Abstract:
In order to compensate tonal changes arising from a multi-path propagation of sound portions during the mixing of multi microphone audio recordings as far as possible it is suggested to form spectral values of respectively overlapping time frames of samples of each a first microphone signal ( 100 ) and a second microphone signal ( 101 ). The spectral values ( 300 ) of the first microphone signal ( 100 ) are distributed with formation of spectral values ( 311 ) of a first sum signal to the spectral values ( 301 ) of a second microphone signal ( 101 ) in a first summing level ( 310 ), whereat a dynamic correction of the spectral values ( 300, 301 ) of one of the two microphone signals ( 100, 101 ) occurs. Spectral values ( 399 ) of a result signal are formed out of the spectral values ( 311 ) of the first sum signal which are subject to an inverse Fourier-transformation and a block junction ( FIG. 3 ).

Description:
FIELD OF THE INVENTION 
     The invention relates to a method for mixing microphone signals of an audio recording with a plurality of microphones. 
     Background and Objects of Invention 
     It is recognized (“Handbuch der Tonstudiotechnik” by Michael Dickreiter et al., ISBN 978-3598117657, pp. 211-212, 230-235, 265-266, 439, 479) to use several microphones instead of a single microphone in order to capture vast acoustic sceneries during the production of audio recordings for canned music, films, broadcasting, sound archives, computer games, multi-media presentations or websites. Therefore the term “multi-microphone audio recording” is generally used. A vast acoustic scenery may be, e.g., a concert hall with an orchestra of several musical instruments. In order to capture tonal details each individual instrument is recorded with an individual microphone positioned closely to the instrument and, in order to record the overall acoustics including the echoes in the concert hall and audience noises (applause in particular), additional microphones are positioned in a greater distance. 
     Another example of a vast acoustic scenery is a drum set consisting of several pulsatile instruments which is recorded in a recording studio. For a “multi-microphone audio recording” individual microphones are positioned near each pulsatile instrument and an additional microphone is installed above the drummer. 
     Such multi-microphone recordings allow for a maximized number of acoustic and tonal details along with the overall acoustics of the scenery to be captured in a high quality and to shape them aesthetically satisfactory. Each microphone signal of the several microphones is usually recorded as a multi-trace recording. During the following mixing of the microphone signals further creative work is done. In special cases it is possible to mix immediately “live” and only record the product of the mixing. 
     The creative goals of the mixing process are generally the balance of volumes of all sound sources, a natural sound and a reality-like spatial impression of the overall acoustics. 
     During the common mixing technique in an audio mixing console or in the mixer function of digital editing systems, a sum of the added microphone signals is produced, conducted by a summing unit (“bus”) which is a technical realization of a common mathematical addition. In  FIG. 1  a single summation in the signal path of a common mixing console or a digital editing system is exemplified. In  FIG. 2  a series connection of summations in the summing unit (“bus”) in the signal path of a common mixing console or a digital editing system is exemplified. The reference numbers of  FIGS. 1 and 2  are as follows: 
       100  a first microphone signal 
       101  a second microphone signal 
       110  a summation level based on an addition 
       111  a sum signal 
       199  a result signal 
       200  an n th  sum signal 
       201  an n+2 th  microphone signal 
       210  an n+1 th  summation level based on an addition 
       211  an n+1 th  sum signal 
     With the multi-microphone audio recording at least two microphone signals contain portions of sound which originate from the same sound source due to the ineluctable multipath propagation of sound. As these portions of sound reach the microphones with varying delays due to their varying sound paths a comb-filter effect occurs with the common mixing technique in the summing unit which can be heard as sound changes and which run counter to the intended natural sound. In the common mixing technique those sound changes based on comb-filter effects can be reduced by an adjustable amplification and a possible adjustable delay of the recorded microphone signals. However, such a reduction is only restrictively possible in case of a multipath propagation of sound from more than a single sound source. In any case a significant adjustment of the mixing console or the digital editing system is required for figuring out the best compromise. 
     In the earlier DE 10 2008 056 704 a down-mixing (so-called “downmixing”) for the production of a two-channel audio format from a multi-channel (e.g., five-channel) audio format is described which projects phantom audio sources. Here two input signals are summed up, wherein a loading with a corrective factor of the spectral coefficients of one of the two input signals to be summed up is conducted; the input signal which is loaded with the corrective factor is prioritized over the other input signal. The determination of the corrective factor as described in DE 10 2008 056 704, however, leads to possibly audible disturbing ambient noises in cases in which the amplitude of the prioritized signal over the non-prioritized signal is low. The likelihood of occurrence of such disturbances is low, but it cannot be manipulated. 
     A method of mixing microphone signals of an audio recording with several microphones is known from WO 2004/084 185 A1 in which spectral values of overlapping time windows of samples of a first microphone signal and a second microphone signal respectively are generated. The spectral values of the first microphone signal are distributed onto the spectral values of the second microphone signal in a first summation level, wherein a dynamic correction of the spectral values of one of the microphone signals is conducted. Spectral values of a result signal are made up of the spectral values of the first summation signal which are subject to an inverse Fourier-transformation and block junction. Thus, for every block of samples individual corrective factors can be determined. The dynamic correction by a signal depending loading of spectral coefficients instead of a common addition reduces unwanted comb-filter effects during multi-microphone mixing which occur in the summing element of the mixing console or editing system due to common addition. However, with this method disturbing ambient noises are audible if the amplitude of the prioritized signal is low compared to that of the non-prioritized signal. 
     The task of the invention is to compensate the tonal change which occurs due to multipath propagation of sound portions during the mixing of multi-microphone recordings as far as possible. 
    
    
     
       BRIEF DESCRIPTION OF THE DRAWINGS 
       The invention is described by means of the embodiments given in the figures wherein: 
         FIG. 1  shows a block diagram of a single summation in a signal path of a common mixing console or a digital editing system. 
         FIG. 2  shows a block diagram of a series connection of summations in a summing unit (“bus”) in a signal path of a common mixing console or a digital editing system. 
         FIG. 3  shows a general block diagram of an arrangement for the conducting of the method according to the invention; 
         FIG. 4  shows a similar block diagram as  FIG. 3 , but with the difference of having the first summing level enhanced by a number of additional summing levels; 
         FIG. 5  shows a block diagram of the first summing level as intended in  FIGS. 3 and 4 ; and 
         FIG. 6  shows a block diagram of a further summing level as intended in  FIG. 4 . 
     
    
    
     The reference numbers of  FIGS. 1 and 2  are as follows: 
       100  a first microphone signal 
       101  a second microphone signal 
       199  a result signal 
       201  an n+2 th  microphone signal 
       300  spectral values of the first microphone signal 
       301  spectral values of the second microphone signal 
       310  a first summing level 
       311  spectral value of a first sum signal 
       320  a block-building and spectral transformation unit 
       330  an inverse spectral transformation and block junction unit 
       399  spectral values of a result signal 
       400  spectral values of an n th  sum signal 
       401  spectral values of an n+2 th  microphone signal 
       410  an n+1 th  summing level 
       411  spectral values of an n+1 th  sum signal 
       500  allocation unit 
       501  spectral values A(k) of the prioritized signal 
       502  spectral values B(k) of the non-prioritized signal 
       510  calculation unit for corrective factor values 
       511  corrective factor values m(k) 
       520  multiplier-summer unit 
       700  an n th  building group consisting of unit  320  and the n+1 th  summing level  410 . 
     DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS 
       FIG. 3  shows a general block diagram of an arrangement for the conduction of the method according to the invention. A first microphone signal  100  and a second microphone signal  101  are lead to a dedicated block building and spectral transformation unit  320  respectively. In units  320  the microphone signals  100  and  101  are first divided into temporally overlapping signal segments, after what the built blocks undergo a Fourier-transformation. This results in the spectral values  300  of the first microphone signal  100  and the spectral values  301  of the second microphone signal  101  respectively at the outputs of blocks  320 . The spectral values  300  and  301  are subsequently fed into a first summing level  310  which creates the spectral values  311  of a first sum signal from the spectral values  300  and  301 . The spectral values  311  form at the same time the spectral values  399  of a result signal, which are first subject to an inverse Fourier-transformation in unit  330 . The so-formed spectral values are subsequently merged into blocks. The hence resulting blocks of temporally overlapping signal segments are accumulated to the result signal  199 . 
     The block diagram shown in  FIG. 4  is constructed similarly to the block diagram in  FIG. 3 , but with the main difference that spectral values  399  are not at the same time the spectral values  311 . In fact, in  FIG. 4  a connection series of one or more equal building groups  700  from each a block building and spectral transformation unit  320  and an n+1 th  summing level  410  is inserted between the spectral values  311  and the spectral values  399 . For simplification purposes  FIG. 4  only shows a single building group  700  of the building group  700  in the block diagram, which is described below, wherein the number index n serves as a serial number. The connection series of building groups  700  mentioned above are to be understood in a way that the spectral values  400  form at the same time the spectral values of the first sum signal  311  at the beginning of the connection series, and the spectral values  411  form at the same time the spectral values of the result signal  399  at the end of the connection series. For all other sections of the connection series the spectral values  411  of a summing level  410  form at the same time the spectral values  400  of the following summing level  410 . An n+2 th  microphone signal  201  is fed into each block building and spectral transformation unit  320  of a building group  700  of the connection series, in which it is divided into segments of temporally overlapping signal sections. The resulting blocks of temporally overlapping signal segments are Fourier transformed, resulting in the spectral values  401  of the n+2 th  microphone signal. The spectral values  400  of the n th  sum signal and the spectral values  401  of the n+2 th  microphone signal are then fed in the n+1 th  summing level  410 , which then produces the spectral values  411  of the n+1 th  sum signal from them. 
       FIG. 5  shows the details of the first summing level  310 . In summing level  310  the spectral values  300  of the first microphone signal  100  and the spectral values  301  of the second microphone signal  101  are fed into an allocation unit  500  in which a prioritization of the output signals  501 ,  502  of the unit  500  occurs depending on the choice of the producer or the user. Two alternative allocations are possible: When prioritizing the output signal  501  the spectral values A(k) of the signal  501  to be prioritized are allocated to the spectral values  301  and the spectral values B(k) of the signal  502  not to be prioritized are allocated to the spectral values  300 . Alternatively, the spectral values A(k) of the signal  501  to be prioritized are allocated to the spectral values  300  and the spectral values B(k) of the signal  502  not to be prioritized. The choice of the allocation of prioritization determines the spatial impression of the overall acoustics, and is made according to the creative demands. A typical possibility is to allocate the signals of those microphones intended to gather the overall acoustics (so-called main microphones) or sum signals formed according to the invention to the prioritized signal path, and to allocate the signals of those microphones placed near the sound sources (so-called supportive microphones) to the non-prioritized signal path. The allocated spectral values A(k) of the signal to be prioritized  501  and the spectral values B(k) of the signal not to be prioritized  502  are then fed into a calculation unit  510  for the corrective factor values m(k), which calculates the corrective factor values m(k) from the spectral values A(k) and B(k) as output signal  511  as follows. Either the corrective factor m(k) is calculated as follows:
 
 eA ( k )=Real( A ( k ))·Real( A ( k ))+Imag( A ( k ))·Imag( A ( k ))
 
 x ( k )=Real( B ( k ))·Real( B ( k ))+Imag( A ( k ))·Imag( A ( k ))
 
 w ( k )= D·x ( k )/ eA ( k )
 
 m ( k )=( w ( k ) 2 +1) (1/2)   −w ( k )
 
     or the corrective factor m(k) is calculated as follows:
 
 eA ( k )=Real( A ( k ))·Real( A ( k ))+Imag( A ( k ))·Imag( A ( k ))
 
 eB ( k )=Real( B ( k ))·Real( B ( k ))+Imag( B ( k ))·Imag( B ( k ))
 
 x ( k )=Real( B ( k ))·Real( B ( k ))+Imag( A ( k ))·Imag( A ( k ))
 
 w ( k )= D·x ( k )/ eA ( k )+ L·eB ( k ))
 
 m ( k )=( w ( k ) 2 +1) (1/2)   −w ( k )
 
     wherein it means that 
     m(k) is the k th  corrective factor 
     A(k) is the k th  spectral value of the signal to be prioritized 
     B(k) is the k th  spectral value of the signal not to be prioritized 
     D is the grade of compensation 
     L is the grade of the limitation of the compensation 
     Grade D of compensation is a numeric value which determines in how far the sound changes due to comb-filter effects are balanced. It is chosen according to the creative demand and the intended tonal effect and is advantageously in the rage of 0 to 1. If D=0 the sound equals exactly the sound of conventional mixing. If D=1 the comb-filter effect is completely removed. For values of D between 0 and 1 the tonal result is accordingly between the ones for D=0 and D=1. 
     Grade L of the limitation of the compensation is a numeric value which determines in how far the probability of the occurrence of disturbing ambient noises is reduced. Said probability is given when the amplitude of the microphone signal to be prioritized is low in contrast to the microphone signal not to be prioritized. L&gt;=0 is valid. If L=0 not reduction of the probability of disturbing ambient noises is given. Grade L is to be chosen that according to experience just as no more ambient noises can be heard. Typically grade L is of the order of 0.5. The bigger grade L the smaller the probability of ambient noises, but the balance of tonal changes as adjusted by D may also be reduced. 
     The spectral value A(k) of the signal to be prioritized  501  is additionally lead to a multiplier  520 , whereas the spectral values B(k) of the signal not to be prioritized  502  is additionally lead into a summer  530 . Furthermore, the corrective factor values m(k) of the output signal  511  are fed into the calculation unit  510  where they are multiplied complexly (according to real part and imaginary part) with the spectral values A(k)  501 . The resulting values of the multiplier  520  are fed into the summer  530  where they are added complexly (according to real part and imaginary part) to the spectral values B(k) of the signal not to be prioritized  502 . This results in the spectral values  311  of the first sum signal of the first summing level  310 . 
     What is important for the prioritization is the multiplication of the corrective factor m(k) with exactly one of the two summands of the addition conducted in the summer  530 . Thus, the complete signal path of this summand is “prioritized” from the microphone signal input to the summer  530 . 
       FIG. 6  shows the details of the n+1 th  summing level  410 . The n+1 th  summing level  410  is similar to the first summing level  310  in its construction, but with the difference that here the spectral values  400  of the n th  sum signal and the spectral values  401  of the n+2 th  microphone signal are fed into the allocation unit  500 ; furthermore, that the result values of the summer  530  form the spectral values of the n+1 th  sum signal. 
     It is apparent that this invention does not only refer to microphone signals but generally to every audio signal facing the problem described above. 
     Accordingly the input signals can be general audio signals which originate from audio recordings, which are available in the form of audio files or sound tracks which were saved for further editing in a storage. 
     Additionally the invention can be implemented in different ways, such as, e.g., a software, which runs on a computer, hardware, a combination thereof and/or a special circuit.