing being variable with frequency across the signal spectrum; and

conditioning the response of a filter for use in a subsequent audio reproduction with said smoothed spectrum.

2. A method according to claim 1, wherein a first spectral transform operation is performed prior to processing the corresponding signal spectrum to effect smoothing thereof, and a second, inverse, spectral transform operation is then performed on the smoothed spectrum.

3. A method according to claim 1, wherein said executing step comprises a first spectral transform operation deriving the corresponding signal spectrum and a second, inverse, spectral transform operation is performed on the signal spectrum, prior to processing the corresponding inverse spectrum to effect smoothing thereof.

4. A method according to claim 1, wherein the audio signal carries information relating to the response of an acoustic environment to a test signal.

5. A method according to claim 1, wherein the degree of smoothing increases with increasing frequency.

6. A method according to claim 5, wherein the degree of smoothing varies continuously over at least a portion of the signal spectrum.

7. A method according to claim 1, further comprising the step of using the smoothed spectrum to condition the response of a filter for use in subsequent audio reproduction in the acoustic environment.

8. A method according to claim 1, wherein said filter is a digital finite impulse response filter.

9. An acoustic environment audio correction system, comprising:

a test signal generator;

a measurement microphone;

a conditionable correction filter; and

processing means for deriving a representation of the response of the acoustic environment at a plurality of frequencies in dependence on a test signal generated by the test signal generator and transmitted within the acoustic environment, for processing the representation of the response to effect smoothing, said smoothing being variable with frequency, and subsequently for conditioning the correction filter in dependence on the smoothed representation of the response.

10. An acoustic environment audio correction system according to claim 9 further including a memory for storing a signal received by the measurement microphone in dependence on the test signal generated by the test signal generator and transmitted within the acoustic environment.

11. An acoustic environment audio correction system according to claim 9 wherein the conditionable correction filter is a digital finite impulse response filter.

12. An acoustic environment audio correction system according to claim 9 wherein the test signal generator is arranged to supply an electrical test signal to an input of a loudspeaker.

13. An acoustic environment audio correction system according to claim 9, wherein said processing means performs a first spectral transform operation prior to processing the corresponding signal spectrum to effect smoothing thereof, and subsequently performs a second, inverse, spectral transform operation on the smoothed spectrum.

14. An acoustic environment audio correction system according to claim 9, wherein said processing means performs a first spectral transform operation deriving the corresponding signal spectrum and a second, inverse, spectral transform operation on the signal spectrum, prior to processing the corresponding inverse spectrum to effect smoothing thereof.