WO2005060307A1 - Vorrichtung und verfahren zum erzeugen eines tieftonkanals - Google Patents
Vorrichtung und verfahren zum erzeugen eines tieftonkanals Download PDFInfo
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- WO2005060307A1 WO2005060307A1 PCT/EP2004/013130 EP2004013130W WO2005060307A1 WO 2005060307 A1 WO2005060307 A1 WO 2005060307A1 EP 2004013130 W EP2004013130 W EP 2004013130W WO 2005060307 A1 WO2005060307 A1 WO 2005060307A1
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- signal
- woofer
- loudspeaker
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04S—STEREOPHONIC SYSTEMS
- H04S3/00—Systems employing more than two channels, e.g. quadraphonic
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
- H04R2201/00—Details of transducers, loudspeakers or microphones covered by H04R1/00 but not provided for in any of its subgroups
- H04R2201/40—Details of arrangements for obtaining desired directional characteristic by combining a number of identical transducers covered by H04R1/40 but not provided for in any of its subgroups
- H04R2201/403—Linear arrays of transducers
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- 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
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- 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
- H04S2420/13—Application of wave-field synthesis in stereophonic audio systems
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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
- H04S7/307—Frequency adjustment, e.g. tone control
Definitions
- the present invention relates to the generation of one or more bass channels, and in particular to the generation of one or more bass channels in connection with a multi-channel audio system, such as, for example, a cellular field synthesis system.
- a multi-channel audio system such as, for example, a cellular field synthesis system.
- wave field synthesis Due to the enormous demands of this method on computer performance and transmission rates, wave field synthesis has so far only rarely been used in practice. It is only the advances in the areas of microprocessor technology and audio coding that allow this technology to be used in concrete applications. The first products in the professional sector are expected next year. The first wave field synthesis applications for the consumer sector are also expected to be launched in a few years.
- Every point that is captured by a wave is the starting point for an elementary wave that propagates in a spherical or circular manner.
- a large number of loudspeakers that are arranged next to each other can be used to simulate any shape of an incoming wavefront.
- the audio signals of each loudspeaker have to be fed with a time delay and amplitude scaling in such a way that the emitted sound fields of the individual loudspeakers overlap correctly. If there are several sound sources, the contribution to each loudspeaker is calculated separately for each source and the resulting signals are added. In a room with reflective walls, reflections can also be reproduced via the loudspeaker array as additional sources.
- the effort involved in the calculation therefore depends heavily on the number of strongly depends on the number of sound sources, the reflection properties of the recording room and the number of speakers.
- the particular advantage of this technique is that a natural spatial sound impression is possible over a large area of the playback room.
- the direction and distance of sound sources are reproduced very precisely. To a limited extent, virtual sound sources can even be positioned between the real speaker array and the listener.
- wave field synthesis works well for environments whose properties are known, irregularities do occur when the nature changes or when the wave field synthesis is carried out on the basis of an environment condition that does not match the actual nature of the environment.
- the technique of wave field synthesis can also be used advantageously to complement a visual perception with a corresponding spatial audio perception. So far, the mediation was an authentic visual impression of the virtual scene v in the foreground in the production in virtual studios.
- the acoustic impression that goes with the image is usually imprinted on the audio signal by manual work steps in what is known as post-production, or is classified as too complex and time-consuming to implement and is therefore neglected. This usually leads to a contradiction of the individual sensations, which leads to the fact that the designed space, i. H.
- wave field synthesis In the audio area, the technique of wave field synthesis (WFS) can be used to achieve a good spatial sound for a large range of listeners.
- wave field synthesis is based on the principle of Huygens, according to which wave fonts can be formed and built up by superimposing elementary waves. According to a mathematically exact theoretical description, an infinite number of sources at infinitely small distances would have to be used to generate the elementary waves. In practice, however, many loudspeakers are finally used at a finite distance apart. Each of these speakers is based on the WFS principle with an audio signal from a virtual source that has a certain delay and one has a certain level. Levels and delays are usually different for all speakers.
- the ellen field synthesis system works on the basis of the Huygens principle and reconstructs a given waveform, for example a virtual source, which is arranged at a certain distance from a presentation area or to a listener in the presentation area by a plurality of single waves.
- the wave field synthesis algorithm thus receives information about the actual position of a single speaker from the speaker array, in order to then calculate a component signal for this single speaker, which this speaker must then ultimately emit so that the listener overlays the speaker signal from one speaker with the speaker signals of the other active ones Loudspeaker carries out a reconstruction in such a way that the listener has the impression that he is not "sonicated” by many individual loudspeakers, but only borrowed from a single loudspeaker at the position of the virtual source.
- each virtual source for each loudspeaker ie the component signal of the first virtual source for the first loudspeaker, the second virtual source for the first loudspeaker, etc.
- the contribution from each virtual source for each loudspeaker is calculated in order to then add up the component signals to finally get the actual loudspeaker signal.
- the overlaying of the loudspeaker signals of all active loudspeakers at the listener would result in the listener not having the impression that he was being exposed to a large array of loudspeakers, but that the sound he hears only comes from three sound sources positioned in special positions that are equal to the virtual sources.
- the component signals are usually calculated by applying a delay and a scaling factor to the audio signal assigned to a virtual source, depending on the position of the virtual source and the position of the loudspeaker, in order to produce a delayed and / or scaled audio signal
- a virtual source that directly represents the loudspeaker signal if only one virtual source is present, or that, after addition with other component signals for the loudspeaker under consideration, from other virtual sources then contributes to the loudspeaker signal for the loudspeaker under consideration.
- Typical wave field synthesis algorithms work regardless of how many speakers are in the speaker array.
- the theory on which the wave field synthesis is based is that any sound field can be exactly reconstructed by an infinitely high number of individual speakers, the individual single speaker being arranged infinitely close to one another. In practice, however, neither the infinitely high number nor the infinitely close arrangement can be realized. Instead, there is a limited number of speakers, which are also arranged at certain predetermined distances from each other. This means that in real systems only an approximation to the actual waveform is achieved, which would take place if the virtual source were actually available, i.e. would be a real source.
- the loudspeaker array can only be viewed when viewing a cinema, e.g. B. is arranged on the side of the cinema screen.
- the wave field synthesis module would generate speaker signals for these speakers, the speaker signals for these speakers normally being the same as for corresponding speakers in a speaker array that extends, not just across the side of a cinema, for example, to the screen is arranged, but is also arranged on the left, right and behind the listening room.
- This "360 °" speaker array will of course create a better approximation to an exact wave field than just a one-sided array, for example in front of the audience.
- a wave field synthesis module typically receives no feedback as to how many loudspeakers are present or whether it is a single-sided or multi-sided or even a 360 ° array or not.
- a wave field synthesis device calculates a loudspeaker signal for one which other speakers still exist ⁇ According speaker based on the position of the speaker and independent of it or do not exist.
- Wave field synthesis devices are also able to emulate several different types of sources.
- a prominent source form is the point source, where the level decreases proportionally 1 / r, where r is the distance between a listener and the position of the virtual source.
- Another source form is a source that emits plane waves. Here the level remains constant regardless of the distance to the listener, since plane waves can be generated by point sources that are arranged at an infinite distance.
- the level change in two-dimensional loudspeaker arrangements corresponds to the natural level change except for a negligible error.
- the absolute level can result from the use of a finite number of loudspeakers instead of the theoretically required infinite number of loudspeakers, as has been explained above.
- the so-called subwoofer principle is used in such existing five-channel systems or seven-channel systems.
- the subwoofer principle is used in multichannel playback systems to save expensive and large woofers.
- a low-frequency channel is used that only contains music signals with frequencies lower than a cut-off frequency of around 120 Hz.
- This low-frequency channel controls a low-frequency loudspeaker with a large diaphragm area, with which high sound pressures are achieved, especially at low frequencies.
- the subwoofer principle takes advantage of the fact that it is very difficult for the human ear to localize low-frequency sounds in the direction.
- an extra bass channel for a special loudspeaker arrangement (spatial arrangement) is already mixed during the sound mixing. Examples of such ultimate channel playback systems are Dolby Digital, Sony SDDS and DTS.
- the subwoofer channel can be mixed regardless of the size of the room to be sounded, since the spatial conditions only change in the scale sense.
- the loudspeaker arrangement remains the same in scale.
- Wave Field Synthesis a large audience area can be covered with sound. Sound events can be digestge ⁇ in their spatial depth. For this purpose, the complete sound field of the individual sound events is reproduced in the audience area. This is done by a large number of speakers. Around 500 or more speaker systems are required for large installations. If you wanted to equip every single speaker system with a powerful woofer, very high costs would arise.
- the number of speaker channels is related to the size of the audience area.
- the number of loudspeaker channels is determined by how densely the loudspeakers are distributed over the circumference of the area to be irradiated. The goodness of the WFS playback system depends on this density.
- the volume depends on the number of loudspeaker channels and the density of the loudspeakers, since all loudspeaker channels add up to form a wave field. The volume of a WFS system is therefore not predetermined.
- the volume of the subwoofer channel is, however, predetermined with the known parameters of the electrical amplifier and the loudspeaker.
- the object of the present invention is to provide a concept for generating a low-frequency channel in a multi-channel reproduction system, which enables a reduction of level artifacts.
- the present invention is based on the knowledge that the low-frequency channel for a low-frequency loudspeaker or that several low-frequency channels for several low-frequency loudspeakers in a multichannel system is not already generated in a sound mixing process that is independent of an actual one Playback space takes place, but that reference is made to the actual playback space by taking into account the predetermined position of the woofer on the one hand and properties of audio objects, which typically represent virtual sources, on the other hand, in order to generate the woofer channel.
- audio objects are assumed, an object description on the one hand and an object signal on the other hand being assigned to an audio object.
- an audio object scaling value is calculated for each audio object signal, which is then used to scale each object signal and then to sum up the scaled object signals to obtain a sum signal.
- the low-frequency channel which is fed to the low-frequency loudspeaker, is then derived from the sum signal.
- a scaling of the audio object signal which originates from a virtual source, which is arranged at a virtual position, is carried out in such a way that an actual volume or an actual amplitude state, based on this virtual source, corresponds to a desired amplitude state at the reference reproduction position ,
- the target amplitude condition depends on the volume of the audio object signal associated with the virtual source and the distance between the virtual position and the reference playback position.
- This calculation of audio object scaling values is carried out for all virtual sources in order to then scale the audio object signals of each virtual source with the corresponding scaling value.
- the scaled audio object signals are then summed up to obtain a sum signal.
- the low-frequency channel is then derived from this sum signal in the case where there is only a single low-frequency loudspeaker. This can be done by simple low-pass filtering.
- the low-pass filtering can already be carried out with the still unscaled audio object signals, so that only low-pass signals are already being processed further, so that the sum signal is already the low-frequency channel itself.
- the extraction of the low-frequency channel only after summing up the scaled object signals in order to obtain the best possible approximation of the volume of the low-frequency signals in the performance room on the one hand and the volume of the mid-range and high-frequency signals in the performance room on the other hand.
- a subwoofer channel is therefore not already mixed during the sound mixing process from the virtual sources, that is to say the sound material for the wave field synthesis. Instead, the mixing takes place automatically during playback in the wave field synthesis system regardless of the size of the system and the number of speakers.
- the volume of the subwoofer signal depends on the number and the extent of the fringed area of the wave field synthesis system. Even prescribed loudspeaker arrangements no longer have to be observed, since the loudspeaker position and number of loudspeakers are included in the generation of the low-frequency channel.
- the present invention is not only limited to wave field synthesis systems, but can generally be applied to any multichannel reproduction systems in which the mixing and generation, that is to say the rendering, of the reproduction channels, that is to say the loudspeaker channels themselves, only occurs in the case of the actual playback takes place.
- Systems of this type are, for example, 5.1 systems, 7.1 systems, etc.
- the low-frequency channel generation according to the invention is preferably combined with a level artifact reduction in order to carry out level corrections in a wave field synthesis system not only for low-frequency channels, but for all loudspeaker channels, in order to be independent of the number and position of the loudspeakers used with respect to the wave field synthesis algorithm used.
- the low-frequency loudspeaker will not be arranged in a reference reproduction position for which an optimal level correction is carried out.
- the sum signal is scaled according to the invention, taking into account the position of the woofer using a loudspeaker scaling value to be calculated.
- This scaling will preferably only be an amplitude scaling and not a phase scaling, taking into account the fact that the ear has no good localization at the low frequencies present in the low-frequency channel, but only shows an exact amplitude / volume perception.
- a phase scaling can be used as scaling, if such is desired in an application scenario.
- a separate woofer is created for each woofer.
- the bass channels of the individual bass speakers preferably differ only in their amplitude, but not in the signal itself. All woofers thus send the same sum signal, but with different amplitude scaling, with the amplitude scaling for a single one Woofer depends on the distance of the individual woofer to the reference playback point.
- the overall volume of all superimposed tie tone channels at the reference playback position is equal to the volume of the sum signal or the loudspeaker of the sum signal corresponds at least within a predetermined tolerance range.
- a separate loudspeaker scaling value is calculated for each individual woofer channel, with which the sum signal is then scaled accordingly in order to obtain the individual woofer channel.
- subwoofer channel is particularly advantageous in that it leads to a significant price reduction, since the individual speakers, for. B. a wave field synthesis system can be built much cheaper, since they do not have to have low-frequency properties. In contrast, only one or a few, such as three to four, subwoofer speakers is sufficient to realize the very low frequencies with high sound pressure through a correspondingly large membrane area.
- the present invention is also advantageous in that the one or more low-frequency channels for any speaker positions and multi-channel formats can be generated automatically, this requiring only little additional effort, particularly in the context of a wave field synthesis system, since the wave field synthesis system carries out a level correction anyway.
- the individual volume and preferably also the delay of each virtual source is first calculated in relation to the reference playback position.
- the audio signal of each virtual source is then scaled and delayed in order to sum up all virtual sources.
- the total volume and delay of the subwoofer are then calculated depending on its distance from the reference point, if the subwoofer is not already located in the reference point.
- the individual volumes of all subwoofers it is preferred to first determine the individual volumes of all subwoofers depending on their distances from the reference point.
- the sum of all subwoofer channels is equal to that of the reference volume at the reference playback position, which preferably corresponds to the center point of the wave field synthesis system.
- Corresponding scaling factors per subwoofer are thus calculated, whereby however first individual volume and delay of each virtual source are determined in relation to the reference point. Then each virtual source is scaled again accordingly and optionally delayed, in order then to sum up all virtual sources to the sum signal, which is then scaled with the individual scaling factors for each subwoofer channel in order to obtain the individual bass channels for the various bass speakers.
- FIG. 1 shows a block diagram of the device according to the invention for level correction in a wave field synthesis system
- FIG. 2 shows a basic circuit diagram of a wave field synthesis environment as can be used for the present invention
- FIG. 3 shows a more detailed illustration of the wave field synthesis module shown in FIG. 2;
- FIG. 4 shows a block diagram of a device according to the invention for determining the correction value according to an exemplary embodiment with a look-up table and, if appropriate, interpolation device;
- FIG. 5 shows a further exemplary embodiment of the device for determining FIG. 1 with target value / actual value determination and subsequent comparison
- 6a is a block diagram of a wave field synthesis module with an embedded manipulation device for manipulating the component signals
- 6b shows a block diagram of a further exemplary embodiment of the present invention with an upstream manipulation device
- FIG. 7a shows a sketch for explaining the desired amplitude state at an optimal point in a demonstration area
- 7b shows a sketch to explain the actual amplitude state at an optimal point in the demonstration area
- 8 shows a basic block diagram of a wave field synthesis system with a wave field synthesis module and loudspeaker array in a demonstration area
- FIG. 9 shows a block diagram of a device according to the invention for generating a low-frequency channel
- FIG. 10 shows a preferred embodiment of the device for providing the bass channel for a plurality of bass speakers
- FIG. 11 shows a schematic representation of a demonstration area with a plurality of individual speakers and two subwoofers.
- both the volume and the delay are calculated by the wave field synthesis algorithm for each speaker channel and each virtual source.
- the position of the individual loudspeaker must be known.
- This scaling of the individual audio object signals for the individual wave field synthesis system loudspeakers is based on the knowledge that the inadequacies of a wave field synthesis system with a (practically realizable) finite number of loudspeakers can at least be alleviated if a level Correction is performed by manipulating either the audio signal associated with a virtual source prior to wave field synthesis or the component signals for various loudspeakers originating in a virtual source after wave field synthesis using a correction value to detect a deviation between a target amplitude condition in a demonstration area and an actual amplitude state in the demonstration area to reduce.
- the target amplitude state results from the fact that depending on the position of the virtual source, and z.
- a target level is determined as an example of a target amplitude state, and that an actual level as an example of an actual amplitude state of the listener is determined. While the target amplitude state is determined independently of the actual grouping and type of the individual speakers only on the basis of the virtual source or their position, the actual situation is calculated taking into account the positioning, type and control of the individual speakers of the speaker array.
- the sound level at the ear of the listener can be determined at the optimum point within the demonstration area on the basis of a component signal from the virtual source, which is emitted via a single loudspeaker.
- the level at the ear of the listener at the optimal point within the demonstration area can also be determined for the other component signals that go back to the virtual source and are emitted via other loudspeakers, in order to then summarize these levels to the actual actual level at the ear of the To receive the handset.
- the transfer function of each individual loudspeaker as well as the level of the signal at the loudspeaker and the distance of the listener at the point under consideration within the demonstration area from the individual loudspeaker can be taken into account.
- the transmission characteristics of the speaker can be assumed to work as an ideal point source.
- the directional characteristic of the individual speaker can also be taken into account.
- a major advantage of this concept is that in an embodiment in which sound levels are considered, only multiplicative scaling occurs, in that for a quotient between the target level and the actual level, which gives the correction value, not the absolute level at the listener or the absolute level of the virtual source is required. Instead, the correction factor depends only on the position of the virtual source (and thus on the positions of the individual speakers) and the optimal point within the demonstration area. However, these values are fixed with regard to the position of the optimal point and the positions and transmission characteristics of the individual loudspeakers and do not depend on a piece being played.
- the concept can be implemented in a computationally efficient manner as a look-up table, in that a look-up table is generated and used, which includes position correction factor-value pairs, for all or a substantial part of possible virtual positions.
- a look-up table is generated and used, which includes position correction factor-value pairs, for all or a substantial part of possible virtual positions.
- no on-line setpoint determination, actual value determination and setpoint / actual value comparison algorithm is to be carried out.
- These algorithms which can be time-consuming in some cases, can be dispensed with if the look-up table is accessed on the basis of a position of a virtual source in order to determine from there the correction factor valid for this position of the virtual source.
- a virtual source with a certain calibration level would be placed in a certain virtual position.
- a wave field synthesis module would calculate the loudspeaker signals for the individual loudspeakers in order to finally measure the level actually arriving at the listener due to the virtual source.
- a correction factor would then be determined such that it at least reduces the deviation from the target level to the actual level or preferably brings it to 0.
- This correction factor would then be stored in the look-up table in association with the position of the virtual source in order to gradually generate the entire look-up table for a specific wave field synthesis system in a special demonstration room, that is to say for many positions of the virtual source.
- manipulation based on the correction factor there are several options for manipulation based on the correction factor.
- the correction factor is not necessarily identical for all component signals got to. However, this is largely preferred in order not to impair the relative scaling of the component signals to one another, which is necessary for the reconstruction of the actual wave situation.
- One advantage is that with relatively simple measures, at least during operation, a level correction can be carried out in such a way that the listener, at least in view of the volume of a virtual source that he perceives, does not notice that the infinitely many loudspeakers that are actually required are not available , but only a limited number of speakers.
- Another advantage is that even if a virtual source moves at a constant distance from the viewer (e.g. from left to right), this source for the viewer, who is sitting in the middle in front of the screen, for example, is always the same loud and is not even louder and once quieter, which would be the case without correction.
- Another advantage is that it offers the option of offering less expensive wave field synthesis systems with a smaller number of loudspeakers, which, however, do not involve any level artifacts, especially for moving sources, ie just as good for a listener with regard to the level problem act like more complex wave field synthesis systems with a large number of speakers. Levels which are too low can also be corrected according to the invention for holes in the array.
- FIG. 9 is either standing alone, ie without level correction
- Individual speakers can be used, or that can preferably be combined with the concept of level artifact correction described later with reference to FIGS. 1-8, in order to also use the correction values used for level artifact correction of the individual speakers as audio object scaling values, which are used in low-frequency channel generation must be used.
- FIG. 9 shows a device for generating a low-frequency channel for a low-frequency loudspeaker, which is arranged at a predetermined loudspeaker position.
- the device shown in FIG. 9 initially comprises a device 900 for providing a plurality of audio objects, an audio object signal 902 and an audio object description 904 being assigned to an audio object.
- the audio object description will typically include an audio object position and possibly the audio object type.
- the audio object description can also directly include an indication of the audio object volume. If this is not the case, then the audio object volume can be easily calculated from the audio object signal itself, for example by squaring and summing over a certain period of time. If the transmission function, frequency response, etc.
- the object description of the audio signal is supplied to means 906 for calculating an audio object scaling value for each audio object.
- the individual audio object scaling values 908 are then, as shown with reference to FIG. 9, fed to a device 910 for scaling the object signals.
- the means 906 for computing the audio object scaling value is designed to calculate an audio object scaling value for each audio object depending on the object description. If it is a source that emits plane waves, the audio object scaling value or the correction factor will be 1, since for such plane-wave audio objects a spacing between the position of this object and the optimal reference playback position is irrelevant, because in this case the virtual position is assumed to be infinite.
- the audio object scaling value becomes dependent on the object volume, which is either in the object description or can be derived from the object signal, and that Distance between the virtual position of the audio object and the reference playback position is calculated.
- the audio object scaling value or correction value such that it is taken into account that it is based on a target amplitude state in the demonstration area, the target amplitude state depending on a position of the virtual source or a type of the virtual source, where the correction value is also based on an actual amplitude state in the demonstration area, which is based on the component signals for the individual speakers on the basis of the virtual source under consideration.
- the correction value is thus calculated in such a way that a manipulation of the audio signal assigned to the virtual source using the correction value reduces a deviation between the desired amplitude state and the actual amplitude state.
- a delay which may be caused by different virtual positions before the summation by means 914, so that the individual audio object signals, which are present as sequences of samples, are shifted with respect to a time reference. in order to take sufficient account of the differences in transit time of the sound signal from the virtual position to the reference reproduction position.
- the scaled and correspondingly delayed object signals are then summed up by the device 914 in samples, in order to obtain a sum signal with a sequence of sum signal samples, which is designated by 916 in FIG. 9.
- This sum signal 916 is fed to a device 918 for providing the low-frequency channel for the one or more subwoofers, which supplies the subwoofer signal or the low-frequency channel 920 on the output side.
- the sound signal emitted by a woofer is not a sound signal with full bandwidth, but with an upper bandwidth limit.
- the cut-off frequency of the sound signal emitted by a woofer is less than 250 Hz and preferably even only 125 Hz.
- the band limitation of this sound signal can take place at different points.
- a simple measure is to supply the woofer with a full bandwidth excitation signal, which is then band-limited by the woofer itself, since it only converts low frequencies into sound signals, but suppresses high frequencies.
- the band limitation can also be carried out in the device 918 for providing the low-frequency channel. gene by low-pass filtering the signal there before a digital / analog conversion, this low-pass filtering being preferred, since it can be carried out on the digital side, so that clear conditions exist regardless of the actual implementation of the subwoofer.
- the low-pass filtering can take place before the device 910 for scaling the object signals, so that the operations which are carried out by the devices 910, 914, 918 are now carried out with low-pass signals and not with signals of the entire bandwidth.
- the low-pass filtering in the device 918 it is preferred to carry out the low-pass filtering in the device 918, so that the calculation of the audio object scaling values, the scaling of the object signals and the summation with signals of full bandwidth are carried out in order for the loudspeakers to match the low tone on the one hand and the mid-tone and Ensure high tones on the other hand.
- 11 schematically shows a wave field synthesis system with a large number of individual speakers 808.
- the individual speakers 808 form an array 800 of individual speakers, which enclose the demonstration area, preferably within the demonstration area the reference playback position or reference point 1100 is located.
- 11 also schematically shows an audio object 1102, which is referred to as a "virtual sound object".
- the virtual sound object 1102 comprises an object description that represents a virtual position 1104. Using the coordinates of the reference point 1100 and the coordinates of the virtual position 1104, the if necessary, can be converted accordingly, the distance D of the virtual sound object 1102 from the reference playback position 100.
- 11 also shows a first woofer 1106 at a first predetermined loudspeaker position 1108 and a second woofer 1110 at a second woofer position 1112.
- the second subwoofer 1110 or any further one in FIG 11 not shown optional additional subwoofers.
- the first subwoofer 1106 is at a distance d1 from the reference point 1100
- the second subwoofer 1110 is at a distance d2 from the reference point.
- a subwoofer n (not shown in FIG. 11) is at a distance dn from the reference point 1100.
- the device 918 for providing the low-frequency channel is designed to, in addition to the sum signal 916, which is denoted by s in FIG. 10, also the distance d1 of the woofer 1, which is denoted by 930, based on the distance d2 of the woofer 2, designated 932, and the distance dn of the woofer n, designated 934.
- the device 918 supplies a first bass channel 940, a second bass channel 942 and an nth bass channel 944. It can be seen from FIG.
- the respective weighting factors are denoted by ai, a 2 , ..., a n .
- the individual weighting factors ai, a 2 , a n depend on the one hand on the distances 930-934 and on the other hand on the general boundary condition that the volume of the bass channels at reference point 100 corresponds to the reference volume, that is to say the desired amplitude state for the bass channel at reference playback position 1100 (FIG. 11).
- the sum of the loudspeaker scaling values ai, a 2 , a n will be greater than 1 in order to take into account the attenuation of the bass channels on the way from the corresponding subwoofer to the reference point. If only a single woofer (e.g. 1106) is provided, the scaling factor ai will also be greater than 1, while no further scaling factors need to be calculated, since there is only a single woofer.
- a level artifact correction device for the loudspeaker array 800 in FIG. 8 and FIG. 11 is shown, which are preferably combined with the low-frequency channel calculation according to the invention, as has been illustrated with reference to FIGS. 9-11 can.
- the wave field synthesis system has a speaker array 800 placed with respect to a demonstration area 802.
- the speaker array shown in Fig. 8 which is a 360 ° array, includes four array sides 800a, 800b, 800c and 800d.
- the demonstration area 802 e.g. B. a cinema hall, it is assumed with regard to the conventions front / rear or right / left that the cinema screen is on the same side of the screening area 802 on which the sub-array 800c is also arranged.
- the viewer who is sitting at the so-called optimal point P in the demonstration area 802, would see the front, that is, the screen.
- Sub-array 800a would then be behind the viewer, while to the left of Viewer would have sub-array 800d and sub-array 800b to the right of the viewer.
- Each loudspeaker array consists of a number of different individual loudspeakers 808, each of which is controlled with its own loudspeaker signals, which are provided by a wave field synthesis module 810 via a data bus 812, which is shown only schematically in FIG. 8.
- the wave field synthesis module is designed to use the information about e.g. B.
- loudspeaker information (LS information)
- loudspeaker signals for the individual loudspeakers 808, each of which is generated by the audio tracks for virtual sources to which positive ons information are assigned are derived according to the known wave field synthesis algorithms.
- the wave field synthesis module can also receive further inputs, such as information about the room acoustics of the demonstration area, etc.
- the following explanations regarding the present invention can in principle be carried out for each point P in the demonstration area.
- the optimum point can thus be anywhere in the demonstration area 802.
- it is preferred to place the optimal point or the optimal line in the middle or at the center of gravity of the wave field synthesis system, which is generated by the loudspeaker sub-arrays 800a, 800b, 800c , 800d is defined to assume.
- FIG. 2 shows a wave field synthesis environment in which the present invention can be implemented.
- the center of a wave field synthesis environment is a wave field synthesis module 200, which comprises various inputs 202, 204, 206 and 208 and various outputs 210, 212, 214, 216.
- Various audio signals for virtual sources are fed to the wave field synthesis module via inputs 202 to 204. So the input 202 receives z.
- the audio signal 1 would be e.g. B. the language of an actor who moves from a left side of the screen to a right side of the screen and possibly additionally away from the viewer or towards the viewer.
- the audio signal 1 would then be the actual language of this actor, while the position information as a function of time represents the current position of the first actor in the recording setting at a certain point in time.
- the audio signal n would be the language of, for example, another actor who moves the same or different than the first actor.
- the current position of the other actor to whom the audio signal n is assigned is communicated to the wave field synthesis module 200 by position information synchronized with the audio signal n.
- different virtual sources exist depending on the recording setting, the audio signal of each virtual source being supplied to the wave field synthesis module 200 as a separate audio track.
- a wave field synthesis module feeds a plurality of loudspeakers LSI, LS2, LS3, LSm by outputting loudspeaker signals via the outputs 210 to 216 to the individual loudspeakers.
- the positions of the individual loudspeakers in a playback setting, such as a cinema, are communicated to the wave field synthesis module 200 via the input 206.
- the wave field synthesis module 200 In the cinema hall there are many individual loudspeakers grouped around the cinema audience, preferably in arrays are arranged such that there are loudspeakers both in front of the viewer, for example behind the screen, and behind the viewer and to the right and left of the viewer.
- other inputs can be communicated to the wave field synthesis module 200, such as information about the room acoustics, etc., in order to be able to simulate the actual room acoustics prevailing during the recording set-up in a cinema hall.
- the loudspeaker signal which is supplied to the loudspeaker LSI via the output 210 will be a superimposition of component signals of the virtual sources, in that the loudspeaker signal for the loudspeaker LSI is a first component which originates from the virtual source 1, a second Component, which goes back to the virtual source 2, as well as an nth component, which goes back to the virtual source n.
- the individual component signals are linearly superimposed, i.e. added after their calculation, in order to simulate the linear superposition at the ear of the listener, who will hear a linear superposition of the sound sources perceivable in a real setting.
- the wave field synthesis module 200 has a strongly parallel structure in that, starting from the audio signal for each virtual source and starting from the position information for the corresponding virtual source, delay information V x and scaling factors SFj are first of all . calculated from the position information and the position of the speaker under consideration, e.g. B. depend on the loudspeaker with the order number j, ie LSj.
- the calculation of a delay information V x and a scaling factor SFj happens based on the position information of a virtual source and the position of the speaker j in question by known algorithms implemented in devices 300, 302, 304, 306.
- the individual component signals are then summed by a summer 320 in order to determine the discrete value for the current time t a of the loudspeaker signal for loudspeaker j which can then be supplied to the loudspeaker for the output (for example the output 214 if the loudspeaker j is the loudspeaker LS3).
- a value that is valid due to a delay and scaling with a scaling factor at a current point in time is first calculated individually for each virtual source, after which all component signals for a loudspeaker are summed due to the different virtual sources. If, for example, there were only one virtual source, the summer would be omitted and the signal present at the output of the summer in FIG. B. correspond to the signal output by the device 310 when the virtual source 1 is the only virtual source.
- a loudspeaker signal is obtained at the output 322 of FIG. 3, which is a superimposition of the component signals for this loudspeaker due to the different virtual sources 1, 2, 3, ..., n.
- An arrangement as shown in FIG. 3 would in principle be provided for each loudspeaker 808 in the wave field synthesis module 810, unless that what is preferred for practical reasons always z. B. 2, 4 or 8 lying speakers can be controlled with the same speaker signal.
- FIG. 1 shows a block diagram of the device according to the invention for level correction in a wave field synthesis system, which has been explained with reference to FIG. 8.
- the wave field synthesis system comprises the wave field synthesis module 810 and the loudspeaker array 800 for supplying sound to the presentation area 802, the wave field synthesis module 810 being designed to receive an audio signal associated with a virtual sound source and source position information associated with the virtual sound source and component signals for them taking into account speaker position information Calculate speakers based on the virtual source.
- the device according to the invention first comprises a device 100 for determining a correction value which is based on a desired amplitude state in the demonstration area, the desired amplitude state depending on a position of the virtual source or a type of the virtual source, and the correction value also on Actual amplitude state is based in the demonstration area, which depends on the component signals for the loudspeakers due to the virtual source.
- the device 100 has an input 102 for obtaining a position of the virtual source when e.g. B. has a point source characteristic, or to obtain information about a type of source when the source z. B. is a source for generating plane waves.
- the distance of the listener from the source is not necessary to determine the actual state, since the source is already located infinitely far from the listener in the model due to the generated plane waves and has a position-independent level.
- the device 100 is designed to output a correction value 104 on the output side, which is a device 106 for manipulation an audio signal associated with the virtual source (obtained via an input 108) or for manipulating component signals for the speakers due to a virtual source (obtained via an input 110).
- an output 112 results in a manipulated audio signal which, according to the invention, then in the wave field synthesis module 200 instead of the original audio signal which is provided at the input 108 is fed in to generate the individual loudspeaker signals 210, 212, ..., 216.
- the other alternative to manipulation has been used, namely the manipulation of the component signals received via input 110 to some extent, component signals manipulated on the output side are obtained which still have to be summed up loudspeaker-wise (device 116), with or without manipulated component signals from other virtual sources, which are provided via further inputs 118.
- the device 116 again delivers the loudspeaker signals 210, 212, ..., 216.
- the alternatives of the upstream manipulation (output 112) or the embedded manipulation (output 114) shown in FIG. 1 are used alternatively to one another can.
- the weighting factor or correction value which is provided via the input 104 in the device 106, is split to a certain extent, so that partly an upstream manipulation and partly an embedded manipulation is carried out.
- the upstream manipulation would thus consist in manipulating the audio signal of the virtual source, which is fed into a device 310, 312, 314 or 316, before it is fed in.
- the embedded manipulation would consist in manipulating the component signals output by devices 310, 312, 314 and 316, respectively, prior to their summation in order to obtain the actual loudspeaker signal.
- FIGS. 6a and 6b show the embedded manipulation by the manipulation device 106, which is drawn in FIG. 6a as a multiplier.
- a wave field synthesis device which for example consists of blocks 300, 310 or 302, 312, or 304, 314 and 306 or 316 of FIG. 3, delivers the component signals n, K ⁇ 2 , K 13 for the loudspeaker LSI or Component signals K nl , K n2 and K n3 for the loudspeaker LSn.
- the first index of Ki j indicates the loudspeaker
- the second index indicates the virtual source from which the component signal originates.
- the virtual source 1 is expressed, for example, in the component signal Kn, ..., K nl .
- the component signals belonging to the source 1 are multiplied , ie the component signals, the index j of which indicates the virtual source 1, take place with the correction factor F x .
- the manipulation device is upstream of the wave field synthesis device and is effective to correct the audio signals of the sources with the corresponding correction factors in order to obtain manipulated audio signals for the virtual sources, which are then fed to the wave field synthesis device in order to obtain the component signals, which are then are summed up by the respective component summation devices in order to obtain the loudspeaker signals LS for the corresponding loudspeakers, such as, for example, the loudspeaker LSi.
- the device 100 for determining the correction value is designed as a look-up table 400, which stores position-correction factor-value pairs.
- the device 100 is preferably further provided with an interpolation device 402, on the one hand to keep the table size of the look-up table 400 within a limited framework, and on the other hand also for current positions of a virtual source that are fed into the interpolation device via an input 404, at least below Use one or more neighboring position correction factors stored in the lookup table Value pairs, which are fed to the interpolation device 402 via an input 406, to generate an interpolated current correction factor at an output 408.
- the interpolation device 402 can also be omitted, so that the device 100 for determining FIG. 1 performs direct access to the lookup table using position information supplied at an input 410 and a corresponding correction at an output 412 - door factor delivers. If the current position information that is assigned to the audio track of the virtual source does not exactly correspond to a position information that can be found in the look-up table, then a simple rounding-off / rounding-up function can also be assigned to the look-up table in order to find the closest one in the Table to take the stored base value instead of the current base value.
- the device for determining can be designed to actually carry out a setpoint-actual value comparison.
- the device 100 of FIG. 1 comprises a setpoint amplitude State determination device 500 and an actual amplitude state determination device 502 for supplying a target amplitude state 504 and an actual amplitude state 506, which are fed to a comparison device 508 which, for example, a quotient of the target amplitude state 504 and the actual Amplitude state 506 is calculated to generate a correction factor 510, which is manipulated by the device 106, which is shown in Fig. 1 is fed for further use.
- the correction value can also be stored in a look-up table.
- the target amplitude state calculation is designed to determine a target level at the optimum point for a virtual source configured at a specific position or in a specific type.
- the target amplitude state determination device 500 does not require any component signals for the target amplitude state calculation, since the target amplitude state is independent of the component signals.
- component signals are fed to the actual amplitude determination device 502, which, depending on the embodiment, can also receive information about the speaker positions and information about speaker transmission functions and / or information about directional characteristics of the speakers by one Determine the current situation as well as possible.
- the actual amplitude state determination device 502 can also be designed as an actual measuring system in order to determine an actual level situation at the optimal point for certain virtual sources at certain positions.
- Fig. 7a shows a diagram for determining a target amplitude state at a predetermined point, which is designated in Fig. 7a with "optimal point" and which is in the demonstration area 802 of Fig. 8.
- virtual source 700 is shown as a point source that generates a sound field with concentric wavefronts.
- the level L v of virtual source 700 is known on the basis of the audio signal for virtual source 700.
- the desired amplitude state or when the amplitude state is a level state the target level at point P in the demonstration area is easily determined obtained in that the level L P at point P is equal to the quotient of L v and a distance r that point P has from virtual source 700.
- the target amplitude state can thus be easily determined by calculating the level L v of the virtual source and by calculating the distance r from the optimal point to the virtual source.
- a coordinate transformation of the virtual coordinates into the coordinates of the screening room or a coordinate transformation of the screening room coordinates of point P into the virtual coordinates must typically be carried out, which is known to those skilled in the field of wave field synthesis.
- the virtual source is an infinitely distant virtual source that generates plane waves at point P
- the distance between point P and the source is not required to determine the desired amplitude state, since this is in any case almost infinite. In this case, only information about the type of source is required.
- the target level at point P is then equal to the level which is assigned to the plane wave field which is generated by the virtual source which is infinitely distant.
- Fig. 7 shows a diagram for explaining the actual amplitude state.
- different loudspeakers 808 are drawn in FIG. 7b, all of which are fed with their own loudspeaker signal which, for. B. has been generated by the wave field synthesis module 810 of FIG. 8.
- each loudspeaker is modeled as a point source that outputs a concentric wave field.
- the law of the concentric wave field is again that the level drops according to 1 / r.
- the signal generated by the loudspeaker 808 directly on the loudspeaker membrane or the level of this signal can be based on the loudspeaker characteristics and the component signal in the loudspeaker signal LSn, which goes back to the virtual source under consideration.
- the distance between P and the loudspeaker membrane of the loudspeaker LSn can be calculated, so that a level for the point P can be obtained on the basis of a component signal which is directed to the virtual source under consideration goes back and has been emitted by the loudspeaker LSn.
- a corresponding procedure can also be carried out for the other loudspeakers of the loudspeaker array, so that there is a number of "partial level values" for point P, which represent a signal contribution from the virtual source under consideration, which has reached the listener at point P from the individual loudspeakers
- a correction value which is preferably multiplicative, but which, in principle, is additive or subtractive could be get.
- the desired level for a point is thus calculated on the basis of certain source forms, that is to say the desired amplitude state. It is preferred that the optimal point or the point in the demonstration area that is being viewed lies in the middle of the wave field synthesis system. At this point it should be pointed out that an improvement is achieved even if the point on which the calculation of the target amplitude state is based does not correspond directly to the point which was used to determine the actual amplitude state is.
- a target amplitude state for any point in the demonstration area and that an actual amplitude condition is also determined for any point in the demonstration area it is in principle sufficient that a target amplitude state for any point in the demonstration area and that an actual amplitude condition is also determined for any point in the demonstration area, but it is preferred that the point to which the actual amplitude condition is related is in a zone around the point for which the Desired amplitude condition has been determined, this zone is preferably less than 2 meters for normal cinema applications. For best results, these points should essentially coincide.
- the level practically generated by superimposition at this point which is called the optimum point in the demonstration area.
- the levels of the individual speakers and / or sources are then corrected with this factor according to the invention.
- FIG. 6b in which the device 914 for summing is drawn in in order to supply the sum signal 916 on the output side, while the scaled object signals 912 are obtained on the input side, which, as can be seen from FIG. 6b, by Scaling the source signals of sources 1, 2, 3 with the corresponding audio object scaling values or correction values F1, F2, F3 can be obtained.
- the version shown in FIG. 6b is preferred, in which scaling or manipulation or correction is already carried out at the audio object signal level and not at the component level, as shown in FIG. 6a. is carried out.
- the concept of correction at component level shown in FIG The inventive concept of low-frequency channel generation can be combined in that at least the calculation of the audio object scaling values F1, F2, ..., Fn only has to be carried out once.
- the subwoofer channel is thus scaled in a manner similar to the scaling of the overall volume of all loudspeakers in the reference point of the wave field synthesis reproduction system.
- the method according to the invention is therefore suitable for any number of subwoofer loudspeakers, all of which are scaled in such a way that they reach a reference volume at the center of the wave field synthesis system.
- the reference volume only depends on the position of the virtual sound source. With the known dependencies on the distance of the sound object from the reference point and the associated attenuation of the volume, the individual volume of the respective sound object for each subwoofer channel is preferably calculated. The delay of each source is calculated from the distance between the virtual source and the reference point of the volume scaling.
- Each subwoofer speaker reproduces the sum of all sound objects converted in this way. How the individual volumes of the subwoofer speakers add up depends on their position. The preferred positioning of Subwooferlautspre- brass, and the selection of the number are far from necessary subwoofers in the already mentioned specialist publications Welti, Todd, “How Many Subwoofers are Enough", 112 th AES Conv. Paper 5602, May 2002, Kunststoff, Germany, Martens, "The impact of decorrelated low-frequency reproduction on auditory spatial imagery: Are two subwoofers better than one?", 16 th AES Conf. Paper, April 1999, Rovaniemi, Finland.
- the method according to the invention for generating a low-frequency channel can be implemented in hardware or in software.
- the method according to the invention for level correction as shown in FIG. 1, can be implemented in hardware or in software.
- the implementation can take place on a digital storage medium, in particular a floppy disk or CD with electronically readable control signals, which can interact with a programmable computer system in such a way that the method is carried out.
- the invention thus also consists in a computer program product with a program code stored on a machine-readable carrier for carrying out the method for level correction when the computer program product runs on a computer.
- the invention can thus be implemented as a computer program with a program code for carrying out the method if the computer program runs on a computer.
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JP2006540333A JP4255031B2 (ja) | 2003-11-26 | 2004-11-18 | 低周波チャネルを生成する装置および方法 |
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- 2004-11-18 WO PCT/EP2004/013130 patent/WO2005060307A1/de active IP Right Grant
- 2004-11-18 JP JP2006540333A patent/JP4255031B2/ja active Active
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2006
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Cited By (6)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
JP2010506521A (ja) * | 2006-10-11 | 2010-02-25 | フラウンホーファー−ゲゼルシャフト・ツール・フェルデルング・デル・アンゲヴァンテン・フォルシュング・アインゲトラーゲネル・フェライン | 再生空間を画定するラウドスピーカアレイのための複数のラウドスピーカ信号の生成装置及びその方法 |
US8358091B2 (en) | 2006-10-11 | 2013-01-22 | Fraunhofer-Gesellschaft Zur Foerderung Der Angewandten Forschung E.V. | Apparatus and method for generating a number of loudspeaker signals for a loudspeaker array which defines a reproduction space |
JP2008219228A (ja) * | 2007-03-01 | 2008-09-18 | Yamaha Corp | 音響再生装置 |
JP2016524883A (ja) * | 2013-06-18 | 2016-08-18 | ドルビー ラボラトリーズ ライセンシング コーポレイション | オーディオ・レンダリングのためのベース管理 |
CN111133411A (zh) * | 2017-09-29 | 2020-05-08 | 苹果公司 | 空间音频上混 |
CN111133411B (zh) * | 2017-09-29 | 2023-07-14 | 苹果公司 | 空间音频上混 |
Also Published As
Publication number | Publication date |
---|---|
EP1671516A1 (de) | 2006-06-21 |
US20060280311A1 (en) | 2006-12-14 |
JP2007512740A (ja) | 2007-05-17 |
DE502004002926D1 (de) | 2007-03-29 |
JP4255031B2 (ja) | 2009-04-15 |
DE10355146A1 (de) | 2005-07-07 |
EP1671516B1 (de) | 2007-02-14 |
CN100588286C (zh) | 2010-02-03 |
CN1906971A (zh) | 2007-01-31 |
US8699731B2 (en) | 2014-04-15 |
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