EP4131264A1 - Verarbeitung von digitalen audiosignalen - Google Patents

Verarbeitung von digitalen audiosignalen Download PDF

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Publication number
EP4131264A1
EP4131264A1 EP21190169.9A EP21190169A EP4131264A1 EP 4131264 A1 EP4131264 A1 EP 4131264A1 EP 21190169 A EP21190169 A EP 21190169A EP 4131264 A1 EP4131264 A1 EP 4131264A1
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EP
European Patent Office
Prior art keywords
sampling frequency
channel
section
insert
working
Prior art date
Legal status (The legal status 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 status listed.)
Withdrawn
Application number
EP21190169.9A
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English (en)
French (fr)
Inventor
Christoph Musialik
Friedemann TISCHMEYER
Current Assignee (The listed assignees may be inaccurate. Google has not performed a legal analysis and makes no representation or warranty as to the accuracy of the list.)
Maat Labs GmbH
Original Assignee
Maat Labs GmbH
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 Maat Labs GmbH filed Critical Maat Labs GmbH
Priority to EP21190169.9A priority Critical patent/EP4131264A1/de
Publication of EP4131264A1 publication Critical patent/EP4131264A1/de
Withdrawn legal-status Critical Current

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Classifications

    • 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
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10H—ELECTROPHONIC MUSICAL INSTRUMENTS; INSTRUMENTS IN WHICH THE TONES ARE GENERATED BY ELECTROMECHANICAL MEANS OR ELECTRONIC GENERATORS, OR IN WHICH THE TONES ARE SYNTHESISED FROM A DATA STORE
    • G10H7/00—Instruments in which the tones are synthesised from a data store, e.g. computer organs
    • G10H7/002—Instruments in which the tones are synthesised from a data store, e.g. computer organs using a common processing for different operations or calculations, and a set of microinstructions (programme) to control the sequence thereof
    • G10H7/006—Instruments in which the tones are synthesised from a data store, e.g. computer organs using a common processing for different operations or calculations, and a set of microinstructions (programme) to control the sequence thereof using two or more algorithms of different types to generate tones, e.g. according to tone color or to processor workload
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10H—ELECTROPHONIC MUSICAL INSTRUMENTS; INSTRUMENTS IN WHICH THE TONES ARE GENERATED BY ELECTROMECHANICAL MEANS OR ELECTRONIC GENERATORS, OR IN WHICH THE TONES ARE SYNTHESISED FROM A DATA STORE
    • G10H7/00—Instruments in which the tones are synthesised from a data store, e.g. computer organs
    • G10H7/02—Instruments in which the tones are synthesised from a data store, e.g. computer organs in which amplitudes at successive sample points of a tone waveform are stored in one or more memories
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10H—ELECTROPHONIC MUSICAL INSTRUMENTS; INSTRUMENTS IN WHICH THE TONES ARE GENERATED BY ELECTROMECHANICAL MEANS OR ELECTRONIC GENERATORS, OR IN WHICH THE TONES ARE SYNTHESISED FROM A DATA STORE
    • G10H2250/00—Aspects of algorithms or signal processing methods without intrinsic musical character, yet specifically adapted for or used in electrophonic musical processing
    • G10H2250/541—Details of musical waveform synthesis, i.e. audio waveshape processing from individual wavetable samples, independently of their origin or of the sound they represent
    • G10H2250/611—Waveform decimation, i.e. integer division of the sampling rate for reducing the number of samples in a discrete-time signal, e.g. by low-pass anti-alias filtering followed by the actual downsampling
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10H—ELECTROPHONIC MUSICAL INSTRUMENTS; INSTRUMENTS IN WHICH THE TONES ARE GENERATED BY ELECTROMECHANICAL MEANS OR ELECTRONIC GENERATORS, OR IN WHICH THE TONES ARE SYNTHESISED FROM A DATA STORE
    • G10H2250/00—Aspects of algorithms or signal processing methods without intrinsic musical character, yet specifically adapted for or used in electrophonic musical processing
    • G10H2250/541—Details of musical waveform synthesis, i.e. audio waveshape processing from individual wavetable samples, independently of their origin or of the sound they represent
    • G10H2250/621—Waveform interpolation
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10H—ELECTROPHONIC MUSICAL INSTRUMENTS; INSTRUMENTS IN WHICH THE TONES ARE GENERATED BY ELECTROMECHANICAL MEANS OR ELECTRONIC GENERATORS, OR IN WHICH THE TONES ARE SYNTHESISED FROM A DATA STORE
    • G10H2250/00—Aspects of algorithms or signal processing methods without intrinsic musical character, yet specifically adapted for or used in electrophonic musical processing
    • G10H2250/541—Details of musical waveform synthesis, i.e. audio waveshape processing from individual wavetable samples, independently of their origin or of the sound they represent
    • G10H2250/631—Waveform resampling, i.e. sample rate conversion or sample depth conversion

Definitions

  • Embodiments of the invention relate to a method for processing a digital audio signal, a non-transitory computer readable medium, a computer program product and an apparatus for processing or generating a digital audio signal.
  • the apparatus can be a digital audio workstation or an audio network or any other digital signal processing system e.g. a video digital processing system.
  • a method for processing a digital audio signal comprises the following method steps: providing a channel for processing the digital audio signal or a part of the digital audio signal, wherein the channel is working on a predefined or predefinable base sampling frequency, characterized in that the base sampling frequency is increased to a working sampling frequency in a predefinable section of the channel and at the end of the section the working sampling frequency is decreased back to the base sampling frequency.
  • the method for processing a digital audio signal can be performed on a digital audio workstation (in the following: DAW), in a virtual sub-mixers within virtual instruments, in virtual instruments as well as on digital hardware mixing desks or digital mixers.
  • DAW digital audio workstation
  • the method can also be used in other multichannel digital processing systems working generally with single sampling frequencies, but using up-sampling or oversampling for dedicated modules or plugins e.g., video processing systems.
  • a channel is a channel of a multiple of channels being used for processing a digital audio signal.
  • a DAW there are used a multiple of channels for processing different aspects of audio signals.
  • one channel can be used for a voice recording and processing.
  • Another channel can be used for an electric guitar, a further channel for a bass.
  • Another channel can be a virtual instrument.
  • the bass drum can be processed in one channel, a snare can be processed in another channel, the hi-hat can be processed in a further channel and so on.
  • only some of the channels, which are to be processed, are up-sampled or the sampling frequency is increased in a predefined or pre-definable section of the channel or the channels.
  • Up-sampling of only a selected number of audio processing channels leads to more efficient performance in terms of audio quality and required computational power.
  • only the voice channel or the voice channels, which are situated in the foreground of the audio signal to be processed need to be processed in a higher quality with respect to some plugins that do produce nonlinearities or disturbing frequencies in an audible range.
  • These channels need to be processed on a higher sampling rate to get rid of these audible sound problems.
  • an embodiment is to up-sample just a section of a channel or some few channels of a multiple of channels in the processing of a digital audio signal.
  • the base sampling frequency is the base sampling frequency of the channel.
  • the increase of the base sampling frequency to the higher working sampling frequency or the up-sampling is performed at the beginning of the section in which the up-sampled frequency is used, such that at least two inserted plugins work on this working sampling frequency or the at least two inserted plugins are triggered or driven by the working sampling frequency.
  • the section of the channel includes at least a part of an insert area, wherein in the insert area plugins are inserted or are insertable into the channel.
  • the section of the channel includes a cluster of at least two successive insert interfaces, in which plugins are inserted or insertable.
  • the cluster includes three or more insert interfaces.
  • the more insert interfaces are used within an oversampled channel the better the efficiency due to the invention.
  • plugins like equalizer, compressor, limiter, and soft-clip or tape emulation are used simultaneously in series. In this case 4 local interpolators and decimators can be saved.
  • the cluster includes all insert interfaces of the insert area.
  • the plugins inserted in such oversampled channel work with the increased working sampling frequency without using their own interpolators and decimators.
  • the most of the current plugins can automatically work with higher than standard (i.e., 44.1 kHz or 48 kHz) sampling frequency, especially if they incorporate internal oversampling filters.
  • the section starts at the beginning of the channel or starts after a signal input control.
  • the sampling frequency is increased.
  • the gain and/or phase of the signal is adjusted or adjustable.
  • the section in which oversampling is performed, starts prior an insert interface of the channel.
  • the insert interface is used to insert plugins or processing modules (which can be software or hardware) into the channel for processing the audio signal.
  • a plugin can be a distortion effect, a compressor, an equalizer and/or any other audio processing functionality.
  • the section ends prior to the output.
  • the oversampled sections can be mixed with other system specific modules like gains, faders, internal equalizers, etc.
  • the section ends after the pre-fader send interface or the post-fader send interface, it is preferred to decrease the sampling frequency of the send signal to the base sampling frequency. This means an extra down-sampling compared to the down-sampling that is performed in the channel.
  • the working sampling frequency is increased by a factor of k to the base sampling frequency, wherein k is an integer that is ususally a multiple of 2, but generally can be any factor.
  • a non-transitory computer readable medium including a computer program, executable by a processor or a gate array for performing a method, which is described above is, provided.
  • the computer program can run on a stand-alone computer or any kind of a dedicated embedded computer like a DSP (digital signal processing) engine included in a digital mixing console or a DAW used a specific DSP platform.
  • a DSP digital signal processing
  • an apparatus for processing a digital audio signal comprising multiple audio processing channels, the multiple audio processing channels are working on a predefined or predefinable base sampling frequency, wherein in a predefinable number of channels the base sampling frequency is increased to a working sampling frequency in a predefinable section of the number of channels and after the section or the end of the section the working sampling frequency is decreased back to the base sampling frequency.
  • the apparatus can be a digital audio workstation, a virtual sub-mixer, within a virtual instrument, a digital hardware mixer or a digital mixer.
  • the methods according to embodiments of the invention and the apparatus according to embodiments of the invention can be used for audio processing or for video processing.
  • the apparatus is configured to perform a method, which is described above.
  • the embodiments of the invention use dedicated channels or paths of the audio signal in the audio processing system or the video signal in a video processing system, which allow oversampling and thus the use of plugins or processing modules without switching the whole system to a higher sampling frequency and/or without using local oversampling filters in each plugin.
  • a plugin can be called a processing module.
  • a selected channel or selected channels within an apparatus for processing a digital audio signal which can be switched to higher sampling frequencies by using for example only one interpolator at the input or somewhere in the beginning of the channel and only one decimeter somewhere later in the channel or at the output of the channel are used.
  • plugins or processing modules set up to a desired or predefinable working sampling frequency or oversampling rate can be placed or inserted without using the own local interpolators and decimators of the plugins.
  • an increase of the sampling frequency has the same meaning as an up-sampling or oversampling of the frequency. The same is the case with respect to a decrease of the sampling frequency and a down-sampling of the frequency.
  • a single up-sampling algorithm or module can be used and a single down-sampling algorithm or module can be used in a channel that has a better quality than up-sampling and down-sampling algorithm, normally used in single plugins. This further increases the quality of the processed audio signal. Furthermore, the latency in the channel is much lower than caused by the series of local oversampling filters as well as the computational power and the amount of memory is lower compared to a system working entirely at a higher sampling frequency.
  • the length of the section, in which the working sampling frequency is used can be different in different channels.
  • the length can also be the same for two or more channels.
  • the increase of the sampling frequency or the oversampling factor k can be different for different channels.
  • the oversampling factors can be selected differently in different channels.
  • the section of the channel includes at least a part of an insert area, wherein in the insert area plugins are inserted or are insertable into the channel.
  • the insert area normally has several inserts or insert interfaces, in which plugins can be inserted.
  • an insert interface can be an insert, an insert slot and an insert point.
  • the section starts prior an insert interface.
  • the section starts at the beginning of the channel or after a signal input control.
  • the section of the channel includes a cluster of at least two successive insert interfaces, in which plugins are inserted or insertable.
  • An embodiment include the feature, that at a first insert interface a plugin is inserted or insertable that is enabled to increase the base sampling frequency to the working sampling frequency and to keep the working sampling frequency at the output of the plugin and at the end of the section at another insert interface another plugin is inserted or insertable that is enabled to decrease the working sampling frequency back to the base sampling frequency.
  • the plugin standard can implement that in the section that is selected to be in an up-sampling mode the plugins are not forced to reduce the sampling frequency to the base sampling frequency of the channel.
  • Embodiments according to the invention can fulfill individual characteristics or a combination of several characteristics.
  • Fig. 1 shows a channel 18 according to a digital audio processing system or an apparatus for processing a digital audio signal in a schematic view.
  • the audio channel 18 can be a mono-channel format, a stereo-channel format, a surround-channel format or any multi-channel format configuration for, especially immersive, audio. It can be a part of a digital audio workstation, a virtual instrument plugin or a digital mixing desk.
  • the input signal 10 is shown.
  • the input signal 10 can have any mono or multi-channel configuration and can come from audio inputs, for example, microphones or line sources or from the hard drive of the audio computer.
  • the signal can also come from an internal or external recording device.
  • the signal comes from the instrument itself.
  • the signal comes to the input control, which usually consists or comprises an input trim (gain) and a phase reverse button.
  • the input control usually consists or comprises an input trim (gain) and a phase reverse button.
  • the up-sampling to a working sampling frequency is performed. This is a good point in the signal chain to apply up-sampling. This may vary in different situations.
  • the up-sampling can also be performed prior the input control 11.
  • the up-sampling could also start after a view initial insert interfaces, which then would work in the base or system or apparatus sampling frequency, for example, 44.1 or 48 kHz.
  • the up-sampling 30 could also take place, for example between the first insert interface 21 and the second insert interface 22 or after another insert interface.
  • Fig. 1 not all insert interfaces are shown, but only the first two and the last two of, for example, ten insert interfaces.
  • the insert interfaces 21, 22, 23, 24 are dedicated to insert audio processing tools such as equalizers, dynamic processing tools and so forth.
  • the total number of insert interfaces varies between systems or apparatuses.
  • the down-sampling to the base sampling frequency can take place at different points in the audio signal processing line or the channel 18.
  • the down-sampling is shown by the reference number 31 and in the example of Fig. 1 , there are shown different places for down-sampling 31.
  • a good point, at which the down-sampling to the base sampling frequency can take place, can be within the insert area 20 between two insert interfaces, for example, the insert interface 22 or 23.
  • insert interfaces 22 and 23 Between the insert interfaces 22 and 23, several other insert interfaces can be positioned.
  • a good point for the down-sampling is after insert interfaces in which plugins are inserted, that profit from working on a higher sampling rate or frequency.
  • Other plugins working satisfactory at the base sampling frequency e.g. have a good sound quality if working on the base sampling frequency, can be inserted into insert interfaces, which are positioned after the down-sampling 31.
  • the down-sampling can take place at other positions as shown in Fig. 1 .
  • insert interfaces 11 and 12 After the fader, a further insert interface or insert interfaces 25 are shown, for example, insert interfaces 11 and 12. After this insert interface 25, in which, for example, plugins with reverb-effects or delay-effects or other effects can be implemented, the post-fader sends 17 is implemented. The signal is send transferred to the panorama 14.
  • the post-fader sends 17 usually is used to send signals to FX devices such as reverbs and delays.
  • the post-fader send is within the up-sampling cluster, which means within the section in which the working sampling frequency is used, it is required to have an extra down-sampling processing of the post-fader sends signal to communicate with other processes in the base sampling frequency of the system or apparatus.
  • the output signal 15 may be mono, stereo or any multi-channel configuration for surround or immersive audio.
  • Fig. 1 shows one embodiment of a channel. However different configurations of channels can be used according to the invention.
  • Fig. 2 shows a schematic flowchart of an embodiment of the invention.
  • the signal is shown, which is put into an interpolator 41, which leads to an up-sampling of the base sampling frequency Fs of the signal 40 to a working sampling frequency k*Fs.
  • an interpolator 41 which leads to an up-sampling of the base sampling frequency Fs of the signal 40 to a working sampling frequency k*Fs.
  • a section 43 or cluster 43 which is processed with the working sampling frequency k*Fs.
  • the plugins P1, P2 ... Pn are inserted into insert interfaces.
  • a decimator 42 is used to down sample the working sampling frequency back to the base sampling frequency.
  • This embodiment shows schematically a channel or a part of a channel of a system or an apparatus, e.g. a digital audio workstation, a digital mixing console or an audio matric processor, which incorporates its own interpolator at the beginning or prior of the oversampled section and its own decimator at the end or after the oversampled section. So the interpolator and decimator are provided by system manufacturer as a part of the system in case of channels which can be switched to oversampling mode.
  • Fig. 3 shows another schematic flowchart according to an embodiment of the invention.
  • an interpolator 41 and a decimator 42 as a plugin.

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  • Engineering & Computer Science (AREA)
  • Physics & Mathematics (AREA)
  • Acoustics & Sound (AREA)
  • Multimedia (AREA)
  • General Engineering & Computer Science (AREA)
  • Computational Linguistics (AREA)
  • Quality & Reliability (AREA)
  • Signal Processing (AREA)
  • Health & Medical Sciences (AREA)
  • Audiology, Speech & Language Pathology (AREA)
  • Human Computer Interaction (AREA)
  • Circuit For Audible Band Transducer (AREA)
EP21190169.9A 2021-08-06 2021-08-06 Verarbeitung von digitalen audiosignalen Withdrawn EP4131264A1 (de)

Priority Applications (1)

Application Number Priority Date Filing Date Title
EP21190169.9A EP4131264A1 (de) 2021-08-06 2021-08-06 Verarbeitung von digitalen audiosignalen

Applications Claiming Priority (1)

Application Number Priority Date Filing Date Title
EP21190169.9A EP4131264A1 (de) 2021-08-06 2021-08-06 Verarbeitung von digitalen audiosignalen

Publications (1)

Publication Number Publication Date
EP4131264A1 true EP4131264A1 (de) 2023-02-08

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Citations (5)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US5789689A (en) * 1997-01-17 1998-08-04 Doidic; Michel Tube modeling programmable digital guitar amplification system
US20010013270A1 (en) * 1999-12-28 2001-08-16 Yoshinori Kumamoto Pitch shifter
US6337999B1 (en) * 1998-12-18 2002-01-08 Orban, Inc. Oversampled differential clipper
US20060050897A1 (en) * 2002-11-15 2006-03-09 Kohei Asada Audio signal processing method and apparatus device
US10147433B1 (en) * 2015-05-03 2018-12-04 Digimarc Corporation Digital watermark encoding and decoding with localization and payload replacement

Patent Citations (5)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US5789689A (en) * 1997-01-17 1998-08-04 Doidic; Michel Tube modeling programmable digital guitar amplification system
US6337999B1 (en) * 1998-12-18 2002-01-08 Orban, Inc. Oversampled differential clipper
US20010013270A1 (en) * 1999-12-28 2001-08-16 Yoshinori Kumamoto Pitch shifter
US20060050897A1 (en) * 2002-11-15 2006-03-09 Kohei Asada Audio signal processing method and apparatus device
US10147433B1 (en) * 2015-05-03 2018-12-04 Digimarc Corporation Digital watermark encoding and decoding with localization and payload replacement

Non-Patent Citations (1)

* Cited by examiner, † Cited by third party
Title
ANONYMOUS: "HP VEE (Visual Engineering Environment) HP E2120C Technical Specifications", HP PUBLICATION NO.: 5965-5498E, 1 January 1997 (1997-01-01), pages 1 - 6, XP055881597 *

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