This application claims priority from US patent application no. 61/805,449 filed on 26 March 2013, the contents of which are to be taken as incorporated herein by this reference. This application is related to and if required claims priority from US patent application nos. 61/805,406, 61/805,432, 61/805,466, 61/805,469, and 61/805,463 all filed on 26 March 2013, the contents of which are to be taken as incorporated herein by these references. This application is also related to and if required claims priority from US patent application no. 61/819,630 filed on 5 May 2013 and US patent application no. 61/903,225 filed on 12 November 2013, the contents of which are to be taken as incorporated herein by these references.

Technical Field

The present invention relates broadly to a method of digitally filtering an audio signal. The invention relates particularly although not exclusively to digitally filtering an audio signal in audio equalisation (EQ). The invention extends to other digital filtering including filtering images and other signals including signals associated with digital communications and processing. Background Art

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.

Summary of Invention

According to one aspect of the present invention there is provided a method of digitally filtering an audio signal, said method comprising the steps of: providing a plurality of sample filters at respective of a plurality of predetermined frequencies;

applying an averaging curve to each of the plurality of sample filters where a width of the averaging curve is adjusted proportional to the frequency of the respective sample filter to provide an averaged sample filter;

combining the averaged sample filters to provide a composite audio filter; filtering the audio signal using the composite audio filter.

Preferably the predetermined frequency of each of the filters includes a range of frequencies.

Preferably the frequency bandwidth covered by the plurality of sample filters is generally representative of the audio signal to be filtered. Preferably the method also comprises the step of increasing the sample rate of the composite audio filter from a predetermined sample rate to an increased sample rate prior to filtering the audio signal.

Preferably the step of increasing the sample rate of the composite audio filter includes:

defining intermediate sample points at the increased sample rate located between neighbouring sample points at the predetermined sample rate;

calculating a weighting for each of the intermediate sample points including the steps of (i) nominating neighbouring audio signals neighbouring an audio signal of the audio filter at respective of the neighbouring sample points (ii) shifting each of the nominated neighbouring audio signals in the time domain between the relevant neighbouring sample point and the intermediate sample point (iii) combining values for the shifted neighbouring audio signals at the intermediate sample point to derive the weighting;

applying the weighting to the audio signal of the composite audio filter at respective of the intermediate sample points. Preferably the nominated neighbouring audio signals is shifted in the time domain substantially midway between the neighbouring sample point and the intermediate sample point. Preferably the step of increasing the sample rate of the composite audio filter includes:

defining intermediate sample points at the increased sample rate located between neighbouring sample points at the predetermined sample rate;

calculating a weighting for each of the intermediate sample points including the steps of (i) providing a hypothetical audio signal of a waveform corresponding to an audio signal of the audio filter and shifted in its time domain to align with the intermediate sample point (ii) expanding the shifted hypothetical audio signal in the time domain (iii) combining values for the expanded hypothetical audio signal at the neighbouring sample points to derive the weighting;

applying the weighting to the audio signal of the filter at respective of the intermediate sample points.

Preferably the shifted hypothetical audio signal is expanded in the time domain by a factor of substantially two (2).

Preferably the step of combining the averaged filters is performed at an adjusted sampling rate wherein the other audio filter includes one or more intervening sample points between adjacent of its neighbouring sample points. More preferably the adjusted sampling rate for applying the audio filter to the other audio filter is inversely proportional to the number of intervening sample points relative to the number of neighbouring sample points for the other filter.

Alternatively the step of increasing the sample rate of the composite audio filter includes:

defining intermediate sample points at the increased sample rate located between neighbouring sample points at the predetermined sample rate;

calculating a weighting for each of the intermediate sample points including the steps of (i) nominating neighbouring audio signals neighbouring an audio signal of the composite audio filter at respective of the neighbouring sample points (ii) combining values for the neighbouring audio signals at the intermediate sample point to derive the weighting;

applying the weighting to the audio signal of the composite audio filter at respective of the intermediate sample points.

Still alternatively the step of increasing the sample rate of the composite audio filter includes:

defining intermediate sample points at the increased sample rate located between neighbouring sample points at the predetermined sample rate;

calculating a weighting for each of the intermediate sample points including the steps of (i) providing a hypothetical audio signal of a waveform corresponding to an audio signal of the composite audio filter and shifted in its time domain to align with the intermediate sample point (ii) determining values for the hypothetical audio signal at the neighbouring sample points (iii) combining the values for the neighbouring sample points to derive the weighting;

applying the weighting to the audio signal of the composite audio filter at respective of the intermediate sample points.

Preferably the weighting is applied across a predetermined number of the neighbouring sample points.

According to another aspect of the invention there is provided a computer or device- readable medium including instructions for digitally filtering an audio signal using a plurality of filters at respective of a plurality of predetermined frequencies, said instructions when executed by a processor cause said processor to:

apply an averaging curve to each of the plurality of filters where a width of the averaging curve is adjusted proportional to the frequency of the respective filter to provide an average filter;

combine the averaged filters to provide a composite audio filter;

filter the audio signal using the composite audio filter.

According to a further aspect of the invention there is provided a system for digitally filtering an audio signal, the system comprising: a plurality of filters at respective of a plurality of predetermined frequencies, a processor configured to:

apply an averaging curve to each of the plurality of filters where a width of the averaging curve is adjusted proportional to the frequency of the respective filter to provide an averaged filter;

combine the averaged filters to provide a composite audio filter;

filter the audio signal using the composite audio filter.

According to yet another aspect of the invention there is provided a method of digitally filtering a signal, said method comprising the steps of:

providing a plurality of filters at respective of a plurality of predetermined frequencies;

applying an averaging curve to each of the plurality of filters where a width of the averaging curve is adjusted proportional to the frequency of the respective filter to provide an averaged filter;

combining the averaged filters to provide a composite filter;

filtering the signal using the composite filter.

Preferably the signal is an electronic signal derived from displacement of a transducer or measurement device.

According to yet a further aspect of the present disclosure there is provided a method of digitally filtering an image, said method comprising the steps of:

providing a plurality of image sample filters at respective of a plurality of predetermined frequencies;

applying an averaging curve to each of the plurality of image sample filters where a width of the averaging curve is adjusted proportional to the frequency of the respective image sample filter to provide an averaged image sample filter;

combining the averaged image sample filters to provide a composite image filter;

filtering the image using the composite image filter.

Preferably the image includes a matrix of pixels to which the composite image filter is applied. Brief Description of Drawings

In order to achieve a better understanding of the nature of the present invention an embodiment of a method of digitally filtering 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 illustrates an averaging curve applied to an impulse response according to an embodiment of the invention;

Figure 3 illustrates another averaging curve applied to another impulse response;

Figure 4 is a graph depicting averaging curves of different widths as a function of the frequency of the impulse response;

Figure 5 illustrates a composite filter provided by combining the averaged impulse responses of figures 2 and 3;

Figure 6 is an enlarged view of the composite filter of figure 5 with an increased sample rate;

Figure 7 shows multiple impulse responses combined to provide a composite bandpass audio filter;

Figure 8 is a frequency response for cos components where the averaging curve is adjusted proportionally;

Figure 9 is the frequency response of a cos component with a non-proportional averaging curve for comparative purposes;

Figure 10 is a schematic of one technique for increasing the sample rate of the filter; Figure 11 is a schematic of one technique for adjusting the sampling rate according to an alternative embodiment of the invention; and

Figure 12 is a schematic of another technique for increasing the sample rate of the filter;

Figure 13 is a schematic of an alternative technique for increasing the sample rate of the composite audio filter;

Figure 14 is a graph depicting weightings for intermediate sample points to be applied to relevant audio values;

Figure 15 is a schematic of another alternative technique for increasing the sample rate of the composite filter; Figure 16 is a frequency response of the composite audio filter according to an embodiment of the invention.

Description of Embodiments

The present invention in a preferred embodiment is directed to a method of digitally filtering an audio signal by applying a composite audio filter. The composite audio filter is obtained by combining a plurality of sample filters to one another where each of the sample filters is provided at respective of a plurality of predetermined frequencies. The method involves applying an averaging curve to each of the sample filters where the width of the averaging curve is adjusted proportional to the frequency of the respective sample filter. This provides an averaged sample filter for each of the plurality of sample filters and the averaged sample filters are combined to provide the composite audio filter. 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, for example in audio equalisation (EQ). 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 the various embodiments of the invention can be applied: i) at the ADC 12 where the digital audio signal undergoes a sample rate increase or over-sampling, which may be performed with weighting;

ii) at the digital signal processor 14 or a digital filter associated with EQ where, for example, the digital signal is filtered with a lowpass filter or bandpass filter;

iii) downstream of the storage memory 16 where the filtered audio signal undergoes a sample rate increase or up-sampling prior to playback. An embodiment of the present disclosure may be embodied in computer program code or software. The digital filter of the digital signal processor 14 is represented by a particular frequency response. The particular frequency response is generally dependent on the impulse response of the filter which is characterised by the software or techniques of the various embodiment of this invention. The present disclosure is intended to cover the basic types of frequency response by which digital filters are classified including lowpass, highpass, bandpass and bandreject or notch filters. The digital filters are broadly categorised as Finite Impulse Response (FIR) or Infinite Impulse Response (IIR) filters.

The composite audio filter generally includes a bank of the sample filters. The bank of filters together define a frequency bandwidth representative of the audio signal or spectrum to be filtered. In this embodiment an impulse response is produced by an impulse fed to the respective sample filters. The impulse response for each of the sample filters may be represented by a sine function according to the equation:

e ^~iqxy Sin[2nx/lpf]

2 πχ

Equation 1

where Ipf is the corner frequency for the lowpass filter, x is the time variable on the x- axis, and e-^ ² represents an averaging curve with q representing the aspect ratio of the averaging curve. It is to be understood that the sine function is the sum of cos components.

Figure 2 illustrates an averaging curve having a width of around four (4) samples applied to an impulse response having a relatively high frequency to provide an averaged impulse response. Figure 3 shows an adjusted averaging curve having a width of around eight (8) samples applied to another impulse response having a relatively low frequency to provide another averaged impulse response. It can be seen that in both cases the width or q of the averaging curve is substantially proportional to the frequency of the corresponding impulse response. This is schematically shown in figure 4 where the width of the averaging curve increases in the z-axis with decreasing frequency in the impulse response. Figure 5 schematically illustrates combining or in this case summing the averaged impulse responses of figures 2 and 3 to provide the composite audio filter. It is to be understood that a[0] is the instance at which the impulse occurs and a[n] designates neighbouring sample points for the impulse response where n is the number of the sample point at the predetermined sample rate. In this embodiment the predetermined sample rate is 44.1 kHz (samples per second) although it will be appreciated that any other sample rate may be used depending on the application.

Figure 6 illustrates an enlarged view of the composite audio filter of Figure 5 with a sample rate increase to the increased sample rate. For illustrative purposes only the predetermined sample rate is increased by a factor of ten (10) with nine (9) intermediate and equally spaced sample points designated a[0a] to a[0i] located between neighbouring sample points such as a[0] and a[1]. The predetermined sample rate may in practice be increased by a factor of up to 1 ,000 where the increased sample rate is 44,100 kHz.

For a bandpass composite filter the averaged impulse responses as illustrated in figure 7 are combined with one another by summing values across an infinite number of sample points to obtain one of the composite audio filters. This can be mathematically defined by the equation: d _lpf

Equation 2 Figure 8 shows a frequency response for cos components of a bandpass filter where the averaging curve is according to an embodiment of the invention adjusted proportionally. Figure 9 illustrates for comparative purposes a frequency response for a cos component of a bandpass filter without adjusting the averaging curve. It can be seen that the filter frequency shifts away from the frequency of the component filter and the filter shape approaches the shape of a lowpass filter at the Nyquist frequency. On the other hand the frequency response of figure 8 with the adjusted averaging curve is more desirable because it is still consistent with a bandpass filter. In some embodiments, the sample rate increase on each of the audio filters is performed by the following two techniques involving:

1. Shifted neighbouring audio signals; and/or

2. Expanded hypothetical impulse response.

In weighting values of the impulse response using the shifted neighbouring audio signals, neighbouring impulse responses are nominated from either side of the intermediate sample point to be determined. Each of the nominated neighbouring signals is then shifted in the time domain substantially midway between the neighbouring sample point and the intermediate sample point. In this example the relevant weighting is calculated by summing values which each of the shifted neighbouring impulse responses contribute at the relevant intermediate sample point. This technique is schematically illustrated in figure 10. In some embodiments, the weighting may be applied across a predetermined number of the neighbouring sample points, for example 1 ,024 sample points.

In using this weighting technique, combining of the audio filters is performed at the adjusted sampling rate so that neighbouring sample points for the audio filter align or correspond with at least each of the intervening sample points of the other audio filter to which it is applied. This involves shifting the audio filter at the adjusted sampling rate relative to the other audio filter. For example, if the other audio filter includes intervening sample points located substantially midway between adjacent of its neighbouring sample points, the adjusted sampling rate for applying the filters to one another is substantially half the predetermined sample rate. Figure 11 schematically illustrates this technique for adjusting the sampling rate.

The sampling rate is adjusted in this embodiment by convolving every other impulse response. This means the uppermost impulse response of figure 11 is convolved with the three (3) impulse responses shown in solid line detail and the other impulse responses shown in broken line detail are effectively ignored. The resulting or composite audio filter is the lowermost impulse response of figure 11 shown in broken line detail and can in this example be represented by the following equations.

^« Equations 3

For a predetermined sample rate of 44.1 kHz the adjusted sampling rate in this example is 22.05 kHz. If the other audio sample filter includes nine (9) intervening sample points between adjacent of its neighbouring sample points the adjusted sampling rate will be one tenth of the predetermined sample rate. This equates to an adjusted sampling rate of 4.41 kHz for a predetermined sample rate of 44.1 kHz. It is understood that adjusting the sampling rate "corrects" for shifting of the nominated neighbouring sample points in calculating weightings for each of the intermediate sample points. The shift in the nominated neighbouring samples in the time domain is generally proportional to the adjustment in the sampling rate in convolving the audio filters. Thus, a shift in the nominated neighbouring samples midway between neighbouring sample point and the intermediate sample point means an adjustment in the sampling rate by a factor of substantially one-half.

In weighting values of the impulse response using the expanded hypothetical impulse response, the relevant impulse response is effectively replicated as a hypothetical impulse response with its time domain shifted to align with the intermediate sample point to be determined. In some embodiments, the hypothetical and impulse response is expanded in its time domain by a factor of substantially 2. In this example the relevant weighting is calculated by summing values for the expanded impulse response at the neighbouring sample points. This technique is schematically illustrated in figure 12. In some embodiments, the weighting may be applied across a predetermined number of the neighbouring sample points, for example 1 ,024 sample points. In alternative embodiments the composite bandpass filter may undergo a sample rate increase by the following two (2) techniques involving i) a hypothetical audio signal, and/or ii) neighbouring audio signals. In weighting values of the impulse response using the hypothetical audio signal, the relevant impulse response is effectively replicated with its time domain shifted to align with the intermediate sample point to be determined. The weighting is calculated by summing values for the hypothetical audio signal at the neighbouring sample points and the weighting is a factor inversely proportional to the sum of these values. The relevant weighting or factor is applied to the impulse response of the filter at respective of the intermediate sample points. This technique is schematically illustrated in Figure 13. The weighting may be calculated across a predetermined number of the neighbouring sample points. The following equation provides for weighting of values for each of the intermediate sample points where 1024 neighbouring sample points are taken into account:

where n is the sample number, q represents the aspect ratio of the averaging and Ipf is the corner frequency for the lowpass filter.

Figure 14 illustrates the weightings for each of the intermediate sample points a[0a] to a[0i] to be applied to the value of the relevant impulse response.

In weighting values of the impulse response using the neighbouring audio signals, neighbouring impulse responses are nominated from either side of the intermediate sample point to be determined. In this example the relevant weighting is calculated by summing values which each of the nominated neighbouring impulse responses contribute at the relevant intermediate sample point. This technique is schematically illustrated in Figure 15. In some embodiments, the weighting may be applied across a predetermined number of the neighbouring sample points, for example 1024 sample points.

Figure 16 depicts the frequency response of a Iowpass filter comprising the sum of cos components with proportional averaging curves. It can be seen that a range of standard Iowpass filters can be constructed with proportional curves. The composite audio filter is in this example a Iowpass filter which approaches the Nyquist frequency. The Nyquist frequencies and above are substantially removed in performing the sample rate increase on the composite audio filter. The composite filter or other composite filters may also function as bandpass or bandreject filter depending on the application.

The composite audio filter is applied to the audio signal with the benefit of increased accuracy at the increased sample rate. Alternatively the sample rate increase may be applied to each of the averaged impulse responses and the composite filter returned to the predetermined sample rate prior to filtering of the audio signal. The composite filter is thus applied to the audio signal at the predetermined sample rate with a virtual sample rate increase which may be less demanding in terms of processor power. Now that several embodiments of the present disclosure have been described it will be apparent to those skilled in the art that the method of digitally filtering an audio signal has at least the following advantages over the prior art:

1. The composite audio filter may be derived by applying adjusted averaging curves to each of the plurality of sample filters which provides a relatively "smooth" filter in its frequency response;

2. The composite filter provides improved filtering in for example EQ;

3. The composite audio filter substantially reduces unwanted resonants inherent in analog and prior digital filters;

4. It provides a frequency response which is smoother and in this respect more akin to an analog filter. Those skilled in the art will appreciate that the invention described herein is susceptible to variations and modifications other than those specifically described. For example, the impulse response may be of practically any waveform. If represented by a mathematical equation, the impulse response is not limited to a sine function but includes other waveforms such as: i) a sine function of absolute values represented in the time domain;

ϋ) a sine function of values from zero (0) to positive infinity only; iii) a sine function (sum of cosine components) for positive values only.

The composite audio filter may be constructed by combining averaged impulse responses of iii) and iv). In this variation, the width of the averaging curve is adjusted proportional to the frequency of the respective impulse response or sample filter. The filter is thus in the time domain represented by positive values only effectively as a 1/2 impulse.

The processing of 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 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. The invention also covers digital filtering of signals associated with digital communications.

The invention in another embodiment is applied to imaging. For example, each of the pixels in a matrix of pixels in the image is processed with a sample rate increase. In increasing the sample rate to include intermediate points, these intermediate points are weighted depending on the influence of neighbouring sample points. 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.