Patent US2220612A successfully refreshed from Google Patents.
Patent US5185805A successfully refreshed from Google Patents.
US5185805A
Expired / Lapsed
US
⏱️ 0.0 years remaining
Tuned deconvolution digital filter for elimination of loudspeaker output blurring
Assignee: Individual
•
Filing: 1990-12-17
•
Granted: 1993-02-09
•
Expires: 2010-12-17
Abstract
Abstract A FIR (finite impulse response) type digital filter operates on digital audio signals in modern sound reproduction systems. It is shown that this operation forces the loudspeaker to produce a sound pressure wave having the original signal waveform. Given a multi-driver speaker, its response to a known broad band analog signal (impulsive) is sampled at least as fast as the Nyquist rate. The result is used to construct a deconvolution filter which compacts, in the least-squares sense, the blurred signal (speaker output) back into its original waveform. Since this anti-blurring process is linear and time invariant, it can be applied to the speaker driving signal as a blur preventive. A fine-tuning procedure utilizing Lagrange's Method of Multipliers modifies the deconvolution process such that the blur-free speaker output achieves a degree of flatness in frequency response beyond what could be attained with a simple deconvolution filter.
Inventors
- 👤 David Chiang
Bibliographic Details
Application Number
—
Jurisdiction
US
Data Source
google_patents
Last Synchronized
2026-11-06 01:33
CPC Classifications
H; H04; H04R; H04R3/00; H04R3/04
Claims Analysis
Total Claims: 5
Claim 1
Independent Claim
161 words
Method of making a finite impulse response filter for deconvolving audio signals to be converted by a given speaker to sound pressure waves comprising the steps of: providing a digital multiplier-accumulator having digital multiplicand inputs for receiving digitized audio signals, M+1 digital multiplier imputs for receiving filter coefficients (h i ; i=0,1, . . . M) and digital outputs for transmitting digitized deconvolved audio signals; generating the digital band-limited impulse response y i , i=0,1, . . . N, by driving the said speaker with the signal sin 2πf h t/2πf h t, wherein the frequency f h is the upper limit of the hearing range, measuring the acoustic output by a microphone and converting to digital data with sampling rate 1/T≧2f h ; calculating, from the values y i , i=0,1, . . . N, the set of coefficients h i , i=0,1, . . . M; and applying said set of coefficients h i to said digital multiplier inputs.
Claim 2
Dependent Claim
161 words
Method of making a finite impulse response filter for deconvolving audio signals to be converted by a given speaker to sound pressure waves comprising the steps of: providing a digital multiplier-accumulator having digital multiplicand inputs for receiving digitized audio signals, M+1 digital multiplier imputs for receiving filter coefficients (h i ; i=0,1, . . . M) and digital outputs for transmitting digitized deconvolved audio signals; generating the digital band-limited impulse response y i , i=0,1, . . . N, by driving the said speaker with the signal sin 2πf h t/2πf h t, wherein the frequency f h is the upper limit of the hearing range, measuring the acoustic output by a microphone and converting to digital data with sampling rate 1/T≧2f h ; calculating, from the values y i , i=0,1, . . . N, the set of coefficients h i , i=0,1, . . . M; and applying said set of coefficients h i to said digital multiplier inputs.
Claim 3
Dependent Claim
85 words
The method of claim 1 wherein said calculating step includes solving the matrix equation [h]=[R].sup.-1 [Y].sup.T [x] where [h]=COL [h 0 , h 1 , . . . h M ] is the filter coefficients [x]=COL [x 0 , x 1 , . . . x N+M ] is the delayed idealized FIR [Y] is the N+M+1 by M+1 matrix formed with the measured speaker impulse (band-limited) response y i , i=0, 1, . . . N [R]=[Y] T [Y] is the sampled autocorrelation matrix
Claim 4
Dependent Claim
41 words
The method of claim 1 further comprising the step of comparing filter performances for different values of delay associated with said vector [x] and selection of an optimum lag D opt which yields the maximally flat response in the frequency domain.
Claim 5
Dependent Claim
79 words
The method of claim 3 further comprising the step of fine tuning the coefficients, and therefore further flattening the speaker frequency response, by solving the matrix equation [h']=[R'].sup.-1 [Y].sup.T [x] where [h']=COL [h' 0 , h' 1 , . . . h' M ] is the improved coefficients [R']=[Y] T [U][Y] is the tuned sampled autocorrelation matrix [U] is the (N+M+1, N+M+1) tuning matrix constructed for the purpose of tuning out the remaining irregularities caused by finite filter length.
References Cited (Backward Citations)
US4888811A
Cited
USPTO Annuity & Maintenance Windows
| Annuity Stage | Window Opens | Payment Deadline | Surcharge Period Until | Estimated Fee | Status |
|---|---|---|---|---|---|
| 3.5 Years (Year 4) | 1996-02-09 | 1996-08-10 | 1997-02-09 | $2000.00 | Window Closed / Lapsed |
| 7.5 Years (Year 8) | 2000-02-09 | 2000-08-10 | 2001-02-09 | $3760.00 | Window Closed / Lapsed |
| 11.5 Years (Year 12) | 2004-02-09 | 2004-08-10 | 2005-02-09 | $7700.00 | Window Closed / Lapsed |