EP1260119B1 - Multikanaltonwiedergabesystem für stereophonische signale - Google Patents

Multikanaltonwiedergabesystem für stereophonische signale Download PDF

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EP1260119B1
EP1260119B1 EP00904860A EP00904860A EP1260119B1 EP 1260119 B1 EP1260119 B1 EP 1260119B1 EP 00904860 A EP00904860 A EP 00904860A EP 00904860 A EP00904860 A EP 00904860A EP 1260119 B1 EP1260119 B1 EP 1260119B1
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Prior art keywords
signals
phase
output
difference
function
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French (fr)
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EP1260119A1 (de
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Jan Abildgaard Pedersen
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Bang and Olufsen AS
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Bang and Olufsen AS
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    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04S—STEREOPHONIC SYSTEMS 
    • H04S5/00—Pseudo-stereo systems, e.g. in which additional channel signals are derived from monophonic signals by means of phase shifting, time delay or reverberation 
    • H04S5/005—Pseudo-stereo systems, e.g. in which additional channel signals are derived from monophonic signals by means of phase shifting, time delay or reverberation  of the pseudo five- or more-channel type, e.g. virtual surround
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04S—STEREOPHONIC SYSTEMS 
    • H04S2400/00—Details of stereophonic systems covered by H04S but not provided for in its groups
    • H04S2400/05—Generation or adaptation of centre channel in multi-channel audio systems

Definitions

  • the present invention relates generally to multi-channel sound reproduction via loudspeakers and more particularly to extraction of appropriate monophonic signal components from a normal stereophonic signal and providing each of these monophonic signals to different loudspeakers in a multi-channel sound reproduction set-up.
  • a specific class of such multi-channel systems utilises some kind of decoding means to translate the signals from two stereophonic sound tracks for instance on a motion picture film or on a gramophone record or compact disc for domestic use into a larger number of signals, each of which is to be provided to separate loudspeakers placed at different positions in the listening room.
  • the extraction of the monophonic signal for the centre loudspeaker is in the above system based on determination of the correlation between the left and right stereophonic signals. These signal components that are highly correlated with each other are extracted from the two channels, added and provided to the centre loudspeaker. There remains the "stereophonic part" of the signals which parts are reproduced via the front left and right loudspeakers as normal stereophonic signals.
  • a system according to US-5426702 is suggested by Aarts.
  • a centre channel signal is derived from the left and right channel signals based on the determination of a direction vector which indicates the direction to the most powerful sound from origo in a coordinate system depicting the magnitude of the left signal along one axis and the magnitude of the right signal along the other axis.
  • two weight factors are derived such that weighted right and left signals are added to form the centre channel signal.
  • the prior art systems derive a purely monophonic signal to be provided to a centre loudspeaker, they still function to a large extent as a normal stereophonic loudspeaker system, i.e. the perceived sound images are the result of a perceptual combination in the brain of the listener of sound signal components originating from the left and right loudspeakers. If signal components from the left and right loudspeaker in such a system are either fully or at least partially correlated. these components will "melt together" in the brain of the listener into one spatially defined sound image, which will often be located somewhere on the line between the two loudspeakers.
  • phantom source This perceived sound image is often termed a "phantom source", and it can be said that in stereophonic sound reproduction systems the formation of the overall perceived sound image basically relies on the formation of phantom sources. If either the left or right channel signal is much stronger than the other, or there is a sufficient time delay between these signals, the phantom source will be located at one of the loudspeakers, i.e. either at the loudspeaker radiating the strongest signal or the loudspeaker leading in time relative to the other. Only in such cases there is a coincidence between the phantom source and the actual physical sound source.
  • the desired perceptual effect of this system also relies on the formation of phantom sources and it furthermore requires that the directional information contained in the left and right input signals are encoded according to a predetermined matrix.
  • a system for deriving a signal for a centre loudspeaker from the left and right channel signals of a stereophonic signal is disclosed in US 5,528,694, the system being primarily intended for use in an audio-visual reproduction system such as a TV set.
  • the centre signal is derived by means of a splitter circuit for splitting off from the left channel signal components that are identical to signal components in the right channel and vice versa. Those signal components of the left and right channels that are not identical are reproduced by left and right loudspeakers of a normal stereophonic set-up. Apart from the monaural center channel signal the total perceived sound image is still formed by phantom sources created by the left and right loudspeakers.
  • a sound element which was intended to be located in the symmetry plane, will thus only be located in this plane, when the listener is also positioned herein.
  • the optimal listening positions are thus confined to a narrow region around the symmetry plane. It would, however, be desirable to extend the listening region to a large region of space, at least in front of the loudspeakers.
  • a localisation error not infrequently encountered in connection with the formation of phantom sources consists of a so-called elevation error, i.e. the phantom source, which ideally should be perceived directly on the line between the left and right loudspeakers, and hence normally approximately at the level of the listener's ears, is actually being perceived above this level.
  • elevation errors can be the result of the presence of small phase differences, which at a specific frequency correspond to similar minor time differences between the signals from the two loudspeakers at the position of the ears of the listener.
  • phase or time differences between two substantially equally powerful signals will produce a combfilter effect cancelling the sound signals at a discrete series of frequencies.
  • these objects are achieved by replacing the phantom sources of a normal stereophonic reproduction system by a number of actual physical sound sources placed at the positions where said phantom sources would be located while listening to the normal stereophonic system from a ideal listening position substantially located in the symmetry plane of the two stereophonic loudspeakers.
  • a method for converting two stereophonic (left and right channel) input signals L(t) and R(t) into N output signals according to the characterising clause of claim 1, where said method according to a preferred embodiment of the invention comprises the following steps:
  • step (3) of the method according to the invention said comparison and application of the specific set of requirements could be carried out on the pair of first residual left and right channel signals provided in step (2) above instead of on the original left and right channel signals. It is advantageous that the procedure described in step (3) is applied, but in a practical implementation it may be necessary or desirable to apply said alternative.
  • a device for carrying out said method, where said device comprises N-2 means for extracting said output signals corresponding to said phantom sources, where each of said N-2 means furthermore provides said pairs of residual left and right channel signals which does not contain any of - or according to a second embodiment only a fraction of - those signal components, that would have contributed to said phantom sources, which pair of residual signals are provided to succeeding means for extraction of the remaining output signals.
  • the extractions of said output signals from the left and right input signals - or from the corresponding residual signals - is according to the invention based on a running comparison, i.e. a comparison as a function of time, of the degree of linear dependency between each of said pairs of separate frequency components of the two input signals.
  • a measure of the degree of linear dependency between left and right signals is thus according to the invention based on a running cross correlation analysis of left and right signal pairs and a succeeding determination of the coherence function, which is a number between 0 and 1, where the value 1 is obtained when the left and right signals are fully correlated and the value 0 is obtained when the left and right signals are fully uncorrelated.
  • the criterion for extraction of a output signal to be provided to one of said N-2 loudspeakers positioned between the left and right loudspeakers is that the coherence function should have a value close to 1, preferably between 0.8 and 1, although other intervals may also be chosen. If it is found that certain left and right signal elements fulfil said coherence criterion, those elements could have contributed to the formation of a phantom source in a normal left and right channel stereophonic system, and will thus according to the invention be represented by an actual physical sound source, i.e. one of the N-2 loudspeakers placed between the outermost left and right loudspeakers.
  • the signal to be provided to this loudspeaker is according to the invention being obtained by a linear combination of the corresponding left and right input signals to that particular processing block, in which the extraction of that particular output signal takes place.
  • Which one of these N-2 loudspeakers actually should be provided with the extracted signal could on principle be determined based on either a comparison - for each pair of frequency components - of the magnitudes of the left and right signals or on a comparison of the relative phase (or time delay) between these frequency components. It is also possible to use combinations of magnitude and phase (or time) differences for extracting a measure for the lateralisation of the phantom source and hence for the appropriate location of a corresponding sound source.
  • the system according to the invention can be said to replace the phantom sources obtained in a normal stereophonic system with a corresponding number of real physical sound sources.
  • phantom sources will only be perceived, if correlated signal components are found in the left and right channels (see for instance: Jens Blauert, "Spatial Hearing", Section 3.1.).
  • the perceived position of the phantom source will depend on both the amplitude difference and the phase difference (or time difference) between the correlated signal components, this dependency being generally a function of frequency, in the left channel relative to the right channel.
  • a time delay corresponds to a linear phase difference, i.e. a phase difference, which is proportional to the frequency.
  • the coherence function ⁇ (f) can be used as a measure of the degree of correlation between the left and right signal.
  • the coherence function is a real number between 0 and 1 indicating the fraction of power in the correlated part of the signals compared to the total signal power, when considering two signals, for instance the left signal L(t) and the right signal R(t) in a normal stereo system.
  • the coherence is 1 when the two signals are fully correlated at that frequency, i.e.
  • Equation (1) gives the coherence function ⁇ (f) at the frequency f, obtained using calculated values of the cross spectrum G LR (f) and the two auto spectra G LL and G RR based on the spectra L(f) and R(f) obtained by FFT analysis of the original pair of signals L(t) and R(t). For more information about the coherence function see for instance Julius S. Bendat and Allan G.
  • Certain requirements must be fulfilled before a part of the left and right signals are extracted and provided to a specific loudspeaker. These requirements comprise upper and lower limits on the amplitude difference between left and right signals, limits on group delay between these signals and as mentioned previously a minimum value of the coherence function. These three requirements together ensure that a phantom source was intended to be formed in the vicinity of a given one of the loudspeakers.
  • Enforcement of the limits can be carried out very sharply as in said first embodiment of the invention or smoothly as in said second embodiment of the invention.
  • a sharp enforcement is obtained by requiring that the value of the coherence function should be at least 0.8 for a signal to be extracted for a specific loudspeaker.
  • a smooth enforcement would be obtained by providing a highly attenuated signal to the particular loudspeaker at a coherence value of for instance 0.7 and letting the signal level increase gradually up to coherence values above 0.9.
  • the sharp limits are used, i.e. the total left and right signals components at a given frequency are extracted after suitable combination hereof and provided as an output signal to the particular loudspeaker.
  • a requirement R comprises three parameters: the minimum value of the coherence function, the range of the amplitude difference (dB) between left and right signal and the range of the group delay (ms) or phase difference (degrees) between left and right signals.
  • N 5
  • three loudspeakers - centre-left, centre and centre-right - are placed substantially equidistantly between the left and right loudspeakers.
  • the different sets of requirements could for instance be the following, although other requirements and/or specific values would also be conceivable:
  • amplitude differences are used to decide between the different loudspeakers. It is as mentioned previously also possible to base the choice between the loudspeakers on group delay differences (or phase differences which are related to group delay differences at a specific frequency) or on combinations of amplitude- and group delay(phase) differences. It should be emphasised that the invention is not limited to the utilisation of amplitude differences for the choice between the different loudspeakers, although a choice based on amplitude differences may be advantageous, because the normal way of producing stereophonic signals (so-called intensity stereophony), i.e.
  • left/right channel signals to be recorded for instance on normal compact discs is to control the lateralisation of the created phantom sources by manipulating the relative amplitudes (levels) of different output sound recordings in an electronic mixing console. Creation of phantom sources by manipulating relative group delays of output signals is normally not used.
  • a fourth requirement is set up in order to handle the special case of left and right signals being in anti-phase, i.e. 180 degrees out of phase. If the left and right channel signals are 180 degree out of phase the corresponding group delay is still 0 ms. Consequently, two otherwise identical signals in the left and right channels but 180 degrees out of phase will fulfil the above three requirements for a signal to be extracted and provided to the centre channel.
  • the extracted output signal is formed as a linear combination of left and right channel signals. According to a preferred embodiment of the invention this linear combination consists of the sum of the left and right channel signals and in the case of 180 degrees phase shifted left and right signals the extracted output signal will thus be equal to zero.
  • the above-mentioned fourth requirement would in this case be unnecessary.
  • the extraction is still based on specific sets of requirements for the coherence function, the amplitude difference and the phase difference corresponding to each of the phantom sources, which in this case generally are only partly replaced by physical sound sources.
  • the fraction of each frequency component to be extracted from the specific input signals is obtained by multiplying these frequency components with a filter function H(z) which is a product of continuos functions the parameters of which are chosen according to the specific sets of requirements, e.g.
  • Gaussian functions (normal distribution density function) of the values of the square of the coherence function, the amplitude difference and the phase difference, where the parameters of these three Gaussian functions (normal distribution density function) (means and variances) correspond to sets of requirements as for instance those used in the first embodiment.
  • the mean value of the three Gausian functions will be 1 (coherence), 0 (amplitude difference) and 0 (phase difference), and the variances will be suitably chosen, so that the product of these three Gaussian functions (normal distribution density function) will only be substantially equal to unity for those signal components that correspond to the particular phantom source. which is to be replaced entirely by a physical sound source.
  • the value of the filter function H(z) can thus be anywhere between 0 and 1, yielding a more smooth enforcement of the requirements for extraction of monophonic output signals than obtained according to the first embodiment.
  • a third embodiment of the invention it is possible to combine said sharp enforcement of the requirements for extraction of monophonic output signals according to the first embodiment and said smooth enforcement according to the second embodiment described above.
  • This can for instance be done by replacement of said filter function H(z) according to the first or second embodiment with a new filter function H(z) formed as a product of a logical function H1 (z;p) with output values of 1 or substantially 0 according to whether the parameters p, which may be the coherence function, the amplitude difference, the phase and/or group delay difference, belongs to the corresponding target intervals according to the first embodiment, and a function H2(z;q) which according to the second embodiment is a product of continuous functions, where q denotes the remaining parameters not contained in said function H1.
  • N 5 i.e. a total of five loudspeakers are used and these loudspeakers are placed in a line in front of a listening area, although the loudspeakers could also have been placed along for instance an arc in front of the listening area.
  • a normal stereophonic loudspeaker set-up is shown.
  • An actual physical sound source located midways between the two loudspeakers is in this set-up being simulated with the aid of two highly correlated electrical signals L(t) and R(t) fed to the loudspeakers.
  • L(t) and R(t) fed to the loudspeakers.
  • the perceived sound image is no longer located at PS as intended but is shifted more or less to the left as indicated at 17 by the area B in the figure.
  • the overall perceived sound image thus depends on the position of the listener, and the "correct" perception of a sound source at PS is thus only obtained in a narrow region around A in the figure.
  • Figure 2 shows one embodiment of a system according to the present invention utilising five loudspeakers 21, 22, 23, 24, 25 placed in front of a row of seats 26, 27, 28 in a listening room.
  • N 5 in this embodiment.
  • a physical sound source midways between the extreme left and right loudspeakers 21 and 25 is not simulated by a phantom source midways between these loudspeakers but by a physical sound signal radiated by the centre loudspeaker 23.
  • This means, that a listener will perceive the sound as originating from the centre loudspeaker 23 no matter where he is located, at least in the whole listening area in front of the loudspeakers.
  • correct spatial reproduction of a given original sound source is being preserved by the system according to the invention no matter what listening position the listener actually chooses.
  • the correct spatial characteristics of the perceived sound image are also preserved, if the listener moves around ir front of the loudspeakers.
  • FIG. 3 shows an embodiment of the system according to the present invention utilising three processing blocks 32, 33, 34 and five loudspeakers 35, 36, 37, 38, 39.
  • a normal intensity stereophonic signal L, R is provided from a stereophonic source 31, exemplified by a CP-player, to the first processing block 32.
  • This processing block 32 extracts in a manner to be described in detail in the following an output centre channel signal c 1 , which is being provided to the centre loudspeaker 37.
  • the output signal c 1 is in processing block 32 being removed from the left and right signals L and R in a manner to be described in detail in connection with the description of fig.
  • This processing block 33 extracts in an analogous manner as block 32 a second output signal c 2 , which is being provided to a loudspeaker placed midways between the left loudspeaker 35 and the overall centre loudspeaker 37.
  • the second output signal c 2 is removed from the signals L' and R' in a manner analogous to the procedure in the preceding block 32, and two new output signals L" and R" are being obtained and forwarded as new input signals to the succeeding processing block 34.
  • the filter H(z) 43 which according to this embodiment of the invention in principle can only take on the two values 1 or 0 at any given frequency, is used to filter both left and right channel signals 41, 42, and thereby to isolate those parts of the left and right channel signals, which fulfils the particular requirements for that output signal, which is to be provided to that channel and removed from the left and right input signals in order to produce the residual left and right channel signals L' and R' respectively.
  • the filter 43 used for the left channel is similar to the filter used for the right channel.
  • the original stereophonic signal might have been produced by panning, i.e. splitting an output signal up into two parts, which are provided to the left and right channels separately.
  • panning i.e. splitting an output signal up into two parts, which are provided to the left and right channels separately.
  • intensity stereophony panning consists of splitting an output signal up into two signals with an appropriate amplitude (intensity) difference between the two signals and adjusting this amplitude difference, so that it corresponds to the desired lateral position of the finally created phantom source.
  • the frequency components of the output signals of the filters H(z) 43 are added in an addition means 45 to produce an output signal 48 and a gain 49 and post delay 410 is applied to this signal to obtain the desired output signal 411.
  • the gain 49 can be used to adjust the output level of the signal radiated from the particular loudspeaker, to which the signal 411 is being provided, in order for instance to preserve total radiated power.
  • the post delay 410 will be explained in the following.
  • the parts of the left and right channel signals L and R which are extracted and provided as an output signal to the particular channel should be removed from the left and right channels, leaving the residual left and right signals L' and R' respectively.
  • This is done by subtracting in subtraction means 44 the output signals from H(z) 43 from delayed versions, delayed in two delay means 48, of the left and right channel signals.
  • This delay is introduced to compensate for the delay of H(z) 43, which should ideally be a linear phase filter, i.e. exhibit a frequency independent delay.
  • each processing block 32, 33, 34 takes those parts of the left and right channel signals, which fulfil the requirements set up for each loudspeaker, and then passes the remaining parts (the residual left and right signals) on the next processing block in the chain.
  • the residual left and right signals which remain after the processing in the last of the preceding blocks 34 have been carried out, are then provided to the left 35 and right 39 loudspeakers respectively. This ensures that if no parts of the left and right channel signals fulfils the requirements set up for any of the intermediate loudspeakers 36, 37, 38, then the signals are reproduced only by the outermost left and right loudspeakers 35, 39 as ordinary stereophonic reproduction.
  • H(z) 43 is calculated independently at different frequencies or in a number of different frequency bands.
  • a hysteresis can be implemented by changing the limits once a requirement has been met, e.g. to (-2.5 dB ⁇ amp(z) ⁇ 2.5 dB). In this case the value of amp(z) needs to change more than 0.5 dB before it can make H(z) shift back to 0.
  • a smoothing 519 might then be applied to the target of H(z) before implementing H(z), e.g. a Gaussian function (normal distribution density function) with frequency dependent width (e.g. 1/3 octave).
  • Figure 5 contains a detailed block diagram of the processing block shown in figure 4.
  • the upper part of figure 5 (reference numerals 51 to 521) and figure 6(a) shows the determination of the function H(z) based on left and right input signals 51, 52 according to the first embodiment of the invention
  • the lower part of figure 5 (reference numerals 522 to 534) corresponds to figure 4 except for the fact that in figure 5 fast convolution (see Oppenheim and Schafer: "Descrete-time-signal-processing", Prentice Hall, 1989, ISBN 0-13-216771-9) is employed to perform convolution by H(z).
  • H(z) carried out in the upper part of figure 5 and in figure 6(a) are based on block operations, e.g. 512 samples at a time. These samples are isolated using time windows 53. After a transformation to the frequency domain has been performed by FFT means 54, three quantities are calculated by means 55, 56 and 57: the instantaneous autospectra G ll and G rr are calculated in 55 and 56 respectively and the instantaneous crossspectrum G lr is calculated in 57. These instantaneous spectra are then turned into a real estimate of these spectra by the application of low pass filtration in each of the filters 58 respectively, one frequency at a time. This is done in this embodiment of the invention using first order IIR filters for each frequency.
  • the foregoing equations (1), (2) and (3) are then used to calculate the desired parameters, i.e. the phase difference is calculated in 511, the coherence function is calculated in 512 and the amplitude difference is calculated in 513.
  • the resulting parameter values are compared with the set of requirements corresponding to the particular output signal, which it is desired to derive, and this is done by comparing in blocks 514, 515 and 516 the output signals from the means 512. 511 and 513 with the specific parameter target ranges corresponding to the particular output signal (c1, c2, c3 ...), which is to be extracted as exemplified by the parameter intervals for the phase, the coherence and the amplitude difference shown in fig.
  • the lower part of figure 5 (522 through 534) is the processing part of the system according to the invention while the upper part of figure 5 (51 through 521) is the analysis part of the system.
  • N 3, i.e. three loudspeakers
  • two possible configurations of the series of processing blocks would be possible.
  • the two input terminals 51, 52 of the analysis part of the block are connected to the corresponding two input terminals 522 and 523 respectively of the processing part of the block.
  • the input signals to the analysis parts of the first block would be the original left and right channel signals L and R
  • the input signals to the analysis part of the next block would be the residual left and right channel signals L' and R' and so on. If, during the production of the original stereo signals, an output signal is being rapidly panned between the left and right channel for instance simulating a rapid shift of the position of a sound source between for instance the centre loudspeaker 37 and the loudspeaker 38 to the right of this, there will initially correctly be extracted an output signal for the centre loudspeaker 37and finally also correctly an output signal for the loudspeaker 38 to the right of the centre loudspeaker.
  • the input terminals 714, 715; 716, 717; 718, 719 to the three analysis parts 73, 74, 75 are all connected to the original left and right channel signals, whereas the three processing blocks 76, 77, 78 extracting output signals for the centre loudspeaker 711, the loudspeaker 710 to the left of the centre loudspeaker 711 and the loudspeaker 712 to the right of the centre loudspeaker 711 are coupled in series as already shown in figure 3.
  • the phase difference, phase is at 61 provided to a means 64 for calculation of the exponent of the corresponding Gaussian function (normal distribution density function), which Gaussian function (normal distribution density function) in the case shown in figure 6(b) corresponds to the extraction of signal components corresponding to a phantom source placed directly midways between the outermost left and right loudspeakers, and hence the mean of this Gausian function (normal distribution density function) is 0.
  • Gaussian function normal distribution density function
  • the squared coherence function is at 62 provided to a means 65 for calculation of the exponent of the second one of said three Gaussian functions (normal distribution density function) and the amplitude difference is at 63 provided to means 66 for calculating the exponent of the third one of said Gaussian functions (normal distribution density function).
  • the three Gaussian functions are hereafter calculated in three identical means 67, the output of each of these being provided to a multiplication means 68, which via a succeeding slew rate limiter 69 and smoothing 610 provides the final filter function H(z), 611, the value of which will be equal to unity for those frequency components which correspond exactly to a phantom source midways between said outermost left and right loudspeakers, and less than unity for frequency components corresponding to a phantom source created somewhat either to the left or to the right of the center loudspeaker or for frequency components, which do not correspond to any phantom source, because the corresponding coherence function differs significantly from unity.
  • the signal provided by the slew rate limiter 69 and succeeding smoothing 610 is hereafter used as a weigthing function, and provided to the multiplication means 526 shown in figure 5.

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Claims (26)

  1. Verfahren zur Umwandlung von zwei Eingangssignalen L(t) und R(t), welche die Signale auf dem linken und rechten Kanal eines stereophonischen Signals bilden, in N Ausgangssignale, die N Ausgangskanäle bilden, wobei N > 2, welches die folgenden Schritte aufweist:
    (A) Basierend auf den Originalsignalen der linken und rechten Kanäle L(t) und R(t), die für die linken und rechten Wandler in einem normalen stereophonischen Wiedergabesystem mit zwei Lautsprechern gedacht sind, und basierend auf einem Vergleich jedes einzelnen Paars linker und rechter Frequenzkomponenten dieser Signale, die z.B. durch eine schnelle Fourier-Transformation dieser linken und rechten Signale bereitgestellt werden, und auf der Anwendung eines ersten spezifischen Satzes von Bedingungen auf das Ergebnis dieser Vergleiche Extrahieren eines ersten Ausgangssignals (c1) als eine Linearkombination aus den Signalkomponenten der linken und rechten Kanäle unter der Bedingung, daß die Beziehung zwischen den Signalkomponenten der linken und rechten Kanäle derart ist, daß diese zur Bildung einer ersten Phantomquelle beitragen würden,
    (B) Bereitstellen eines Paars erster Restsignale der linken und rechten Kanäle (L', R'), wobei dieses Paar nicht die skalierten Versionen von Frequenzkomponenten enthält, die in dem vorhergehenden Schritt (A) extrahiert wurden,
    (C) Basierend auf den Originalsignalen der linken und rechten Kanäle L(t) und R(t) und auf die gleiche Weise wie weiter oben basierend auf einem Vergleich jeder einzelnen Frequenzkomponente dieser Signale und auf der Anwendung eines zweiten spezifischen Satzes von Bedingungen auf das Ergebnis dieser Vergleiche Extrahieren eines zweiten Ausgangssignals (c2) als eine Linearkombination aus den Restsignalen der linken und rechten Kanäle unter der Bedingung, daß die Beziehung zwischen den Originalsignalkomponenten der linken und rechten Kanäle derart ist, daß diese zur Bildung einer zweiten Phantomquelle beitragen würden, die an einer anderen Position als die erste Phantomquelle angeordnet ist,
    (D) Bereitstellen eines Paars zweiter Restsignale der linken und rechten Kanäle (L", R"), wobei dieses Paar nicht die skalierten Versionen von Frequenzkomponenten enthält, die in den vorhergehenden Schritten (A) und (C) extrahiert wurden,
    (E) Wiederholen der vorhergehenden Schritte für eine ausreichende Anzahl von Malen und jedes Mal mit anderen Bedingungssätzen, um in der Lage zu sein, ein Maximum von N-2 Ausgangssignalen (c1, c2, c3, c4 ...) zu extrahieren, die N-2 Phantomquellen entsprechen, die durch die Originalsignale der linken und rechten Kanäle L(t) und R(t) gebildet werden könnten,
    (F) Bereitstellen eines Paars von letzten Restsignalen der linken und rechten Kanäle (L"', R"'), wobei dieses Paar nicht die skalierten Versionen von Frequenzkomponenten enthält, die in irgendeinem der vorhergehenden Schritte extrahiert wurden,
    (G) Bereitstellen des ersten, zweiten, etc. Ausgangssignals (c1, c2, c3 ...) an N-2 elektroakustische Wandler (36, 710; 37, 711; 38, 712), wobei die Position jedes dieser Wandler dem bestimmten Satz von Bedingungen entspricht, der bei der Extraktion des Ausgangssignals (c1, c2, c3, ...) für diesen bestimmten Wandler verwendet wurde,
    (H) Bereitstellen des letzten Restsignals des linken Kanals (L"') an einen elektroakustischen Wandler (35, 79), der links von allen anderen N-2 Wandlern angeordnet ist, und Bereitstellen des letzten Restsignals des rechten Kanals (R"') an einen elektroakustischen Wandler (39, 713), der rechts von allen anderen N-2 Wandlern angeordnet ist.
  2. Verfahren nach Anspruch 1, dadurch gekennzeichnet, daß der Vergleich des Originalsignals des linken Kanals L(t) und des Originalsignals des rechten Kanals R(t) die Bestimmung der Kohärenzfunktion (γ) der Originalsignale L(t) und R(t), der Amplitudendifferenz (Amp) zwischen den Originalsignalen L(t) und R(t) und der Phasendifferenz (oder der Gruppenlaufzeitdifferenz) (Phase oder τ) zwischen den Originalsignalen L(t) und R(t) bei jeder Frequenzkomponente aufweist.
  3. Verfahren nach Anspruch 2, dadurch gekennzeichnet, daß die Kohärenzfunktion (γ), die Amplitudendifferenz (Amp) und die Phasen- oder Gruppenlaufzeitdifferenz Funktionen der Frequenz sind und basierend auf dem Kreuzspektrum GLR(f) und den zwei Autospektra GLL (f) und GRR(f) gemäß den folgenden Gleichungen berechnet werden: γ ( f ) = | G L R ( f ) | G L L ( f ) ⋅ G R R ( f )
    Figure imgb0015
     Amp ( f ) = G L L ( f ) G R R ( f )
    Figure imgb0016
     Phase ( f ) = Winkel ( G L R ( f ) )
    Figure imgb0017
     τ ( f ) = d ( stetige_Phase ( f ) ) 2 π ⋅ d f
    Figure imgb0018
  4. Verfahren nach einem der vorhergehenden Ansprüche, dadurch gekennzeichnet daß die genannten Bedingungssätze jeweils ein Zielintervall der Kohärenzfunktion (γ), ein Zielintervall der Amplitudendifferenz (Amp) und ein Zielintervall der Phasen- oder Gruppenlaufzeitdifferenz (Phase, τ) aufweisen, wobei diese Zielintervalle Funktionen der Frequenz sein können.
  5. Verfahren nach einem der vorangehenden Ansprüche, dadurch gekennzeichnet, daß die Extraktion von Ausgangssignalen (c1, c2, c3 ...) auf einem Vergleich der Kohärenzfunktion (γ), der Amplitudendifferenz (Amp) und der Phasen-oder Gruppenlaufzeitdifferenz (Phase, τ) bei jeder Frequenzkomponente mit dem jeweiligen der Zielintervalle basiert, so daß ein spezifisches der Ausgangssignale (c1, c2, c3 ...) nur extrahiert wird, wenn die Kohärenzfunktion (γ), die Amplitudendifferenz (Amp) und die Phasen- oder Gruppenlaufzeitdifferenz (Phase, τ) alle den spezifischen Zielintervallen für dieses spezifische Ausgangssignal (c1, c2, c3 ...) entsprechen.
  6. Verfahren nach einem der vorangehenden Ansprüche, dadurch gekennzeichnet, daß die Extraktion eines gegebenen der Ausgangssignale (c1, c2, c3 ...) zum Beispiel für schnelle Fourier-Transformierte eines gegebenen Paars von Eingangssignalen (L, R; L', R'; L", R"; ...) durchgeführt wird, wobei das gegebene Paar von Eingangsignalen im Fall des ersten extrahierten Ausgangssignals (c1) die Originalsignale der linken und rechten Kanäle (L, R) sind, im Fall des zweiten extrahierten Ausgangssignals (c2) die ersten Restsignale der linken und rechten Kanäle (L', R') sind, im Fall des dritten extrahierten Ausgangssignals (c3) die zweiten Restsignale der linken und rechten Kanäle (L", R") sind, etc., wobei die schnellen Fourier-Transformierten eines gegebenen Paars von Eingangssignalen mit gleichen Filterfunktionen H(z) multipliziert werden, die durch den Vergleich der bestimmten Kohärenzfunktion (γ), der bestimmten Amplitudendifferenz (Amp) und der bestimmten Phasen- oder Gruppenlaufzeitdifferenz (Phase, τ) mit ihren Zielwerten, die dem bestimmten der zu extrahierenden Ausgangssignale (c1, c2, c3 ...) entsprechen, gebildet werden, und wobei die multiplizierten Versionen der schnellen Fourier-Transformierten einer umgekehrten schnellen Fourier-Transformierten (527) unterzogen werden, so daß die zwei sich ergebenden Zeitbereichsignale (535, 536) nach ihrer einzelnen geeigneten Skalierung schließlich addiert (529) werden können, um eine erste Version (c1' , c2', c3' ...) dieses bestimmten Ausgangssignals (c1, c2, c3 ...) als eine Linearkombination des gegebenen Eingangssignalpaares zu bilden, wobei die Schritte der schnellen Fourier-Transformation, der Multiplikation und der schnellen Fourier-Umkehrtransformation zum Beispiel Verfahrensschritte des als schnelle Faltung bekannten Verfahrens sind.
  7. Verfahren nach Anspruch 6, dadurch gekennzeichnet, daß die Ausgangssignale (c1, c2, c3 ...) durch eine Verstärkung (530) gebildet werden, der eine nachgelagerte Verzögerung (531) der ersten Version (c1', c2', c3', ...) der Ausgangssignale (c1, c2, c3 ...) folgt.
  8. Verfahren nach Anspruch 6, dadurch gekennzeichnet, daß die Filterfunktion H(z) eine logische UND-Funktion ist, d.h. eine Funktion mit Ausgangswerten von 1 oder im wesentlichen 0, welche durch einen Vergleich der Kohärenzfunktion, der Amplitudendifferenz und der Phasen- oder Gruppenlaufzeitdifferenz bei jeder Frequenzkomponente mit entsprechenden Zielintervallen erhalten wird, welche dem einen der Ausgangssignale, das abgeleitet werden soll, entsprechen, wobei H(z) durch eine der beiden Gleichungen gegeben ist:
    H(z)=(γ1 < γ(z) < γ2) UND (Amp1 < Amp(z) < Amp2) UND (Phase1 < Phase(z) < Phase2) oder
    H(z)=(γ1 < γ(z) < γ2) UND (Amp1 < Amp(z) < Amp2) UND (Gruppenlaufzeit1 < Gruppenlaufzeit(z) < Gruppenlaufzeit2) UND (-Phase,max < Phase(z) < +Phase,max, wobei Phase,max weniger als 180 Grad, bevorzugt etwa 170 Grad ist).
  9. Verfahren nach Anspruch 6, dadurch gekennzeichnet, daß die Filterfunktion H(z) bei jeder Frequenzkomponente ein Produkt stetiger Funktionen der Werte der Kohärenzfunktion, der Amplitudendifferenz, der Phasendifferenz und/oder der Gruppenlaufzeitdifferenz ist, wobei die Parameter dieser Funktionen entsprechend Sätzen von Zielintervallen gewählt werden, welche dem bestimmten der Ausgangssignale entsprechen, das extrahiert werden soll.
  10. Verfahren nach Anspruch 9, dadurch gekennzeichnet, daß die stetigen Funktionen Gaußfunktionen (Normalverteilungsdichtefunktion) der Werte des Quadrats der Kohärenzfunktion, der Amplitudendifferenz, der Phasendifferenz und/oder der Gruppenlaufzeitdifferenz sind, wobei die Parameter dieser Gaußfunktionen (Normalverteilungsdichtefunktion) (Mittel und Streuung) den Sätzen von Zielintervallen entsprechen, welche dem bestimmten der Ausgangssignale entsprechen, das extrahiert werden soll.
  11. Verfahren nach Anspruch 8, 9 oder 10, dadurch gekennzeichnet, daß die Filterfunktion H(z) gebildet wird als ein Produkt einer logischen Funktion H1(z; p) mit Ausgangswerten von 1 oder im wesentlichen 0, je nachdem, ob der Parameter p, welcher die Kohärenzfunktion, die Amplitudendifferenz, die Phasendifferenz und/oder die Gruppenlaufzeitdifferenz sein kann, zu den entsprechenden Zielintervallen gehört, und einer Funktion H2(z; q), die ein Produkt stetiger Funktionen nach Anspruch 10 oder 11 ist, wobei q die restlichen nicht in der Funktion H1 enthaltenen Parameter bezeichnet.
  12. Verfahren nach einem der vorangehenden Ansprüche, dadurch gekennzeichnet, daß die Bestimmung der ersten Restsignale der linken und rechten Kanäle (L', R'), der zweiten Restsignale der linken und rechten Kanäle (L", R") etc. durchgeführt wird, indem die zwei einer schnellen Fourier-Umkehrtransformation unterzogenen (527) Signale (535, 536) jeweils von verzögerten (525) Versionen der linken und rechten Eingangssignale (522, 523) subtrahiert werden, wobei die Eingangssignale (522, 523) im Fall des ersten Ausgangssignals (c1) die Originalsignale der linken und rechten Kanäle (L, R) sind, im Fall des zweiten Ausgangssignals (c2) die ersten Restsignale der linken und rechten Kanäle (L', R') sind, im Fall des dritten Ausgangssignals (c3) die zweiten Restsignale der linken und rechten Kanäle (L", R") sind, etc.
  13. Verfahren nach Anspruch 1, dadurch gekennzeichnet, daß die Vergleiche zwischen Frequenzkomponenten, die einem gegebenen Ausgangssignal (c1, c2, c3 ...) entsprechen, und dem entsprechenden Verarbeitungsblock bei jeder einzelnen Frequenzkomponente der Signale auf der Bestimmung der Kohärenzfunktion (γ), der Amplitudendifferenz (Amp) und der Phasen- oder Gruppenlaufzeitdifferenz (Phase, τ) zwischen den Eingangssignalen (L, R; L', R'; L", R" ...) basieren.
  14. Verfahren nach einem der vorangehenden Ansprüche, dadurch gekennzeichnet, daß die elektroakustischen Wandler Lautsprecher sind.
  15. Vorrichtung zum Umwandeln von zwei Originaleingangssignalen L(t) und R(t), welche die Signale auf dem linken und rechten Kanal eines stereophonischen Signals bilden, in N Ausgangssignale, die N Ausgangskanälen entsprechen, wobei N > 2, wobei die Vorrichtung eine Einrichtung zum Extrahieren der Ausgangssignale (c1, c2, c3 ...) basierend auf dem momentanen Grad der linearen Abhängigkeit zwischen Signalelementen in den zwei Eingangssignalen und unter Nutzung von Bedingungssätzen bezüglich charakteristischer Differenzen zwischen den zwei Eingangssignalen, wobei diese Bedingungen für jedes der Eingangssignale (c1, c2, c3 ...) spezifisch sind und wobei die Vorrichtung außerdem N-2 Blöcke (32, 33, 34; 76, 77, 78) mit jeweils zwei Eingangssignalen aufweist, wobei jeder der Blöcke eines der Ausgangssignale (c1, c2, c3 ...) extrahiert und wobei jeder der Blöcke (32, 33, 34; 76, 77, 78) außerdem zwei Restausgangssignale (L, R; L', R'; L", R" ...) bereitstellt, wobei die Restausgangssignale nicht die skalierten Versionen von Frequenzkomponenten enthalten, die als die Ausgangssignale (c1, c2, c3 ...) extrahiert wurden, dadurch gekennzeichnet, daß die Blöcke (32, 33, 34; 76, 77, 78) derart nacheinander in Reihe geschaltet sind, daß der erste der Blöcke (32; 76) die Originaleingangssignale R(t) und L(t) als Eingangssignale empfängt, ein erstes der Ausgangssignale (c1) extrahiert und ein erstes Paar der Restausgangssignale (L', R') bereitstellt, und der zweite der Blöcke (33; 77) die Restausgangssignale (L', R') als Eingangssignale empfängt, ein zweites der Ausgangssignale (c2) extrahiert und ein zweites Paar der Restausgangssignale (L", R") bereitstellt, und der dritte der Blöcke (34; 78) das zweite Paar der Restausgangssignale (L", R") als Eingangssignale empfängt, ein drittes der Ausgangssignale (c3) extrahiert und ein drittes Paar der Restausgangssignale (L"', R"') bereitstellt, etc., bis ein Maximum von N-2 Ausgangssignalen (c1, c2, c3 ...) extrahiert wurde, und daß das Paar letzter Restausgangssignale (L"', R"'), das nach der Extraktion des letzten der Ausgangssignale (c3) übrig ist, als zwei getrennte Ausgangssignale von dieser Vorrichtung verwendet wird.
  16. Vorrichtung nach Anspruch 15, dadurch gekennzeichnet, daß der Grad der linearen Abhängigkeit zwischen Frequenzkomponenten basierend auf der Bestimmung der Kohärenzfunktion (γ) der Originaleingangssignale L(t) und R(t) und der Bestimmung der Amplitudendifferenz (Amp) zwischen den Originaleingangssignalen L(t) und R(t) und auf der Phasen- oder Gruppenlaufzeitdifferenz (Phase oder τ) zwischen den Originaleingangssignalen L(t) und R(t) bei jeder einzelnen Frequenzkomponente der Signale berechnet wird.
  17. Vorrichtung nach Anspruch 15, dadurch gekennzeichnet, daß der Grad der linearen Abhängigkeit zwischen Frequenzkomponenten in einem bestimmten Analyseblock (73, 74, 75) basierend auf der Bestimmung der Kohärenzfunktion (γ), der Amplitudendifferenz (Amp) und auf der Phasen- oder Gruppenlaufzeitdifferenz (Phase oder τ) zwischen den Eingangssignalen (L, R; L', R'; L", R" ...) zu dem entsprechenden Verarbeitungsblock bei jeder einzelnen Frequenzkomponente der Signale berechnet wird.
  18. Vorrichtung nach Anspruch 16 oder 17, dadurch gekennzeichnet, daß die Vorrichtung eine Einrichtung zur Bestimmung der Kohärenzfunktion (γ), der Amplitudendifferenz (Amp) und der Phasen- oder Gruppenlaufzeitdifferenz (Phase oder τ) basierend auf berechneten Werten für die Autospektra GLL (f) und GRR (f) und auf dem Kreuzspektrum GLR (f) gemäß den folgenden Gleichungen aufweist: γ ( f ) = | G L R ( f ) | G L L ( f ) ⋅ G R R ( f )
    Figure imgb0019
     Amp ( f ) = G L L ( f ) G R R ( f )
    Figure imgb0020
     Phase ( f ) = Winkel ( G L R ( f ) )
    Figure imgb0021
     τ ( f ) = d ( stetige_Phase ( f ) ) 2 π ⋅ d f
    Figure imgb0022
  19. Vorrichtung nach einem der vorangehenden Ansprüche 15 bis 18, dadurch gekennzeichnet, daß Sätze von Bedingungen bezüglich charakteristischer Differenzen zwischen den zwei Eingangssignalen für jeden der zwei Blöcke (32, 33, 34; 76, 77, 78) ein Zielintervall der Kohärenzfunktion (γ), der Amplitudendifferenz (Amp) und der Phasen- oder Gruppenlaufzeitdifferenz (Phase oder τ) aufweisen, wobei die Zielintervalle spezifisch für diesen bestimmten Block sind und wobei die Zielintervalle Funktionen der Frequenz sein können.
  20. Vorrichtung nach einem der vorangehenden Ansprüche 15 bis 19, dadurch gekennzeichnet, daß jeder der Blöcke (32, 33, 34; 76, 77, 78) eine Einrichtung zum Durchführen eines Vergleichs zwischen der Kohärenzfunktion (γ), der Amplitudendifferenz (Amp) und der Phasen- oder Gruppenlaufzeitdifferenz (Phase, τ) mit dem jeweiligen der Zielintervalle, und eine Einrichtung mit der Wirkung aufweist, daß ein spezifisches der Ausgangssignale (c1, c2, c3 ...) nur extrahiert wird, wenn die Kohärenzfunktion (γ), die Amplitudendifferenz (Amp) und die Phasen- oder Gruppenlaufzeitdifferenz (Phase, τ) alle den spezifischen Zielintervallen für dieses spezifische Ausgangssignal (c1, c2, c3 ...) entsprechen.
  21. Vorrichtung nach einem der vorangehenden Ansprüche 15 bis 20, dadurch gekennzeichnet, daß jeder der Blöcke (32, 33, 34; 76, 77, 78) die Extraktion des spezifischen Ausgangssignals (c1, c2, c3 ...) für diesen Block durchführt, indem er in einer geeigneten Multiplikationseinrichtung (526) die einer schnellen Fourier-Transformation unterzogenen Eingangssignale zu diesem spezifischen Block mit einer Filterfunktion H(z) multipliziert, wobei die Filterfunktion für die zwei Eingangssignale in diesen bestimmten Block die gleiche ist, wobei diese Filterfunktion H(z) auf dem genannten Vergleich basiert und die gefilterten Eingangssignale danach an eine schnelle Fourier-Umkehrtransformationseinrichtung (527) geliefert werden und dadurch ein Paar von Signalen (535, 536) bereitgestellt wird, das an eine Additionsvorrichtung (529) geliefert wird, deren Ausgangssignal an eine Verstärkungseinrichtung (530) und danach an eine Verzögerungseinrichtung (531) geliefert wird, deren Ausgangssignal das gewünschte Ausgangssignal (c1, c2, c3 ...) dieses bestimmten Blocks ist.
  22. Vorrichtung nach Anspruch 21, dadurch gekennzeichnet, daß die Filterfunktion H(z) als das Ausgangssignal einer logischen UND-Funktion (517) bereitgestellt wird, wobei dieses Ausgangssignal gemäß dem folgenden Ausdruck:
    H(z)=(γ1 < γ(z) < γ2) UND (Amp1 < Amp(z) < Amp2) UND (Phase1 < Phase(z) < Phase2) oder
    H(z)=(γ1 < γ(z) < γ2) UND (Amp1 < Amp(z) < Amp2) UND (Gruppenlaufzeitl < Gruppenlaufzeit (z) < Gruppenlaufzeit2) UND (-Phase,max < Phase(z) < +Phase,max, wobei Phase,max weniger als 180 Grad, bevorzugt etwa 170 Grad ist) entweder den Wert 1 oder im wesentlichen 0 annimmt.
  23. Vorrichtung nach Anspruch 21, dadurch gekennzeichnet, daß die Filterfunktion H(z) bei jeder Frequenzkomponente ein Produkt stetiger Funktionen der Werte der Kohärenzfunktion, der Amplitudendifferenz, der Phasendifferenz und/oder der Gruppenlaufzeitdifferenz ist, wobei die Parameter dieser Funktionen entsprechend Sätzen von Zielwerten für die Kohärenzfunktion, die Amplitudendifferenz, die Phasendifferenz und/oder die Gruppenlaufzeitdifferenz gewählt werden, welche dem bestimmten der Ausgangssignale entsprechen, das extrahiert werden soll.
  24. Vorrichtung nach Anspruch 23, dadurch gekennzeichnet, daß die stetigen Funktionen Gaußfunktionen (Normalverteilungsdichtefunktion) der Werte des Quadrats der Kohärenzfunktion, der Amplitudendifferenz, der Phasendifferenz und/oder der Gruppenlaufzeitdifferenz sind, wobei die Parameter dieser Gaußfunktionen (Normalverteilungsdichtefunktion) (Mittel und Streuung) den Sätzen von Zielwerten entsprechen, welche dem bestimmten der Ausgangssignale entsprechen, das extrahiert werden soll.
  25. Vorrichtung nach Anspruch 22, 23 oder 24, dadurch gekennzeichnet, daß die Filterfunktion H(z) gebildet wird als ein Produkt einer logischen Funktion H1 (z; p) mit Ausgangswerten von 1 oder im wesentlichen 0, je nachdem, ob der Parameter p, welcher die Kohärenzfunktion, die Amplitudendifferenz, die Phasendifferenz und/oder die Gruppenlaufzeitdifferenz sein kann, zu den entsprechenden Zielintervallen gehört, und einer Funktion H2(z; q), die ein Produkt stetiger Funktionen nach Anspruch 10 oder 11 ist, wobei q die restlichen nicht in der Funktion H1 enthaltenen Parameter bezeichnet.
  26. Vorrichtung nach einem der Ansprüche 15 bis 17, dadurch gekennzeichnet, daß die Restausgangssignale (L', R'; L", R"; L"', R"' ...) in jedem der Blöcke (32, 33, 34; 76, 77, 78) erhalten werden, indem in einer geeigneten Subtraktionseinrichtung (528) die von der schnellen Fourier-Umkehrtransformationseinrichtung (527) unter Verwendung des schnellen Faltungsverfahrens bereitgestellten Ausgangssignale (535, 536) von den Eingangssignalen in diesen bestimmten Block (32, 33, 34; 76, 77, 78) subtrahiert werden, nachdem diese in der Verzögerungseinrichtung (525) verzögert wurden, um die Verarbeitungsverzögerung in der schnellen Fourier-Transformationseinrichtung (524) und in der schnellen Fourier-Umkehrtransformationseinrichtung (525) zu kompensieren.
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DE60028089D1 (de) 2006-06-22
EP1260119A1 (de) 2002-11-27

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