WO2016179647A1 - Attaque normalisée dans un traitement de signal numérique - Google Patents

Attaque normalisée dans un traitement de signal numérique Download PDF

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Publication number
WO2016179647A1
WO2016179647A1 PCT/AU2016/050342 AU2016050342W WO2016179647A1 WO 2016179647 A1 WO2016179647 A1 WO 2016179647A1 AU 2016050342 W AU2016050342 W AU 2016050342W WO 2016179647 A1 WO2016179647 A1 WO 2016179647A1
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WO
WIPO (PCT)
Prior art keywords
audio signal
normalised
threshold value
transient components
values
Prior art date
Application number
PCT/AU2016/050342
Other languages
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 WO2016179647A1 publication Critical patent/WO2016179647A1/fr

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Classifications

    • H—ELECTRICITY
    • H03—ELECTRONIC CIRCUITRY
    • H03G—CONTROL OF AMPLIFICATION
    • H03G7/00—Volume compression or expansion in amplifiers
    • H03G7/007—Volume compression or expansion in amplifiers of digital or coded signals
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L21/00—Speech or voice signal processing techniques to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
    • G10L21/02—Speech enhancement, e.g. noise reduction or echo cancellation
    • G10L21/0316—Speech enhancement, e.g. noise reduction or echo cancellation by changing the amplitude
    • G10L21/0324—Details of processing therefor
    • G10L21/0332—Details of processing therefor involving modification of waveforms

Definitions

  • the present invention relates broadly to a method of processing a digital signal and relates particularly, although not exclusively, to a method of compressing 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 or filter, and a computer system for processing a digital signal or filter.
  • 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 compression of the signal.
  • a method of digitally processing an audio signal comprising the steps of:
  • values of the modified audio signal are adjusted for a smooth transition into the normalised attack transient component. More preferably values of the modified audio signal are adjusted for a smooth transition out of the normalised attack transient component.
  • attack transient components are normalised by adjusting their respective peak values to a maximum standardised value.
  • the step of normalising the attack transient components involves applying a normalising algorithm to each of the transient components to progressively adjust its values above the threshold value and toward the maximum predetermined values. More preferably application of the normalising algorithm adjusts said transient values of the transient components substantially proportional to the adjustment of the peak values to the maximum predetermined values.
  • the method also comprises the step of gradually releasing the normalised transient component from its respective maximum predetermined value to the predetermined threshold value. More preferably a release algorithm is applied to each of the normalised transient components to effect their gradual release to the threshold value.
  • the method also comprises the step of combining the
  • a computer or device-readable medium including instructions for processing an audio signal, said instructions when executed by a processor cause said processor to:
  • a computer or device-readable medium including instructions for processing an audio signal, said instructions when executed by a processor cause said processor to:
  • a computer system for processing an audio signal comprising a processor configured to:
  • a computer system for processing an audio signal comprising a processor configured to:
  • attack transient components of the audio signal above a predetermined threshold value; normalise the attack transient components by adjusting at least their respective peak values to maximum predetermined values;
  • 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 signal compression and associated signal processing steps according to one embodiment of the present invention.
  • FIG. 3 is a schematic illustration of signal compression and associated signal processing steps according to another embodiment of the invention.
  • 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
  • FIG 2 there is provided a method of digitally processing an audio signal S, the method comprising the steps of: 1 . applying a predetermined time delay x in compressing the audio signal S to a predetermined threshold value T, see figure 2a;
  • the compressed audio signal CS with the predetermined time delay x is in the form of a modified audio signal, see figure 2b;
  • the modified audio signal includes one or more attack transient
  • attack transient components ATC are each normalised by adjusting their respective peak value PV to a maximum predetermined value or normalised value N, see figure 2c.
  • the normalised value N is the same standardised maximum value for each of the attack transient components ATC.
  • the method may also comprise the step of gradually releasing the normalised transient component NTC from the maximum value N to the predetermined threshold value T, see figure 2d. The gradual release is for a predetermined release time y.
  • the normalised signal NS is thus derived from these processing steps
  • the time delay x and the release time y are generally the same for each of the attack transient components ATC and the normalised transient components NTC respectively.
  • the time delay x and/or the release time y may vary depending on characteristics of the audio signal S, for example its frequency or wavelength.
  • the signal S or its representative components such as its cosine and/or sine components, may at high frequencies (relatively short wavelengths) be compressed with relatively short time delays with relatively short release times.
  • the signal S or its representative waveform components at relatively low frequencies (relatively long wavelengths) may be compressed with relatively long time delays and release times.
  • the step of normalising the attack transient components ATC involves applying a normalising algorithm to each of the components.
  • the normalising algorithm is configured to progressively adjust transient values TV of the attack transient component ATC above the threshold value T and toward the maximum or normalised value N.
  • the normalising algorithm may adjust the transient values such as TV substantially proportional to adjustment of the peak value PV to the normalised value N.
  • the values of the modified audio signal may be adjusted for a smooth transition into the normalised attack transient component.
  • the gradual release of the normalised transient component NTC may be performed by application of a release algorithm.
  • the release algorithm can be applied to each of the normalised transient components NTC to effect their gradual release to the threshold value T.
  • the values of the released components together with other values of the audio signal may be adjusted for a smooth transition out of the normalised attack transient component.
  • attack transient components ATC are obtained by applying a predetermined time delay " in compressing the audio signal S' to the predetermined threshold value T'.
  • This time delay " may be fixed or may vary depending on characteristics of the audio signal S ⁇ for example its frequency or wavelength.
  • the step of normalising the attack transient components ATC is in line with the preceding embodiment involving the application of a normalising algorithm to each of the components.
  • This method may also comprise the step of gradually releasing the normalised transient component NTC from the maximum value N' to the predetermined threshold value T'. This gradual release may similarly be performed by application of a release algorithm.
  • the method may comprise an additional step of combining the normalised signal NS derived from the preceding processing steps with the audio signal S.
  • the signals may be combined or blended at equal or other ratios.
  • the signal may additionally be processed before and/or after the
  • sample rate increases to the signal using for example techniques disclosed in the applicant's international patent application no.'s
  • PCT/AU2014/000325 directed to audio filtering at adjusted sampling rates
  • PCT/AU2014/000321 directed to audio filtering with proportional or adjusted averaging curves
  • PCT/AU2015/000197 directed to digital filtering with sample zoning and the combination of sample values depending on the frequency of the zone
  • 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 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 processing of signals associated with digital communications.
  • the invention in another embodiment is applied to imaging.
  • the signal(s) or filter(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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  • Compression, Expansion, Code Conversion, And Decoders (AREA)

Abstract

L'invention concerne un procédé de traitement numérique d'un signal audio, ledit procédé comprenant les étapes suivantes : appliquer un décalage de temps prédéterminé pour comprimer le signal audio en une valeur de seuil prédéterminée pour obtenir un signal audio modifié comprenant une ou plusieurs composantes transitoires d'attaque au-dessus de la valeur de seuil ; normaliser les composantes transitoires d'attaque par réglage au moins de leurs valeurs de crête respectives à des valeurs prédéterminées maximales pour obtenir un signal normalisé.
PCT/AU2016/050342 2015-05-08 2016-05-08 Attaque normalisée dans un traitement de signal numérique WO2016179647A1 (fr)

Applications Claiming Priority (2)

Application Number Priority Date Filing Date Title
US201562159034P 2015-05-08 2015-05-08
US62/159,034 2015-05-08

Publications (1)

Publication Number Publication Date
WO2016179647A1 true WO2016179647A1 (fr) 2016-11-17

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Family Applications (1)

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PCT/AU2016/050342 WO2016179647A1 (fr) 2015-05-08 2016-05-08 Attaque normalisée dans un traitement de signal numérique

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WO (1) WO2016179647A1 (fr)

Citations (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US6091013A (en) * 1998-12-21 2000-07-18 Waller, Jr.; James K. Attack transient detection for a musical instrument signal
US20080212795A1 (en) * 2003-06-24 2008-09-04 Creative Technology Ltd. Transient detection and modification in audio signals
US20140185833A1 (en) * 2012-12-27 2014-07-03 Canon Kabushiki Kaisha Audio processing apparatus and method

Patent Citations (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US6091013A (en) * 1998-12-21 2000-07-18 Waller, Jr.; James K. Attack transient detection for a musical instrument signal
US20080212795A1 (en) * 2003-06-24 2008-09-04 Creative Technology Ltd. Transient detection and modification in audio signals
US20140185833A1 (en) * 2012-12-27 2014-07-03 Canon Kabushiki Kaisha Audio processing apparatus and method

Non-Patent Citations (5)

* Cited by examiner, † Cited by third party
Title
"Chapter 5. Digital Audio Processing Part I: Sec. 5.1-5.3", DEPARTMENT OF COMPUTER SCIENCE, NATIONAL TSING HUA UNIVERSITY., 17 August 2014 (2014-08-17), XP055330587, Retrieved from the Internet <URL:http://cv.cs.nthu.edu.tw/upload/courses/14/uploads/CS3570_Chapter5_part1.pdf> *
MASRI, P ET AL.: "Improved Modelling of Attack Transients in Music Analysis- Resynthesis", PROCEEDINGS OF THE 1996 INTERNATIONAL COMPUTER MUSIC CONFERENCE, 1996, pages 100 - 103, XP007901916 *
PARKER, J. ET AL.: "Linear Dynamic Range Reduction of Musical Audio Using an Allpass Filter Chain", IEEE SIGNAL PROCESSING LETTERS, vol. 20, no. 7, July 2013 (2013-07-01), pages 669 - 672, XP011510988 *
ZAUNSCHIRM, M. ET AL.: "A Sub-Band Approach to Modification of Musical Transients", COMPUTER MUSIC JOURNAL, vol. 36, no. 2, June 2012 (2012-06-01), pages 23 - 36, XP055330588 *
ZOLZER, U.: "Digital Audio Signal Processing", 2008, Hamburg, Germany *

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