AUDIO PROCESSING WITH MODIFIED CONVOLUTION

• US 20170250676A1
• Filed: 09/26/2015
• Published: 08/31/2017
• Est. Priority Date: 09/26/2014
• Status: Abandoned Application

• Current Assignee: Lachlan Barratt
• Original Assignee: Lachlan Barratt
• Inventors: BARRATT, Lachlan
• Application Number:
  US15/514,488
• Publication Number:
  US 20170250676A1
• Time in Patent Office: Days
• CPC Class Codes: H03H 17/0248 Filters characterised by a particular frequency response or filtering method
• CPC Class Codes: H03H 17/06 Non-recursive filters
• CPC Class Codes: H03H 17/0621 with input-sampling frequency and output-delivery frequency which differ, e.g. extrapolation; Anti-aliasing
• CPC Class Codes: H03H 17/0657 where the output-delivery frequency is higher than the input sampling frequency, i.e. interpolation

Abstract

A method of processing a digital signal includes providing a digital filter including neighbouring sample points and performing a sample rate increase on the digital filter to provide intermediate sample points between adjacent neighbouring sample points, said intermediate points being populated dependent on a weighted influence determined in the time domain of a predetermined number of the neighbouring sample points. The digital filter is applied to the signal where: i) one of the neighbouring sample points of the filter is applied to a corresponding sample point of the signal; ii) offset and neighbouring sample points of the signal are defined either side of the corresponding sample point, said offset points being offset in the time domain relative to the respective neighbouring sample points of the filter; and iii) the neighbouring sample points of the filter are applied to respective of the offset and neighbouring sample points of the signal.

Claims

1. A method of processing a digital signal or filter, said method comprising the steps of:

providing a digital filter or signal including a plurality of neighbouring sample points;

performing a sample rate increase on the digital filter or signal to provide a plurality of intermediate sample points between adjacent of the neighbouring sample points, said intermediate points being populated dependent on a weighted influence determined in the time domain of a predetermined number of the neighbouring sample points;

applying the digital filter to the signal or vice versa where;

i) one of the neighbouring sample points of the filter or signal is applied to a corresponding sample point of the signal or filter, respectively;

ii) offset and neighbouring sample points of the signal or filter are defined either side of the corresponding sample point of the signal or filter, said offset points being offset in the time domain relative to the respective neighbouring sample points of the filter or signal; and

iii) the neighbouring sample points of the filter or signal are applied to respective of the offset and neighbouring sample points of the signal or filter, respectively.

2. A method as defined in claim 1 also comprising an initial step of selecting an initial digital filter having a sample resolution substantially the same as the signal, and expanding the selected initial filter to effect offset of the offset sample points of the signal relative to the respective neighbouring sample points of the filter.

3. A method as defined in claim 1 wherein the step of applying the digital filter to the signal involves applying said filter at an adjusted sampling rate proportional to the offset of the sample points of the signal relative to the respective neighbouring sample points of the filter.

4. A method as defined in claim 1 wherein the filter is at least in part represented by an impulse response produced by an impulse fed to said filter and wherein the neighbouring sample points of the impulse response are offset in the time domain relative to the respective sample points of the signal.

20. A method as defined in claim 1 also comprising an initial step of including additional sample points in the audio signal between the neighbouring sample points to effect time-stretching of the audio signal prior to its application to the filter.

22. A method as defined in claim 1 wherein the step of performing a sample rate increase involves calculating the weighted influence for each of the intermediate sample points by:

(i) representing the signal or the 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) determining values for each of the representative waveforms at the intermediate sample point;

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

23. A method as defined in claim 1 wherein the step of performing a sample rate increase involves calculating the weighted influence for each of the intermediate sample points by:

representing the signal or the filter at one of 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 to obtain a representative waveform at the neighbouring sample point;

(iii) shifting the representative waveform in its time domain to align with 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.

24. A method as defined in claim 1 wherein the step of performing a sample rate increase involves calculating the weighted influence for each of the intermediate sample points by:

(i) representing the signal or the 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 points and 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 point to derive the weighted influence.

25. A method as defined in claim 1 wherein the step of performing a sample rate increase involves calculating the weighted influence for each of the intermediate sample points by:

(i) representing the signal or the filter at one of 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 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.

5. (canceled)

6. A method of processing a digital signal, said method comprising the steps of:

providing another digital signal including a plurality of neighbouring sample points;

performing a sample rate increase on the other digital signal to provide a plurality of intermediate sample points between adjacent of the neighbouring sample points, said intermediate points being populated dependent on a weighted influence determined in the time domain of a predetermined number of the neighbouring sample points;

applying the other signal to the signal where;

i) one of the neighbouring sample points of the other signal is applied to a corresponding sample point of the signal;

ii) offset and neighbouring sample points of the signal are defined either side of the corresponding sample point of the signal, said offset points being offset in the time domain relative to the respective neighbouring sample points of the other signal; and

iii) the other of the neighbouring sample points of the other signal are each applied to respective of the offset and neighbouring sample points of the signal.

7. A method as defined in claim 6 also comprising an initial step of expanding the other signal in its time domain to effect offset of the offset sample points relative to the respective neighbouring sample points of the signal.

8. A method as defined in claim 6 further comprising the step of representing the other signal in the time domain by at least its cosine and sine components.

9. A method as defined in claim 8 wherein the cosine and/or sine components are replaced at least in part with square-shaped wave components for valve-type emulation on application to the signal.

10. A method as defined in claim 8 wherein the cosine and/or sine components are replaced at least in part with triangular-shaped wave components for transistor-type emulation on application to the signal.

12. A method as defined in claim 6 wherein the other signal includes random noise represented by discrete values of a mathematical function to be applied to the signal.

13. A method as defined in claim 12 wherein one or more of the discrete values of the random noise is replaced with either a zero (0) or one (1).

14. A method as defined in claim 6 wherein the step of applying the other signal to the signal involves applying said other signal at an adjusted sampling rate proportional to the offset of the offset sample points.

15. A method as defined in claim 6 also comprising a step of deriving or constructing a modulation envelope to be applied to the signal as the other signal.

11. (canceled)

16. (canceled)

17. (canceled)

18. (canceled)

19. (canceled)

21. A method as defined in claim 19 wherein the method involves applying the signal to the filter at an adjusted sampling rate to effect pitch-shifting of another audio signal to be filtered by the digital filter.

26. (canceled)

27. A non-transitory computer or device-readable medium including instructions for processing a digital signal or filter, said instructions when executed by a processor cause said processor to:

provide a digital filter or signal including a plurality of neighbouring sample points;

perform a sample rate increase on the digital filter or signal to provide a plurality of intermediate sample points between adjacent of the neighbouring sample points, said intermediate points being populated dependent on a weighted influence determined in the time domain of a predetermined number of the neighbouring sample points;

apply the digital filter to the signal or vice versa where

i) one of the neighbouring sample points of the filter or signal is applied to a corresponding sample point of the signal or filter, respectively;

ii) offset and neighbouring sample points of the signal or filter are defined either side of the corresponding sample point of the signal or filter, said offset points being offset in the time domain relative to the respective neighbouring sample points of the filter or signal; and

iii) the neighbouring sample points of the filter or signal are applied to respective of the offset and neighbouring sample points of the signal or filter, respectively.

28. (canceled)

29. (canceled)

30. (canceled)