EP0559530A1 - Verfahren und Vorrichtung für künstliche Raumklangeffekte von digitalen Audiosignalen - Google Patents

Verfahren und Vorrichtung für künstliche Raumklangeffekte von digitalen Audiosignalen Download PDF

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EP0559530A1
EP0559530A1 EP93400503A EP93400503A EP0559530A1 EP 0559530 A1 EP0559530 A1 EP 0559530A1 EP 93400503 A EP93400503 A EP 93400503A EP 93400503 A EP93400503 A EP 93400503A EP 0559530 A1 EP0559530 A1 EP 0559530A1
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delay
signals
delayed
module
elementary
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EP93400503A
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French (fr)
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EP0559530B1 (de
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Jean-Marc Jot
Antoine Chaigne
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Orange SA
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France Telecom SA
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    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04S—STEREOPHONIC SYSTEMS 
    • H04S7/00—Indicating arrangements; Control arrangements, e.g. balance control
    • H04S7/30—Control circuits for electronic adaptation of the sound field
    • H04S7/305—Electronic adaptation of stereophonic audio signals to reverberation of the listening space
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04S—STEREOPHONIC SYSTEMS 
    • H04S1/00—Two-channel systems
    • H04S1/007—Two-channel systems in which the audio signals are in digital form

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  • the present invention relates to a method and a system for artificial spatialization of audio-digital signals.
  • Artificial reverberators are used in the music and film industry to superimpose a room effect on recordings made in the studio, or even to modify the acoustic properties of a listening room.
  • the response to an impulse sound excitation from a listening room shows, as shown in FIG. La, that the typical echogram includes the direct sound followed by the first echoes or early echoes temporally locatable by ear, then finally a continuum perceived on the contrary as a sound trail.
  • This sound trail called late reverberation is characteristic of the listening room itself, because it is, as a first approximation, independent of the relative positions and the extent of the sources and listeners, which is not the case for the first echoes.
  • a reverberator usually comprises, as shown in FIG. 1b, an FIR filter (digital filter with finite impulse response) simulating the first echoes, and a reverberant filter, formed by a recursive network of digital delays and capable of reproducing the characteristic properties of late reverberation.
  • FIR filter digital filter with finite impulse response
  • the basic elementary structures of the majority of commercial reverberators consist in the use of filters, known as comb filters and all-pass filters. These filters are widely known in the state of the art.
  • the comb filter has a drawback, in the frequency domain, coming from the periodicities of its spectral response resulting in a coloration perceived as a metallic stamp. The same applies to the all-pass filter when the input signal is not stationary, as in the case of speech and music signals.
  • the two aforementioned filters also have the drawback, in the time domain, of having a low density of echoes of their impulse response, which causes the phenomenon known as rolling (flutter) in the transients.
  • MR SCHROEDER proposed to use in cascade a parallel association of comb filters, called sum of combs, and a series association of all-pass filters, as shown in Figure 1c, confer publication "Natural sounding artificial reverberation", J. Audio.Eng.Soc. 10 (3): 219-223, 1962.
  • the assignment to each comb of the same reverberation time Tr implies a choice of the loop gain gi linked to the duration of the delay mi.
  • M.R. SCHROEDER proposed to associate a serial all-pass filter in cascade with the sum of combs.
  • the all-pass filter makes it possible to increase the density of echoes without appreciably modifying the timbre of the reverberation, defined by the comb filters associated in parallel.
  • Another object of the present invention is also a method and a system for artificial spatialization of an audio-digital signal making it possible to satisfy both the criteria of modal density in the spectral domain and of temporal density of the echoes.
  • Another object of the present invention is to allow, both at the level of the method and of the artificial spatialization system object of the invention, a separate control of the reverberation time, of the spectral envelope of the response of the simulated hearing room and the modal density, in fact reflecting the size of the simulated hearing room.
  • Another object of the present invention is also a method and a system for artificial spatialization of an audio-digital signal, both mono and stereophonic, allowing in the latter case a control of the directions of origin of the early echoes.
  • Another object of the present invention is finally a method and a system of artificial spatialization simultaneous of several sources, with controls of each early echo and clarity for each of the sources.
  • the method and the system of artificial spatialization in real time of an audio-digital signal x (k) to generate a spatialized audio-digital signal y (k) consists in, respectively allows to carry out, starting from elementary signals xi (k ) replicates the digital audio signal a plurality of different delays to generate a plurality of delayed elementary signals and a linear combination between the delayed elementary signals to obtain a plurality of combined delayed elementary signals, at least one of each of the combined delayed elementary signals being added to at least one elementary signal xi (k) prior to the delay thereof.
  • the delayed elementary signals are subjected to a weighted summation with the audio-digital signal x (k) to generate the spatialized audio-digital signal y (k).
  • the audio-digital signal is denoted x (k), this signal consisting of a series of samples of an encoded audio-digital signal.
  • the audio-digital signal x (k) is duplicated into elementary signals xi (k) obtained from the audio-digital signal by corresponding weighting bi adapted.
  • the elementary signals xi (k) are each subjected to a different delay to generate a plurality of delayed elementary signals.
  • Each delayed elementary signal is denoted seri and corresponds to the elementary signal xi (k) considered.
  • a linear combination between the delayed elementary signals, seri is carried out in order to obtain a plurality of combined delayed elementary signals, denoted serci .
  • serci a plurality of combined delayed elementary signals
  • the weighted summation is, a first part, represented by the application to each delayed elementary signal seri a corresponding weighting coefficient, noted above, and summation of all the delayed elementary signals, seri , and, on the other hand, summation of the set to the weighted audio-digital signal x (k) to which the weighting coefficient d has been applied to generate the spatialized audio-digital signal y (k).
  • the method which is the subject of the present invention consists, in order to simulate a phenomenon of late reverberation, in accordance with a particularly advantageous aspect thereof, in effecting the above-mentioned linear combination by unitary looping.
  • unitary looping means a looping for which the plurality of combined delayed elementary signals, serci , has the same energy as the delayed elementary signals, seri , ie ⁇ seri 2 - ⁇ serci 2.
  • the method which is the subject of the invention consists in carrying out, with each different delay, an attenuation, denoted Hi ( ⁇ ), of the delayed elementary signal, seri , this attenuation being a function of the audio pulse ⁇ above.
  • this attenuation is a decreasing monotonic function of the reverberation time Tr ( ⁇ ) the simulation of which is desired and proportional to each delay.
  • ⁇ i defined as the absorbing delay
  • ⁇ i actually designates the value of each delay increased by the phase delay provided by the corresponding attenuation Hi ( ⁇ ), ⁇ i denoting the sum of all the absorbing delays.
  • This phase delay is in fact negligible compared to the value of each delay and will therefore be considered as such in the following description.
  • z ⁇ 1 represents the unit delay operator and D (z) is defined by:
  • a first constraint can be imposed, that is to say an identical decay time for all the resonance modes.
  • the transfer matrix A is unitary, that is to say in the case where the plurality of combined delayed elementary signals, serci , has the same energy as the delayed elementary signals, seri , all the aforementioned poles are on the unit circle of the complex plane.
  • the modulus of each of the poles then being equal to unity, the decay time is infinite for each of the associated resonance modes, and the impulse response can be represented by a sum of unshocked sinusoids.
  • the modal density is always equal to the total duration of the delays.
  • the process which is the subject of the present invention then consists in varying the reverberation time, while respecting the identical modulus constraint for all the poles. Such a variation is obtained by assigning an attenuation ki to each of the previously mentioned delays.
  • T is the audio digital signal sampling period, ⁇ being expressed in dB.
  • the equality constraint of the pole module is respected when, from a reference filter, as defined above, an attenuation is assigned to each delay, which is proportional to the duration of it.
  • the proportionality factor ⁇ is linked to the reverberation time Tr by equation (6) previously mentioned.
  • the method which is the subject of the present invention thus allows the control of the simulated reverberation time, this control being valid whatever the structure of the reference filter, and also guarantees the absence of parasitic colorations in the presence of transient signals.
  • the insertion of absorbent filters has the effect of modifying the spectral envelope of the response finally obtained, because, confer [JOT, CHAIGNE, 91], the energy of each resonance mode is proportional to the decay time thereof.
  • the spectral balance of the response thus obtained is obtained by the spectral correction t (z), this spectral correction being inversely proportional to the reverberation time Tr ( ⁇ ) in the frequency range of the digital audio signal processed.
  • the impulse response of the process which is the subject of the present invention is temporally dilated by a homothety. of ratio ⁇ , but the average energy of the reverberated signal, in any given frequency band, is not modified.
  • Such a multiplication in fact simulates a homothety of ratio ⁇ on the dimensions of the simulated hearing room, and has the effect of modifying the frequencies resonance while multiplying the reverberation time by ⁇ at any frequency. Dividing the reverberation time by ⁇ to bring it back to the initial situation has the effect of dividing the energy of the spatialized signal by the same quantity ⁇ .
  • the correction spectral t (z) checks the relation (2) previously mentioned in the description.
  • the loopback transfer matrices verifying the above-mentioned relation (8) thus make it possible to obtain a maximum echo density for a given number N of delays with, however, a minimal calculation cost, that is to say 2.N additions -multiplications as shown in Figure 2c.
  • the total duration of the delays being fixed by the size of the room for which the reverberation must be simulated, the number N of delays determines the time necessary for the temporal density of the echoes to be built in the impulse response.
  • each elementary signal seri delayed is reduced by the sum weighted by the ratio 2 / N of the delayed elementary signals.
  • each elementary signal xi (k) is added for example to a delayed elementary signal, seri , the resulting sum being subjected to the corresponding delay ⁇ i, absorbing delay, and all the delayed elementary signals being summed for give the sum of the delayed elementary signals, this sum being reinjected after weighting by the coefficient -2 / N to the audio-digital input signal x (k).
  • this consists in carrying out a time shift t1, ti, tN, of the instants of arrival at the level of the looping of the elementary signals, this temporal offset of the instants of arrival thus having the effect of causing a separation of the elementary signals due to the aforementioned offset.
  • the elementary signals denoted for example xi (k) are then shifted in time by the difference of two successive shift instants.
  • the method which is the subject of the present invention as shown in FIG. 2d consists, with the elementary signals xi (k) now being offset, choosing an offset difference between the largest and the smallest of the arrival instants, symbolized by t1 and tN in FIG. 2d, less than the smallest value of the absorbent delays ⁇ i previously mentioned.
  • Controlling the clarity and direction of source of echoes from monophonic sources in such a situation is particularly advantageous, in particular in the case where these monophonic sources are other than source elements of the corresponding stereophonic recording, ie that is to say that the aforementioned monophonic sources are elements of the source of the stereophonic signals subjected to the spatialization process in accordance with the object of the present invention.
  • Such a situation may be encountered, in particular, during the recording or retransmission of a stereophonic recording of a concert given by a symphonic orchestra in which one or more concerting instruments, and in particular the playing of these, want to be highlighted.
  • the shifted elementary monophonic signals are then injected into the looping applied to the stereophonic signals subjected to the reverberation process simulated by summing, before looping, to the delayed elementary stereophonic signals.
  • the system which is the subject of the present invention comprises delay channels, denoted Vi, each consisting for example successively of a multiplier element, denoted 1i, a summing element, 2i, a retarding element , 3i, and a multiplier element, 5i, in cascade, each delay channel being connected to a summing element, denoted 6i, bearing the reference of the index of the corresponding delay channel, except possibly with regard to the delay of order 1, V1.
  • the audio-digital signal x (k) is thus duplicated into elementary signals xi (k) feeding each delay channel, Vi, and a summing element 9 allows, after weighting of the audio-digital signal, x (k), by an element multiplier 8 to deliver the spatialized signal y (k), the summing element further receiving the weighted sum of the delayed elementary signals, seri , delivered by each delay channel, Vi, this weighted sum being further submitted, via from the spectral correction element 7, to a spectral correction verifying the relation (2) previously mentioned in the description.
  • each delay element, 3i contained in each delay channel, Vi, there is associated an absorbent element, denoted 4i, whose transfer function causes a Hi attenuation ( ⁇ ) of each delayed elementary signal, this attenuation being a monotonic decreasing function of the reverberation time Tr ( ⁇ ) and proportional to each delay generated by each corresponding delay element 3i.
  • the artificial spatialization system which is the subject of the present invention as shown in FIG. 3a constitutes a reverberant filter formed by a reference filter, as mentioned previously in the description, in which inserted, for each attenuation channel Vi, an attenuation function by the element 4i, under the conditions of relation to the reverberation time Tr (..) and to the delay, noted z -mi , as previously mentioned in the description.
  • the reference filter is entirely characterized by the durations of the delays z -mi , the coefficients bi, ci having been defined, which can be chosen to be irrational with one another so as to avoid overlapping echoes, and such that their sum is proportional to a dimension characteristic of the phenomenon of the room to be simulated.
  • loop transfer transfer matrices thus retained make it possible to produce loopings which are characterized by the fact that the input of each delay, that is to say each summing element 2i, receives the output signal of another delay, by a bijective correspondence, reduced by the sum multiplied by 2 / N of the output signals of the N delays.
  • This class of loopback matrices and the corresponding loopbacks make it possible to maximize the echo density, and are in fact distinguished from each other only by the choice of the matrix JN in the above-mentioned relation (8).
  • the monophonic reverberators as shown in FIG. 3b and 3c can, if necessary, cause a parasitic echo whose arrival date corresponds to the sum of the durations of the absorbent delays ⁇ i.
  • the amplitude of this parasitic echo decreases when the number N of delays increases and this echo merges into the reverberation when N> 12.
  • this parasitic echo is not present at the output of each of the N absorbent delays 34i, but arises from the interference between these signals.
  • FIGS. 3d and 3e allow the abovementioned interference phenomenon to be eliminated, by splitting and placing in phase opposition, at the input or at the output of the reverberant filter of the input, respectively output, split signals.
  • the elementary signals are split into elementary signals of odd rank x2p-1 (k), even x2p (k), and put in phase opposition by means of a first summing element, 22a, respectively second subtractor element, 22b, corresponding, the corresponding delayed elementary signals being of course summed by the corresponding summing elements 6i and the reinjection weighted by the multiplier element 23 being carried out at the first, 22a, respectively second, 22b, summing element , respectively subtractor.
  • the summation of the above-mentioned signals of even rank, respectively odd, respectively, is carried out by the summing elements 6 1a , of odd rank, respectively 6 2a of even rank, and the loopback is carried out by means of an additional summing output element split 6 1b , respectively 6 2b , the summing element 6 1b receiving the signals delivered by the summing element 6 1a , respectively 6 2a , and delivering the sum signal to the multiplying element 23, while the subtracting element 6 2b receives the signals delivered by the summing element 6 1a , respectively 6 2a , and delivers the spatialized audio-digital signal y (k).
  • the system which is the subject of the present invention makes it possible to avoid any phenomenon of coloration of the reverberated signal.
  • the system which is the subject of the present invention comprises a module for processing the first echoes, noted 20, and the reverberant filter proper, noted 30, which corresponds substantially to the reverberant filter. shown in Figure 3a.
  • the module of the first echoes 20 makes it possible to control the instants of arrival ti independently of the delay times of the reverberant filter proper.
  • the role of the coefficients bi of the multiplier elements 1i of the first echo module 20 is slightly modified compared to the case of FIG. 3a.
  • the absorbent delay values ⁇ i caused by the absorbent delay elements 34i can then be chosen taking into account the values ti of the arrival times as already mentioned in connection with FIG. 2d.
  • the reference filters of FIGS. 3a and 4 are strictly equivalent, but in the presence of the attenuation elements 4i, the two systems differ in the fact that in FIG.
  • system which is the subject of the present invention is not limited to the sole processing of monophonic audio-digital signals.
  • the reverberant filter proper 30 of FIG. 5a is arranged so that it comprises a plurality of N delay channels, divided into N / 2 delay channels relative to the left channel, and making it possible to generate successively N / 2 left elementary signals, denoted xi (k) g, then analogously to the reverberant filter shown in FIG. 3a or 4, N / 2 left delayed elementary signals, seri g.
  • each echo module synthesizes N / 2 stereophonic echoes of which the amplitude, the date of arrival and the direction of provenance are controlled.
  • the direction of origin of each echo is defined by the time and energy difference between the left and right channels.
  • the first echo assigned to each source plays the role of direct sound for this source.
  • gain adjustment multiplier elements g bear the reference 24d, 24g, these elements allowing adjustment of the corresponding gain, in order to avoid possible saturation phenomena.
  • FIG. 5c there is shown the system object of the present invention in which the looping of the reverberant filter itself is carried out, for example, as shown in Figure 3c, the subdivision between channel delay, Vi, of even rank, respectively odd, that is to say at the output of each absorbing delay of even or odd rank corresponding to reconstruct the right channels, respectively left, of the stereo output signal.
  • the stereophonic input signal has not been shown, so as not to overload the drawing, but corresponds substantially to that of FIG. 3c.
  • the corresponding reverberation filter is controlled by 4 completely independent parameters: size of the hearing room defined by a characteristic dimension thereof, reverberation time Tr ( ⁇ ) at low frequencies, ratio Tr at high frequencies / Tr at low frequencies , and cutoff frequency of the reverberated signal.
  • the reverberant filter proper was produced using digital calculation means comprising a DSP 56000 calculator receiving the stereophonic source signal at input and a calculator element of the same type producing the control modules for the first echoes of Figure 5c, for example.
  • This second calculator element allows the signals from several mono sources to be read and transmits the channels of the echo bus to the reverberant filter. Note that even if the number of monophonic sources is greater, four echo modules are sufficient for realistic spatialization. Note that the monophonic sources are then divided into four groups, each of which is assigned to an echo module.
  • the relation 13 in fact constitutes an approximation of the relation 16.
  • the family of reverberant filters constituting the spatialization systems of an audio-digital signal object of the present invention considerably improves the quality of the reverberation compared to the known structure, known as in sum of combs. It allows in particular to quickly obtain a high density of echoes in the temporal response for a number N of reduced delays.
  • 8 delays are sufficient, that is to say 8 delay channels, where 40 comb filters would be necessary.
  • the simulation of a large room requires that the modal density, therefore the sum of the durations of the absorbing delays ⁇ i, be of the order of one second. It is therefore advisable to increase the number of delays to at least 12, in order to increase the echo density at the start of the time response.
  • the real-time simulation of the reverberation in all cases can be carried out by means of the computing capacity of a DSP 56000 microcomputer and that in particular this type of calculator allows, in the case of the simultaneous spatialization of several monophonic sources, to process 4 monophonic sources if the number of channels of the echo BUS is 12.
  • This embodiment makes it possible, for example, to separately control for each source the amplitude, the instant of arrival and the direction of provenance of the direct sound and the first 5 reflections.
  • the use of 3 DSP 56000 type computers makes it possible to spatialize 6 monophonic sources by controlling for each the first 8 echoes.
  • FIGS. 7a, 7b and 7c A particularly advantageous use of a system which is the subject of the present invention will now be described in conjunction with FIGS. 7a, 7b and 7c.
  • the absolute values of the coefficients aji can take only two absolute values. Indeed, N of them have the absolute value 1- (2 / N), and all the others have the absolute value 2 / N. Consequently, when the number N of delays becomes large, a small number of loopback paths is preponderant compared to the others. This has the effect of delaying the moment when, in the impulse response, all the echoes have similar amplitudes. As a result, the temporal density is perceived as insufficient at the start of the impulse response, although the theoretical echo density is high.
  • a PN is a unit matrix, because product of a block-diagonal matrix formed by the unit matrices Aj, and of a permutation matrix noted JPN. This permutation corresponds to the exchange of the indices i and j in the numbering of the delays ⁇ ji, it is such that if all the matrices Aj are equal to the same matrix A, then the matrix A PN can be written:
  • the looping matrix AB PN then appears as a matrix obtained by unitary assembly of unitary blocks, this looping matrix AB PN being designated by "unitary matrix by blocks”.
  • this consists in choosing the matrices Aj and Bi within the family defined by the preceding relation (8).
  • each of the P loopings defined by the matrices Aj can be carried out in 2.N operations
  • each of the N interleaving defined by the matrices Bi can be carried out in 2.P operations, for a total of 4.NP operations for make a reverberant filter including NP delays.
  • a reverberant filter consisting of N.P delays is obtained, as described above, by association in parallel and interleaving of the loops of P reverberant filters each consisting of N delays.
  • the P starting reverberant filters are identical to that of FIG. 3b and the interlacing of the P loopings is itself carried out as the looping of FIG. 3b.
  • Fig. 7c shows that the reverberant filter thus produced can also be seen as the paralleling of N reverberant filters with P inputs and P outputs, the assembly being "looped back" on itself as shown in FIG. 3b.
  • the total number of additions-multiplications necessary for the loopback and the calculation of the output signal y (k) is approximately equal to 4.N.P.
  • the looping matrices Aj and the interleaving matrices Bi are all equal to the matrix: where, to simplify the writing, the signs + and - signify +1 and -1 respectively.
  • the looping matrix, denoted AA16, of the reverberant filter with 16 delays thus produced is unitary by blocks, and all of its coefficients have the same value.
  • the particularly efficient character of the method and of the system which are the subject of the present invention results in particular from the independence between the control of the aforementioned parameters, this independence being essential from the perceptual point of view, but also in order to allow the simulation of spatialization in a real room from measurements made in it.

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  • Engineering & Computer Science (AREA)
  • Multimedia (AREA)
  • Physics & Mathematics (AREA)
  • Acoustics & Sound (AREA)
  • Signal Processing (AREA)
  • Reverberation, Karaoke And Other Acoustics (AREA)
  • Stereophonic System (AREA)
EP93400503A 1992-03-03 1993-02-26 Verfahren und Vorrichtung für künstliche Raumklangeffekte von digitalen Audiosignalen Expired - Lifetime EP0559530B1 (de)

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FR9202528A FR2688371B1 (fr) 1992-03-03 1992-03-03 Procede et systeme de spatialisation artificielle de signaux audio-numeriques.
FR9202528 1992-03-03

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CN112201267A (zh) * 2020-09-07 2021-01-08 北京达佳互联信息技术有限公司 一种音频处理方法、装置、电子设备及存储介质

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