EP1843636A1 - Verfahren zur automatischen Entzerrung eines Tonsystems - Google Patents

Verfahren zur automatischen Entzerrung eines Tonsystems Download PDF

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EP1843636A1
EP1843636A1 EP07003712A EP07003712A EP1843636A1 EP 1843636 A1 EP1843636 A1 EP 1843636A1 EP 07003712 A EP07003712 A EP 07003712A EP 07003712 A EP07003712 A EP 07003712A EP 1843636 A1 EP1843636 A1 EP 1843636A1
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Prior art keywords
loudspeakers
loudspeaker
frequency
sound
equalizing
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French (fr)
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EP1843636B1 (de
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Markus Christoph
Leander Scholz
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Harman Becker Automotive Systems GmbH
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Harman Becker Automotive Systems GmbH
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    • 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
    • H04R2499/00—Aspects covered by H04R or H04S not otherwise provided for in their subgroups
    • H04R2499/10—General applications
    • H04R2499/13—Acoustic transducers and sound field adaptation in vehicles

Definitions

  • the present invention relates to a method for automatically equalizing a sound system.
  • Said sound system includes at least two groups of loudspeakers supplied with electrical sound signals to be converted into acoustical sound signals,
  • the method according to the present invention for automatically adjusting such sound system to a target sound comprises the steps of: individually supplying each group with the respective electrical sound signal; individually assessing the deviation of the acoustical sound signal from the target sound for each group of loudspeakers in at least one listening position; and adjusting at least two groups of loudspeakers to a minimum deviation from the target sound by equalizing the respective electrical sound signals supplied to said groups of loudspeakers, wherein the assessment step includes receiving in the listening position the acoustical sound signal from a certain group of loudspeakers, wherein the total assessment over all listening positions is derived from the assessments at the at least one listening position weighted with a location specific factor, and wherein each position specific factor comprises an amplitude specific factor and a phase specific factor.
  • an automatic, e.g., iterative method for equalizing the magnitude and phase of the transfer function of all of the individual loudspeakers of a sound system, e.g., in a motor vehicle which determines all of the necessary parameters for equalizing without any manual actions and thus, e.g., provides appropriate filtering in a digital signal processing system.
  • the advantageous effect of the invention results from the completely automatic matching of the transfer function of the sound system to a predetermined target function, in which case the number and frequency range of the loudspeakers which are used for the sound system may be variable.
  • the following example describes the procedure and the investigations in order to create an algorithm which is also referred to in the following text as AutoEQ, for automatically adjusting, e.g., of equalizing filters in accordance with the present invention.
  • Two procedures are investigated which are disclosed in detail further below, together with a sequential method and a method taking account of the maximum interval between a measured level profile and a predetermined target function. The results obtained are used to derive a method, which is then used for automatic equalizing, that is to say without any manual influence on the parameters involved.
  • the major tonal sensitivities to be taken into account in this case which comprise psycho-acoustic parameters of human perception of sounds, are the location capability, the tonality and the staging.
  • the location capability which is also referred to as localization, denotes the perceived location of a hearing event, as a result, for example from the superimposition of stereo signals.
  • the tonality results from the time arrangement and the harmony of sounds and the ratio of the background noise to the useful signal that is presented, for example, stereophonic audio signals.
  • Staging is used to refer to the effect of perception of the point of origin of a complex hearing event that is composed of individual hearing events, such as that which results from an orchestra, in which case individual hearing events, for example instruments, always have their own location capability.
  • the location capability of phantom sound sources which are produced by stereophonic audio signals depends on a plurality of parameters, the delay-time difference of arriving sound signals, the level difference of arriving sound signals, the inter-aural level difference of an arriving sound between the right and left ear (inter-aural intensity difference IID), the inter-aural delay time difference of an arriving sound between the right and left ear (inter-aural time difference ITD), the head related transfer function HRTF, and on specific frequency bands in which levels have been raised, with the spatial directional localization in terms of front, above and to the rear depending solely on the level of the sound in these frequency bands without their being any delay-time difference or level difference in the sound signals at the same time in the latter case.
  • inter-aural intensity difference IID inter-aural level difference of an arriving sound between the right and left ear
  • ITD inter-aural delay time difference of an arriving sound between the right and left ear
  • HRTF head related transfer function
  • the major parameters for spatial-acoustic perception are the inter-aural time difference ITD, the inter-aural intensity difference IID and the head related transfer function HRTF.
  • the ITD results from delay-time differences between the right and left ear in response to a sound signal arriving from the side, and may assume orders of magnitude of up to 0.7 milliseconds. If the speed of sound is 343 m/s, this corresponds to a difference of about 24 centimetres in the path length of an acoustic signal, and thus to the anatomical characteristics of a human listener. In this case, the hearing evaluates the psycho-acoustic effect of the law of arrival of the first wavefront. At the same time it is evident for a sound signal which arrives at the head at the side, that the sound pressure which is applied to the ear which is spatially further away is less (IID) owing to sound attenuation.
  • the auricle of the human ear is shaped such that it represents a transfer function for received audio signals into the auditory system.
  • the auricles thus have a characteristic frequency response and phase response for a given sound signal incidence angle.
  • This characteristic transfer function is convolved with the sound which is entering the auditory system and contributes considerably to the spatial hearing capability.
  • a sound which reaches the human ear is also changed by further influences. These changes are caused by the environment of the ear, that is to say the anatomy of the body.
  • HRTF head related transfer function
  • correlated signals are offered via two physically separated loudspeakers, forming a so-called phantom sound source between the two loudspeakers.
  • the expression phantom sound source is used because a hearing event is perceived where there are no loudspeakers as a result of the superimposition and addition of two or more sound signals produced by different loudspeakers.
  • the sound source (phantom sound source) is located as being on the loudspeaker base, that is to say in the centre.
  • a phantom sound source can move between the loudspeakers as a result of delay-time and/or level differences between the two loudspeaker signals.
  • Level differences of between 15 and 20 dB and delay-time differences of between 0.7 and 1 ms, up to a maximum of 2 ms are required to shift the phantom sound source to the extreme on one side, depending on the signal.
  • the asymmetric seat position (driver, front-seat passenger, front and rear row or rows of seats) for loudspeaker configuration in a vehicle leads to sounds arriving neither with the same phase nor with the same delay time with respect to the position of a single listener. This primarily changes the spatial sensitivity, although the tonality and localization are also adversely affected.
  • the staging propagates on both sides unequally in front of the listener. Although delay-time correction with respect to an individual listener position would be possible, this is not desirable since this would automatically lead to matching specifically for one individual seat, with a disadvantageous effect on the remaining seats in the motor vehicle.
  • the spatial directional localization also depends on the level of the sound in specific frequency bands, without there being any delay-time difference or level difference between the sound signals at the same time (for example a mono signal arriving from the front).
  • a mid-frequency of 1 kHz and above 10 kHz narrowband test signal
  • test subjects locate a signal that is offered as being behind them, while an identical sound event with a mid-frequency of 8 kHz is localized as being above. If a signal contains frequencies of around 400 Hz or 4 kHz, then this enhances the impression that the sound has come from in front, and thus the presence of a signal.
  • the matching process was carried out taking into account the Blauert direction-determining bands mentioned above and taking account of individual loudspeaker groups in the sound system.
  • the procedure is in this case similar to the known procedure by acousticians for adjustment of an optimum hearing environment. This procedure is characterized in that groups of mutually associated loudspeakers are processed successively in order to determine their contribution to a desired required frequency response (sequential method).
  • the required frequency response which is used as a reference in this case and is also referred to in the following text as the target function of the level and phase profile over the frequency, is determined during hearing trials.
  • a sound system with all of the individual loudspeakers is simulated in laboratory conditions (low-echo room) as in the situation, for example when producing sound in passenger compartments in motor vehicles.
  • a significant group of trial subjects is in this case offered various sound signals which comprise music of different styles, such as classical, rock, pop, etc.
  • the trial subjects reproduce their subjective hearing impression (tonality, localization capability, presence, staging, etc.) for different settings of the parameters of the sound system, such as cut-off frequencies of the crossover filters of the loudspeakers, the level profile in the various spectral ranges and thus loudspeaker groups (woofers, medium-tone speakers, tweeters) or the phase angle of the sound signals arriving at the location of the test subjects.
  • complex sound systems now allow hearing environments to be created which have desired individual features and which thus, for example, can be associated by trained listeners with specific manufacturers of sound systems and/or, for example, loudspeakers.
  • the loudspeaker groups which have been mentioned further above and have been mentioned for the equalizing of a sound system in order to achieve an optimum listening environment in this case, by way of example, comprise the groups of sub-woofers, woofers, rear, side, front and centre, and the phases of these loudspeaker groups, for example front left and front right, are matched by the equalizing process such that signals from the respective loudspeaker groups arrive as far as possible in the same phase as the left and right ear, thus making it possible to achieve the best-possible location capability effect.
  • the process of adjustment of the tonality is started once the phases of the individual, independent loudspeaker groups have been matched.
  • the individual loudspeaker groups are first of all equalized separately with respect to the level, corresponding to the sum target function. This results in all of the medium-high-tone loudspeaker pairs sounding similar. Excessive levels in an individual loudspeaker group and/or in an individual spectral range would reduce the so-called sweet spot, that is to say that spatial area in which the listening experience is at its best in terms of the stated parameters, since the localization is fixed on that loudspeaker group which actually produces the highest level for the signal being reproduced at that time.
  • the levels of these individual groups are then matched to one another. This is done in a simple form by changing the maxima of the measured sound levels of the individual broadband loudspeaker groups to a common level value. This can be done by reducing the levels of specific loudspeaker groups, increasing the levels of specific loudspeaker groups or by a mixture of these techniques. In each case, care is taken to ensure that none of the loudspeaker groups is overdriven by raising the level, which could result in undesirable effects, such as non-linear distortion, while excessive reduction in the level would no longer ensure adequate transmission of all of the frequency components associated with this loudspeaker group.
  • the levels for matching of the bass channels which are likewise predistorted in the previous equalizing process, are in this case determined using a somewhat modified method, to be precise by relating the sum function of all of the loudspeaker groups for the medium-tone range to a target function.
  • the levels of the bass channels are dealt with differently during the matching process.
  • the level, averaged over the frequency range of the respective loudspeaker group, of this loudspeaker group can also be used as a measure for the extent to which the individual loudspeaker groups must be matched to one another, that is to say must be changed to a common, medium level value. In this case, care is taken, as mentioned above, to ensure that this matching process does not lead to undesirable effects such as excessively high or excessively low sound levels from the individual loudspeaker groups.
  • volume and loudness that are used in this context relate to the same sensitivity variable and differ only in their units. They take account of the frequency-dependent sensitivity of the human ear.
  • the psycho-acoustic variable loudness indicates how loud a sound event at a specific level, with a specific spectral composition and for a specific duration is perceived to be subjectively.
  • the loudness is doubled when a sound is perceived as being twice as loud and thus allows comparison of different sound events with respect to the perceived volume.
  • the unit for assessment and measurement of loudness is in this case the sone.
  • a sone is defined as the perceived volume of a sound event of 40 phons, that is to say the perceived volume of a sound event which is perceived as being equally loud to a sinusoidal tone at the frequency of 1 kHz with a sound pressure level of 40 dB.
  • volume as perceived by people in this case depends on the sound pressure level, the frequency spectrum and the behaviour of the sound over time and is likewise used for modelling of masking effects.
  • standardized measurement methods for loudness measurement also exist according to DIN 45631 and ISO 532 B.
  • Figure 2 illustrates curves of equal volume.
  • the frequency is plotted logarithmically on the abscissa, and the level L of the offered narrowband sounds is plotted along the ordinate.
  • L N whose unit is the phon, and associated loudnesses N whose unit is the sone
  • tones or noises with the same sound pressure level L are perceived as being quieter at low and high frequencies than at medium frequencies.
  • the illustration in Figure 2 has been taken from E. Zwicker and R. Feldtkeller, Das Ohr als jos josenempf briefly [The ear as an information receiver], S. Hirzel Verlag, Stuttgart, 1967 .
  • the A-assessment is a frequency-dependent correction of measured sound levels, by means of which the physiological hearing capability of the human ear is simulated, with the level values which result from this assessment being stated using dB(A) as the units.
  • dB(A) the level values which result from this assessment
  • a considerably different matching process is obtained, however, by further subdividing the frequency range into subgroups rather than making use of the relatively coarse subdivision of the offered frequency band, as is initially carried out by means of the individual loudspeaker groups. This prevents any level peaks in closely bounded frequency ranges in a loudspeaker group resulting in a corresponding reduction of all of the frequency ranges represented by this loudspeaker group.
  • This subdivision can, in this case, be carried out in fractions of thirds for example, or in regions which are oriented to the characteristics of the human hearing. This subdivision will be described in more detail further below.
  • the sum function itself which is obtained from the addition of the individual, equalized ranges and groups is equalized in a further process step.
  • the procedure is in this case once again similar to the known procedure by acousticians for adjustment of an optimum hearing environment, that is to say the sequential processing of loudspeaker groups.
  • the group with the greatest influence on the profile of the sum level is first of all changed such that this results in a profile that is as close as possible to the required frequency response.
  • This change to the loudspeaker group with the greatest influence is carried out within previously defined limits, which once again ensure that none of the loudspeaker groups is overdriven by raising the level, which could result in undesirable effects such as non-linear distortion, while excessively reducing the level could mean that adequate transmission of all frequency components associated with this loudspeaker group was no longer ensured.
  • staging and spatial sensitivity can be influenced by the change in the sequence of processing of the groups, with desirably good staging being achieved in particular when the volumes of the various loudspeaker groups are changed with respect to one another. If, by way of example, front-seat passengers were to be given the hearing impression that the staging is perceived further in front, the rear and/or the side loudspeakers would have to be reduced and/or the front loudspeakers or the centre loudspeaker would have to have their or its levels raised.
  • the perceived location of the staging is initially too far upwards or downwards, or else too far forwards or backwards, the desired effect can be achieved, that is to say the perceived location of the staging can be optimized as desired, by appropriate moderate level changes in the area of the Blauert direction-determining bands (see Figure 1).
  • moderate level changes in the area of the Blauert direction-determining bands or if individual loudspeaker groups are raised or lowered in order to optimize the staging, a subsequent change in the sum level which has already been matched to the required frequency response and thus a renewed, possibly undesirable, discrepancy from the required frequency response, can result.
  • the sequential processing is defined in advance in a specific manner, according to the invention.
  • the procedure according to the invention comprises definition of the sequence of processing of the individual loudspeaker groups for adjustment of the equalizing, in advance, in such a way that this empirically ensures that the discrepancy from the approximation that has already been achieved to the required frequency response is minimized.
  • the equalizing be carried out in the following sequence of loudspeaker groups: sub-woofer, woofer, rear, side, centre and front.
  • Variations in this fixed predetermined sequence can in this case be defined depending on the situation with regard to the current acoustic environment and the preference for a specific acoustic configuration. For example, from experience, it is possible in this case to interchange the rear and side as well as the centre and front loudspeakers in the sequence with the desired staging still being produced in this case as well, but allowing variations in the overall impression of the acoustic environment. This allows good staging to be achieved by skilful choice, defined in advance, of the sequence of processing of the loudspeaker groups during the procedure per se, without excessively changing the sum level which has already been matched to the required frequency response.
  • the aim is to carry out an equalizing process which is as independent as possible of position, for acoustic presentation in motor vehicles.
  • This means that the aim of the equalizing process should not only result in a sweet spot as such but should also cover the region of optimum presentation, covering as large a spatial area as possible, while providing spatial areas of optimum presentation that are as large as possible at the respective positions of the driver and front-seat passenger as well as in the rear row or rows of seats.
  • crossover filters also referred to as frequency filters
  • these crossover filters must be adjusted as a first step before carrying out any equalizing process on the entire sound system.
  • manual adjustment such as this can be carried out quickly and effectively if, as in the present case, the physical data for the loudspeakers and their installation state are known.
  • FIR filters finite impulse response filters
  • IIR filters infinite impulse response filters
  • FIR filters are characterized in that they have an extremely linear frequency response in the transmission range, a very high cut-off attenuation, linear phase and constant group delay time, have a finite impulse response and operate in discrete time steps, which are normally governed by the sampling frequency of an analogue signal.
  • y(n) is the initial value of the time n and is calculated from the sum, weighted with the filter coefficients b i , of the N most recently sampled input values x(n-N) to x(n).
  • the desired transfer function and thus the filtering of the signal are achieved by the definition of the filter coefficients b i .
  • IIR filters In contrast to FIR filters, IIR filters also use already calculated initial values in the calculation (recursive filters) and they are characterized in that they have an infinite impulse response, no initial oscillations, no level drop and a very high cut-off attenuation.
  • the disadvantage in comparison to FIR filters is that IIR filters do not have a linear phase response, as is often highly desirable in acoustic applications. Since the calculated values in the case of IIR filters become very small after a finite time, however, the calculation can in practice be terminated after a finite number of sample values n, and the computation power complexity is considerably less than that required for FIR filters.
  • y(n) is the initial value of the time n and is calculated from the sum, weighted with the filter coefficients b i , of the sampled input values x(n) added to the sum, weighted with the filter coefficients a i of the initial values y(n).
  • the desired transfer function is once again achieved by the definition of the filter coefficients a i and b i .
  • IIR filters may in this case be unstable, but have a higher selectivity for the same implementation complexity.
  • the filter chosen is that which best satisfies the required conditions taking into account the requirements and computation complexity associated with them.
  • crossover filters in the form of IIR filters be used.
  • FIR filters were thus used as the basis for the crossover filters in the following text, in which case these crossover filters are adjusted before carrying out the automatic equalizing process according to the invention (AutoEQ) with their parameters first of all being transferred to the subsequent AutoEQ algorithm so that the phase distortion in the transmitted signals caused by these IIR filters can be taken into account in the calculation of the equalizing filters for phase matching, as described further above, for the location capability, and, if necessary, can be compensated for appropriately.
  • the channel gains of the individual loudspeaker groups should likewise also be set before the start of an automatic equalizing process. This may be done manually or automatically.
  • the step-by-step procedure for automatic matching in one preferred embodiment is described, by way of example, as follows:
  • the maximum values of the levels and/or the mean values of the levels can optionally also be assessed for the method steps 1 to 6 as described above, before matching with the A-assessed level.
  • the A-assessment represents a frequency-dependent correction of measured sound levels which simulates the physiological hearing capability of the human ear.
  • FIR filters In contrast to the use of crossover filters, FIR filters, whose advantages have already been described further above, are used in the implementation of the filters as determined for the automatic equalizing (AutoEQ algorithm) in the amplifier of a sound system. Since, depending on the embodiment and in particular when they have a wide bandwidth, these FIR filters can result in stringent requirements for the computation power of a digital signal processor on which they are carried out, the psycho-acoustic characteristics of the human hearing are made use of again in this case, as well. According to the invention this is achieved in that the filtering is carried out by means of FIR filters via a filter bank, with the bandwidth of the filters increasing as the frequency increases, in a manner which corresponds to the frequency-dependent, integrating characteristic of the human hearing.
  • the modelling of the psycho-acoustic hearing sensitivities is in this case based on fundamental characteristics of the human hearing, in particular of the inner ear.
  • the human inner ear is incorporated in the so-called petrous bone, and is filled with incompressible lymph fluid.
  • the inner ear is in the form of a worm (cochlea) with about 2.5 turns.
  • the cochlea in turn comprises channels which run parallel, with the upper and lower channel being separated by the basilar lamina.
  • the cortical organ with the hearing sense cells is located on this lamina.
  • the human hearing comprises different sound stimuli which fall in limited frequency ranges. These frequency bands are referred to as critical frequency groups or else as the critical bandwidth CB.
  • the frequency group width has its basis in the fact that the human hearing combines sounds which occur in specific frequency ranges, in terms of the psycho-acoustic hearing sensitivities which result from these sounds, to form a common hearing sensitivity. Sound events which are within a frequency group in this case produce different influences than sounds which occur in different frequency groups. Two tones at the same level within one frequency group are, for example, perceived as being quieter than if they were in different frequency groups.
  • the frequency groups have a bandwidth of 100 Hz.
  • the frequency groups have a bandwidth which corresponds to about 20% of the mid-frequency of the respective frequency group ( Zwicker, E.; Fastl, H. Psycho-acoustics - Facts and Models, 2nd edition, Springer-Verlag, Berlin/Heidelberg/New York, 1999 ).
  • tonality a hearing-oriented non-linear frequency scale which is referred to as tonality, with the Bark as the unit.
  • the non-linear relationship between the frequency and tonality originates from the frequency/location transformation on the basilar lamina.
  • the tonality function has been stated by Zwicker ( Zwicker, E.; Fastl, H. Psycho-acoustics - Facts and Models, 2nd edition, Springer-Verlag, Berlin/Heidelberg/New York, 1999 ) on the basis of monitoring threshold and loudness investigations, in tabular form.
  • 24 frequency groups can actually be arranged in a row in the audibility frequency range from 0 to 16 kHz, so that the associated tonality range is 0 to 24 Bark.
  • a filter bank is preferably formed from individual FIR filters whose bandwidth is in each case 1 Bark or less.
  • FIR filters are used for automatic equalizing as investigations progress and in order to produce embodiments, possible alternatives exist which, for example, comprise rapid convolution, the PFDFC algorithm (Partition Frequency Domain Fast Convolution Algorithm), WFIR filters, GAL filters or WGAL filters.
  • MaxMag For automatic equalizing of the levels and/or amplitudes of the sound system, two different methods were investigated, which are referred to in the following text as "MaxMag” and “Sequential".
  • MaximumMag searches in the manner described further above in all of the available independent loudspeaker groups to find that which, in terms of its maximum or average level, is furthest away from the target function of the frequency profile and thus provides the greatest contribution to approximation to the target function by raising or lowering the level.
  • the value which is set for the selected loudspeaker group within the permissible limit values is that which allows the greatest possible approximation to the target function and, following this, the loudspeaker group which is selected and whose level is changed is that which now has the greatest level difference from the target function from the group of loudspeaker groups whose levels have not yet been matched.
  • This method is continued until either the target function is reached with sufficient accuracy or the dynamic limits of the overall system, that is to say the permissible reductions or increases (limit values) by equalizers are exhausted within the respective loudspeaker groups.
  • the sequential method processes the existing loudspeaker groups successively in a previously defined sequence, in which case the user can produce the described influence on the mapping of the staging by the previous definition of the sequence.
  • the automatic algorithm also attempts to achieve the best approximation to the target function just by the equalizing of the first loudspeaker group within the permissible limits (dynamic range).
  • each group no longer reaches its maximum dynamic limits at each frequency location but may now only act at the restricted dynamic range.
  • the algorithm uses the ratio of the signal vectors of the relevant group to the existing sum signal vector at this frequency location as a weighting parameter. This avoids the first groups provided for processing being excessively (over a broad bandwidth) attenuated.
  • the self-scaling target function which is oriented on the minimum of the sum function and then scales the target function such that the minimum value of the sum transfer function in a predetermined frequency range is located exactly by the maximum permissible increase below the target function, this indicated the strengths and weaknesses of the two versions "MaxMag" and "Sequential".
  • this procedure can lead to the level profile of the first loudspeaker group, which is modified by equalizing using the described "sequential" method, being raised or lowered more than proportionally over a broad bandwidth while, in contrast, the other loudspeaker groups which are processed using the "sequential” method, are not subject to any changes, or only to minor changes, since the target function has already been largely approximated by the equalizing of the first loudspeaker group.
  • the first loudspeaker in the defined sequence may experience a major increase or attenuation as the result of this procedure, with the following loudspeaker groups remaining largely unchanged, so that the frequency range which is represented by the first loudspeaker group is more than proportionally amplified or attenuated, which could lead to a considerable discrepancy from the desired sound impression.
  • the "sequential" method was thus subsequently modified such that a single loudspeaker group may now no longer be raised or lowered within its theoretical maximum available dynamic range, but only within a dynamic range which is less than this.
  • This reduced dynamic range is calculated from the original maximum dynamic range by weighting this original maximum dynamic range with a factor which is obtained from the ratio of the overall level of the relevant loudspeaker group to the totaled overall level from all of the loudspeaker groups in this frequency range in the relevant loudspeaker group, so that this factor is always less than unity and results in a restriction to the maximum dynamic range which can be regulated out for the relevant loudspeaker group.
  • This reliably avoids the level profiles of the first loudspeaker groups that are processed in the sequence previously determined being undesirably strongly raised or lowered in the course of the automatic equalizing process.
  • the target function to be achieved is raised or lowered over its entire level profile (parallel shifting of the level profile without changing the frequency response, also referred to in the following text as scaling), such that, in predetermined frequency ranges, the interval between this target function and the sum function of the level profile of all the loudspeaker groups to be considered and to be adjusted by the automatic equalizing process is not greater than the maximum increase or decrease as determined using the above method in the level profile of the individual loudspeaker groups.
  • the specified frequency ranges in which the level profiles of the target function and sum function of all the loudspeaker groups are compared may, for example be oriented to the transmission bandwidths of the loudspeaker groups being used, but preferably to the Bark scale, as explained further above, that is to say in the region of frequency-group wide frequency ranges or partial ranges, thus once again taking account of the physiological hearing capability of the human hearing in this case in particular tone level perception and volume sensitivity (loudness).
  • the results of the loudspeaker settings achieved by the two “sequential” and “MaxMag” methods on the basis of the embodiment described above were obtained by hearing trials with suitable subjects, that is to say subjects with experience in the assessment of sound environments produced by sound systems. In this case, these trials were carried out in order to assess the major parameters of the hearing impression, such as location capability, tonality and staging for in each case four seat positions in the passenger compartment of a motor vehicle. These seat positions comprise the driver, front-seat passenger, rear left and rear right.
  • an automated process can result in an unsuitable processing sequence of the loudspeaker groups. For example, it is possible for a situation to occur in which the automated algorithm for equalizing first of all identifies, in the case of the loudspeaker group for the front loudspeakers, the greatest contribution for the desired approximation to the target function, and correspondingly strongly raises or lowers its level profile.
  • the front loudspeakers in particular contribute a major proportion to, for example, good staging and, furthermore, this relates to their transmission quality, they are relatively unproblematic in comparison to other loudspeaker groups in the sound system by virtue of the installation location and the loudspeaker quality which can thus be used.
  • further loudspeaker groups which may have disturbing spectrum components that have an adverse effect on the location capability will no longer be included in the automatic equalizing process, resulting in the parameters becoming worse, in the manner which has been mentioned.
  • the target function to be achieved is raised or lowered over its entire level profile (scaling, parallel shifting of the level profile without variation of the frequency response), such that the interval between this target function and the sum function of the level profile of all the loudspeaker groups to be considered and to be adjusted by the automatic equalizing process is no greater in predetermined frequency ranges than the maximum permissible increase or decrease in the level profile of the individual loudspeaker groups in the respective frequency range.
  • the target function to be approximated by the equalizing process is aligned by virtue of this scaling in its absolute position at the minimum level of the sum function of the level profile of all the loudspeaker groups to be considered, which generally leads to a reduction, which in some cases is considerable, in this target function to be approximated, since the sum function of the level profile of all the loudspeaker groups to be considered normally has a highly fluctuating profile with pronounced maxima, and, in particular, minima.
  • pre-equalizing of the levels of the individual loudspeaker groups (not the sum function) to the target function of the level profile, with this pre-equalizing process being coordinated with the equalizing of the phases as already described further above and as carried out even before the equalizing, in which the phases are matched by equalizing such that signals from the respective loudspeaker groups arrive as far as possible in phase at the left ear and at the right ear.
  • the equalizing values which are determined in the course of the pre-equalizing process may in this case be used as initial values for the subsequent, final equalizing by means of the "sequential" method.
  • the levels of the loudspeaker groups as approximated to the target function in a first step by means of the pre-equalizing process must, however, be matched to one another within their frequency ranges which are bounded by the respectively associated crossover filters.
  • the level difference between the groups is also amplified by the pre-equalizing process, because the dynamic range of the equalizer is designed such that major reductions, but only slight increases, are permitted. If the frequency response of a group differs to a major extent from the target function, a considerable level reduction must therefore be expected. Major level increases are therefore not permissible, because they will be perceived as disturbing, particularly in conjunction with high filter Q factors.
  • the desired result of the described method is obtained in that, once the equalizing steps have been carried out, the transmission response of all the loudspeaker groups is maintained over a broad bandwidth and the loudspeaker groups each in their own right make a contribution to the overall sound impression, which leads to good tonality and the largest possible sweet spot at all four passenger locations under consideration.
  • the resultant sum transfer function that is to say the addition of the level profiles over all of the loudspeaker groups, is approximated by the step of pre-equalizing in its own right to the target function of the desired level frequency response to such an extent that this target function need no longer be reduced to such a major extent in the scaling process with respect to the sum function minima, which are in consequence less pronounced.
  • this is once again a precondition for the use according to the invention of one of the two methods already described (“sequential” and “MaxMag”) for automatic equalizing of the sum of the level profiles of all the loudspeaker groups in the sound system, in order, in the end, also to obtain a balanced sound impression at all seat positions.
  • a loudspeaker-specific crossover filter would admittedly make it possible for each loudspeaker in a loudspeaker group, normally a loudspeaker pair, to be operated with maximum efficiency in its frequency range, but loudspeaker environments or installation conditions which are not the same can result in situations in which the transmission range of one loudspeaker in a loudspeaker group differs to a major extent from that of another loudspeaker in the same loudspeaker group. If the crossover filters in a situation such as this were designed on a loudspeaker-specific basis, this could likewise lead to undesirable spatial shifts in the resultant sound impression.
  • the loudspeaker-specific pre-equalizing both of the phase response and of the magnitude frequency response, as well as the matching of the channel gain fine matching of the sum transfer function is now carried out, that is to say of the sum of the level profiles of all the loudspeakers involved, to the target function.
  • the process based on the "MaxMag” method is in this case preferred to the process based on the "sequential" method.
  • the pre-equalizing process is now carried out on a loudspeaker-specific basis, only a small number of narrowband frequency ranges of individual loudspeakers now need to be modified by the filter algorithm in order to achieve the desired approximations of the target function, and the broadband and major level changes produced by the equalizing filters, which in the past when using the "MaxMag” method have led to the undesirable results in terms of the location capability, no longer occur.
  • the results of the hearing trials confirm that, for using the loudspeaker-specific pre-equalizing process, a good localization capability is now achieved even with the process for automatic equalizing based on the "MaxMag” method, in which case the tonality was also additionally improved by the previous loudspeaker-specific pre-equalizing process.
  • the crossover filters were adjusted manually in the course of the previous investigations, for simplicity reasons.
  • an approach is searched for in order to carry out this adjustment process automatically as well, since the aim of the present invention is to develop automatic equalizing, which is as comprehensive as possible and covers all aspects, of a sound system in a motor vehicle, including the adjustment of the crossover filters in the automatic equalizing process, as well.
  • the following disclosure relating to the automatic adjustment of the crossover filters is based on the assumption that Butterworth filters of a sufficient order are, in principle, sufficient for the desired delineation of the respective frequency response of the relevant loudspeaker.
  • the empirical values of acousticians, maintained over many years, for the equalizing of sound systems show that fourth-order filters are adequate both for high-pass and low-pass filters in order to achieve the desired crossover filter quality.
  • a higher-order filter would result in advantages, for example by having a steeper edge gradient, however the amount of computation time required for this purpose for implementation in digital signal processors would rise in a corresponding manner at the same time.
  • Fourth-order Butterworth filters are therefore used in the following text.
  • the transfer function of the left rear loudspeaker measured binaurally using the described measurement method and averaged over the recordings at the driver's seat and the front-seat passenger's seat, is shown in comparison to the target function being used in the top left of Figure 3.
  • a suitable upper cut-off frequency of a crossover low-pass filter can be determined quite easily in the present case.
  • the difference is formed between the target function and the transfer function of the respective loudspeaker as determined after the pre-equalizing process.
  • the result associated with the example under discussion is shown in the illustration at the bottom left in Figure 3.
  • This difference transfer function which is also referred to for short in the following text as the difference, is then investigated in the next step, to determine the frequency of this difference function at which it is within, above, or below a specific, predetermined limit range.
  • the human hearing inter alia has a frequency resolution related to the frequency
  • the difference transfer function as calculated from the measured data and the target function was introduced into a level difference function, which had been smoothed by averaging, before evaluation of whether the limit range had been overshot or undershot.
  • the mean value at the respective frequency is in this case preferably calculated from empirical values over a range with a width of 1/8 third octave band (in the following mentioned just as "third"). This means that the frequency resolution of the smoothed level difference function is high at low frequencies and decreases as the frequency increases. This corresponds to the fundamental frequency-dependent behaviour of the human hearing to whose characteristics the illustration of the level difference function in Figure 3 is thus matched.
  • the level difference spectrum is then smoothed once again in a further processing step with the aid of a simple first-order IIR low-pass filter in the direction from low to high frequencies and in the direction from high to low frequencies in order to eliminate bias problems and smoothing-dependent frequency shifts resulting from them.
  • the level difference spectrum processed in this way is now compared by the automatic algorithm with the range limits (in this case +/-6 dB), and this is used to form a value for the trend of the profile of the level difference spectrum.
  • the value "1” for this trend denotes that the upper range limit has been exceeded at the respective frequency of the level difference spectrum
  • the value "-1" indicates that the lower range limit of the level difference spectrum has been undershot at the respective frequency
  • the value "0" for the trend indicates level values of the level difference spectrum at the respective frequency which are within the predetermined range limits.
  • the level difference spectra are initially unknown in an automated method, that is to say when using an automatic algorithm, it is possible for a situation to occur in which predetermined range limits are exceeded within a relatively narrow spectral range when, for example, the loudspeaker and/or the space into which sound is being emitted have/has a narrowband resonance point, and the profile of the level difference spectrum then falls again below the predetermined range limit (situations of the same type can also occur when the predetermined range limits are undershot). In situations such as these, the previously described method cannot determine clear cut-off frequencies for the crossover filters.
  • the level values determined by averaging using a filter in each case with a width of 1/8 third are thus investigated for the frequency of successive overshoots and undershoots of the predetermined range limits. Only when a specific minimum number (which can be predetermined in the algorithm) of related overshoots and undershoots of the predetermined range limits is overshot at successive frequency points is this interpreted by the algorithm as reliable overshooting or undershooting of the predetermined range limits, and thus as a frequency position of a cut-off frequency of the crossover filter.
  • this minimum number of associated level values which overshoot or undershoot the range limits (+/-6 dB) is typically about 5-10 level values.
  • this extreme value of the largest found and related level overshoot or level undershoot range is in this case below a specific cut-off frequency (for example about 1 kHz), and if this extreme value furthermore also has a negative value (minimum), then the decision is made to use a high-pass filter for the sought crossover filter.
  • a specific cut-off frequency for example about 1 kHz
  • this extreme value furthermore also has a negative value (minimum)
  • the decision is made to use a high-pass filter for the sought crossover filter.
  • a search is now carried out, starting from the frequency of the minimum, in the direction of higher frequencies within the level difference function as determined after pre-equalizing for its first intersection with the 0 dB line. This frequency denotes the filter cut-off frequency of the crossover high-pass filter.
  • the decision is made to use a low-pass filter for the sought crossover filter.
  • a specific cut-off frequency for example about 10 kHz
  • this extreme value furthermore also has a negative value (minimum)
  • the decision is made to use a low-pass filter for the sought crossover filter.
  • a search is now carried out starting from the frequency of the minimum in the direction of lower frequencies within the level difference function as determined after pre-equalizing, for its first intersection with the 0 dB line. This frequency denotes the filter cut-off frequency of the crossover low-pass filter.
  • crossover filter cut-off frequencies for all of the broadband loudspeakers in the medium and high-tone range of the sound system to be regulated and to be equalized are determined and set in the manner described above.
  • the crossover filter cut-off frequencies of the narrowband low-tone loudspeakers must be dealt with separately, in further steps, and are restricted here just to logical range limits which, however, still need not represent final values.
  • This prior stipulation is important for the described method because, once all of the crossover filter cut-off frequencies have been set, the complete automatic equalizing process (AutoEQ) is carried out once again in order to achieve a more accurate approximation to the target function, with the crossover filters being taken into account, in a second run.
  • AutoEQ complete automatic equalizing process
  • the crossover filters of all of the broadband loudspeakers have been defined and the crossover filters of the narrowband loudspeakers in the low-tone range have been preset to suitable values, the search for better filter cut-off frequency values for the low-tone loudspeakers can be started. This procedure is necessary because the frequency transition from the narrowband loudspeakers for low-tone reproduction to the broadband loudspeakers depends on the nature and number of the low-tone loudspeakers being used and thus cannot easily be determined in a comparable manner.
  • the crossover filter cut-off frequencies of the woofers are in this case always defined and determined in the same way and a distinction is just drawn in the calculation of the crossover filter cut-off frequencies for the sub-woofer between the two situations mentioned above.
  • the crossover filter cut-off frequencies of the sub-woofer are in this case calculated in the same way as that for the woofer stereo pair in the situation in which only one subwoofer and no woofer stereo pair is used. Only in the situation in which a woofer stereo pair is also present in addition to the sub-woofer is the way in which the crossover filter cut-off frequencies of the sub-woofer are calculated changed.
  • the upper spectral range within which a search is carried out for a minimum distance in this case results from the upper filter cut-off frequency of the woofer loudspeakers, which has already been determined prior to this, that is to say during the search for the crossover filter cut-off frequencies of the broadband loudspeakers.
  • the upper limits of the lower spectral range for searching for the cut-off frequency results from twice the value of the lower limit.
  • the procedure for determination of the crossover filter cut-off frequencies for the relevant loudspeakers or loudspeaker groups is identical in the situation in which the sound system either comprises only a single sub-woofer loudspeaker, or a stereo pair formed from woofer loudspeakers.
  • the following text explains and describes the transfer functions and level profiles of a single sub-woofer or of a woofer stereo pair, as well as the procedure for determination of the associated crossover filter cut-off frequencies.
  • the filter cut-off frequency or the filter cut-off frequencies of the sought crossover filter for the woofer loudspeakers has or have its or their frequency varied within the permissible limits of the lower or upper spectral range, respectively, for as long as it is possible in this way to reduce the magnitude of the mean value, formed from the profile of the difference between the sum magnitude frequency response and the target function (distance).
  • the filter cut-off frequency of the upper crossover filter is reduced at most until the filter cut-off frequency of the lower crossover filter is reached, or is increased at most until the maximum permissible filter cut-off frequency of the low-tone loudspeakers (about 500 Hz) is reached.
  • the filter cut-off frequency of the lower crossover filter is reduced at most until the minimum permissible filter cut-off frequency of the low-tone loudspeakers (about 10 Hz) of the lower crossover filter is reached or is increased at most until the filter cut-off frequency of the upper crossover filter is reached.
  • this method leads to crossover filters whose filter cut-off frequencies are set such that they have reached either their minimum or their maximum permissible range limits, or are located within the frequency range predetermined by these range limits and are set such that the magnitude of the mean value of the distance between the lower range limits of the lower spectral range and the upper range limits of the upper spectral range is minimized.
  • This is illustrated, once again by way of example, in the two lower illustrations in Figure 4, with the left-hand illustration once again showing the magnitude frequency responses of the transfer function and the right-hand illustration showing the frequency responses of the level functions.
  • this method is used when the sound system either has only a single sub-woofer loudspeaker for low-tone reproduction or has only one stereo pair, formed from woofer loudspeakers.
  • the following text describes the procedure for determination of the cut-off frequencies of the crossover filters for the situation in which the sound system comprises not only the stereo pair as described above, formed from woofer loudspeakers, but at the same time, in addition to this, a sub-woofer loudspeaker as well.
  • the method according to the invention is in this case dependent on the filter cut-off frequencies of the crossover filters for the stereo pair that is formed from woofer loudspeakers in this situation being calculated in advance and being already available, since these are used as input variables for determination of the filter cut-off frequencies of the crossover filter for the sub-woofer.
  • its upper cut-off frequency is first of all set as a start value to the value of the upper cut-off frequency of the upper crossover filter of the woofer loudspeakers, and the already previously determined lower filter cut-off frequency is used to determine the new lower and upper range limits for the permissible filter cut-off frequencies in the same way as that which has already been described for the woofer loudspeakers.
  • the filter cut-off frequencies of the crossover filters for the sub-woofer loudspeaker are now found in a different way than would be the case if the sub-woofer were to be the only loudspeaker responsible for reproduction of the low frequencies of the sound system.
  • the sum magnitude frequency responses are in each case determined for this purpose with and without inclusion of the subwoofer loudspeaker and the corresponding target functions are determined for each of these two sum magnitude frequency responses, and the respectively associated difference transfer functions are calculated. These are then once again averaged using the described methods and are in each case changed to the appropriate level function.
  • the top left illustration in Figure 5 in this case shows the magnitude frequency responses of the target function, of the difference function as well as of the sum function including the sub-woofer and the range limits derived from this for the permissible upper and lower spectral range for the filter cut-off frequencies of the crossover filters for the sub-woofer loudspeaker.
  • the top right illustration in Figure 5 in contrast shows the unaveraged and averaged level functions of the differences, in each case with and without a sub-woofer. As can be seen from this, the difference function is increased by inclusion of the sub-woofer loudspeaker, that is to say the discrepancy is undesirably increased.
  • the filter cut-off frequencies of the crossover filters for the sub-woofer loudspeaker must therefore be changed by the algorithm in order once again to achieve a distance which is at least just as short from the target function, as was the case without consideration of the sub-woofer.
  • This iterative method is continued until the system including the sub-woofer is at a distance from the target function which is at most just as great as was the case previously for the sound system without a sub-woofer.
  • the difference between the sound system without a sub-woofer loudspeaker, as previously determined in the processing step, and the target function is used as a reference for this iteration.
  • the complete automatic algorithm of the equalizing process is carried out once again, but with the previously determined cut-off frequencies of the crossover filters remaining fixed, and not being modified again in this repeated run.
  • the impulse responses are determined using the crossover filters defined in the meantime, first of all for all of the individual loudspeakers in the sound system, as well as for all the loudspeakers jointly - once with and once without a sub-woofer - before running through the algorithm for automatic equalizing (AutoEQ) once again, that is to say once the phase equalizing and loudspeaker-specific pre-equalizing have already been carried out.
  • AutoEQ automatic equalizing
  • Figure 6 shows the measured transfer functions for the front left and front right individual loudspeakers (Front-Left and FrontRight in Figure 6), for the left side and right side individual loudspeakers (SideLeft and SideRight in Figure 6), for the rear left and rear right individual loudspeakers (RearLeft and RearRight in Figure 6), for the woofer individual loudspeakers on the left and right (WooferLeft and WooferRight in Figure 6), the centre loudspeaker (Center in Figure 6), the sub-woofer loudspeaker (Sub in Figure 6), and for all of the loudspeakers jointly without any sub-woofer loudspeaker (Broadband-Sum+Woofer in Figure 6) and for all of the loudspeakers jointly including a sub-woofer loudspeaker (Complete Sum), in this case all in comparison to the defined target function (Target Function in Figure 6).
  • the settings and values determined in the first run through the AutoEQ algorithm are likewise used
  • the left illustration in Figure 8 shows the principle for the measurements of the binaural transfer functions for the front left and front right positions in the passenger compartment, using the example of the centre loudspeaker C, which in this case represents an example of the presentation of mono signals. Furthermore, the left illustration in Figure 8 shows the two front left FL_Pos and front right FR_Pos measurement positions and, associated with them, the positions simulated by the measurement microphones for the left ear L and the right ear R in each case at these measurement points.
  • the transfer function from the centre loudspeaker C to the left ear position L of the front left measurement position FL_Pos is annotated H_FL_Pos_CL
  • the transfer function from the centre loudspeaker C to the right ear position R of the front left measurement position FL_Pos is annotated H_FL_Pos_CR
  • the transfer function from the centre loudspeaker C to the left ear position L of the front right measurement position FR_Pos is annotated H_FR_Pos_CL
  • the transfer function from the centre loudspeaker C to the right ear position R of the front right measurement position FR_Pos is annotated H_FR_Pos_CR.
  • the localization of mono signals depends essentially on inter-aural level differences IID and inter-aural delay-time differences ITD, which are formed by the transfer functions H_FL_Pos_CL and H_FL_Pos_CR on the left front seat position, and by the transfer functions H_FR_Pos_CL and H_FR_Pos_CR on the right front seat position, respectively.
  • the right-hand illustration in Figure 8 shows the principle of the measurements of the binaural transfer functions for the front left and front right positions in the passenger compartment, using the example of the front loudspeaker pair FL (front left loudspeaker) and FR (front right loudspeaker), which in this case represent examples of the presentation of stereo signals.
  • the right-hand illustration in Figure 8 once again shows the two measurement positions, front left FL_Pos and front right FR_Pos, as well as the associated positions which are modelled by the measurement microphones respectively for the left ear L and the right ear R at these measurement points.
  • the transfer function from the front left loudspeaker FL to the left ear position L at the front left measurement position FL_Pos is annotated H_FL_Pos_FLL
  • the transfer function from the front left loudspeaker FL to the right ear position R at the front left measurement position FL_Pos is annotated H_FL_Pos_FLR
  • the transfer function from the front left loudspeaker FL to the left ear position L of the front right measurement position FR_Pos is annotated H_FR_Pos_FLL
  • the transfer function from the front left loudspeaker FL to the right ear position R at the front right measurement position FR_Pos is annotated H_FR_Pos_FLR
  • the transfer function from the front right loudspeaker FR to the left ear position L at the front left measurement position FL_Pos is annotated H_FL_Pos_FRL
  • the transfer functions for the further loudspeaker groups which are arranged in pairs and comprise the woofer, the loudspeakers arranged at the side and the rear loudspeakers, are obtained in a corresponding manner.
  • the addition of the sum transfer functions and sum levels resulting from these transfer functions and the weightings of the measurement points, for the complete sum transfer function of the sound system, can easily be derived from the exemplary description of the situations for mono signals and stereo signals shown in Figure 8, and will therefore not be described in detail here.
  • the respective binaural transfer functions in the form of impulse responses of the sound system and of its individual loudspeakers and loudspeaker groups are, however, measured not only at the two front seat positions but also at the two rear positions, in the case of a vehicle which has a second row of seats.
  • the algorithm can be extended to, for example, the seat positions in a third row of seats, for example as in minibuses or vans, by appropriate distribution of the weighting of the components for the seat positions at any time.
  • the invention is not restricted to vehicle interior but is also applicable with all kinds of rooms, for example living rooms, concert halls, ball rooms, arenas, railway stations, airports, etc. as well as under open air conditions.
  • the large number of measured transfer functions of a single loudspeaker must be combined at the left and right ear positions at the respective seat positions to form a common transfer function, in order to obtain a single representative transfer function for each individual loudspeaker in the sound system, for processing in the algorithm for automatic equalizing.
  • the weighting with which the transfer functions at the various seat positions are in each case included in the addition process for the transfer function can in this case be chosen differently depending on the vehicle interior (vehicle type) and preference for individual seat positions.
  • the following text describes a procedure which has been used in the course of the investigations relating to the present invention, although the algorithm according to the invention is not restricted to this procedure.
  • the respective components at the various seat position are weighted, to be precise, both for the magnitude frequency response and for the phase frequency response, at the various seat positions.
  • the annotations for a vehicle interior with two rows of seats are in this case as follows:
  • H_FLL ⁇ * H_FL_Pos_FLL + ⁇ * H_FR_Pos_FLL + ⁇ * H_RL_Pos_FLL + ⁇ * H_RR_Pos_FLL * e j * ⁇ ⁇ * H_FL_Pos_FLL + ⁇ * H_FR_Pos_FLL + ⁇ * H_RL_Pos_FLL + ⁇ * H_RR_Pos_FLL and for the microphone which in each case represents the right ear and the left front loudspeaker FL
  • H_FLR ⁇ * H_FL_Pos_FLR + ⁇ * H_FR_Pos_FLR + ⁇ * H_RL_Pos_FLL
  • H_FLL FFT h_FLL und
  • H_FRL FFT h_FRL und
  • the resultant, loudspeaker-dependent transfer function was made dependent exclusively on the measurements at the driver's position (generally front left), to be precise by combination of the phase frequency responses of the left and right microphones. None of the other phase frequency responses of the other seat positions were included. This stipulation was made in order initially to restrict the amount of effort associated with this, and in particular that relating to the hearing tests with a significant number of test subjects. More detailed investigations will have to be carried out relating to this in order to determine whether other constellations (weightings) of the superimposition of the phase frequency responses cannot be found which lead to a further improvement in the hearing impression. For example, one approach would be to use a position in the centre of the passenger compartment or else the position between the two front seats as the only point for recording the impulse responses for calculation of the equalizing filters for the phase response.
  • PhaseFL - arctan Im ⁇ FL old / Re ⁇ FL old
  • PhaseFR - arctan Im ⁇ FR old / Re ⁇ FR old
  • the sum transfer function is calculated on the basis of the real and imaginary parts before the equalizing of the sum transfer function is then carried out using the "MaxMag" method, as described in the following text:
  • the filters which have been modified in the course of the iteration process are optionally smoothed again for the pre-equalizing (preferably matched to the hearing, non-linearly, for example with 1/8 third filtering), are then transformed to the time domain using the "frequency sampling” method, and finally optionally have their length limited before being transformed back to the spectral domain, in this way resulting in the final filters for the magnitude equalizing.
  • the FIR filters for the equalizing of the phases are in this case determined using the following method.
  • phaseEQ - arctan Im / Re
  • RePhaseEQ cos PhaseEQ
  • ImPhaseEQ sin PhaseEQ
  • RePhaseEQ RePhase RePhaseEQ ⁇ end - 1 : - 1 : 2
  • ImPhaseEQ ImPhase ImPhaseEQ ⁇ end - 1 : - 1 : 2
  • H_PhaseEQ H_PhaseEQ * H_Delay
  • a position specific equalizing may be based only on sound picked up in said position in view of only those loudspeaker positions which are relevant for said listening position. Further, channel (group) specific equalizing is applied in each position to the effect that only adjacent loudspeaker positions are used for the equalization in order to maintain symmetry.
  • the front channels may include, e.g., the front left and right channels (FL, FR) as well as the center speaker. Those speakers are only relevant for the front left and front right listening positions with respect to cross-over frequency, gain, amplitude, and phase.
  • Figure 9 shows in a diagram an exemplary spectral weighting function for measurements at different positions (FL_Pos+FR_Pos+RL_Pos+RR_Pos)/4 and (FL_Pos+FR_Pos)/2 over frequency.
  • the sound levels may vary depending on the particular position and frequency. Improvements addressing this situation may be reached by a bass management system. Measurements showed that problems especially with woofers and subwoofers arranged in the rear of a car occur in a frequency range of 40Hz to 90Hz which corresponds to a wave length of one half of the length of a vehicle interior indicating that this is because of a standing wave. In particular, measurements of the unsigned amplitude over frequency showed that the unsigned amplitude at the front seats are different from the ones at the rear seats, i.e., at the rear seats a maximum and at the front seats a minimum may occur.
  • the difference between front and rear seats may be up to 10dB especially if the subwoofer is arranged in the trunk of a car (see figure 11).
  • a different position, e.g., under the front seats, of the subwoofer may provide some improvement, the bass management system according improves the sound even more, not only in view of the front-rear mode but also the left-right mode.
  • the bass management system of the present invention creates the same or at least a similar sound pressure at different locations by, i.a., adapting the phase over frequency for one or more of the low frequency loudspeakers. If this successfully took place, it is no problem to adapt the amplitude over frequency to the target function, since all loudspeakers only have to be weighted with an overall amplitude equalizing function to get amplitude over frequency being equal to the target function at all positions.
  • the level over frequency of one position or the average level over frequency of all positions may be taken as a reference wherein subsequently the distance of each individual position to the reference is determined. The individual distances are added leading to a first cost function which stands for the overall distance from the reference mentioned above. To minimize the first cost function, it is investigated what phase shift has what influence to the cost function.
  • a very simple approach is to choose a first group of loudspeakers (which may be only one loudspeaker) or a first channel serving as the reference to which a second group of loudspeakers (which also may be only one loudspeaker) or a second channel is adapted in terms of phase such that the cost function is minimized.
  • a cost function over phase is derived which shows the dependency of the distance from the phase. Determining the minimum of this cost function leads to the phase shift that has to be applied to the respective group or channel in order to reach a maximum reduction of the cost function and, accordingly, a maximum equalization of the sound levels of all positions.
  • Figure 12 illustrates how the cost function is changed by changing the mean sound pressure level.
  • the optimum phase shift is not changed since the original cost function and the modified cost function have their overall minimum at the same position.
  • the phase shift per frequency change (e.g., 1Hz) may be restricted to a certain maximum phase shift, e.g., ⁇ 10°.
  • a certain maximum phase shift e.g., ⁇ 10°.
  • the local minimum is determined for each frequency (e.g., 1 Hz steps) which then is used as a new phase value in the phase equalization process.
  • Figure 16 illustrates the corresponding equalizing phase-frequency response for the front right loudspeaker with respect to the reference signal.
  • a new reference signal is derived through superposition of the old reference signal with the new phase equalized loudspeaker group (or channel).
  • the new reference signal serves as a reference for the next loudspeaker to be investigated.
  • each group of loudspeakers (or channel) can be used as a reference the front left position may be preferred since most car stereo systems will have a loudspeaker in this particular position.
  • Figure 18 illustrates the sound pressure levels over frequency at four positions in the interior of a vehicle with the already mentioned difference between front and rear seats.
  • Figure 19 shows the sound pressure levels over frequency upon filtering the respective electrical sound signals according to the above mention method using the phase equalizing function with no phase limitation.
  • Figure 20 illustrates the case of applying such a phase limitation of ⁇ 10° per frequency step.
  • Figure 21 shows the performance of the bass management system as sound pressure level over frequency using a FIR filter with 4096 taps.
  • a reference is derived from the average amplitude over frequency of all positions under investigation. Said reference is then adapted to a target function by means of an amplitude equalization function which is the same for all positions to be investigated.
  • the target function may be, for example, the manually modified sum amplitude response of the auto equalization algorithm that, in turn, follows automatically its respective target function.
  • the resulting target function for the bass management system is depicted "Target" in figures 22 and 23.
  • FIR filters in general have been used in the examples above, all kind of digital filtering may be used. However, emphasis is put to minimal phase FIR filters which showed the best performance, particularly, in view of the acoustical results as well as the filter length.
  • Figure 25 illustrates the signal flow in a system exercising the methods described above.
  • two stereo signal channels a left channel L and a right channel R
  • a sound processor unit SP generating five channels thereof.
  • Said five channels are a front right channel FR, a rear right channel RR, a rear left RL, a front left channel FL, and a woofer and/or subwoofer channel LOW.
  • Each of said five channels is supplied to a respective equalizer unit EQ_FR, EQ_RR, EQ_RL, EQ_FL, and EQ_LOW for amplitude and phase equalization.
  • the equalizer units EQ_FR, EQ_RR, EQ_RL, EQ_FL, and EQ_LOW are controlled via a equalizer control bus BUS_EQ by a control unit CONTROL which also performs the basic sound analysis for controlling other units of the system.
  • the equalizer units EQ_FR, EQ_RR, EQ_RL, EQ_FL, and EQ_LOW comprise preferably minimal phase FIR filters.
  • Such other units are, e.g., controllable crossover filter units CO_FR, CO_RR, CO_RL, and CO_FL having a controllable crossover frequency and being connected downstream of the respective equalizer units EQ_FR, EQ_RR, EQ_RL, and EQ_FL for splitting each respective input signal into two output signals, one in the high frequency range and the other in the mid frequency range.
  • controllable crossover filter units CO_FR, CO_RR, CO_RL, and CO_FL having a controllable crossover frequency and being connected downstream of the respective equalizer units EQ_FR, EQ_RR, EQ_RL, and EQ_FL for splitting each respective input signal into two output signals, one in the high frequency range and the other in the mid frequency range.
  • the signals from the crossover filter units CO_FR, CO_RR, CO_RL, and CO_FL are supplied via respective controllable switches S_FR_H, S_RR_H, S_RL_H, S_FL_H, S_FR_M, S_RR_M, S_RL_M, and S_FL_M as well as controllable gain units G_FR_H, G_RR_H, G_RL_H, G_FL_H, G_FR_M, G_RR_M, G_RL_M, and G_FL_M to loudspeakers LS_FR_H, LS_RR_H, LS_RL_H, LS_FL_H, LS_FR_M, LS_RR_M, LS_RL_M, and LS_FL_M.
  • the signal from the equalizer unit EQ_LOW is supplied via two controllable switches S_LOW1 and S_LOW2 as well as respective controllable gain units G_LOW1 and G_LOW2 to (sub-)woofer loudspeakers LS_LOW1 and LS_LOW2.
  • the controllable switches S_FR_H, S_RR_H, S_RL_H, S_FL_H, S_FR_M, S_RR_M, S_RL_M, S_FL_M, S_LOW1, S_LOW2 and the controllable gain units G_FR_H, G_RR_H, G_RL_H, G_FL_H, G_FR_M, G_RR_M, G_RL_M, G_FL_M, G_LOW1, G_LOW2 are controlled by the control unit CONTROL via control bus BUS_S or BUS_G, respectively.
  • two microphones MIC_L and MIC_R are arranged in a dummy head DH which is located in the room where the loudspeakers are located.
  • the signals from the microphones MIC_L and MIC_R are evaluated as described herein further above wherein, during the analysis procedure, a certain group of loudspeakers (including groups having only one loudspeaker) may be switched on while the other groups are switched of by means of the controlled switches S_FR_H, S_RR_H, S_RL_H, S_FL_H, S_FR_M, S_RR_M, S_RL_M, S_FL_M, S_LOW1, S_LOW2.
  • the groups may be switched on sequentially according to a given sequence or dependant on the deviation from a target function.

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  • Physics & Mathematics (AREA)
  • Engineering & Computer Science (AREA)
  • Acoustics & Sound (AREA)
  • Signal Processing (AREA)
  • Stereophonic System (AREA)
  • Circuit For Audible Band Transducer (AREA)
EP20070003712 2006-04-05 2007-02-23 Verfahren zur automatischen Entzerrung eines Tonsystems Active EP1843636B1 (de)

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WO2010000807A2 (en) * 2008-07-03 2010-01-07 Bang & Olufsen A/S A system and a method for configuring af loudspeaker system
US9344822B2 (en) 2011-07-08 2016-05-17 Dolby Laboratories Licensing Corporation Estimating nonlinear distortion and parameter tuning for boosting sound
CN111970607A (zh) * 2019-05-20 2020-11-20 宏碁股份有限公司 扬声器调整方法与使用此方法的电子装置
CN115604646B (zh) * 2022-11-25 2023-03-21 杭州兆华电子股份有限公司 一种全景深空间音频处理方法

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SG11201605692WA (en) * 2014-01-16 2016-08-30 Sony Corp Audio processing device and method, and program therefor

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US5581621A (en) 1993-04-19 1996-12-03 Clarion Co., Ltd. Automatic adjustment system and automatic adjustment method for audio devices
DE19720217A1 (de) 1996-11-20 1998-05-28 Korea Electronics Telecomm Vorrichtung und Verfahren zum automatischen Ausgleich eines Mehrkanal-Audiosystems
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US5581621A (en) 1993-04-19 1996-12-03 Clarion Co., Ltd. Automatic adjustment system and automatic adjustment method for audio devices
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WO2010000807A2 (en) * 2008-07-03 2010-01-07 Bang & Olufsen A/S A system and a method for configuring af loudspeaker system
WO2010000807A3 (en) * 2008-07-03 2010-06-24 Bang & Olufsen A/S A system and a method for configuring af loudspeaker system
US9344822B2 (en) 2011-07-08 2016-05-17 Dolby Laboratories Licensing Corporation Estimating nonlinear distortion and parameter tuning for boosting sound
CN111970607A (zh) * 2019-05-20 2020-11-20 宏碁股份有限公司 扬声器调整方法与使用此方法的电子装置
CN111970607B (zh) * 2019-05-20 2022-07-05 宏碁股份有限公司 扬声器调整方法与使用此方法的电子装置
CN115604646B (zh) * 2022-11-25 2023-03-21 杭州兆华电子股份有限公司 一种全景深空间音频处理方法

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