EP1260119B1 - Systeme de reproduction sonore multivoie pour signaux stereophoniques - Google Patents

Systeme de reproduction sonore multivoie pour signaux stereophoniques 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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EP1260119A1 (fr
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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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Abstract

L'invention concerne une reproduction multivoie de signaux sonores, plus spécifiquement, d'une dérivation d'un nombre de signaux sonores de sortie à partir d'une paire de signaux stéréophoniques, tels que chacun de ces signaux de sortie puisse être reproduits via des haut-parleurs placés à l'emplacement de sources fantômes qui auraient été créées par lesdits signaux stéréophoniques si elles avaient été prévues associées à une paire de haut-parleurs dans un montage stéréophonique normal. L'invention consiste donc à remplacer lesdites sources fantômes par des sources sonores physiques réelles qui rendent l'emplacement d'écoute dans un local moins critique que dans un montage stéréophonique normal.

Claims (26)

  1. Procédé pour convertir deux signaux d'entrée L(t) et R(t), constituant les signaux des canaux gauche et droit d'un signal stéréophonique, en N signaux de sortie constituant N canaux de sortie, où N > 2, comprenant les étapes consistant à:
    (A) à partir des signaux originaux des canaux gauche L(t) et droit R(t), destinés aux transducteurs gauche et droit dans un système de reproduction stéréophonique normal à deux enceintes, et à partir d'une comparaison de chaque paire séparée de composantes fréquentielles gauche et droite de ces signaux, fournies, par exemple, par une transformée de Fourier rapide desdits signaux gauche et droit, et de l'application d'un premier ensemble spécifique de spécifications au résultat de ces comparaisons, extraire un premier signal de sortie (c1) sous la forme d'une combinaison linéaire desdits signaux des canaux gauche et droit, à condition que la relation entre lesdites composantes de signaux des canaux gauche et droit soit telle que ceux-ci contribueraient à la formation d'une première source fantôme;
    (B) fournir une paire de premiers signaux résiduels des canaux gauche et droit (L', R'), laquelle paire ne contenant pas ces versions mises à l'échelle des composantes fréquentielles qui ont été extraites à l'étape précédente (A),
    (C) à partir des signaux originaux des canaux gauche L(t) et droit R(t), et à partir, de la même manière que ci-dessus, de la comparaison de chaque composante fréquentielle séparée de ces signaux, à partir de l'application d'un deuxième ensemble spécifique de spécifications au résultat de ces comparaisons, extraire un deuxième signal de sortie (c2), sous la forme d'une combinaison linéaire desdits signaux résiduels des canaux gauche et droit, à condition que la relation entre lesdites composantes de signaux originaux des canaux gauche et droit soit telle que ceux-ci contribueraient à la formation d'une deuxième source fantôme située à une position différente de ladite première source fantôme;
    (D) fournir une paire de deuxièmes signaux résiduels des canaux gauche et droit (L", R"), laquelle paire ne contenant pas ces versions mises à l'échelle de composantes fréquentielles qui ont été extraites aux étapes précédentes (A) et (C);
    (E) répéter les étapes précédentes un nombre de fois suffisant et, à chaque fois, avec des ensembles de spécifications différents, afin de pouvoir extraire un maximum de N-2 signaux de sortie (c1, c2, c3, c4...) correspondant à N-2 sources fantômes, lesquelles peuvent être constituées des signaux originaux des canaux gauche L(t) et droit R(t);
    (F) fournir une paire de signaux résiduels finaux des canaux gauche et droit (L"', R"'), laquelle paire ne contenant pas ces versions mises à l'échelle de composantes fréquentielles qui ont été extraites à l'une quelconque des étapes précédentes;
    (G) fournir lesdits premier, deuxième, etc., signaux de sortie (c1, c2, c3...) à N-2 transducteurs électroacoustiques (36, 710; 37, 711; 38, 712), la position de chacun de ces transducteurs correspondant à l'ensemble particulier de spécifications utilisé au niveau de l'extraction du signal de sortie (c1, c2, c3...) pour ce transducteur particulier;
    (H) fournir ledit signal résiduel final du canal gauche résiduel final (L'") à un transducteur électroacoustique (35, 79) placé à la gauche de tous les N-2 autres transducteurs, et fournir ledit signal résiduel final du canal droit (R"') à un transducteur électroacoustique (39, 713) placé à la droite de tous les N-2 autres transducteurs.
  2. Procédé selon la revendication 1, caractérisé en ce que ladite comparaison du signal original du canal droit L(t) et du signal original du canal droit R(t) comprend la détermination au niveau de chaque composante fréquentielle de la fonction de cohérence (γ) desdits signaux originaux L(t) et R(t), la différence d'amplitude (amp) entre lesdits signaux originaux L(t) et R(t) et la différence de phase (ou de temps de propagation de groupe) (phase ou τ) entre lesdits signaux originaux L(t) et R(t).
  3. Procédé selon la revendication 2, caractérisé en ce que ladite fonction de cohérence (γ), ladite différence d'amplitude (amp) et ladite différence de phase, ou de temps de propagation de groupe, sont fonctions de la fréquence, et sont calculées à partir du spectre croisé GLR(f) et des deux auto-spectres GLL(f) et GRR(f), selon les équations suivantes: γ ( f ) = | G L R ( f ) | G L L ( f ) ⋅ G R R ( f )
    Figure imgb0023
     amp ( f ) = G L L ( f ) G R R ( f )
    Figure imgb0024
     phase ( f ) = angle ( G L R ( f ) )
    Figure imgb0025
     τ ( f ) = − d ( phase_continue ( f ) ) 2 π ⋅ d f
    Figure imgb0026
  4. Procédé selon l'une quelconque des revendications précédentes, caractérisé en ce que lesdits ensembles de spécifications comprennent chacun un intervalle cible de ladite fonction de cohérence (γ), un intervalle cible de ladite différence d'amplitude (amp), et un intervalle cible de ladite différence de phase ou de temps de propagation de groupe (phase, τ), lesquels intervalles cibles peuvent être fonctions de la fréquence.
  5. Procédé selon l'une quelconque des revendications précédentes, caractérisé en ce que ladite extraction de signaux de sortie (c1, c2, c3...) repose sur une comparaison au niveau de chaque composante fréquentielle de ladite fonction de cohérence (γ), de ladite différence d'amplitude (amp) et de ladite différence de phase ou de temps de propagation de groupe (phase ou τ), avec celui respectif desdits intervalles cibles, de telle sorte que celui spécifique desdits signaux de sortie (c1, c2, c3...) est uniquement extrait si ladite fonction de cohérence (γ), ladite différence d'amplitude (amp) et ladite différence de phase ou de temps de propagation de groupe (phase ou τ) correspondent tous aux intervalles cibles spécifiques à ce signal de sortie spécifique (c1, c2, c3...).
  6. Procédé selon l'une quelconque des revendications précédentes, caractérisé en ce que ladite extraction de l'un donné desdits signaux de sortie (c1, c2, c3...) est réalisée au moyen, par exemple, de transformées de Fourier rapides d'une paire de signaux d'entrée donnée (L, R; L', R'; L", R"...), où ladite paire donnée de signaux d'entrée, dans le cas du premier signal de sortie extrait (c1), est composée des signaux originaux des canaux gauche et droit (L, R), dans le cas du deuxième signal de sortie extrait (c2), est composée des signaux résiduels des canaux gauche et droit (L', R'), dans le cas du troisième signal de sortie extrait (c3), est composée des deuxième signaux résiduels des canaux gauche et droit (L", R"), etc., où lesdites transformées de Fourier rapides d'une paire de signaux d'entrée donnée sont multipliées par des fonctions de filtre égalisateur H(z) formées par ladite comparaison de la fonction de cohérence déterminée (γ), de la différence d'amplitude déterminée (amp) et de la différence de phase ou de temps de propagation de groupe déterminé(e) (phase ou τ), avec lesdites valeurs cible de celle-ci, correspondant à celui particulier desdits signaux de sortie (c1, c2, c3...) qui doit être extrait, et où les versions multipliées desdites transformées de Fourier rapides sont soumise à une transformation de Fourier rapide inverse (527), de sorte qu'après chaque calibrage individuel approprié de celles-ci, les deux signaux résultants dans le domaine temporel (535, 536) peuvent finalement être additionnés (529) pour constituer une première version (c1', c2', c3'...) de ce signal de sortie particulier (c1, c2, c3...) sous la forme d'une combinaison linéaire de ladite paire donnée de signaux d'entrée, où lesdites étapes de transformée de Fourier rapide, la multiplication et la transformée de Fourier inverse sont des étapes procédurales de la méthode connue sous le nom de CONVOLUTION RAPIDE, par exemple.
  7. Procédé selon la revendication 6, caractérisé en ce que lesdits signaux de sortie (c1, c2, c3,...) sont formés par amplification (530) suivie par un retard postérieur (531) de ladite première version (c1', c2', c3'...) des signaux de sortie (c1, c2, c3...).
  8. Procédé selon la revendication 6, caractérisé en ce que ladite fonction de filtrage H(z) est une fonction ET logique, c'est-à-dire une fonction avec des valeurs de sortie à 1 ou sensiblement à 0, obtenues par comparaison au niveau de chaque composante fréquentielle de ladite fonction de cohérence, ladite différence d'amplitude et ladite différence de phase ou de temps de propagation de groupe avec les intervalles cibles correspondants correspondant à celui particulier desdits signaux de sortie à dériver, où H(z) est donné soit par l'équation:
    H(z) = (γ1 < γ(z) < γ2) ET (amp1 < amp (z) < amp2) ET (phase1 < phase (z) < phase2) ou par l'équation:
    H(z) = (γ1 < γ(z) < γ2) ET (amp1 < amp(z) < amp2) ET (temps de propagation de groupe1 < temps de propagation de groupe(z) < temps de propagation de groupe 2) ET (-phase,max < phase(z) < +phase,max, où phase,max est inférieur à 180 degrés, de préférence approximativement à 170 degrés).
  9. Procédé selon la revendication 6, caractérisé en ce que ladite fonction de filtrage H(z) est un produit au niveau de chaque composante fréquentielle de fonctions continues des valeurs de la fonction de cohérence, de la différence d'amplitude, de la différence de phase et/ou de la différence de temps de propagation de groupe, où les paramètres de ces fonctions sont choisis en fonction des ensembles d'intervalles cibles correspondant à celui particulier desdits signaux de sortie à extraire.
  10. Procédé selon la revendication 9, caractérisé en ce que lesdites fonctions continues sont des fonctions gaussiennes (fonction de densité de distribution normale) des valeurs du carré de la fonction de cohérence, de la différence d'amplitude, de la différence de phase et/ou de la différence de temps de propagation de groupe, où les paramètres de ces fonctions gaussiennes (fonction de densité de distribution normale) (moyennes et variances) correspondent aux ensembles d'intervalles cibles correspondant à celui particulier desdits signaux de sortie à extraire.
  11. Procédé selon la revendication 8, 9 ou 10, caractérisé en ce que ladite - fonction de filtrage H(z) est formée sous la forme d'un produit d'une fonction logique H1(z; p) avec des valeurs de sortie à 1 ou sensiblement à 0 selon que les paramètres p, qui peuvent être la fonction de cohérence, la différence d'amplitude, la différence de phase et/ou de temps de propagation de groupe, appartiennent aux intervalles cibles correspondants, et une fonction H2(z;q) qui est le produit de fonctions continues selon la revendication 10 ou 11, où q désigne les paramètres restant non contenus dans ladite fonction H1.
  12. Procédé selon l'une quelconque des revendications précédentes, caractérisé en ce que la détermination desdits premiers signaux résiduels des canaux gauche et droit (L', R'), des deuxièmes signaux résiduels des canaux gauche et droit (L", R"), etc., est exécutée par soustraction (528) desdits deux signaux (535, 536) soumis à une transformation de Fourier rapide inverse (527), respectivement à partir des versions retardées (525) des signaux d'entrée gauche et droit (522, 523), lesquels signaux d'entrée (522, 523) dans le cas du premier signal de sortie (c1) sont les signaux originaux des canaux gauche et droit (L, R), dans le cas du deuxième signal de sortie (c2) sont les premiers signaux résiduels des canaux gauche et droit (L', R'), dans le cas du troisième signal de sortie (c3) sont les deuxièmes signaux résiduels des canaux gauche et droit (L", R"), etc.
  13. Procédé selon la revendication 1, caractérisé en ce que lesdites comparaisons entre des composantes fréquentielles correspondant à un signal de sortie donné (c1, c2, c3...), reposent sur la détermination de la fonction de cohérence (γ), sur la différence d'amplitude (amp) et sur la différence de phase ou de temps de propagation de groupe (phase ou τ) entre les signaux d'entrée (L, R; L', R'; L" R"...) pour le bloc de traitement correspondant au niveau de chaque composante fréquentielle séparée des signaux.
  14. Procédé selon l'une quelconque des revendications précédentes, caractérisé en ce que lesdits transducteurs électroacoustiques sont des enceintes.
  15. Dispositif pour convertir deux signaux d'entrée originaux L(t) et R(t) constituant les signaux dans les canaux gauche et droit d'un signal stéréophonique, en N signaux de sortie correspondant à N canaux de sortie, où N > 2, où ledit dispositif comprend un moyen d'extraction desdits signaux de sortie (c1, c2, c3...) reposant sur le degré instantané de dépendance linéaire entre des éléments de signaux dans lesdits deux signaux d'entrée, et utilisant des ensembles de spécifications concernant les différences caractéristiques entre lesdits deux signaux d'entrée, lesdites spécifications étant spécifiques à chacun desdits signaux de sortie (c1, c2, c3...), et où ledit dispositif comprend, en outre, N-2 blocs (32, 33, 34; 76, 77, 78) chacun avec deux signaux d'entrée, où chacun desdits blocs extrait l'un desdits signaux de sortie (c1, c2, c3...), et où chacun desdits blocs (32, 33, 34; 76, 77, 78) fournit, en outre, deux signaux résiduels de sortie (L', R', L", R"; L"', R"'...), lesquels signaux résiduels de sortie ne contiennent pas ces versions mises à l'échelle des composantes fréquentielles, qui ont été extraites en tant que dits signaux de sortie (c1, c2, c3...), caractérisé en ce que lesdits blocs (32, 33, 34; 76, 77, 78) sont couplés en série l'un après l'autre, de telle sorte que le premier desdits blocs (32; 76), en tant que signaux d'entrée, reçoit lesdits signaux d'entrée originaux L(t) et R(t), extrait un premier desdits signaux de sortie (c1), et fournit une première paire desdits signaux résiduels de sortie (L', R'), et le deuxième bloc desdits blocs (33; 77), en tant que signaux d'entrée, reçoit lesdits signaux résiduels de sortie (L', R'), extrait un deuxième signal desdits signaux de sortie (c2), et fournit une deuxième paire de signaux résiduels de sortie (L", R"), et le troisième bloc desdits blocs (34; 78), en tant que signaux d'entrée, reçoit ladite deuxième paire de signaux résiduels de sortie (L", R"), extrait un troisième signal desdits signaux de sortie (c3), et fournit une troisième paire de signaux résiduels de sortie (L"', R"'), etc., jusqu'à ce qu'un maximum de N-2 signaux de sortie (c1, c2, c3...) aient été extraits, et que la paire de signaux résiduels finaux de sortie (L"', R"'), qui restent après l'extraction de celui final desdits signaux de sortie (c3) sont utilisé en tant que deux signaux de sortie séparés dudit dispositif.
  16. Dispositif selon les revendications 15, caractérisé en ce que ledit degré de dépendance linéaire entre composantes fréquentielles est évalué en fonction de la détermination de la fonction de cohérence (γ) desdits signaux d'entrée originaux L(t) et R(t) et de la détermination de la différence d'amplitude (amp) entre lesdits signaux d'entrée originaux L(t) et R(t) et de la différence de phase, ou de temps de propagation de groupe (phase ou τ), entre lesdits signaux d'entrée originaux L(t) et R(t) au niveau de chaque composante fréquentielle séparée des signaux.
  17. Dispositif selon la revendication 15, caractérisé en ce que ledit degré de dépendance linéaire entre composantes fréquentielles dans un bloc d'analyse particulier (73, 74, 75) est évalué en fonction de la détermination de la fonction de cohérence (γ), de la différence d'amplitude (amp) et de la différence de phase ou de temps de propagation de groupe (phase ou τ) entre les signaux d'entrée (L, R; L', R'; L", R"...) pour le bloc de traitement correspondant au niveau de chaque composante fréquentielle séparée des signaux.
  18. Dispositif selon la revendication 16 ou 17, caractérisé en ce que ledit dispositif comprend un moyen de détermination de ladite fonction de cohérence (γ), de ladite différence d'amplitude (amp) et de ladite différence de phase ou de temps de propagation de groupe (phase ou τ) en fonction des valeurs calculées des auto-spectres GLL(f) et GRR(f) et du spectre croisé GLR(f) selon les équations suivantes: γ ( f ) = | G L R ( f ) | G L L ( f ) ⋅ G R R ( f )
    Figure imgb0027
     amp ( f ) = G L L ( f ) G R R ( f )
    Figure imgb0028
     phase ( f ) = angle ( G L R ( f ) )
    Figure imgb0029
     τ ( f ) = − d ( phase_continue ( f ) ) 2 π ⋅ d f
    Figure imgb0030
  19. Dispositif selon l'une quelconque des revendications précédentes 15 à à 18, caractérisé en ce que lesdits ensembles de spécifications concernant les différences caractéristiques entre lesdits deux signaux d'entrée pour chacun desdits blocs (32, 33, 34; 76, 77, 78) comprennent un intervalle cible de ladite fonction de cohérence (γ), de ladite différence d'amplitude (amp) et de ladite différence de phase ou de temps de propagation de groupe (phase ou τ), lesquels intervalles cibles sont spécifiques à ce bloc particulier, et lesquels intervalles cibles peuvent être fonctions de la fréquence.
  20. Dispositif selon l'une quelconque des revendications précédentes 15 à 19, caractérisé en ce que chacun desdits blocs (32, 33, 34; 76, 77, 78) comprend un moyen pour obtenir une comparaison entre ladite fonction de cohérence (γ), ladite différence d'amplitude (amp) et ladite différence de phase ou de temps de propagation de groupe (phase ou τ), avec celui respectif desdits intervalles cibles, et un moyen ayant pour effet que celui spécifique desdits signaux de sortie (c1, c2, c3...) est uniquement extrait si ladite fonction de cohérence (γ), ladite différence d'amplitude (amp) et ladite différence de phase ou de temps de propagation de groupe (phase ou τ) correspondent tous aux intervalles cibles spécifiques à ce signal de sortie spécifique (c1, c2, c3...).
  21. Dispositif selon l'une quelconque des revendications précédentes 15 à 20, caractérisé en ce que chacun desdits blocs (32, 33, 34; 76, 77, 78) effectue l'extraction du signal de sortie spécifique (c1, c2, c3...) pour ce bloc à l'aide d'une multiplication, par une fonction de filtrage H(z), dans le moyen multiplicateur (526) approprié, des signaux d'entrée de ce bloc spécifique sur lesquels a été appliquée la transformée de Fourier rapide, laquelle fonction de filtrage est la même pour lesdits deux signaux d'entrée pour ce bloc particulier, laquelle fonction de filtrage H(z) repose sur ladite comparaison et fournissant, par la suite, lesdits signaux d'entrée filtrés à un moyen de transformée de Fourier rapide inversée (527), et fournissant par conséquent une paire de signaux (535, 536), lesquels sont fournis à un moyen additionneur (529), dont le signal de sortie est fourni à un moyen de gain (530) et, par la suite, à un moyen retardateur (531), dont le signal de sortie est le signal de sortie souhaité (c1, c2, c3...) pour ce bloc particulier.
  22. Dispositif selon la revendication 21, caractérisé en ce que ladite fonction de filtrage H(z) est prévue comme signal de sortie d'un moyen de ET logique (517), ce signal de sortie prenant soit la valeur 1 soit, sensiblement, la valeur 0 en fonction de l'expression suivante:
    H(z) = (γt < γ(z) < γ2) ET (amp1 < amp(z) < amp2) ET (phase 1 < phase(z) < phase2) ou par
    H(z) = (γ1 < γ(z) < γ2) ET (amp1 < amp(z) < amp2) ET (temps de propagation de groupe1 < temps de propagation de groupe(z) < temps de propagation de groupe 2), ET (-phase,max < phase(z) < +phase, max, où phase,max est inférieur à 180 degrés, de préférence, approximativement 170 degrés).
  23. Dispositif selon la revendication 21 caractérisé en ce que ladite fonction de filtrage H(z) est un produit, au niveau de chaque composante fréquentielle, de fonctions continues des valeurs de la fonction de cohérence, de la différence d'amplitude, de la différence de phase et/ou de différence de temps de propagation de groupe, où les paramètres de ces fonctions sont choisis en fonction des ensembles de valeurs cibles pour ladite fonction de cohérence, ladite différence d'amplitude, ladite différence de phase et/ou différence de temps de propagation de groupe correspondant à celui en particulier desdits signaux de sortie à extraire.
  24. Dispositif selon la revendication 23, caractérisé en ce que lesdites fonctions continues sont des fonctions gaussiennes (fonctions de densité de distribution normale) des valeurs du carré de la fonction de cohérence, de la différence d'amplitude, de la différence de phase et/ou de différence de temps de propagation de groupe, où les paramètres de ces fonctions gaussiennes (fonctions de densité de distribution normale) (moyennes et variances) correspondent aux ensembles de valeurs cibles correspondant à celui particulier desdits signaux de sortie à extraire.
  25. Dispositif selon la revendication 22, 23 ou 24, caractérisé en ce que ladite fonction de filtrage H(z) est formée sous la forme d'un produit d'une fonction logique H1(z; p) avec des valeurs de sortie à 1 ou à sensiblement 0, selon que les paramètres p, qui peuvent être la fonction de cohérence, la différence d'amplitude, la différence de phase et/ou de temps de propagation de groupe, appartiennent aux intervalles cibles correspondants, et une fonction H2(z; q) qui est un produit des fonctions continues selon la revendication 10 ou 11, où q désigne les paramètres restants non contenus dans ladite fonction H1.
  26. Dispositif selon l'une quelconque des revendications 15 à 17, caractérisé en ce que lesdits signaux résiduels de sortie (L', R'; L", R"; L"', R"';...) dans chacun desdits blocs (32, 33, 34; 76, 77, 78) sont obtenus par soustraction, dans un moyen soustracteur (528) approprié, desdits signaux de sortie (535, 536) fournis à partir dudit moyen de transformation de Fourier rapide inverse (527), utilisant ladite méthode de CONVOLUTION RAPIDE, à partir des signaux d'entrée pour ce bloc particulier (32, 33, 34; 76, 77, 78) après que ceux-ci ont été retardés dans un moyen retardateur (525) pour compenser le retard de traitement dans ledit moyen de - transformation de Fourier rapide (524) et dans ledit moyen de transformation de Fourier rapide inverse (525).
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AU2000226583A1 (en) 2001-08-27
WO2001062045A1 (fr) 2001-08-23
EP1260119A1 (fr) 2002-11-27
DE60028089D1 (de) 2006-06-22

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