Multi-band, digital audio noise filter

• US 5,416,847 A
• Filed: 02/12/1993
• Issued: 05/16/1995
• Est. Priority Date: 02/12/1993
• Status: Expired due to Term

• Current Assignee: Disney Enterprises Incorporated (The Walt Disney Company)
• Original Assignee: The Walt Disney Company
• Inventors: Boze, Steven E.
• Primary Examiner(s): Isen, Forester W.
• Application Number:
  US08/017,133
• Time in Patent Office: 823 Days
• Field of Search: 381/98, 381/68.2, 381/94, 381/68.4
• US Class Current: 381/94.3
• CPC Class Codes: H04H 60/04 Studio equipment; Interconnection of studios

Abstract

This disclosure provides a multi-band, digital audio noise filter that is especially useful in the restoration of motion picture film soundtracks. More particularly, the preferred embodiment presented herein utilizes a remote fader board having eight faders which permit a user to control thresholds for sixty-four frequency bins. These faders are monitored by a MOTOROLA 56000-series microprocessor, which accepts digitized audio input signals, performs a Fast Fourier Transform upon a 128 sample window to yield signal contribution for each of the sixty-four frequency bins, and derives FIR filter coefficients for noise attenuation. The digitized audio input signals, which have been stored in a circular input buffer, are convolved with the FIR filter and output as the digitized output signals of the restored motion picture soundtrack.

Claims

1. A multi-band, digital audio noise filter that permits a user to convert a digital audio input signal into a digital audio output signal, the filter comprising:

a signal processor that processes the digital audio input signal to obtain the digital audio output signal by multiplying each of a plurality of filter coefficients with each of a plurality of time-based digital samples which compose a finite-length window of the digital audio input signal and by summing the results to obtain a digital value that is an output value of the digital audio output signal;

a user-interface that permits the user to vary contribution-thresholds for at least several of a plurality of frequency bins that correspond to composition frequencies of the digital audio input signal;

a filter generator that repeatedly updates said filter coefficients in dependence upon current input values of each of said digital audio input signal and said contribution-threshold for said plurality of frequency bins, said filter generator including

FFT means for receiving the digital audio input signal and applying thereto a Fast Fourier Transform to produce, in response thereto, at least one FFT value for each of the plurality of frequency bins, said FFT values each representing the contribution of harmonics to the digital audio input signal from frequencies within said frequency bins, index generating means for comparing, for each of said plurality of frequency bins, said FFT values which correspond to each frequency bin with a corresponding user-set signal threshold level, and for generating an attenuation index for each of said plurality of frequency bins in response thereto, said attenuation index representing an attenuation of harmonics within the frequency bin if said corresponding FFT value is less than said corresponding user-set signal threshold level, and,

IFFT means coupled to said index generating means for generating and updating, in response to said attenuation index for each of said plurality of frequency bins, said plurality of filter coefficients.

2. A filter according to claim 1, wherein said FFT means includes frequency windowing means for smoothing frequencies of the digital audio input signal, such that distinct frequencies of said digital audio input signal fall substantially within one of said frequency bins.

3. A filter according to claim 2, wherein said IFFT means includes filter windowing means for smoothing frequencies represented in time by said filter coefficients, such that attenuation represented by said filter coefficients is applied substantially only to frequencies of the digital audio input signal which fall within frequency bins that correspond to the particular attenuation.

4. A filter according to claim 1, wherein said signal processor includes means for convolving a finite impulse response filter with said plurality of time-based digital samples which compose a finite-length window of the digital audio input signal to obtain therefrom said digital value that is an output value of the digital audio output signal.

5. A filter according to claim 1, wherein said attenuation index, for each frequency bin'"'"'s corresponding FFT values, is selected to correspond to no attenuation of the digital audio input signal if said FFT values are not less than said corresponding user-set threshold, and is otherwise selected to pass corresponding composition frequencies of said digital audio input signal in an amount proportional to a ratio of said corresponding FFT value to said corresponding user-set threshold.

6. A filter according to claim 1, wherein said user-interface includes a fader control having a predefined number of faders that each control at least one of said contribution-thresholds, at least one fader controlling simultaneous selection of a plurality of said contribution-thresholds, said fader control having a digital electronic signal output coupled to said filter generator that represents a position-setting of each of said faders.

7. A filter according to claim 6, wherein each of said predefined number of faders control simultaneous selection of an equal number of said contribution-thresholds.

8. A filter according to claim 1, wherein said user-interface includes:

a threshold-control that permits the user to select a contribution-threshold for at least a predefined one of said frequency bins; and

,

a slope control that permits the user to select a rate of change between frequency bins and thereby select said contribution-thresholds, in relation to said contribution-thresholds for said predefined one, for said plurality of frequency bins.

9. A filter according to claim 1, wherein said FFT means includes adding means for summing time-based digital samples from each of left and right audio channels and for applying the Fast Fourier Transform thereto.

10. A filter according to claim 1, wherein:

said index generating means includes multiplying means for multiplying said attenuation index (for each frequency bin) with real and imaginary values to yield a corresponding scaled real and imaginary value pair; and

said IFFT means includes means coupled to receive said scaled real and imaginary value pair for each of said plurality of frequency bins and for applying to said scaled real and imaginary value pair said inverse Fast Fourier Transform to thereby generate and update said plurality of filter coefficients.

11. A filter according to claim 1, wherein:

said index generating means further includes FFT value averaging means for averaging said FFT values with at least one old FFT value, for each frequency bin, to produce average FFT values; and

,

said index generating means compares, for each of said plurality of frequency bins, corresponding ones of said averaged FFT values with said corresponding user-set signal threshold level.

12. A filter according to claim 11, wherein said FFT value averaging means includes weighting means for applying a first weight to said FFT values and a second weight to said old FFT values, for each frequency bin, the sum of said first and second weights being approximately one, said weights being varied as a function of frequency bin.

13. A filter according to claim 1, wherein:

said signal processor includes convolution software that directs a microprocessor-based system to process the digital audio input signal by convolving in the time domain a finite impulse response filter with said finite length window; and

,

said filter generator includes filter configuration software that directs a microprocessor-based system to update said filter coefficients.

14. A multi-band, digital audio noise filter that permits a user to convert an audio input signal into a digital audio output signal, the filter comprising:

a user-interface whereby the user may vary contribution-thresholds for at least several of a plurality of frequency bins that correspond to frequencies which combine to form the digital audio input signal;

a digitizing mechanism that is coupled to receive the audio input signal, to sample said audio input signal, and to produce, in response thereto, a digitized audio input signal;

a microprocessor-based system having

a first connector that couples said microprocessor-based system to said user-interface to receive therefrom said contribution-thresholds,

convolution means for convolving a window of samples of said digitized audio input signal with a digital filter having filter coefficients and for producing therefrom the digital audio output signal,

a second connector that couples said digitizing mechanism to said convolution means such that said convolution means receives said digitized audio input signal, and

filter configuration software that directs said microprocessor-based system to periodically and repeatedly

sample said user-interface so as to receive sampled, digitized threshold values representative of said user-set contribution-thresholds,

apply a Fast Fourier Transform to said digitized audio input signal to produce therefrom estimates of signal contribution for each of said plurality of frequency bins,

compare said estimates, for each of said plurality of frequency bins, with a corresponding one of said contribution-thresholds,

derive attenuation indices for each of said plurality of frequency bins in response to the comparison, and

perform an inverse Fast Fourier Transform in response to said attenuation indices so as to periodically and repeatedly update said filter coefficients of said digital filter, and

convolution software that directs said convolution means to convolve said digitized audio input signal with said digital filter to produce said digital audio signal output.

15. A filter according to claim 14, wherein said digitizing mechanism includes a motion picture film soundtrack reading mechanism that reads the audio input signal from motion picture film, said digitizing mechanism generating therefrom said digitized audio input signal.

16. A filter according to claim 14, wherein:

said microprocessor-based system includes a first microprocessor;

said convolution means includes a second microprocessor coupled to said digitizing mechanism to receive therefrom said digitized audio input signal, said second microprocessor being directed by said convolution software to convolve said window with said digital filter and to produce from the convolution the digital audio output signal; and

,

wherein said filter configuration software directs said first microprocessor to communicate said filter coefficients of said digital filter to said second microprocessor.

17. A filter according to claim 14, wherein said microprocessor-based system includes a single microprocessor that is directed by both said convolution software and said filter configuration software.

18. A filter according to claim 14, wherein said filter configuration software includes software that directs said microprocessor based system to:

multiply a pair of real and imaginary values by a corresponding one of said attenuation indices to yield a scaled value pair for each of said plurality of frequency bins; and

,

perform said inverse Fast Fourier Transform upon said scaled value pair so as to periodically and repeatedly update said digital filter.

19. A filter according to claim 14, wherein said filter configuration software includes software that directs said microprocessor based system to:

apply to said digitized audio input signal a frequency window that smoothes frequencies of the digital audio input signal, such that distinct frequencies of said digital audio input signal fall substantially within one of said frequency bins; and

,

apply to said filter coefficients a filter window that smoothes frequencies represented in time by said filter coefficients, such that attenuation represented by said filter coefficients is applied substantially only to frequencies of said digitized audio input signal which fall within frequency bins that correspond to the particular attenuation.

20. A filter according to claim 14, wherein said filter further comprises soundtrack writing means for writing the digital audio output signal onto motion picture film as a soundtrack.

21. A method of reducing noise in at least one audio channel without the need of a test signal dedicated to noise measurement or a period of silence on an audio input signal, the method utilizing an analog-to-digital converter, data registers (which may be defined in random access memory), a microprocessor-based system that convolves (in the time-domain) a digital filter with the audio input signal to produce a digital audio output signal, and a user-interface adapted to create at least one electronic signal that represents user-variations of contribution-thresholds for at least several of a plurality of frequency bins, wherein the microprocessor-based system has random access memory and is coupled to the user-interface so as to receive the electronic signals, and wherein the digital filter is alterable in response to the user-settings of the contribution-thresholds for each of the plurality of frequency bins, the method comprising the steps of:

applying the analog-to-digital converter to the audio input signal to produce a sequence of digital samples that form a digital audio input signal representative of the audio input signal;

using the microprocessor-based system to

obtain the contribution-thresholds by monitoring the electronic signals that represent the user-settings,

apply a Fast Fourier Transform to a window of digital samples of the digital audio input signal to produce at least one FFT value for each of the plurality of frequency bins, for each of the plurality of frequency bins, compare the corresponding contribution-threshold with the corresponding FFT values,

generate attenuation indices for each of the plurality of frequency bins in response to the comparison, the attenuation indices representing attenuation to be applied to all harmonic components falling within corresponding ones of the plurality of frequency bins,

use a inverse Fast Fourier Transform to derive, from the attenuation indices, coefficients of the digital filter, and

repeat each of the above steps to thereby periodically update coefficients of the digital filter; and

,

using the microprocessor-based system to apply the digital filter to the audio input signal to produce the audio output signal.

22. A method according to claim 21, wherein:

the step of obtaining the contribution-thresholds includes

upon power-up, using the microprocessor-based system to define default contribution-thresholds for each of the plurality of frequency bins and to store the default contribution-thresholds in the registers,

using the user-interface to select at least one frequency bin from among the plurality of frequency bins,

using the user-interface to vary the contribution-thresholds that correspond to the selected frequency bins, and

causing the microprocessor-based system to store new contribution-thresholds that correspond to the selected frequency bins in registers; and

wherein the step of comparing the FFT values with the contribution-threshold includes the step of reading the contribution-threshold from the registers.

23. A method according to claim 21, wherein:

the step of obtaining the contribution-thresholds includes the steps of

using the microprocessor-based system to define in the random access memory the registers as each corresponding to specific frequency bins, and

repeatedly and periodically reading the user-interface to sample user-settings, writing the sampled contribution-thresholds represented thereby into the registers to define contribution-thresholds for the comparison step; and

the step of comparing the contribution-threshold with the FFT values includes the step of reading the contribution-threshold from the corresponding register defined in random access memory.

24. A method according to claim 21, wherein:

the step of obtaining the contribution-thresholds includes the steps of

using the microprocessor-based system to define in random access memory the registers as corresponding to specific frequency bins,

repeatedly and periodically reading the user-interface to sample user-settings of a key contribution-threshold that corresponds to at least one of the frequency bins, and to sample a change value representative of change in magnitude between contribution-thresholds corresponding to contiguous frequency bins,

determining contribution-thresholds for each of the plurality of frequency-bins in response to the key threshold-contribution and the change value, and

writing the contribution-thresholds thereby determined into the registers to define contribution-thresholds for the comparison step; and

the step of comparing the contribution-threshold with the FFT values includes the step of reading the contribution-threshold from the corresponding register defined in random access memory.

25. A method according to claim 21, utilizing at least two audio channels including a left audio channel and a right audio channel, each having left and right audio input signals, respectively, to produce corresponding digital audio output signals, wherein the method includes the steps of:

applying the analog-to-digital converter to each audio input signal to produce corresponding sequences of digital samples that form digital audio signals representative of each audio input signal;

using the microprocessor-based system to

average together digital samples of each channel that correspond to the same time of sampling, and

apply the Fast Fourier Transform to a window of averages to produce the FFT values;

using the microprocessor-based system to continuously apply the digital filter to each audio output signal to produce the corresponding digital audio output signals.

26. A method according to claim 21, wherein:

the step of using the microprocessor based system to apply the Fast Fourier Transform includes producing at least one pair of values representing real and imaginary FFT products, respectively;

the step of comparing includes comparing at least one of the real and imaginary FFT products, for each of the plurality of frequency bins, with the corresponding contribution-threshold; and

,

the step of using the inverse Fast Fourier Transform includes,

for each frequency bin, multiplying a real and imaginary value pair with the corresponding one of the attenuation indices to obtain a pair of attenuation values for each frequency bin, and

applying the inverse Fast Fourier Transform to all of the pairs of attenuation values to derive the coefficients of the digital filter.

27. A method according to claim 21, wherein the method further comprises the steps of:

using the microprocessor-based system to retain in the random access memory previous FFT values;

using the microprocessor-based system to obtain present FFT values by application of the Fast Fourier Transform;

using the microprocessor-based system to average together corresponding ones of the present FFT values and the previous FFT values to obtain averaged values to be used in the comparison step for comparison with the contribution-thresholds; and

,

repeating each of these averaging steps by writing into the random access memory the averaged values for future use as previous FFT values.

28. A method according to claim 27, wherein the step of using the microprocessor-based system to obtain averaged values includes the step of scaling each of present FFT values and previous FFT values for each frequency bin, wherein the sum of a first scalar, corresponding to present FFT values, and a second scalar, corresponding to past FFT values, is substantially equal to one, and wherein the relative contribution of present FFT values to the averaged values is relatively low for low frequencies.

29. A method according to claim 21, wherein the method further comprises the steps of:

using the microprocessor-based system to smooth frequencies of the digital audio input signal, such that distinct frequencies of said digital audio input signal fall substantially within one of the frequency bins; and

,

using the microprocessor-based system to smooth frequencies represented in time by the filter coefficients, such that attenuation represented by said filter coefficients is applied substantially only to frequencies of the digital audio input signal which fall within frequency bins that correspond to the particular attenuation.

30. A method according to claim 21, the method utilizing a motion picture film soundtrack reading device and a motion picture film soundtrack writing device to be used with motion picture film, and a digital-to-analog converter, the method further comprising the steps of:

using the motion picture film soundtrack reading device to obtain the audio input signal;

applying the audio output signal to the digital-to-analog converter to produce an analog audio output signal; and

,

using the motion picture film soundtrack writing device to write the analog audio output signal to motion picture film as a restored soundtrack.

31. A multi-band filter that attenuates noise in a prerecorded sound track in response to threshold settings interactively supplied from a user that concurrently listens to the audio output, comprising:

a user-interface that permits the user to independently vary thresholds for at least several of a plurality of frequency bins that represent harmonics of the sound track;

a signal processor that receives an input signal representing the sound track, applies a filter to it to provide attenuation to it, and generates therefrom the audio output;

a filter generator that repeatedly determines coefficients of the filter applied by said signal processor, said filter generator including

a Fast Fourier Transform stage that receives the input signal and applies to the input signal a Fast Fourier Transform to produce, in response thereto, at least one value for each of frequency bin,

an index generating stage that receives the thresholds from the user-interface, compares said at least one value for each bin with a corresponding threshold, and derives an attenuation index that represents attenuation of harmonics within the frequency bin if said corresponding FFT value is less than said corresponding user-set threshold, and,

an Inverse Fast Fourier Transform stage that processes said attenuation index for each of said plurality of frequency bins, to derive the coefficients of the filter; and

an audio player that permits the user to concurrently listen to the audio output and interactively adjust the thresholds for the frequency bins.