EP4324222A1 - Dispositif et procédé de génération d'un premier signal de commande et d'un second signal de commande par linéarisation et/ou par extension de bande passante - Google Patents

Dispositif et procédé de génération d'un premier signal de commande et d'un second signal de commande par linéarisation et/ou par extension de bande passante

Info

Publication number
EP4324222A1
EP4324222A1 EP22718230.0A EP22718230A EP4324222A1 EP 4324222 A1 EP4324222 A1 EP 4324222A1 EP 22718230 A EP22718230 A EP 22718230A EP 4324222 A1 EP4324222 A1 EP 4324222A1
Authority
EP
European Patent Office
Prior art keywords
signal
converter
audio signal
audio
designed
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.)
Pending
Application number
EP22718230.0A
Other languages
German (de)
English (en)
Inventor
Klaus Kaetel
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.)
Kaetel Systems GmbH
Original Assignee
Kaetel Systems 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 Kaetel Systems GmbH filed Critical Kaetel Systems GmbH
Publication of EP4324222A1 publication Critical patent/EP4324222A1/fr
Pending legal-status Critical Current

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Classifications

    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R1/00—Details of transducers, loudspeakers or microphones
    • H04R1/20—Arrangements for obtaining desired frequency or directional characteristics
    • H04R1/22—Arrangements for obtaining desired frequency or directional characteristics for obtaining desired frequency characteristic only 
    • H04R1/225—Arrangements for obtaining desired frequency or directional characteristics for obtaining desired frequency characteristic only  for telephonic receivers
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R3/00—Circuits for transducers, loudspeakers or microphones
    • H04R3/12—Circuits for transducers, loudspeakers or microphones for distributing signals to two or more loudspeakers
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R1/00—Details of transducers, loudspeakers or microphones
    • H04R1/20—Arrangements for obtaining desired frequency or directional characteristics
    • H04R1/22—Arrangements for obtaining desired frequency or directional characteristics for obtaining desired frequency characteristic only 
    • H04R1/28—Transducer mountings or enclosures modified by provision of mechanical or acoustic impedances, e.g. resonator, damping means
    • H04R1/2869—Reduction of undesired resonances, i.e. standing waves within enclosure, or of undesired vibrations, i.e. of the enclosure itself
    • H04R1/2873—Reduction of undesired resonances, i.e. standing waves within enclosure, or of undesired vibrations, i.e. of the enclosure itself for loudspeaker transducers
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04S—STEREOPHONIC SYSTEMS 
    • H04S7/00—Indicating arrangements; Control arrangements, e.g. balance control
    • H04S7/30—Control circuits for electronic adaptation of the sound field
    • H04S7/301—Automatic calibration of stereophonic sound system, e.g. with test microphone
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R3/00—Circuits for transducers, loudspeakers or microphones
    • H04R3/04—Circuits for transducers, loudspeakers or microphones for correcting frequency response
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R5/00—Stereophonic arrangements
    • H04R5/02—Spatial or constructional arrangements of loudspeakers
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R5/00—Stereophonic arrangements
    • H04R5/04—Circuit arrangements, e.g. for selective connection of amplifier inputs/outputs to loudspeakers, for loudspeaker detection, or for adaptation of settings to personal preferences or hearing impairments
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04S—STEREOPHONIC SYSTEMS 
    • H04S2420/00—Techniques used stereophonic systems covered by H04S but not provided for in its groups
    • H04S2420/05—Application of the precedence or Haas effect, i.e. the effect of first wavefront, in order to improve sound-source localisation
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04S—STEREOPHONIC SYSTEMS 
    • H04S3/00—Systems employing more than two channels, e.g. quadraphonic
    • H04S3/02—Systems employing more than two channels, e.g. quadraphonic of the matrix type, i.e. in which input signals are combined algebraically, e.g. after having been phase shifted with respect to each other
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04S—STEREOPHONIC SYSTEMS 
    • H04S5/00—Pseudo-stereo systems, e.g. in which additional channel signals are derived from monophonic signals by means of phase shifting, time delay or reverberation 

Definitions

  • the present invention relates to electroacoustics and more particularly to concepts for generating and reproducing audio signals.
  • acoustic scenes are recorded using a set of microphones. Each microphone outputs a microphone signal.
  • a microphone signal For example, for an orchestral audio scene, 25 microphones may be used.
  • a sound engineer performs a mixing of the 25 microphone output signals into, for example, a standard format such as a stereo format, a 5.1, a 7.1, a 7.2, or other appropriate format.
  • a stereo format for example, two stereo channels are created by the sound engineer or an automatic mixing process.
  • a 5.1 format mixing results in five channels and one subwoofer channel.
  • a mix is made into seven channels and two subwoofer channels in a 7.2 format, for example.
  • a mixed result is fed to electrodynamic loudspeakers.
  • two speakers exist, with the first speaker receiving the first stereo channel and the second speaker receiving the second stereo channel.
  • a 7.2 playback format for example, there are seven loudspeakers in predetermined positions and two subwoofers that can be placed relatively arbitrarily. The seven channels are routed to their respective speakers, and the two subwoofer channels are routed to their respective subwoofers.
  • European patent EP 2692154 B1 describes a set for capturing and playing back an audio scene in which not only the translation is recorded and played back, but also the rotation and also the vibration. Therefore, a sound scene reproduced not only by a single detection signal or a single mixed signal, but by two detection signals or two mixed signals which are simultaneously recorded on the one hand and reproduced simultaneously on the other hand. This ensures that different emission characteristics are recorded from the audio scene compared to a standard recording and are reproduced in a playback environment.
  • a set of microphones is placed between the acoustic scene and an (imaginary) auditorium to capture the “conventional” or translational signal, characterized by high directivity or high goodness.
  • a second set of microphones is placed above or to the side of the acoustic scene to record a low-Q or low-directivity signal intended to represent the rotation of the sound waves as opposed to translation.
  • corresponding loudspeakers are placed in the typical standard positions, each having an omnidirectional array to reproduce the rotational signal and a directional array to reproduce the "conventional" translational sound signal.
  • European patent EP 2692144 B1 discloses a loudspeaker for reproducing, on the one hand, the translatory audio signal and, on the other hand, the rotary audio signal.
  • the loudspeaker thus has an omnidirectionally emitting arrangement on the one hand and a directionally emitting arrangement on the other hand.
  • European patent EP 2692151 B1 discloses an electret microphone which can be used to record the omnidirectional or the directional signal.
  • European patent EP 3061262 B1 discloses an earphone and a method for manufacturing an earphone that generates both a translational sound field and a rotary sound field.
  • European patent application EP 3061266 A0 intended to be granted, discloses a headphone and a method for producing a headphone designed to to generate the "conventional" translational sound signal using a first transducer, and to generate the rotary sound field using a second transducer arranged perpendicularly to the first transducer.
  • the recording and playback of the rotational sound field in addition to the translational sound field leads to a significantly improved and thus high-quality audio signal perception, which almost conveys the impression of a live concert, although the audio signal is reproduced through loudspeakers or headphones or earphones.
  • a disadvantage of the concept described is that the recording of the additional signal, which reproduces the rotation of the sound field, represents an additional expense.
  • pieces of music be it classical pieces or pop pieces, in which only the conventional translational sound field has been recorded. These pieces are typically still highly compressed in their data rate, such as in accordance with the MP3 standard or the MP4 standard, which contributes to an additional deterioration in quality which, however, is normally only audible to experienced listeners.
  • the object of the present invention is to create an improved concept for generating or reproducing a first control signal for a first converter and a second control signal for a second converter.
  • the present invention is based on the finding that a synthetic generation of the rotation signal is possible when an audio piece with more than one channel, ie already with two stereo channels, for example, or even more channels, exists.
  • an at least approximate difference at least one approximation of the difference signal or rotation signal is obtained according to the invention, which can then be used to control an omnidirectional or a transducer with a lower directional effect, in order to thereby also generate a rotation component from a signal that was actually recorded purely in a translatory manner derived and reproduced in the sound field.
  • the approximate difference signal is manipulated by a signal manipulator to obtain the second drive signal for a rotary converter.
  • the signal manipulation takes place in particular by delaying the combination signal and/or frequency-selectively amplifying or attenuating the combination signal in order to at least partially counteract a non-linear converter characteristic over the frequency of the second converter, ie the rotary converter.
  • a bandwidth expansion stage is provided to improve reception quality, preferably for the first control signal for the (normal) translational converter and, depending on the implementation, also for the third control signal for the second (conventional) translational converter.
  • the fourth control signal for the additional rotary converter is again preferably delayed and/or linearized by a linearization filter in order to at least partially compensate for the typically strongly non-linear frequency response of the rotary converter.
  • the audible range which is e.g. B. extends up to 20 kHz, targeted, but on the non-audible range is about lying.
  • sound energy in the non-audible range is emitted above 20 kHz, with the signal for the sound energy in the non-audible range being derived from the audible sound signal by bandwidth expansion either of a non-harmonic nature or, preferably, of a harmonic nature has been.
  • this synthetically generated non-audible spectrum is amplified instead of attenuated in order to achieve that the typical conventional translatory sound transducers still emit enough sound energy in the non-audible range, although the emission efficiency at frequencies above 30 to 40 kHz typically decreases. However, it is preferred to emit sound signals up to 80 kHz.
  • the particularly good propagation properties of such natural signals are due to the fact that the audio signals have a particularly powerful harmonic component that reaches up to very high frequencies, which is used for the aforementioned air preconditioning. It is similar, for example, for certain percussion instruments in the orchestra, such as a triangle. Although this does not produce a particularly high sound pressure level, it is particularly clear even at a considerable distance, e.g. B. can also be heard well in the back rows of a concert hall.
  • a delay is applied to delay the rotation signal relative to the translation signal to take advantage of the precedence effect or the Haas effect.
  • the necessary delay of around 10 to 40 ms means that the sound source can be localized by a listener according to the principle of the first wave front on the basis of the translational signal, which carries the directional information.
  • the rotational signal does not disturb the perception of direction, but at the same time leads to a high-quality and lifelike audio signal experience due to the excitation of rotating sound velocity vectors in the sound field by the corresponding second or fourth transducer, which reproduces the second or fourth control signal. Due to the Haas effect, the listener believes that the rotating components of the sound field come from the source whose translational sound field had just reached the listener's ear.
  • the linearization filter is only designed to reduce peaks at least partially or preferably completely, but to leave the cancellations almost untouched in order to avoid potentially disruptive artefacts by avoiding the strong amplification in the cancellations that would otherwise be necessary.
  • the device for generating the first drive signal for the first transducer and the second drive signal for the second transducer preferably also includes means for generating a drive signal for the third and the fourth transducer in order to achieve stereo reproduction via loudspeakers, for example. If more than two channels are to be reproduced, additional control signals are generated, e.g. B. for egg NEN left rear speaker, a right rear speaker and a center speaker. Then both a converter for the translational sound and a converter for the ro tatory sound will be provided at each point of the standardized speaker output for mats, and the inventive synthetically generated control signal for the rotary sound is ermit mined for each speaker position or by derived from one and the same manipulated combination signal, depending on the wall of the corresponding embodiment.
  • additional control signals are generated, e.g. B. for egg NEN left rear speaker, a right rear speaker and a center speaker. Then both a converter for the translational sound and a converter for the ro tatory sound will be provided at
  • an interface is provided that receives the first electrical signal, such as a left channel, and a second electrical signal, such as for a right channel. These signals are fed to a signal processor to reproduce the first electrical signal for the first transducer and the second electrical signal for a third transducer. These converters are the conventional converters.
  • the signal processor is designed to calculate the at least approximate difference between the first electrical signal and the second electrical signal and to determine a third electrical signal for a second transducer or a fourth electrical signal for a fourth transducer from this difference.
  • the signal processor is designed to output the first electrical signal for the first transducer and the second electrical signal for the third transducer, and to calculate a first, at least approximately, difference from the first electrical signal and the second electrical signal, and to calculate a second at least approximate difference from the first electrical signal and the second electrical signal, and to calculate a third electrical signal for the second transducer based on the first at least approximate difference and a fourth to output an electrical signal for the fourth transducer based on the second at least approximately difference.
  • the difference is an exact difference in which the second signal is changed by 180° and added to the first signal.
  • this signal is the first, at least approximately, difference
  • the different, second, at least approximately, difference is what results when the first signal is phase-shifted by 180°, i.e. given a “minus” value and added to the unchanged second signal becomes.
  • the first difference is at least approximately calculated and that this is subjected to a phase shift of 180°, for example, in order to calculate the second at least approximately difference.
  • the second, at least approximately, difference is then determined directly from the first, at least approximately, difference.
  • both differences can be determined independently of one another, namely both from the original first and second electrical signals, ie the left and the right input signal.
  • the difference is ideally a value obtained when the first channel is subtracted from the second channel or vice versa.
  • an at least approximate difference also results and is useful in certain exemplary embodiments if the phase shift is not 180° but is greater than 90° and less than 270°. In the more preferred range, which is smaller, the phase shift has a phase value between 160° and 200°.
  • one of the two signals can also be subjected to a phase shift equal to or different from 180° before the difference is formed and, if necessary, have also been subjected to frequency-dependent processing before the addition, such as by equalizer processing or frequency-selective processing or non-frequency selective amplification.
  • Further processing which can be carried out either before or after forming the difference, consists of high-pass filtering. If a high-pass filtered signal is combined with the other signal, for example with an angle of 180°, this also represents at least an approximate difference Producing transducers that are separate from the conventional transducers can be approximated by not changing the magnitudes of the two signals and varying the phase between the two signals between an angle between 90 and 270°.
  • an angle of 180° can be used.
  • the amplitudes of the signals can be varied in a frequency-selective or non-frequency-selective manner.
  • a combination of frequency-selectively or non-frequency-selectively varied amplitudes of the two electrical signals together with an angle between 90 and 270° also leads to a rotation excitation signal that is useful in many cases for the separate rotation converters, i.e. the second converter on the left side and the second transducer on the right.
  • the difference signal for one side and the different difference signal for the other side are preferably used for speakers that are away from the listener's head.
  • Each of these loudspeakers then has at least two transducers fed with different signals, with the first loudspeaker for the "left side” having a first transducer fed with the original left signal or a possibly delayed left signal, while the second converter is fed with the signal derived from the first at least approximately difference.
  • the individual converters of the second loudspeaker for the “right side” are then controlled accordingly.
  • the signal processor or the interface has a down-converter for the first electrical signal, i.e. for the left channel, and a further down-converter for the second electrical signal Signal, i.e. for the right channel, upstream.
  • the signal is an original microphone signal, such as an ambisonics signal with several components
  • each down-converter is designed to calculate a left or right channel from the ambisonics signal, which is then used by the signal processor to calculate the third electrical signal and the fourth electrical signal on the basis of at least approximate differences.
  • FIG. 1 shows a device for generating a first drive signal and a second drive signal according to an embodiment of the present invention
  • FIG. 2 shows a more detailed representation of the signal manipulator of FIG. 1 according to a preferred exemplary embodiment
  • FIG. 3 shows a detailed illustration of the signal combiner from FIG. 1 according to a preferred exemplary embodiment and an illustration of the integration of a bandwidth expansion stage for each drive signal for a translational converter;
  • FIG. 4 shows an alternative implementation of the device for generating with a different arrangement of the bandwidth expansion stages with respect to FIG.
  • 5a shows a schematic representation of the effect of a bandwidth expansion stage according to an embodiment
  • 5b shows a schematic representation of an effect of a bandwidth expansion stage according to a further embodiment
  • FIG. 6 shows a schematic representation of the loudspeaker side of a loudspeaker system for a 2-channel output format
  • FIG. 7a shows an exemplary non-linear frequency response of a converter with a comb filter effect
  • FIG. 7b shows a schematic frequency response of a linearization filter in order to at least partially linearize the frequency response of FIG. 7a;
  • 8a is a schematic representation of another non-linear frequency response of a rotary transducer
  • 8c shows a schematic representation of a linearized frequency response based on the linearization filter and the rotary sound transducer used.
  • 1 shows a device for generating a first control signal 411 for a first converter and a second control signal 412 for a second converter.
  • the device comprises an input interface 100 for delivering a first audio signal 111 for a first audio channel and a second audio signal for a second audio channel.
  • the device also includes a signal combiner 200 for determining a combination signal from the first audio signal 111 and the second audio signal 112, which includes an approximate difference between the first audio signal 111 and the second audio signal 112. This combination signal is shown at 211 .
  • the signal combiner is also designed to generate a further combination signal 212 which also represents a difference between the first and the second audio signal and is derived from the first audio signal and the second audio signal or from the first combination signal 211 .
  • the second combination signal 212 differs from the first combination signal 211 and in particular differs by 180 degrees, ie has an opposite sign.
  • combination signal 211 is fed to a signal manipulator 300, which is designed to manipulate the combination signal in order to obtain a manipulated combination signal from it, which is shown at 311 and corresponds to second control signal 412.
  • the second control signal 412 is thus transmitted by the signal manipulator using the output interface 400 and output or stored through the output interface.
  • the output interface is designed to also output the first control signal 411 for the first converter in addition to the second control signal for the second converter.
  • the first drive signal 411 is obtained from the output interface directly from the input interface and corresponds to the first audio signal 111, or is derived from the first audio signal by the output interface 400, such as using a bandwidth expansion stage, i.e. a spectral enhancer, which will be presented later becomes.
  • the signal manipulator 300 is designed to delay the combination signal, i.e. to feed it into a delay stage, or to amplify or attenuate the combination signal in a frequency-selective manner, i.e. to feed it into a linearization filter, in order to achieve a non-linear converter characteristic above the frequency of the second converter at least partially counteract it.
  • the output interface is designed to feed the first audio signal 111 into a bandwidth expansion stage in order to receive the first output signal 411.
  • the device for generating a first control signal 411 and a second control signal 412 therefore comprises three aspects that can be used together or independently of one another.
  • the first aspect is that the manipulated signal has been generated from the combined signal using a delay, exploiting the Haas effect.
  • the second aspect is that the signal manipulator 300 uses the linearization filter in order to at least partially compensate for a highly nonlinear frequency response of the “rotary” converter in the sense of “predistortion”.
  • the third aspect is that the signal manipulator performs some other kind of manipulation, such as attenuation or high-pass filtering or other processing, whereby the output interface performs a bandwidth expansion for the first audio signal.
  • This bandwidth expansion using a bandwidth expansion stage is special in that at least part of a spectrum of the first audio signal is converted into a frequency range above 20 kHz using an amplification factor greater than 1 or equal to 1, i.e. without amplification, in order to convert the first Obtain drive signal that includes the frequency range above 20 kHz.
  • This is in contrast to conventional bandwidth extension, which is typically designed to extend a signal bandlimited to perhaps 4 or 8 kHz to a frequency range up to perhaps 16 or 20 kHz, further employing attenuation to provide a sloping performance characteristic
  • the bandwidth expansion according to the invention is different in that spectral values are determined for a frequency range above 20 kHz, i.e.
  • This audio signal experience consists in the fact that the air, which transmits the sound energy in the audible range, is “conditioned” to a certain extent, so that certain signals that are very rich in harmonics are clearly audible despite a great distance, such as the cry of a parrot in the jungle or a triangle in an orchestra.
  • all three aspects are implemented, as will be presented later. However, only one aspect of the three aspects can be implemented, or only any two aspects of the three aspects.
  • the first input signal 102 and the second input signal 104 which are input into the input interface 100, represent a left audio channel and a right audio channel.
  • the first audio signal 411 and the second audio signal 412 then represent the control signals for the first and the second converter , which are placed on the left with respect to a listening position.
  • the device for generating is then also designed to also generate the control signals for the right-hand side, that is to say the third control signal 413 for a third converter and the fourth control signal 414 for the fourth converter.
  • the third control signal 413 is formed analogously to the first control signal 411 and the fourth control signal 414 is formed analogously to the second control signal 412 .
  • the first control signal 411 and the third control signal 413 are fed to the conventional translational transducers, and the control signals 412 and 414 are fed to "rotary" transducers, i.e. transducers that emit a sound field with rotating sound velocity vectors, as is still the case with reference to Fig 6 is shown.
  • FIG. 2 shows a preferred implementation of the signal manipulator 300 in order to calculate the second control signal 311/412 from the combination signal 211. Furthermore, Fig.
  • the signal combiner in preferred exemplary embodiments includes a variable attenuator 301, a delay stage 302, and a linearization filter 303. It should be noted that blocks 301, 302, 303 can be in any order. A single element can also be present, which combines the functionalities of the linearization filter, the delay and the damping. The damping can be adjusted, or is set to a pre-defined values between
  • the signal manipulator 300 is designed to subject the further combination signal 212 to an attenuation by an attenuation stage 321 , to subject it to a delay 322 and to feed it into a linearization filter 323 .
  • All three elements can also be integrated in a single filter that implements the constant attenuation, typically over the entire frequency range, the delay, which is also constant over the entire frequency range, and a linearization filter, which attenuates or amplifies at least in a frequency-selective manner. It should be pointed out that only a subset of the elements, i.e. only e.g. B. damping and linearization without delay delay, or only a delay without damping and linearization, or only a damping without delay and linearization can be used. In preferred embodiments, all three aspects are implemented.
  • a delay is used for the delay, which is so large that a precedence effect or A Haas effect or a first wavefront effect occurs.
  • the signal for the rotary converter ie the second control signal 412, is delayed in such a way that a listener first perceives the wave front based on the first control signal 411 and therefore localizes the left channel.
  • the rotational component which is essential for audio quality but does not contain any special information regarding localization, is then perceived somewhat later and is not perceived as a separate signal due to the Haas effect.
  • Useful delay values for the delay stage 302 or 322 are preferably between 10 and 40 ms and more preferably between 25 ms and 35 ms and in particular 30 ms.
  • the signal combiner 200 includes a phase shifter 201, a downstream attenuator 202 and an adder 203. Furthermore, the first audio signal 111 and the second audio signal 112 are used. The first audio signal 111 is phase shifted by the phase shifter 201 and, depending on the setting of the attenuator 202, is attenuated and then added to the first audio signal 112 in order to obtain the further combination signal 212.
  • the signal combiner 200 comprises a further adder 223, a further phase shifter 221 and a further attenuator 222, with the second audio signal 112 is phase-shifted by the phase shifter 221 and the phase-shifted signal is optionally attenuated and then combined with the first audio signal 111 .
  • the phase shifters 201 or 221 carry out a phase shift of 180 degrees, which is preferred, and if the attenuators 202, 222 are set in such a way that the attenuation is equal to zero, i.e. these potentiometers are “fully turned up”, then the combination signal 211 is that Result of the subtraction of the second audio signal 112 from the first audio signal 111, i.e. if the first audio signal 111 is the left channel and the right audio signal 112 is the right channel, then the combination signal 211 is equal to L - R.
  • the further combination signal 212 is equal R - L in this example.
  • phase shift is particularly easy to achieve by plugging in a corresponding connector carrying the audio signal “upside down”.
  • Various phase shifts other than 180 degrees ie in a range from 150 to 210 degrees preferably, can be achieved by correct phase shifter elements and can be advantageous in certain implementations.
  • a subtraction factor x between zero and 1 can thus be formed, as will be explained in FIG.
  • the output interface 400 includes a first bandwidth expansion stage 402 and a second bandwidth expansion stage 404.
  • the first bandwidth expansion stage 402 is designed to allow the first audio signal 111 to have a bandwidth expansion in the non-audible range above 20 kHz under, while the bandwidth expansion stage 404 is designed to also subject the second audio signal, for example the right channel, to a bandwidth expansion in the inaudible range above 20 kHz.
  • both control signals 411, 413 are now also provided with signal energy at frequencies above 20 kHz, with these signal components preferably being present in the control signals up to 40 kHz and particularly preferably even up to 80 kHz or more.
  • FIG. 3 shows an implementation in which only bandwidth expansion is performed on the translational signal
  • bandwidth expansion can also be performed on the rotary signal, as illustrated at 304 and 324 in FIG is.
  • a bandwidth expansion in the input interface 100 could also be provided.
  • a bandwidth expansion stage 121 is provided for a first input signal 102 in order to generate the first audio signal 111 from the first input signal 102 .
  • the input interface 100 is provided in order to generate the second audio signal 112 from the second input signal 104 .
  • these two audio signals now have a frequency range that goes far beyond 20 kHz.
  • bandwidth expansion is already carried out in the input interface, further bandwidth expansions are not necessary in the output interface 400, as has been shown in FIG. 3, or in the signal manipulator elements 300a, 300b, since all signals in the subsequent signal processing have a high bandwidth.
  • an implementation as shown in FIG. 3 is preferred, in which only the control signals for the translational converters, ie the first control signal 411 and the third control signal 413, are subjected to the bandwidth expansion genes, as the high frequencies are particularly important for propagation. Therefore, all other processing stages in the input interface, in the signal combiner and in the signal manipulator can be performed on the band-limited signal, which saves processing resources because all elements except the bandwidth expansion stages 402, 404 in FIG. 3 can work with band-limited signals .
  • FIG. 5 shows a first implementation of the bandwidth expansion stage 402, 404 or the optional elements 121, 122 or 304, 324 from FIG kHz, ie in the inaudible range, which goes up to 80 kHz in FIG. 5a.
  • a harmonic bandwidth expansion is preferably undertaken, with each frequency in the range between 10 and 20 kHz of the audio signal being multiplied by a factor of 2, for example, by a frequency range between 20 kHz and 40 kHz to create.
  • amplification is performed by means of an amplifier 407 implementing a gain greater than 1, as shown by the dotted line in Figure 5a.
  • the harmonic bandwidth extension unit 404 together with the amplifier 407 thus generates a signal component in the corresponding audio signal which is between 20 and 40 kHz and has even greater signal energy than the baseband range which is between 10 and 20 kHz.
  • another transponer 406 is provided, which multiplies the frequencies by 4, the output signal being multiplied again with a gain factor greater than 1, preferably, with this amplifier having the gain greater than 1 is shown at 408 in Figure 5a.
  • the frequency axis is broken at the appropriate points, since the range between 40 kHz and 80 kHz is twice as long as the range between 20 kHz and 40 kHz, which in turn is twice as long as the range between 10 kHz and 20 kHz due to the harmonic bandwidth expansion by the elements 404, 406.
  • transposition factors that are odd i.e. that can be equal to 1, 3, 5 and 7, can in principle be used, it has been shown that Even-numbered transposition factors, such as those achieved by the transposers 404, 406, produce a more realistic audio signal impression.
  • the baseband cannot be attenuated and amplified, ie taken as it is.
  • the amplifier 408 amplifies more for the range between 40 and 80 kHz than the amplifier 407 for the range between 20 kHz and 40 kHz.
  • Fig. 5a shows a first implementation of the bandwidth extension
  • Fig. 5b provides a second implementation of the bandwidth extension, which works due to the technique of "mirroring", i.e. mirroring of the transposed spectral range at the cross-over frequency (crossover frequency), what is advantageous in that with a non-constant signal curve in the baseband, as shown in FIG. 5b, at the transpositi onsstelle, ie at 20 kHz when a gain factor of 1 is used, no discontinuity occurs.
  • the mirroring or upsampling can easily be carried out in the time domain by inserting one or more zeros as additional sample values in an audio signal between two sample values.
  • Fig. 6 shows a speaker system that summarizes a first converter 521 for the first drive signal 411 and a second converter 522a, 522b for the second drive signal 412 to. Furthermore, the loudspeaker system also has a third converter 523 for the third drive signal 413 and a fourth converter 524a, 524b for the fourth drive signal 414. All drive signals can be amplified by respective amplifiers 501, 502, 503, 504, such as input from a user interface via a volume control.
  • the converters 521, 523 represent the translatory and to a certain extent conventional converters, which, however, are distinguished in comparison to normal converters in that they can also emit sound energy in the range above 20 kHz, whereby they should preferably emit up to 80 kHz or more. The decreasing efficiency at higher frequencies is compensated for by the amplification due to the amplifying elements 407, 408.
  • the rotary converters 522a, 522b, for example, or 524a, 524b, for example, are implemented in a preferred embodiment, which is shown in FIG as shown in Fig. 6, face each other. There can be no distance or only a small distance between the front sides, ie between the membranes, so that the membranes can deflect and generate sound in the intermediate area between the membranes, which can exit as rotation along the edges of the membranes.
  • Such a transducer is very efficient when generating rotating sound, ie a sound field with rotating sound velocity vectors.
  • the frequency response is highly non-linear.
  • the linearization filter 303, 323 is therefore provided in order to generate a signal via “pre-distortion”, so to speak, that when it is output by the non-linear frequency response of the converter 522a, 522b or 524a, 524b, it is relatively linear Has transmission or signal characteristics.
  • FIG. 7a shows an exemplary comb spectrum as can occur in converters for rotary signals.
  • Fig. 7b shows an exemplary frequency response of the linearization filter 303, 323.
  • the linearization filter In the preferred implementation of the linearization filter, only the peaks 701, 702, 703, 704, 705 are lowered, while the notches 706 to 710 are "left", so that in the frequency ranges in which the cuts are located, the frequency response of the linearization filter is at the 0 dB reference line and in the range of the peaks the Exaggerations are at least partially lowered, namely by 6 dB, if the over-elevation itself has a height of 6 dB, as is shown in the exemplary frequency response in Fig. 7a.
  • the linearization filter is also designed to provide a high-pass characteristic with respect to a cut-off frequency f g , which is only shown schematically in FIG. 7b and is of the order of between 100 and 500 Hz and is preferably 200 Hz. This means that the first increase 711 in FIG. 7a is completely attenuated.
  • FIG. 8a shows an alternative frequency response of a rotary sound transducer, which can result from the design of the rotary sound transducer as shown in FIG.
  • Very strong peaks and very strong slumps are shown.
  • the linearization is designed in particular in such a way that again only the peaks, which are shown hatched in FIG. 8a, are to be damped, while the dips are to be left almost as they are.
  • the entire “linearized” frequency response is shown schematically in FIG. 8c, where it can be seen that the linearized frequency response is not completely linearized, but when FIG. 8c is compared to FIG. 8a, it is much more linear because the strong peaks have been cut off are.
  • phase shifters 506, 508 are preferably also installed in the rotary sound transducers, which, depending on the implementation, provide a phase shift of 180 degrees and which, however, can be set to other values, but which are preferably between 150 and 210 degrees.
  • FIG. 3 it was pointed out that the attenuators 202, 222 can be adjusted to obtain only an approximate difference. This is in Fig. 6 at "L - x R" and "R - x L" shown.
  • the corresponding attenuator 202, 222 is set to an attenuation of zero, i.e. to no attenuation, the factor x in FIG Factor x for example 0.5. If, on the other hand, the attenuator 202, 222 is set to full attenuation, no more difference formation takes place, and the first converter 522a, 522b only emits the left-hand signal. However, it is preferred to set an attenuation of the attenuator 202, 222 to at most 0.25 in order for the corresponding signal to be a difference signal even though the subtracted channel is reduced in amplitude or power or energy compared to the channel being subtracted from is.
  • the device for generating the first control signal and the second control signal and in particular also for generating the third and the fourth control signal is implemented as a signal processor or as software, for example in a mobile device such as a mobile phone for the control signals for each speaker and then output over a wireless interface.
  • the transducers as shown in Fig. 6, including the amplifiers 502 to 504, together with the device as shown in Fig. 1, are implemented in a speaker unit which additionally comprises transducer 521 and transducer 522a, 522b included in a special carrier. Then this speaker unit can be used as it is, e.g. B. placed at a left playing position with respect to a listening position.
  • loudspeaker units can also be used for channels other than the two stereo channels, such as for a center channel, for a left rear channel, for a right rear channel in the case of a 5.1 system.
  • a converter for rotary sound and a converter for translatory sound can also be installed at corresponding further positions, such as a ceiling loudspeaker, which are controlled with the separate control signals.
  • a preferred embodiment of the present invention resides within a cellular phone.
  • the control device is loaded, for example, as a hardware element or as an app or as a program on the mobile phone.
  • the mobile phone is designed to be located from any source, local or on the internet can be to catch the first audio signal and the second audio signal or multi-channel signal and to generate the control signals depending thereon.
  • These signals are transmitted from the mobile phone to the sound generator with the sound generator elements either by cable or wirelessly, for example using Bluetooth or WLAN.
  • the sound generator elements it is necessary for the sound generator elements to have a battery supply or, in general, a power supply in order to achieve appropriate amplification for the received wireless signals, for example according to the Bluetooth format or according to the WLAN format.
  • aspects have been described in the context of a device, it should be understood that these aspects also represent a description of the corresponding method, so that a block or component of a device can also be understood as a corresponding method step or as a feature of a method step . Similarly, aspects described in connection with or as a method step also constitute a description of a corresponding block or detail or feature of a corresponding device.
  • Some or all of the method steps may be performed by hardware apparatus (or using a Hard ware apparatus), such as a microprocessor, a programmable computer or an electronic circuit. In some embodiments, some or more of the key process steps can be performed by such an apparatus.
  • embodiments of the invention may be implemented in hardware or in software. Implementation can be performed using a digital storage medium such as a floppy disk, DVD, Blu-ray Disc, CD, ROM, PROM, EPROM, EEPROM or FLASH memory, hard disk or other magnetic or optical memory, on which electronically readable control signals are stored, which can interact with a programmable computer system in such a way that the respective method is implemented. Therefore, the digital storage medium can be computer-readable.
  • a digital storage medium such as a floppy disk, DVD, Blu-ray Disc, CD, ROM, PROM, EPROM, EEPROM or FLASH memory, hard disk or other magnetic or optical memory, on which electronically readable control signals are stored, which can interact with a programmable computer system in such a way that the respective method is implemented. Therefore, the digital storage medium can be computer-readable.
  • Some exemplary embodiments according to the invention thus comprise a data carrier which has electronically readable control signals which are capable of interacting with a programmable computer system in such a way that one of the methods described herein is carried out.
  • exemplary embodiments of the present invention can be implemented as a computer program product with a program code, the program code being effective to carry out one of the methods when the computer program product runs on a computer.
  • the program code can also be stored on a machine-readable carrier, for example.
  • exemplary embodiments include the computer program for performing one of the methods described herein, the computer program being stored on a machine-readable medium.
  • an exemplary embodiment of the method according to the invention is therefore a computer program that has a program code for performing one of the methods described herein when the computer program runs on a computer.
  • a further exemplary embodiment of the method according to the invention is therefore a data carrier (or a digital storage medium or a computer-readable medium) on which the computer program for carrying out one of the methods described herein is recorded.
  • a further exemplary embodiment of the method according to the invention is therefore a data stream or a sequence of signals which represents the computer program for carrying out one of the methods described herein.
  • the data stream or sequence of signals may be configured to be transmitted over a data communications link, such as the Internet.
  • Another embodiment includes a processing device, such as a computer or programmable logic device, configured or adapted to perform any of the methods described herein.
  • a processing device such as a computer or programmable logic device, configured or adapted to perform any of the methods described herein.
  • a further exemplary embodiment according to the invention comprises an apparatus or a system which is designed to transmit a computer program for carrying out at least one of the methods described herein to a recipient.
  • the transmission can take place electronically or optically, for example.
  • the recipient may be a computer, mobile device, storage device, or similar device.
  • the device or the system can, for example, comprise a file server for transmission of the computer program to the recipient.
  • a programmable logic device e.g., a field programmable gate array, an FPGA
  • a field programmable gate array may cooperate with a microprocessor to perform any of the methods described herein.
  • the methods are performed on the part of any hardware device. This can be universally replaceable hardware such as a computer processor (CPU) or hardware specific to the process such as an ASIC.

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  • Physics & Mathematics (AREA)
  • Engineering & Computer Science (AREA)
  • Acoustics & Sound (AREA)
  • Signal Processing (AREA)
  • Health & Medical Sciences (AREA)
  • Otolaryngology (AREA)
  • General Health & Medical Sciences (AREA)
  • Stereophonic System (AREA)

Abstract

L'invention se rapporte à un dispositif permettant de générer un premier signal de commande (411) pour un premier transducteur (521) et un second signal de commande (412) pour un second transducteur (522a, 522b), comprenant les éléments suivants : une interface d'entrée (100) permettant d'émettre en sortie un premier signal audio (111) pour un premier canal audio et un second signal audio pour un second canal audio ; un combineur de signal (200) permettant de déterminer un signal de combinaison (211) à partir du premier signal audio (111) et du second signal audio (112), ledit signal de combinaison comprenant une différence approximative entre le premier signal audio (111) et le second signal audio (112) ; un manipulateur de signal (300) permettant de manipuler le signal de combinaison afin d'obtenir le second signal de commande (412) ; et une interface de sortie (400) permettant d'émettre en sortie ou de stocker le premier signal de commande (411) fondé sur le premier signal audio (111) ou d'émettre en sortie ou de stocker le second signal de commande (412) ; le manipulateur de signal (300) étant conçu pour retarder (302) le signal de combinaison (211) ou pour amplifier ou atténuer (303) le signal de combinaison (211) en fonction de la fréquence afin de contrebalancer les caractéristiques de transducteur non linéaires supérieures à la fréquence du second transducteur (522a, 522b), ou le dispositif étant conçu pour convertir au moins une partie d'un spectre du premier signal audio ou du signal de combinaison en une plage de fréquence supérieure à 20 kHz afin d'obtenir le premier signal de commande (411) présentant la plage de fréquence supérieure à 20 kHz.
EP22718230.0A 2021-04-13 2022-04-07 Dispositif et procédé de génération d'un premier signal de commande et d'un second signal de commande par linéarisation et/ou par extension de bande passante Pending EP4324222A1 (fr)

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DE102021203640.6A DE102021203640B4 (de) 2021-04-13 2021-04-13 Lautsprechersystem mit einer Vorrichtung und Verfahren zum Erzeugen eines ersten Ansteuersignals und eines zweiten Ansteuersignals unter Verwendung einer Linearisierung und/oder einer Bandbreiten-Erweiterung
PCT/EP2022/059307 WO2022218822A1 (fr) 2021-04-13 2022-04-07 Dispositif et procédé de génération d'un premier signal de commande et d'un second signal de commande par linéarisation et/ou par extension de bande passante

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EP (1) EP4324222A1 (fr)
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DE102021205545A1 (de) 2021-05-31 2022-12-01 Kaetel Systems Gmbh Vorrichtung und Verfahren zum Erzeugen eines Ansteuersignals für einen Schallerzeuger oder zum Erzeugen eines erweiterten Mehrkanalaudiosignals unter Verwendung einer Ähnlichkeitsanalyse
WO2023166109A1 (fr) 2022-03-03 2023-09-07 Kaetel Systems Gmbh Dispositif et procédé de réenregistrement d'un échantillon audio existant

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US5862228A (en) * 1997-02-21 1999-01-19 Dolby Laboratories Licensing Corporation Audio matrix encoding
KR101421589B1 (ko) * 2007-12-28 2014-07-22 삼성전자주식회사 휴대용 단말기의 음향 보정 필터 설계 방법 및 장치
AT507622B1 (de) 2009-03-19 2010-09-15 Weingartner Bernhard Dipl Ing Ohraufliegender kopfhörer
EP2692144B1 (fr) 2011-03-30 2017-02-01 Kaetel Systems GmbH Haut-parleur
US9271102B2 (en) * 2012-08-16 2016-02-23 Turtle Beach Corporation Multi-dimensional parametric audio system and method
DE102013221754A1 (de) 2013-10-25 2015-04-30 Kaetel Systems Gmbh Kopfhörer und verfahren zum herstellen eines kopfhörers
DE102013221752A1 (de) 2013-10-25 2015-04-30 Kaetel Systems Gmbh Ohrhörer und verfahren zum herstellen eines ohrhörers
CA2953674C (fr) * 2014-06-26 2019-06-18 Samsung Electronics Co. Ltd. Procede et dispositif permettant de restituer un signal acoustique, et support d'enregistrement lisible par ordinateur
WO2019191611A1 (fr) * 2018-03-29 2019-10-03 Dts, Inc. Commande de plage dynamique de protection de centre

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DE102021203640A1 (de) 2022-10-13
WO2022218822A1 (fr) 2022-10-20
DE102021203640B4 (de) 2023-02-16
TW202245483A (zh) 2022-11-16
CN117882394A (zh) 2024-04-12

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