Creating digital signal processing (DSP) filters to improve loudspeaker transient response
View Patent ↗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.
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.