WO2004047485A1 - Systeme de restitution audio et procede de restitution d'un signal audio - Google Patents

Systeme de restitution audio et procede de restitution d'un signal audio Download PDF

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Publication number
WO2004047485A1
WO2004047485A1 PCT/EP2003/013110 EP0313110W WO2004047485A1 WO 2004047485 A1 WO2004047485 A1 WO 2004047485A1 EP 0313110 W EP0313110 W EP 0313110W WO 2004047485 A1 WO2004047485 A1 WO 2004047485A1
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WIPO (PCT)
Prior art keywords
audio
loudspeaker
playback
module
wave field
Prior art date
Application number
PCT/EP2003/013110
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German (de)
English (en)
Inventor
Frank Melchior
Thomas Röder
Michael Beckinger
Sandra Brix
Thomas Sporer
Haymo Kutschbach
Berthold Schlenker
Carsten Land
Original Assignee
Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
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Application filed by Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. filed Critical Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
Priority to JP2004552700A priority Critical patent/JP4620468B2/ja
Priority to EP03782222A priority patent/EP1576847B1/fr
Priority to DE50303069T priority patent/DE50303069D1/de
Publication of WO2004047485A1 publication Critical patent/WO2004047485A1/fr
Priority to US11/099,156 priority patent/US7706544B2/en

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Classifications

    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R5/00—Stereophonic arrangements
    • H04R5/02—Spatial or constructional arrangements of loudspeakers
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R3/00—Circuits for transducers, loudspeakers or microphones
    • H04R3/12—Circuits for transducers, loudspeakers or microphones for distributing signals to two or more loudspeakers
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R1/00—Details of transducers, loudspeakers or microphones
    • H04R1/20—Arrangements for obtaining desired frequency or directional characteristics
    • H04R1/32—Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only
    • H04R1/40—Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only by combining a number of identical transducers
    • H04R1/403—Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only by combining a number of identical transducers loud-speakers
    • 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

Definitions

  • the present invention relates to audio playback systems and, more particularly, to practical audio playback systems for variable size playback rooms such as cinemas, the audio playback systems being based on wave field synthesis.
  • 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. In a few years, the first wave field synthesis applications for the consumer sector will also be launched.
  • Every point that is captured by a wave is the starting point of 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 must be fed with a time delay and amplitude scaling in such a way that the radiated sound fields of the individual loudspeakers overlap correctly. If there are several sound sources, the contribution to each source becomes Speakers calculated separately and the resulting signals added. If the sources to be reproduced are in a room with reflecting walls, then reflections must also be reproduced as additional sources via the loudspeaker array. The effort involved in the calculation therefore depends heavily on the number of sound sources, the reflection properties of the recording room and the number of speakers.
  • the advantage of this technique lies in the fact 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.
  • 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.
  • An environmental condition can be described by the impulse response of the environment.
  • Wave field synthesis thus enables correct mapping of virtual sound sources over a large reproduction range. At the same time, it offers the sound engineer and sound engineer new technical and creative potential when creating complex soundscapes.
  • Wave field synthesis (WFS or sound field synthesis), as developed at the TU Delft in the late 1980s, represents a holographic approach to sound reproduction.
  • the Kirchhoff-Helmholtz integral serves as the basis for this. This says that any sound fields can be generated within a closed volume by means of a distribution of monopole and dipole sound sources (loudspeaker arrays) on the surface of this volume. Details can be found in MM Boone, ENG Verheijen, PF v.
  • a synthesis signal for each loudspeaker of the loudspeaker array is calculated from an audio signal which emits a virtual source at a virtual position, the synthesis signals being designed in terms of amplitude and phase in such a way that a wave resulting from the superimposition of the individual the loudspeaker output in the loudspeaker array results in the sound wave that corresponds to the wave that would come from the virtual source at the virtual position if this virtual source at the virtual position were a real source with a real position.
  • synthesis signals there are multiple virtual sources in different virtual locations.
  • the calculation of the synthesis signals is carried out for each virtual source at each virtual position, so that typically one virtual source results in synthesis signals for several loudspeakers. Seen from a loudspeaker, this loudspeaker thus receives several synthesis signals that go back to different virtual sources. An overlay of these sources, which is possible due to the linear superposition principle, results then the playback signal actually sent from the speaker.
  • the fully rendered and analog-to-digital converted playback signals for the individual loudspeakers could, for example, be transmitted from the wave field synthesis via two-wire lines.
  • Central unit can be transmitted to the individual speakers. This would have the advantage that it is almost guaranteed that all loudspeakers work synchronously, so that no further measures would be necessary for synchronization purposes.
  • the wave field synthesis central unit could only ever be manufactured for a special reproduction room or for reproduction with a fixed number of loudspeakers.
  • a separate wave field synthesis central unit can be built for each playback room, which is not acceptable due to the individual production in terms of price.
  • a maximally equipped wave field synthesis central unit could be set up, which is controllable with regard to the connectable loudspeakers, i.e. with regard to the number of analog signal outputs, but which includes internal computing processors which is designed for the maximum number of analog outputs, i.e. connectable loudspeakers.
  • Such a system would result in audio reproducing systems for smaller display rooms also having almost the same price as audio reproducing systems for very large display rooms, which should not be acceptable for operators of small display rooms.
  • the medium to small display rooms are of interest to providers of audio display systems, with the "smallest" display rooms, which represent, for example, domestic living rooms or smaller restaurants, also being mentioned here.
  • the object of the present invention is to provide an audio reproduction concept which has a higher market acceptance.
  • the present invention is based on the knowledge that audio playback systems which are to achieve market acceptance must be scalable.
  • scalability must not only take place in terms of the computing power provided, but must also be reflected in the price of the audio playback system impact.
  • this means that an audio playback system for a large playback space may cost more than an audio playback system for a small playback space.
  • an audio playback system for a small playback room must cost significantly less than an audio playback system for a large playback room.
  • the price differences were insignificant, since the price differences were only due to the number of individual loudspeakers, which, however, was offered at low cost due to the fact that a large number of loudspeakers were provided and due to novel integration concepts into the structure that included the reproduction space can be.
  • the audio playback system is divided into a central wave field synthesis module and into many individual loudspeaker modules that are decentrally connected to the central wave field synthesis module.
  • the central wave field synthesis module receives an audio signal with a plurality of audio tracks and on the one hand calculates the synthesis signals and on the other hand the channel information for the channels from the virtual positions to the real speaker positions.
  • the central wave field synthesis module is further configured to supply each speaker with one or more synthesis signals that are to be reproduced by the speaker in question, and channel information for the audio channels from the virtual positions of the virtual sources, from which the one or more synthesis signals originate, to the affected one Deliver speakers.
  • a considerable data rate transmission limitation can already be achieved here, since experience shows that it is very rare for every loudspeaker to receive synthesis signals whose Energy content is greater than a certain threshold.
  • the central wave field synthesis module according to the invention thus already has the option of supplying only the synthesis signals to a decentralized loudspeaker module and also only the channel information for the synthesis signals which are important for the individual loudspeakers.
  • the loudspeaker modules according to the invention are decentralized and directly coupled to the loudspeaker or preferably arranged in close proximity to the loudspeaker.
  • Each loudspeaker module comprises a receiver for receiving the one or 'more synthesis signals for the respective loudspeaker, as well as the synthesis signals associated with channel information.
  • each loudspeaker module comprises a rendering device for calculating a reproduction signal for the loudspeaker using the synthesis signals and the channel information for the supplied synthesis signals.
  • each loudspeaker module also comprises a signal processing device with possibly a digital amplifier, a further digital signal processing device and finally a digital-to-analog converter for generating an analog loudspeaker signal which is to be supplied to the loudspeaker concerned, on the basis of the reproduction signal.
  • a plurality of transmission links are provided for connecting the central wave field synthesis module and the decentralized loudspeaker modules, one transmission link each extending from the central wave field synthesis module to the individual loudspeaker.
  • the operation of the rendering is very computationally expensive, which, with regard to the necessary circuit hardware in the form of, for example, a DSP or a hard-wired circuit, contributes considerably to the costs, in particular when considering the multiplier which is provided for each individual loudspeaker.
  • the rendering device works using channel impulse responses as channel information and thus performs computationally intensive convolution, which can either be carried out directly in the time domain or is carried out in the frequency domain, which requires transformations in the frequency domain and transformations from the frequency domain, which are required together with the actual multiplication operation in the frequency domain lead to considerable effort.
  • a rendering unit does not only have to render a single synthesis signal, but always a large number of synthesis signals, which normally corresponds to the number of virtual sources.
  • the concept according to the invention means that operations which can be carried out in a decentralized manner are shifted out of the central wave field synthesis module into the decentralized loudspeaker modules in such a way that, in the best case, only the operations in the central wave field synthesis module which are equally important for all loudspeakers are carried out during all operations that affect only one loudspeaker, or several loudspeakers that are connected to a loudspeaker module, are also implemented decentrally in the loudspeaker module.
  • the costs for the central wave synthesis module can thus be considerably reduced, but at the expense of the loudspeaker modules, the price of which can no longer be neglected, due to the operation of the audio rendering which is mainly carried out in the loudspeaker modules.
  • the audio reproduction system according to the invention is now scalable both in terms of performance and in terms of price. This opens up the possibility of offering a central wave field synthesis module for a large number of display rooms at a reduced price, such that the costs for the overall system, which result from the costs for the central unit and the decentralized loudspeaker modules, now correspond strongly to the number of loudspeakers set up and thus the size of the playback room.
  • an operator of a large display room will still have to pay a certain price for a display system for his large display room.
  • an operator of a smaller playback room will be able to purchase an audio playback system at a significantly lower price, since the number of loudspeakers and thus the number of complex and costly loudspeaker modules is considerably reduced compared to the large playback room.
  • the audio playback system according to the invention thus makes it possible to offer audio playback systems for smaller playback rooms at considerably reduced prices compared to large playback rooms, so that market acceptance is hoped for due to the reduced price in the very competitive market of audio / video components.
  • the central wave field synthesis unit is designed to be able to process cinema films recorded in the conventional audio format for cinema films, conventional recording formats being, for example, the 5.1 surround format or 7.1. Format or 10.2 format.
  • conventional recording formats being, for example, the 5.1 surround format or 7.1. Format or 10.2 format.
  • a film includes six audio tracks, ie audio tracks for the "left rear", “right rear”, “front left”, “front right” and “front center” channels, as well as the bass channel (subwoofer channel ).
  • a playback of such a movie with respect to audio technology conventional in the audio playback system according to the invention can be achieved in that the Audio tracks can be placed as virtual sources in virtual positions, which can be selected according to the needs of the sound engineer or the operator of the playback room.
  • FIG. 1 shows a conceptual diagram of the audio reproduction system according to the invention
  • FIG. 2 shows a block diagram of the central wave field synthesis module according to the invention
  • FIG. 3 shows a block diagram of a decentralized loudspeaker module according to the invention
  • Fig. 4 is a block diagram of a preferred embodiment
  • FIG. 5 shows a schematic diagram of a compatible reproduction with a large sweet spot
  • FIG. 6 shows a schematic diagram for the formation of a plurality of synthesis signals for a loudspeaker, each of which is to be supplied with channel information in order to obtain the playback signal for the loudspeaker LSi; and
  • Fig. 7 shows a schematic diagram of a channel from a virtual source to a real loudspeaker, showing the quantities that can have an influence on the channel.
  • the audio playback system is basically divided into two parts, as shown in FIG. 1.
  • One part is the central wave field synthesis module 10.
  • the other part is composed of individual loudspeaker modules 12a, 12b, 12c, 12d, 12e, which are connected to actual physical loudspeakers 14a, 14b, 14c, 14d, 14e in the manner shown in FIG. 1 is shown.
  • the number of loudspeakers 14a-14e is in the range above 50 and typically even well above 100 in typical applications. If each speaker is assigned its own speaker module, the corresponding number of speaker modules is also required. Depending on the application, however, it is preferred to address a small group of loudspeakers next to one another from a loudspeaker module.
  • a loudspeaker module which is connected to four loudspeakers, for example, feeds the four loudspeakers with the same playback signal, or whether corresponding different synthesis signals are calculated for the four loudspeakers, so that such a loudspeaker module actually consists of there are several individual loudspeaker modules, but these are physically combined in one unit.
  • each transmission path 16a-leee there is a separate transmission path 16a-leee between the wave field synthesis module 10 and each individual loudspeaker module 12a-12e, each transmission path being coupled to the central wave field synthesis module and a separate loudspeaker module.
  • a serial transmission format that delivers a high data rate is preferred, such as a so-called Firewire transmission format or a USB data format. Data transfer rates of over 100 megabits per second are advantageous.
  • the data stream that is transmitted from the wave field synthesis module 10 to a loudspeaker module is accordingly formatted in accordance with the selected data format in the wave field synthesis module and with a
  • synchronization information which is provided in conventional serial data formats.
  • This synchronization information is extracted by the individual loudspeaker modules from the data stream and used to analyze the individual loudspeaker modules with regard to their reproduction, i.e. ultimately to the analog-digital conversion for obtaining the analog loudspeaker signal and the sampling (resampling) provided for this purpose. to synchronize.
  • the central wave field synthesis module operate as a master and that all loudspeaker modules operate as clients, with the individual data streams via the various transmission links 16a-16e all receiving the same synchronization information from the central module 10.
  • the central wave field synthesis module initially comprises an input device 20 which is basically designed to receive an audio signal at an input, the audio signal having a plurality of audio tracks, each audio track being assigned an audio source position.
  • the audio source position is an indication of the position of a loudspeaker with respect to a listener in the playback room in accordance with a standardized audio format, such as, for. B. 5.1 to achieve a compatible playback.
  • the audio signal can have a larger number of audio tracks, which are already available as signals suitable for wave field synthesis and represent audio sources or audio objects in a real recording position, which are reproduced with respect to the audio signal reproduction as virtual sources in the playback space using the wave field synthesis.
  • the input device 20 is also used as the main control unit, which advantageously has further functionalities. In particular, it has the functionality of a decoding module, as is usually used in cinemas. As an alternative or in addition, the input device 20 is also designed as a DVD decoder which supplies the separate audio channels or audio tracks. Alternatively, the playback device 20 is also designed as an MPEG-4 decoding module, which already provides audio tracks 21 and corresponding audio source information 22 intended for wave field synthesis.
  • the audio tracks 21 each relate to audio signals from audio objects in a recording setting, to the position of the audio objects in the recording setting, to properties of audio objects, in particular with regard to the size of the audio object or the density with regard to the acoustic properties of the audio object ,
  • the information about the recording room or the recording environment should serve to give the listener not only a visual but also an audio impression of the recording situation.
  • the visitor should also notice from the reproduced sound whether, for example, a recording scene of a movie is taking place outdoors or e.g. B. in a small space, such as a submarine. While a recording scenario in the open air delivers relatively "dry" audio signals because the recording environment shows hardly any or no reflections, this situation will be completely different in a submarine, for example.
  • the central wave field synthesis module further comprises a device 24 for determining channel information on the one hand and wave field synthesis signals on the other hand for the individual loudspeakers.
  • a device 25 for converting the audio source positions 22 into virtual positions for the wave field synthesis is also provided.
  • the device 24 is designed to determine audio channel information for each audio channel from a virtual position to a loudspeaker position, the virtual position from the
  • Audio source position associated with the audio track depends (means 25) so that for each channel from each virtual position to each speaker
  • Audio channel information is available. Furthermore, the
  • Device 24 is configured to use the
  • the central wave field synthesis module in FIG. 2 further comprises means 26 for supplying synthesis signals to one or more loudspeakers.
  • the device 26 is also designed to transmit channel information for the transmitted synthesis information from the central wave field synthesis module via the corresponding transmission links to the individual loudspeaker modules, so that audio rendering can take place there.
  • the device 24 also supplies channel information for each synthesis signal or interpolates from calculated channel information and the device 26 for Provides so that the same can initiate a transmission to the individual speaker modules.
  • the device 26 is preferably designed to filter out insignificant synthesis signals and thus neither to transmit the insignificant synthesis signals nor the associated channel information in order to save data transmission capacities. So often occurs the case that a virtual source leads to significant synthesis signals only for some speakers, while for all other speakers in the speaker array synthesis signals can also be calculated based on the theory of wave field synthesis. B. are relatively small in terms of their performance in a certain period of time and can therefore be neglected in terms of a reduced amount of data transfer.
  • the device 24 includes functionalities to be used to preprocess the audio signals.
  • the device 24 controls the individual loudspeaker modules in particular in such a way that it either directly or in conjunction with the device 26 introduces synchronization information into the data streams transmitted to the individual loudspeaker modules and thus achieves central synchronization of all loudspeaker modules with the central wave field synthesis module.
  • the central wave field synthesis module is designed to carry out all processing operations that are the same for all reproduction channels, while according to the inventive concept, the processing operations that are different for the individual loudspeakers or the individual reproduction channels are carried out decentrally.
  • the device 24 is also designed to simulate wave field synthesis information for stereo Signals, 5.1 signals, 7.2 signals, 10.2 signals, etc. with a view to compatible playback.
  • the standard positions of loudspeakers with respect to a playback space for the standardized audio format are used as audio source positions.
  • FIG. 5 shows a playback room 50, a speaker array 52 that extends around the playback room, and a plurality of virtual sources 53a-53e that, as can be seen in FIG. 5, are positioned at virtual positions outside of the Playroom 50 are located.
  • the device 24 is designed in connection with the device 25 from FIG. 1 in order to calculate virtual positions which can be controlled manually from the audio source information, that is to say the standard position information for such a 5.1 signal, for example.
  • the virtual positions e.g. B. to infinity
  • the speaker array 52 sonicates the playback room 50 with plane waves.
  • the so-called sweet spot i.e. the area in a reproduction room in which an optimal sound impression is obtained, is considerably enlarged compared to a common situation in which real 5.1 speakers are placed in the reproduction room.
  • the virtual sources can also be placed at finite virtual positions and modeled as point sources, this option having the advantage that the sound impression has a more pleasant effect on the cinema viewer / listener.
  • Flat waves have the property that the listener has the impression that he is sitting in a very large room, which leads in particular to an unpleasant sensation when, for example, a submarine scene is currently taking place on the screen.
  • conventional films with, for example, 5.1 audio tracks have no information about acoustic Features of the recording setting include. In such a case, it is therefore preferred to find a compromise between the plane waves, that is to say the virtual sources at an infinite position or the virtual sources at a finite position.
  • the audio playback system also provides the possibility of varying the virtual positions of the virtual loudspeakers 53a-53e depending on the film scene. For example, if a scene is taking place outdoors, the speakers can be positioned indefinitely. On the other hand, if a scene takes place in a small room, the loudspeakers can be positioned closer to the playback room 50.
  • input device 20 is configured to sample the audio tracks associated with the video signal by a certain time "delay" before the video signals, such that after processing in the wave field synthesis module in the individual loudspeaker modules, the sound belonging to a point in time is sampled simultaneously with the video signal belonging to a point in time.
  • the negative “delay” must at least be dimensioned in such a way that sound and image are emitted in the audio reproduction system according to the invention in a manner that is associated with one another. If the negative delay is dimensioned somewhat larger, the signals can already be calculated and, for example, can be output from the loudspeaker modules to the loudspeakers by means of a corresponding synchronization signal which ensures the synchronism of image and sound.
  • Information about the playback room can either be determined on the basis of the geometrical nature of the playback room, or can be measured in the playback room using the loudspeakers and special microphone arrays, with control and evaluation for this being possible via an adaptation module 28 for the playback room.
  • the acoustic properties of the display room differ significantly from those when there are no people in the display room.
  • the adaptation module 28 for the reproduction room further comprises a microphone array which can be used to measure the properties of the reproduction. Furthermore, the adaptation module 28 for the reproduction space comprises algorithms to find the position of speaker arrays in the reproduction space. In addition, preprocessing of measurement results is carried out here in order to carry out an optimal inversion of the room and loudspeaker properties, the adaptation module 28 preferably being controlled by the device 24 for this purpose.
  • the adaptation module 28 for the playback room is only required for the system structure. However, if continuous adaptation to a changed situation in the playback room is desired, the adaptation module 28 can also be used continuously during operation.
  • the additional WFS information that is to say the properties of, for example, the audio objects and the properties of the recording space, are extracted from the input audio signal and via a WFS information line 29 fed to the device 24 so that this information can be taken into account in the channel information calculation.
  • the central WFS module is also designed to carry out preprocessing of the WFS-prepared audio signals.
  • the device 24 and / or the device 26 is intended to achieve the synchronization between image and sound, for which, as has been explained, time codes are introduced into the preferably serial data streams for the individual loudspeaker modules.
  • the channel information calculation device 24 is also responsible for driving the adaptation module 28 in order to control the measurement of the acoustic properties of the reproduction space, if desired, either before playback or during playback.
  • the multiplexer / transmission stage 26 is designed to insert synchronization information, which is generated either by the device 24, by the control device 20 or in the device 26 itself, into the data streams to the loudspeaker modules, to which those for the individual speakers required synthesis signals and necessary channel information are supplied.
  • the device 24 for calculating the channel information and for calculating the synthesis signals must also be provided with the speaker locations in the special reproduction room in order to calculate the individual synthesis signals and the individual channel information for the individual speakers. This is symbolically represented in FIG. 2 by a line 30.
  • the loudspeaker module first comprises a receiver / decoder block 31 in order to receive the data stream from the selection device and to extract associated channel information 31b and synchronization information 31c from the same synthesis signals 31a.
  • the loudspeaker module shown in Fig. 3 further includes, as a central unit, an audio rendering device 32 for calculating a playback signal for the speaker using the one or more synthesis signals and using the channel information associated with the synthesis signals.
  • a loudspeaker module comprises a signal processing device 33 with a digital / analog converter for generating an analog loudspeaker signal which is fed to the loudspeaker LSi 34 concerned in order to generate a sound signal.
  • the signal processing device 33 and in particular the resampler, which cooperates with the digital / analog converter, is supplied via the synchronization information (31c) extracted from the data stream by the receiver 31 in order to be synchronous to the central wave field synthesis module and thus synchronous to all other loudspeaker modules 24 of Fig. 1 calculated on the To overlay loudspeakers overlaid with channel information and channel signals.
  • the loudspeaker module shown in FIG. 3 is thus characterized by the combination of a digital receiver, a further signal processing device and a digital-to-analog converter, wherein in particular a digital amplifier can also be provided in the signal processing device 33.
  • a digital amplifier can also be provided in the signal processing device 33.
  • the signal can also be amplified after the digital / analog conversion, although digital amplification is preferred due to the more precise possibility of synchronization. It is further preferred to couple the loudspeaker 34 to the signal processing device 33 via a short analog line.
  • the corresponding lines of all loudspeakers have the same length or have length differences which are within a predetermined tolerance limit, since the synchronization is preferably on digital side is performed so that with very different line lengths between the loudspeaker modules and the loudspeaker a desynchronization could occur, which could already lead to audible artifacts or to a loss of the sound impression that is to be created by the wave field synthesis.
  • channel impulse responses in the time domain or in the frequency domain are transmitted as channel information.
  • the audio rendering device 32 is designed to carry out a convolution of the individual synthesis signals with the channel information assigned to the synthesis signals.
  • This convolution can actually be implemented as convolution in the time domain, or can be implemented in the Frequency range can be performed by multiplying the analysis signal in the frequency range with the channel transfer function.
  • FIG. 4 shows a preferred embodiment of the audio rendering device 32 and comprises for each synthesis signal S j i (t) a time-frequency conversion block 34a, 34b, 34c, and for each branch a multiplier 35a, 35b, 35c for multiplying the transformed a synthesis signal with the transform of a channel impulse response H j i (f), a summer 36 and a final frequency-time conversion device 37, which are connected as shown in FIG. 4.
  • the arrangement shown in Fig. 4 is characterized in that it is reduced in terms of processing effort by the summation of the synthesis signals, which are already acted on by the corresponding channel transmission functions, takes place in the frequency domain, so that for each speaker module regardless of the number the synthesis signals only a single frequency-time converter is available.
  • the time-frequency transformation of the synthesis signals s- ⁇ can be carried out completely in parallel, or, if there is sufficient time, also serial / parallel or completely serial.
  • the audio rendering device 32 shown in FIG. 3 is also designed to receive special program information from the central wave field synthesis module shown in FIG. 2.
  • the multiplexer / transmitter stage 26 has a special output in order to supply the program information to the loudspeaker modules.
  • the program information can also be multiplexed into the data stream with synthesis signals and channel information, although this is not absolutely necessary.
  • channel information is described as channel impulse responses and transmitted to the individual loudspeaker modules, it is preferred not to transmit the entire impulse response in the sense of a data rate saving, but rather only samples of the impulse response that lie in a front area of the impulse response, the envelope of which is still an amount has a threshold.
  • impulse responses typically have large values at small points in time and gradually take on smaller values and finally have a so-called "reverberation tail", which is important for the sound impression, but whose samples are no longer particularly large, and In this case, it is preferred not to transmit the reverberation tail, whose envelope is below the threshold value, on the basis of its sample values, but only to transmit base values for the envelope
  • That are required by the audio rendering device 32 are then generated according to the invention in that the audio rendering device generates a random sequence of zeros and ones, the amplitude of which is weighted for the envelope with the transmitted base values it is preferred to transfer only a few base values and to interpolate between the base values, and then use the interpolated envelope to weight the random 0/1 sequence.
  • the random 0/1 sequence is preferably implemented by positive voltage values for a "1" and negative voltage values for a "0".
  • the information that the audio rendering device receives channel information that is actual samples up to a certain value and then is only support values for the envelope is transmitted via the program information input shown in FIG. 3 or is fixed agreed.
  • the wave field synthesis module further comprises a WFS mixing console, not shown in FIG.
  • the procedure on which the generation of synthesis signals is based is discussed below with reference to FIG. 6.
  • a system with three virtual sources at three virtual positions 60, 61, 62 and a speaker LSi 63 at a real speaker position, which is known to the central WFS module, is considered.
  • the virtual positions of the virtual sources 60, 61, 62 are known to the central wave field synthesis module either from the fact that they are supplied in a WFS-processed input signal or from the fact that they are derived by means of audio source positions by the means 25 for calculating the virtual positions.
  • the synthesis signals s 2 ⁇ , s 2 ⁇ and s 3 i are the signals which the loudspeaker 63 must emit and which go back to the respective virtual positions 60, 61, 62. From this it can be seen that, as has been stated, each loudspeaker will emit the overlay of several synthesis signals.
  • a channel ji is also defined between each virtual position and each loudspeaker can be described by an impulse response, a transfer function or any other channel information, as shown with reference to FIG. 7. All desired properties can be packaged in the channel description, in order then to apply the channel information for the corresponding channel assigned to a synthesis signal to the synthesis signals which are calculated by the wave field synthesis module. If the channel information is given in the form of an impulse response that describes the channel, the loading is a convolution. If the signals are in the frequency range, the application is a multiplication. Alternative channel information can also be used depending on the embodiment.
  • FIG. 7 shows which information can be used to influence a channel 70 from a virtual source 71 to a real loudspeaker 72.
  • the virtual position of the virtual source 71 is included in the channel information, for example the channel impulse response.
  • Properties of the virtual source are also included, such as. B. size, density, etc. B. a small triangle must be described and modeled differently than a large timpani.
  • the properties of the recording space are included in the channel transmission function.
  • Further influencing components are a system distortion of the entire audio reproduction system, which, for example, contains loudspeaker distortions or non-idealities of the loudspeakers.
  • the channel information also includes information about the playback space in order to compensate for the acoustic properties of the playback space. If, for example, the reproduction room is known to have a wall opposite a loudspeaker that is reflecting and whose reflection is to be suppressed, however, the corresponding loudspeaker is underneath Taking this information into account is controlled in such a way that it contains a signal which is 180 degrees out of phase with the reflected signal and has a corresponding amplitude, so that an extinguishing reflection occurs and the wall becomes acoustically transparent, ie no longer for a listener due to the reflections is identifiable.
  • the channel information can also be used to set a specific target reproduction acoustics.
  • the method according to the invention for reproducing an audio signal 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 cooperate with a programmable computer system such that the method is carried out.
  • the invention thus also consists in a computer program product with program code stored on a machine-readable carrier for carrying out the method according to the invention 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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  • Physics & Mathematics (AREA)
  • Engineering & Computer Science (AREA)
  • Acoustics & Sound (AREA)
  • Signal Processing (AREA)
  • Health & Medical Sciences (AREA)
  • General Health & Medical Sciences (AREA)
  • Otolaryngology (AREA)
  • Stereophonic System (AREA)
  • Circuit For Audible Band Transducer (AREA)
  • Stereophonic Arrangements (AREA)
  • Signal Processing Not Specific To The Method Of Recording And Reproducing (AREA)

Abstract

L'invention concerne un système de restitution audio comprenant un module de synthèse de champ électromagnétique (10) et une pluralité de modules de haut-parleurs (12a 12e) disposés de manière décentrée. Des signaux de synthèse pour les haut-parleurs individuels, ainsi que des informations concernant les canaux, qui sont associées au signaux de synthèse, sont calculés dans le module de synthèse de champ électronique central. Les signaux de synthèse pour un haut-parleur, conjointement avec les informations associées concernant les canaux sont ensuite transmis aux modules de haut-parleurs correspondants par l'intermédiaire d'une voie de transmission (16a 1§6e). Chaque module de haut-parleurs reçoit les signaux de synthèse et les informations associées concernant les canaux, conçus pour le haut-parleur qui est associé au module de haut-parleurs. Il s'effectue dans les modules de haut-parleurs une restitution audio décentralisée, ainsi qu'une conversion numérique/analogique, afin de produire les signaux de haut-parleurs analogiques effectifs, de manière décentralisée, au voisinage proche de chaque haut-parleur. La répartition en un module de synthèse de champ électromagnétique central et en une pluralité de modules de haut-parleurs décentralisés permet de produire des systèmes de restitution audio, modulables en termes de prix, afin de pouvoir proposer des systèmes de différentes dimensions extrêmement variables, qui soient modulables en termes de prix.
PCT/EP2003/013110 2002-11-21 2003-11-21 Systeme de restitution audio et procede de restitution d'un signal audio WO2004047485A1 (fr)

Priority Applications (4)

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JP2004552700A JP4620468B2 (ja) 2002-11-21 2003-11-21 オーディオ信号を再生するためのオーディオ再生システムおよび方法
EP03782222A EP1576847B1 (fr) 2002-11-21 2003-11-21 Systeme de restitution audio et procede de restitution d'un signal audio
DE50303069T DE50303069D1 (de) 2002-11-21 2003-11-21 Audiowiedergabesystem und verfahren zum wiedergeben eines audiosignals
US11/099,156 US7706544B2 (en) 2002-11-21 2005-04-05 Audio reproduction system and method for reproducing an audio signal

Applications Claiming Priority (2)

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DE10254404.2 2002-11-21
DE10254404A DE10254404B4 (de) 2002-11-21 2002-11-21 Audiowiedergabesystem und Verfahren zum Wiedergeben eines Audiosignals

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JP2006507727A (ja) 2006-03-02
EP1576847A1 (fr) 2005-09-21
DE10254404A1 (de) 2004-06-17
JP4620468B2 (ja) 2011-01-26
DE50303069D1 (de) 2006-05-24
DE10254404B4 (de) 2004-11-18
EP1576847B1 (fr) 2006-04-19
ATE324021T1 (de) 2006-05-15

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