Creating digital signal processing (DSP) filters to improve loudspeaker transient response

• US 8,081,766 B2
• Filed: 03/06/2006
• Issued: 12/20/2011
• Est. Priority Date: 03/06/2006
• Status: Active Grant

• Current Assignee: Eaw North America, Inc.
• Original Assignee: LOUD Technologies, Inc.
• Inventors: Gunness, David W.
• Primary Examiner(s): Mei, Xu
• Assistant Examiner(s): LAO, LUNSEE
• Application Number:
  US11/368,554
• Publication Number:
  US 20070223713A1
• Time in Patent Office: 2,115 Days
• Field of Search: 381/96, 381/58, 381/59, 381/97, 381/150, 381/337, 381/339, 381/340, 381/343, 381/98, 381/99, 181/144, 181/199, 181/198
• US Class Current: 381/59
• CPC Class Codes: H04R 29/003 of the moving-coil type
• CPC Class Codes: H04R 3/08 of electromagnetic transducers

Abstract

A method is provided for creating a series of digital signal processing (DSP) filters to improve the transient response of a loudspeaker, wherein the loudspeaker is formed of multiple components. The method includes generally six steps. The first step involves identifying a substantially linear, time-invariant, and spatially-consistent loudspeaker mechanism causing transient response distortion. The second step involves characterizing the identified mechanism. The third step involves determining the characterized mechanism'"'"'s two-port response. The fourth step involves establishing a target response for the characterized mechanism. The fifth step involves calculating an ideal filter to achieve the target response. The sixth step involves designing a cost-reduced filter based on the ideal filter.

Claims

1. A method of creating a digital signal processing (DSP) filter to improve the transient response of a loudspeaker, wherein the loudspeaker is formed of multiple components, the method comprising:

identifying a loudspeaker mechanism causing transient response distortion,

wherein the transient response distortion of the identified loudspeaker mechanism is substantially linear;

wherein the transient response distortion of the identified loudspeaker mechanism does not vary over time;

wherein the transient response distortion of the identified loudspeaker mechanism does not vary with respect to a direction away from the loudspeaker; and

wherein the identified loudspeaker mechanism includes a physical behavior of a loudspeaker component;

characterizing the identified mechanism by mathematically modeling the physical behavior of the loudspeaker component and by performing electrical impedance measurements of the loudspeaker component;

determining the characterized mechanism'"'"'s two-port response;

establishing a target response for the characterized mechanism;

calculating an ideal filter to achieve the target response; and

designing a cost-reduced filter based on the ideal filter to thereby form a loudspeaker mechanism algorithm (LMA) filter.

2. The method of claim 1, wherein the identified loudspeaker mechanism further has a limited amount of unit-to-unit variability.

3. The method of claim 1, wherein the identified loudspeaker mechanism is a non-minimum phase system.

4. The method of claim 1, wherein the identified loudspeaker mechanism is selected from a group consisting of:

transient smear due to a compression driver phase plug configuration, acoustical horn resonances, and mechanical radial resonances in loudspeaker cones.

5. The method of claim 1, wherein characterizing the identified mechanism includes performing mechanical or acoustical transfer function measurements.

6. The method of claim 1, wherein the step of determining the characterized mechanism'"'"'s two-port response comprises determining a frequency response of the mechanism.

7. The method of claim 1, wherein the step of determining the characterized mechanism'"'"'s two-port response comprises determining an impulse response of the mechanism.

8. The method of claim 1, wherein the step of designing a cost-reduced filter to achieve the target response comprises selectively employing one or more of a Finite Impulse Response (FIR) filter, Infinite Impulse Response (IIR) filter, and biquadratic (biquad) filter.

9. A method of creating a series of digital signal processing (DSP) filters to improve the transient response of a loudspeaker, wherein the loudspeaker is formed of multiple components, the method comprising:

(a) for each component;

(1) identifying a loudspeaker mechanism causing transient response distortion, wherein;

(i) the identified loudspeaker mechanism includes a physical behavior of the component;

(ii) the transient response distortion of the identified loudspeaker mechanism is substantially linear;

(iii) the transient response distortion of the identified loudspeaker mechanism does not vary over time; and

(iv) the transient response distortion of the identified loudspeaker mechanism does not vary with respect to a direction away from the loudspeaker;

(2) mathematically modeling the physical behavior of the component and performing electrical impedance measurements of the component to characterize the identified mechanism;

(3) determining the characterized mechanism'"'"'s two-port response;

(4) establishing a target response for the characterized mechanism;

,

(5) calculating an ideal filter to achieve the target response; and

(6) designing a cost-reduced filter based on the ideal filter to thereby form a loudspeaker mechanism algorithm (LMA) filter; and

(b) for all components;

(7) applying minimum phase filters to equalize multiple frequency ranges;

(8) applying linear phase crossover filters; and

(9) repeating any of the steps (1)-(8) above to achieve a combined loudspeaker response that exhibits reproduction accuracy.

10. The method of claim 9, further comprising, for each component, repeating steps (1)-(6) to form multiple LMA filters that each address a particular mechanism associated with the component.

11. The method of claim 9, wherein step (6) of designing a cost-reduced filter to achieve the target response comprises selectively employing one or more of a Finite Impulse Response (FIR) filter, Infinite Impulse Response (IIR) filter, and biquadratic (biquad) filter.

12. The method of claim 9, wherein step (7) of applying minimum phase filters comprises employing biquadratic (biquad) filters.

13. A loudspeaker system comprising multiple components and a series of digital signal processing (DSP) filters created to improve the loudspeaker'"'"'s transient response, wherein the DSP filters comprise loudspeaker mechanism algorithm (LMA) filters that are each configured to correct a loudspeaker mechanism causing transient response distortion;

wherein the transient response distortion of the identified loudspeaker mechanism is substantially linear;

wherein the transient response distortion of the identified loudspeaker mechanism does not vary over time;

wherein the transient response distortion of the identified loudspeaker mechanism does not vary with respect to a direction away from the loudspeaker; and

wherein each of the LMA filters is configured according to a method comprising;

characterizing the identified loudspeaker mechanism by mathematical modeling and electrical impedance measurements.

14. The loudspeaker system of claim 13, wherein at least one of the LMA filters is configured to correct a loudspeaker mechanism that has a limited amount of unit-to-unit variability.

15. The loudspeaker system of claim 13, wherein at least one of the LMA filters is configured to correct a loudspeaker mechanism that is a non-minimum phase system.

16. The loudspeaker system of claim 13, wherein at least one of the LMA filters is configured to correct a loudspeaker mechanism selected from a group consisting of:

transient smear due to a compression driver phase plug configuration, acoustical horn resonances, and mechanical radial resonances in loudspeaker cones.

17. The loudspeaker system of claim 13, wherein at least one of the LMA filters is configured based on one or more of a Finite Impulse Response (FIR) filter, Infinite Impulse Response (IIR) filter, and biquadratic (biquad) filter.

18. The loudspeaker system of claim 13, wherein the method by which each of the LMA filters is configured further comprises:

determining the characterized mechanism'"'"'s two-port response;

establishing a target response for the characterized mechanism;

calculating an ideal filter to achieve the target response; and

designing a cost-reduced filter based on the ideal filter to thereby form a loudspeaker mechanism algorithm (LMA) filter.