Tuned deconvolution digital filter for elimination of loudspeaker output blurring

• US 5,185,805 A
• Filed: 12/17/1990
• Issued: 02/09/1993
• Est. Priority Date: 12/17/1990
• Status: Expired due to Fees

• Current Assignee: David Chiang
• Original Assignee: David Chiang
• Inventors: Chiang, David
• Primary Examiner(s): Isen, Forester W.
• Assistant Examiner(s): TONG, NINA C
• Application Number:
  US07/628,635
• Time in Patent Office: 785 Days
• Field of Search: 381/96, 381/98, 381/59
• US Class Current: 381/96
• CPC Class Codes: H04R 3/04 for correcting frequency response

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.

Claims

1. 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 2π

f_h t/2π

f_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.

2. The method of claim 1 wherein said calculating step includes solving the matrix equation

3. 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.

4. 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