WO2016179648A1 - Contrôle de valeurs dynamiques dans des signaux numériques - Google Patents

Contrôle de valeurs dynamiques dans des signaux numériques Download PDF

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Publication number
WO2016179648A1
WO2016179648A1 PCT/AU2016/050343 AU2016050343W WO2016179648A1 WO 2016179648 A1 WO2016179648 A1 WO 2016179648A1 AU 2016050343 W AU2016050343 W AU 2016050343W WO 2016179648 A1 WO2016179648 A1 WO 2016179648A1
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WO
WIPO (PCT)
Prior art keywords
signal
frequency filter
sample points
digital signal
neighbouring
Prior art date
Application number
PCT/AU2016/050343
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English (en)
Inventor
Lachlan BARRATT
Original Assignee
Barratt Lachlan
Priority date (The priority date is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the date listed.)
Filing date
Publication date
Application filed by Barratt Lachlan filed Critical Barratt Lachlan
Publication of WO2016179648A1 publication Critical patent/WO2016179648A1/fr

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    • G—PHYSICS
    • G11—INFORMATION STORAGE
    • G11B—INFORMATION STORAGE BASED ON RELATIVE MOVEMENT BETWEEN RECORD CARRIER AND TRANSDUCER
    • G11B20/00—Signal processing not specific to the method of recording or reproducing; Circuits therefor
    • G11B20/10—Digital recording or reproducing
    • G11B20/10009—Improvement or modification of read or write signals

Definitions

  • 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.
  • 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).
  • ADC analog to digital converter
  • DAC digital to analog converter
  • 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.
  • 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.
  • 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:
  • 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;
  • 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
  • Preferably application of the modulation curve to the digital signal involves convolving the modulation curve with the signal.
  • application of the modulation curve to the signal involves application according to a predetermined algorithm.
  • 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.
  • the demodulated signal is substantially absent of dynamic values.
  • a computer or device readable medium including instructions for processing a digital signal, said instructions when executed by a processor cause said processor to:
  • a computer system for processing a digital signal comprising a processor configured to:
  • 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
  • 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.
  • FIG. 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.
  • ADC analog to digital converter
  • 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.
  • DAC digital to analog converter
  • 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.
  • 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.
  • 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.
  • 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.
  • FIG. 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
  • 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.
  • the frequency filter 14 is represented by a sine function which is the sum of the cosine components in the time domain.
  • /p is the corner frequency for a lowpass filter
  • x is the time variable on the x- axis
  • q represents the aspect ratio of the averaging curve.
  • the frequency filter may be represented by other waveforms constructed from components represented by mathematical functions including:
  • Absolute values of a cosine function in the time domain substantially limited to half a waveform cycle at its mid-point.
  • 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
  • PCT/AU2014/000317 describes in some detail waveforms constructed from
  • the waveform components may also be adjusted by for example applying a mathematical function to each of the components or the waveforms themselves.
  • 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.
  • 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.
  • 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.
  • 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.
  • this dot product methodology involves:
  • 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:
  • Figure 5A schematically illustrates representative waveforms at
  • 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 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.
  • 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].
  • 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.
  • 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.
  • 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.
  • 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:
  • 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 filter or composite filter is generally a low frequency filter at less than an audible frequency of around 20 Hz.
  • 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.
  • the computer 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.
  • the invention also covers digital processing of signals associated with digital communications.
  • the invention in another embodiment is applied to imaging.
  • 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.
  • 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.
  • FFT fast fourier transform

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  • Engineering & Computer Science (AREA)
  • Signal Processing (AREA)
  • Complex Calculations (AREA)

Abstract

L'invention concerne un procédé de traitement numérique d'un signal, ledit procédé comprenant les étapes consistant à : appliquer un filtre de fréquence au signal numérique pour dériver une courbe de modulation représentant des changements de valeur du signal; appliquer la courbe de modulation au signal numérique pour démoduler les changements de valeur du signal afin d'obtenir un signal démodulé.
PCT/AU2016/050343 2015-05-08 2016-05-08 Contrôle de valeurs dynamiques dans des signaux numériques WO2016179648A1 (fr)

Applications Claiming Priority (2)

Application Number Priority Date Filing Date Title
US201562159027P 2015-05-08 2015-05-08
US62/159,027 2015-05-08

Publications (1)

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WO2016179648A1 true WO2016179648A1 (fr) 2016-11-17

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Citations (5)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US6289367B1 (en) * 1998-11-16 2001-09-11 Texas Instruments Incorporated Digital signal processing circuits, systems, and method implementing approximations for logarithm and inverse logarithm
US6640237B1 (en) * 1999-07-27 2003-10-28 Raytheon Company Method and system for generating a trigonometric function
US20090116357A1 (en) * 2007-11-01 2009-05-07 Canon Kabushiki Kaisha Reproducing apparatus
US20120213375A1 (en) * 2010-12-22 2012-08-23 Genaudio, Inc. Audio Spatialization and Environment Simulation
WO2014153604A1 (fr) * 2013-03-26 2014-10-02 Barratt Lachlan Paul Filtres audio utilisant des fonctions sinusoïdales

Patent Citations (5)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US6289367B1 (en) * 1998-11-16 2001-09-11 Texas Instruments Incorporated Digital signal processing circuits, systems, and method implementing approximations for logarithm and inverse logarithm
US6640237B1 (en) * 1999-07-27 2003-10-28 Raytheon Company Method and system for generating a trigonometric function
US20090116357A1 (en) * 2007-11-01 2009-05-07 Canon Kabushiki Kaisha Reproducing apparatus
US20120213375A1 (en) * 2010-12-22 2012-08-23 Genaudio, Inc. Audio Spatialization and Environment Simulation
WO2014153604A1 (fr) * 2013-03-26 2014-10-02 Barratt Lachlan Paul Filtres audio utilisant des fonctions sinusoïdales

Non-Patent Citations (1)

* Cited by examiner, † Cited by third party
Title
ORFANDIS, S. J.: "Introduction to Signal Processing", 2010, ISBN: 0-13-209172-0, article "Interpolation, Decimation, and Oversampling - Chapter 12", pages: 632 - 712, XP055329968 *

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