Description Technical Field

[0001 ] The present invention relates broadly to a method of processing a digital signal and relates particularly, although not exclusively, to a method of modulating a digital signal such as an audio signal. The invention also relates broadly to a computer or device-readable medium for processing a digital signal, and a computer system for processing a digital signal. The invention extends to other digital processing including processing images and other signals including signals associated with digital communications.

Background of Invention

[0002] In digital recording and playback an analog signal representative of audio is converted into a digital signal which lends itself to manipulation and storage. The conversion is performed in an analog to digital converter (ADC). The stored digital signal can be converted back to an analog signal in a digital to analog converter (DAC). The analog signal is played back using conventional audio equipment such as amplifiers and speakers. The digital signal can be manipulated prior to the DAC to improve its quality before playback. This manipulation includes audio EQ where selected parts of the frequency spectrum of the audio are filtered to, for example, compensate for irregularities in the frequency response. The audio may also be filtered to resolve problems from its conversion into a digital signal or back to an analog signal. This manipulation in the digital domain also includes modulation of the signal to, for example, remove or otherwise control dynamic values such as volume.

Summary of Invention

[0003] According to a first aspect of the present invention there is provided a method of digitally processing a signal, said method comprising the steps of:

applying a frequency filter to the digital signal to derive a modulation curve representing value changes in the signal;

Substitute Sheet

(Rule 26) RO/AU applying the modulation curve to the digital signal to demodulate value changes in the signal to provide a demodulated signal.

[0004] Preferably the method also comprises a preliminary step of modifying the signal to obtain a corresponding modified signal having positive values only wherein the frequency filter is applied to the modified signal.

[0005] Preferably the method also comprises the step of increasing the sample rate of the frequency filter from a predetermined sample rate including neighbouring sample points to an increased sample rate including intermediate sample points between adjacent of the neighbouring sample points, said sample rate being increased by populating each of the intermediate sample points depending on a weighted influence of a predetermined number of the neighbouring sample points. More A method as defined in claim 3 wherein the weighted influence is calculated for each of the intermediate sample points by:

(i) representing the frequency filter at the predetermined number of the neighbouring sample points at least in part by its cosine components, each component represented by absolute values of a cosine function in the time domain substantially limited to half a waveform cycle at its mid-point;

(ii) combining the half-cycle cosine components at each of the

neighbouring sample points to obtain representative waveforms located at respective of the neighbouring sample points;

(iii) shifting each of the representative waveforms in their time domain substantially midway between the respective neighbouring sample point sand the intermediate sample point;

(iv) determining values for each of the shifted representative waveforms at the intermediate sample point;

(v) combining the determined values at the intermediate sample point to derive the weighted influence.

[0006] Alternatively the weighted influence is calculated for each of the

intermediate sample points by:

(i) representing the frequency filter at one of the predetermined number of the neighbouring sample points at least in part by its cosine components, each

Substitute Sheet

(Rule 26) RO/AU component represented by absolute values of a cosine function in the time domain substantially limited to half a waveform cycle at its mid-point;

(ii) combining the half-cycle cosine components to obtain a representative waveform at the neighbouring sample point;

(iii) shifting the representative waveform in its time domain substantially midway between the neighbouring sample point and the intermediate sample point;

(iv) determining values for the shifted representative waveform at each of the predetermined number of the neighbouring sample points;

(v) combining the determined values at the neighbouring sample points to derive the weighted influence.

[0007] Preferably the frequency filter with the weighted sample rate increase is expanded in the time domain prior to its application to the digital signal. More preferably the method further comprises the step of constructing a composite frequency filter by combining the expanded frequency filter with the weighted sample rate with another frequency filter at a corresponding sample rate increase. Even more preferably the step of combining said filters involves multiplying corresponding values for the neighbouring and intermediate sample points of the respective expanded frequency filter and the other frequency filter

[0008] Preferably application of the modulation curve to the digital signal involves convolving the modulation curve with the signal. Alternatively application of the modulation curve to the signal involves application according to a predetermined algorithm.

[0009] Preferably application of the frequency filter to the digital signal involves convolving the frequency filter with said signal. More preferably the step of applying the frequency filter involves applying a relatively low frequency filter at less than an audible frequency.

[0010] Preferably the demodulated signal is substantially absent of dynamic values.

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(Rule 26) RO/AU [001 1 ] According to a second aspect of the invention there is provided computer program code which when executed implements the method of any one of the preceding statements.

[0012] According to a third aspect of the invention there is provided a computer or device readable medium including instructions for processing a digital signal, said instructions when executed by a processor cause said processor to:

apply a frequency filter to the digital signal to derive a modulation curve representing value changes in the signal;

apply the modulation curve to the digital signal to demodulate value changes in the signal to provide a demodulated signal.

[0013] According to a fourth aspect of the invention there is provided a computer system for processing a digital signal, said computer system comprising a processor configured to:

apply a frequency filter to the digital signal to derive a modulation curve representing value changes in the signal;

apply the modulation curve to the digital signal to demodulate value changes in the signal to provide a demodulated signal.

Brief Description of Drawings

[0014] In order to achieve a better understanding of the nature of the present invention a preferred embodiment of a method of digitally processing an audio signal will now be described by way of example only, with reference to the accompanying drawings in which:

Figure 1 is a schematic of application of embodiments of the invention in digital audio recording and playback;

Figure 2 is a schematic illustration of digital signal modulation and associated signal processing steps according to one embodiment of the present invention;

Figure 3 is a schematic illustration of convolution of a frequency filter with a digital signal according to an embodiment of the invention;

Substitute Sheet

(Rule 26) RO/AU Figures 4A and 4B are illustrations of alternative waveforms or components thereof represented by mathematical functions representing exemplary frequency filters;

Figures 5A and 5B are schematics of different techniques for increasing the sample rate of the frequency filter.

Detailed Description

[0015] Figure 1 shows application of the various embodiments of the invention in the course of digital audio recording and playback. The analog audio signal 10 is converted to a digital audio signal at an analog to digital converter (ADC) 12. The digital audio signal may then be subject to signal processing at digital processor 14. The processed digital signal is down-sampled and stored at storage memory 16 before a sample rate increase to increase its resolution prior to playback. The relatively high resolution digital audio signal is then converted back to an analog signal 20 at a digital to analog converter (DAC) 18. It will be understood that various embodiments of the invention can be applied at the digital signal processor 14 where, for example, the digital signal undergoes processing to remove or otherwise control dynamic values such as volume.

[0016] In the embodiment shown in figure 2 there is provided a method of digitally processing a digital signal S, the method comprising the steps of:

1 . a preliminary and optional step of modifying the digital signal S of figure 2a to obtain a corresponding modified signal MS of the lower representation of figure 2b having positive values only;

2. applying a frequency filter designated F at the upper representation of figure 2b to the modified signal MS to derive a modulation curve MC shown as the upper representation in figure 2c representing a map of value changes in the signal S;

3. applying the modulation curve MC to the digital signal S in figure 2c to

provide a demodulated signal DS shown in figure 2d.

Substitute Sheet

(Rule 26) RO/AU [0017] In step 3 of the preferred embodiment the modulation curve MC is applied to the digital signal S by multiplying values of the modulation curve MC with corresponding values of the signal S, see figure 2c. Alternatively the modulation curve MC may be convolved with the signal S. This convolution may involve conventional practices or alternatively may adopt the techniques described in the applicant's modified convolution US provisional patent application nos. 61 /974,326 and 62/056,343. The disclosures of these provisional patent applications are to be considered included herein by nature of these references. In an alternative embodiment the modulation curve may be applied to the signal according to a predetermined algorithm. In either case the demodulated signal DS obtained by application of the modulation curve to the digital signal is substantially absent of dynamic values such as volume changes outside a predetermined threshold.

[0018] In step 2 of this embodiment the filter F is applied to the modified signal MS by convolving the filter F with the signal MS, see figure 2b.

[0019] Figure 3 schematically illustrates one technique for applying frequency filter 14 to digital signal 15 (without modification) in convolution. The frequency filter is represented by filter waveform 14 which includes a mid-point sample 16 and a plurality of neighbouring sample points such as 18a to 18d located either side of the mid-point sample 16 (shown by solid dots). The digital signal 15 includes a

corresponding sample point 20 which aligns in the time domain with the mid-point sample 16 of the frequency filter 14. The digital signal 15 also includes neighbouring sample points 18A to 18D located either side of the corresponding sample point 20 (shown by solid dots). The corresponding sample points 18A to 18D are offset in the time domain relative to the respective neighbouring sample points 18a to 18d of the frequency filter 14.

[0020] For simplicity the frequency filter 14 is represented by a sine function which is the sum of the cosine components in the time domain. where /p is the corner frequency for a lowpass filter, x is the time variable on the x- axis, and represents an averaging curve with q representing the aspect ratio of the averaging curve.

Substitute Sheet

(Rule 26) RO/AU [0021 ] In alternative embodiments the frequency filter may be represented by other waveforms constructed from components represented by mathematical functions including:

1 . Sine functions for absolute time values represented in the time domain;

2. Sine functions for time values represented in the time domain from zero (0) to positive infinitely only;

3. Sine functions for values represented in the time domain from zero (0) to positive infinity only;

4. Absolute values of a cosine function in the time domain substantially limited to half a waveform cycle at its mid-point.

[0022] Figures 4A and 4B illustrate the component waveforms of items 1 and 4, respectively, summed across a relevant frequency range. The component waveforms of item 2 are effectively half the component waveforms of item 1 . The component waveform of item 3 are similarly the waveforms of equation 1 but for positive time values only

[0023] The applicant's co-pending International patent application

PCT/AU2014/000317 describes in some detail waveforms constructed from

components represented by sine functions for absolute time values. The disclosures of this PCT application are to be considered included herein by nature of this reference.

[0024] The waveform components may also be adjusted by for example applying a mathematical function to each of the components or the waveforms themselves. In one embodiment the waveforms or their components may be modified by applying an averaging curve having its width adjusted proportional to the wavelength of the respective waveforms. This modification of the waveform components is discussed in the applicant's co-pending International patent application No. PCT/AU2014/000321 the disclosures of which are to be considered included herein by nature of this reference.

Substitute Sheet

(Rule 26) RO/AU [0025] The frequency filter of this embodiment may be constructed from cosine components represented by absolute values of a cosine function in the time domain substantially limited to half a waveform cycle at its mid-point. The half-cycle cosine components are summed across the relevant frequency range to obtain the relevant waveform, see for example figure 4B. In this case the half-cycle cosine waveform components are each weighted for substantially equal contribution to the filter. This weighting is approximated as equal areas under each of the half-cycle cosine component curves.

[0026] In this example the application of the frequency filter 14 to the digital signal 15 involves a modified form of convolution based on dot products of values at neighbouring and intermediate sample points of the frequency filter and the signal, respectively. In considering a limited numbering of sample points and with reference to figure 3 this dot product methodology involves:

1 . Multiplying the mid-point sample 16 of the frequency filter 14 with the

corresponding sample point 20 of the signal 15;

2. Multiplying each of the neighbouring sample points 18a to 18d of the

frequency filter 14 with their respective offset neighbouring sample points 18A to 18D of the signal 15 for a predetermined number of sample points;

3. Multiplying each of the intermediate sample points 22a to 22d of the

frequency filter 14 with their respective offset intermediate sample points 22A to 22D of the signal 15;

4. Summing the product results from the multiplications of steps 1 to 3;

5. Shifting or stepping the frequency filter 14 at its adjusted sampling rate wherein the mid-point sample 16 of the frequency filter 14 aligns in the time domain with the offset sample point 18A of the signal 15;

6. Repeating steps 1 to 4 with the frequency filter 14 at this position;

7. Continuing this modified convolution at the adjusted sampling rate stepping across a predetermined number of sample points in the signal 15.

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(Rule 26) RO/AU [0027] The frequency filter 14 and the digital signal 15 have each undergone the same sample rate increase. For simplicity each of the frequency filter 14 and digital signal 15 have been shown with a single intermediate sample point such as 22a and 22A (shown by hollow dots) respectively between adjacent of the neighbouring sample points, such as 16/18a and 18A/20. In practice it will be understood that depending on the sample rate increase or required resolution there will be many more intermediate sample points.

[0028] In this embodiment the sample rate increase for the frequency filter and/or the digital signal is performed by populating each of the intermediate sample points depending on a weighted influence of a predetermined number of the neighbouring sample points. For example 1 ,024 neighbouring sample points may be taken into account with 512 sample points either side of the intermediate sample point being populated. The weighted influence may be calculated for each of the intermediate sample points by any one of the following exemplary techniques involving:

1 . Representative waveforms at respective of neighbouring sample points where values are combined at the intermediate sample points;

2. A representative waveform at the intermediate sample point where values are combined at the neighbouring sample points;

3. Representative waveforms at respective neighbouring sample points

shifted midway to the intermediate sample point with values combined at the intermediate sample point;

4. A representative waveform at the neighbouring sample points shifted

midway to the intermediate sample point with values combined from the neighbouring sample points.

[0029] Figure 5A schematically illustrates representative waveforms at

respectively neighbouring sample points but shifted midway toward the intermediate sample point under consideration. In this variation, values are combined for each of the shifted representative waveforms at the intermediate sample point. The digital signal S at n[1] \s for simplicity represented with three weighted waveform

components designated wc[1, 1] wc[1,2] wc[1,3]. In practice the signal S will be

Substitute Sheet

(Rule 26) RO/AU represented by many more waveform components sufficient to cover the frequency content of the signal. Each of the waveform components are in this embodiment represented by absolute values of a cosine function in the time domain substantially limited to half a waveform cycle at its mid-point. The waveform components such as wc[1, 1], wc[1,2] wc[1, 3] are combined or in this exampled summed at the relevant neighbouring sample point n[1]\.o obtain a representative waveform designated rw[1]. The shifted representative waveform for each of the neighbouring sample points n[1] and n[2] \s shown in hidden line detail and designated as srw[1] and srw[2] respectively. The values are determined at the intermediate sample point //7/ for each of the waveforms srw[1] and srw[2] and these values designated as ///, 1] and i[1,2] respectively. These steps are repeated for each of the intermediate sample points in order to populate the digital signal or filter at the increased sample rate.

[0030] Figure 5B schematically illustrates the technique where the representative waveform is shifted midway toward the intermediate sample point and the values combined from the neighbouring sample points. The shifted representative waveform is shown in hidden line detail and designated as srw[1]. The values to be combined for the shifted representative waveform srw[1] at the respective neighbouring sample points / ^" // and [2] are designated as srw[1, 1] and srw[1,2].

[0031 ] In this embodiment and as shown in figure 3 the sample points of the digital signal 15 may be offset relative to the respective sample points of the frequency filter 14 or vice versa by expansion of a filter waveform representative of an initial digital filter (not shown). The sample points of the initial digital filter or waveform align in the time domain with respective and corresponding sample points of the signal 15 and thus is of the same sample resolution. In this example the initial filter may be expanded in its time domain by a factor or multiplier of between two (2) and ten (10). The filter 14 has undergone its weighted sample rate increase to the increased or required resolution prior to its expansion. The expansion factor is in this example proportional to the sample rate increase.

[0032] In this embodiment the digital filter 14 is applied to the signal 15 at an adjusted sampling rate. This adjusted sampling rate compensates for and is substantially proportional to the offset of the sample points 18A to 18D of the signal 15 relative to the respective neighbouring sample points 18a to 18d of the filter 14. In

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(Rule 26) RO/AU this example the filter 14 is thus applied to the signal 15 at a fraction of the sample resolution of the initial filter.

[0033] The frequency filter may be processed before and/or after modified convolution as earlier-described. This processing includes but is not limited to the application of:

1 . One or more filters in windowing, see for example the applicant's

International patent application no. PCT/AU2014/000321 ;

2. Sample rate increases to the signal and/or filter using for example

techniques disclosed in the applicant's International patent application no.'s PCT/AU2014/000318 and PCT/AU2014/000319 together with US provisional patent application no. 62/056,349;

3. Sample zoning and the combination of sample values depending on the frequency of the zone as disclosed in the applicant's International patent application no. PCT/AU2015/000197.

[0034] The frequency filter applied to the digital signal may be constructed as a composite frequency filter by combining multiple frequency filters. Each of the frequency filters preferably undergoes a sample rate increase prior to construction of the composite frequency filter. The techniques disclosed in the applicant's

international patent application no. PCT/AU2014/000319 may be adopted in performing this sample rate increase and the disclosures of this PCT application are to be considered included herein by reference of this reference. The techniques described in combining filter components in the applicant's international patent application no. PCT/AU2014/000321 are also to be considered included herein by nature of this reference. The filter or composite filter is generally a low frequency filter at less than an audible frequency of around 20 Hz.

[0035] Is to be understood that the methods and techniques described can be implemented as computer-readable instructions stored on a computer-readable medium. The computer-readable instructions can be executed by a processor of practically any computer system including desktop, portable, tablet, hand-held, and/or any other computer device.

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(Rule 26) RO/AU [0036] It is also to be understood that the present invention extends to computer- readable media for carrying or having computer-executable instructions stored thereon. The computer-readable media include RAM, ROM, EEPORM, CD-ROM or other optical disc storage, magnetic disc storages, or any other medium which carries or stores program code means in the form of computer-executable instructions. In the event of information being transferred or provided over a network or another communications connection to a computer, the computer is to be understood as viewing the connection (hardwired, wireless, or a combination thereof) as a computer- readable medium.

[0037] Those skilled in the art will appreciate that the invention described herein is susceptible to variations and modifications other than those specifically described. The processing of digital or audio signals need not be limited to acoustics but extends to other sound applications including ultrasound and sonar. The invention also extends beyond audio signals to other digital signals including signals derived from a physical displacement such as that obtained from measurement devices, for example a strain gauge or other transducer which generally converts displacement into an electronic signal.

[0038] The invention also covers digital processing of signals associated with digital communications. The invention in another embodiment is applied to imaging. In imaging the demodulation of value changes may apply to contrast and/or brightness; black-and-white and/or one or more colour bands; an image over space and/or an image in a video over time; or any combination of these digital signals.

[0039] The signal(s) may be constructed or represented by fast fourier transform (FFT) algorithms rather than the trigonometric functions described in the preferred embodiments, such as the cosine and/or sine components.

[0040] All such variations and modifications are to be considered within the scope of the present invention the nature of which is to be determined from the foregoing description.

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(Rule 26) RO/AU