Abstract:
A method of generating an audio output signal according to a downward compatible sound format, the method including: generating a sum signal by combining a first input channel signal with a second input channel signal; and dynamically correcting the sum signal using samples of the first and second input channel signals from overlapping time windows.

Description:
BACKGROUND OF THE INVENTION 
     The Relevant Technology 
       [0001]    For regular broadcasting, internet, and the home area, besides two channel stereo and mono, the 5.1 sound format is also well established. Through the additional available sound formats there is an increased effort in audio production, in particular the effort of recording and mixing the respective sound formats. Also the compatibility to playback devices needs to be guaranteed, thus they need to be able to playback every sound format independent of the number of audio channels. 
         [0002]    One possibility is the transmission of the sound format comprising the greatest number of audio channels and if necessary an automatic conversion of the signal by the receiver to a sound format with a smaller number of audio channels (automatic downmix). 
         [0003]    It is also possible to generate the material in all formats during the audio production and broadcast those signals simultaneously (simulcast). In this case each sound format can be generated separately. However, this kind of mixing requires considerable production effort. In most cases this requires either additional manpower, a noticeable higher time effort or multiple sets of equipment (e.g. in the case of a live broadcast). Therefore the resulting volume of production is hardly acceptable. Alternatively—as in the approach described earlier—an automatic downmix can be done. 
         [0004]    Such methods to automatically transform a sound format already exist, but further improvements are necessary in order to achieve a qualitatively satisfying result for a wide spectrum of basic raw material. 
         [0005]    Automatic downmix methods can be categorised roughly into active and passive methods. Active methods adapt the automatic transformation depending on the basic raw material, where passive methods work independent of a signal. A known passive downmix method is the based on the broadcast reference ITU-R BS.775 and is illustrated in  FIG. 1 . 
         [0006]    Based on a five channel sound format with the sound channels 
         [0007]    left channel (L) 
         [0008]    right channel (R) 
         [0009]    centre channel (C) 
         [0010]    rear left channel (Ls) 
         [0011]    rear right channel (Rs), 
         [0000]    the known downmix method is designed to lower the level of the centre channel (C), as well as the rear left channel (Ls) and the rear right channel (Rs) by −3 dB using a damping function  50 ,  60  or  70 . The −3dB lowered centre channel is distributed via the sum function  10  or  20  to the left channel and the right channel, while forming a first sum signal (output sum function  10 ) and a second sum signal (output sum function  20 ). The −3dB lowered level of the rear and the rear right signal (Ls) and (Rs) are distributed via the sum function  30  and  40  to the first and second sum signal to form the left and right channel (L 0 ) and (R 0 ) of the desired two channel sound format. 
         [0012]    For the active method the sum functions according to the block diagram of  FIG. 1  are checked with respect to the properties of the summed audio signal and corrected, where needed in order to avoid unwanted sound results. Therefore a company called Coding Technology has suggested a downmix algorithm based on the ITU downmix according to  FIG. 1 . In the downmix algorithm, the energy content of all sum signals are analyzed in 28 frequency bands/partial bands and are compared with the energy content of the five channel audio format. In this way, increases and decreases of the energy content can be determined and compensated by correcting the amplitude in the affected partial bands. A change in the tone colour via the comb filter effect can be limited in this way. The correction only proceeds up to a meaningful level as the suffixing signal would cause an infinite correction factor. The downmix algorithm can cause shifts of the phantom sound source between the resulting left and right channels of the two channel sound format and in particular independent of the original position of the phantom sound source in the five channel source material. 
         [0013]    In order to reduce such shifts of the phantom sound source, a company called Lexicon has suggested method Logic  7 , where next to the downmix there is also the possibility of an upmix. The multi channel sound can be downmixed to a mono signal as well as to a stereo signal. Furthermore, it is possible, for example, to decode up to 8 channels out of a stereo downmix Therefore the fraction of a centre channel downmix is controlled via variable coefficients and the fraction of the rear right and rear left channels are adapted with further coefficients. For the left channel a fraction of 0.91 of the rear left channel is used with a fraction of −0.38 of the rear right channel. The mixing of the right channel proceeds accordingly. With this method the levels of both rear channels stay unchanged. Through a phase shift of 90° a later separation of both rear channels from the left and right channels are possible. But sound tone changes as of comb filter effects of the phase shift cannot be limited with the method Logic  7 . 
     
    
     
       BRIEF DESCRIPTION OF THE DRAWINGS 
         [0014]    Various embodiments of the present invention will now be discussed with reference to the appended drawings. 
           [0015]      FIG. 1  illustrates a conventional downmix method; 
           [0016]      FIG. 2  is a general block diagram showing a method of generating a downward compatible sound format according to one embodiment of the present invention; and 
           [0017]      FIGS. 3-6  are block diagrams showing various embodiments of analysis and correction algorithms that can be used in the method illustrated in  FIG. 2 . 
       
    
    
     DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS 
       [0018]    The object of the invention is to largely compensate for the shift of the phantom sound source, the change in level difference between the coherent and incoherent signal parts as well as the sound tone changes. 
         [0019]    The underlying idea of the invention is while forming the first (L′) and second (R′) sum signals, to dynamically correct each of the spectral values of overlapping time windows with (k) samples of the left channel (L) and right channel (R). Furthermore while forming the third and fourth sum signals, the spectral values of overlapping time windows with (k) samples of the first (L′) and second (R′) sum signals are each dynamically corrected. 
         [0020]    The invention is explained further while referring to the embodiment shown in  FIGS. 2 to 6 . It shows: 
         [0021]      FIG. 2  is a general block diagram showing a method according to one embodiment of the invention;  FIGS. 3 to 6  are flow charts for the analysis and correction blocks for the intended functions. 
         [0022]    The block diagram shown in  FIG. 2  is similar to the block diagram in  FIG. 1  but with a significant difference. For the sum functions  100  and  200  to form the first and second sum signals L′ and R′ as well as for the sum functions  300  and  400  to form the left and right signals L IRT  and R IRT  of the two channel sound format the sum functions are analysed and corrected (see Analysis and correction blocks  1 - 4 ) in addition to the summation. The lowering of the centre signal C as well as the rear right and rear left signals Ls and Rs is carried out in block diagram  2  in a similar manner to that discussed above regarding the block diagram of  FIG. 1  (e.g. −3dB via a damping function  50 ,  60  or  70 ). However, one could think of dampings other than −3dB in particular depending on the genre or content of the five channel source signal. 
         [0023]    The functional structures of the analysis in correction blocks  100 ,  200 ,  300  and  400  in  FIG. 2  are shown respectively in  FIGS. 3 ,  4 ,  5 , and  6 . 
         [0024]    In  FIG. 3 , Analysis and Correction  1  (block  100 ) is designed to carry out a first transformation of the input left and centre signals L and C to spectral values, e.g. via FFTs, as shown in step  101 . The formed spectral values  1 (k), c(k) are added in the sum function shown in step  102 . The absolute value S 1 (k) of the sum of the spectral values is assessed in step  103  according to if the absolute value S 1 (k) is greater than a desired value A soll, l (k). The desired value A soll, l (k) is determined according to the following: 
         [0000]      A soll, l (k)=√{square root over (|l(k)| 2 +|c(k)| 2 )}{square root over (|l(k)| 2 +|c(k)| 2 )}
 
         [0025]    If the absolute value S 1 (k) is greater than A soll, l (k), then the value l′(k) of the left channel is determined according to step  104  as: 
         [0000]      l′(k)=A soll, l(k)+(|l(k)+c(k)|−A   soll, l (k))* n, 
 
         [0000]    where n is a factor greater than 0.1 and less than 0.4. 
         [0026]    If the absolute value S 1 (k) is not greater than the desired value A soll, l (k), then the spectral value l′(k) of the left channel is determined according to step  105 , in which the spectral value l(k) is multiplied by a factor m 1 (k). The factor m 1 (k) is greater than 1 and is used to adapt the value similar to the aforementioned factor n. The product m 1 (k)*l(k) is added to the spectral value c(k) of the centre channel (i.e., m 1 (k)*l(k)+c). 
         [0027]    In the end, the level adapted signal l′(k) determined either according to m 1 (k)*l(k)+c(k) or A soll, l (k)+(ll(k)+c(k)l−A soll, l (k))*n, as discussed above, is then put through an inverse transformation, as shown in step  106 , to determine the first sum signal L′. 
         [0028]    In  FIG. 4 , Analysis and Correction  2  (block  200 ) is designed to carry out a first transformation of the input right and centre signals R and C to spectral values, e.g. via a FFTs, as shown in step  201 . The formed spectral values r(k) and c(k) are added in the sum function shown in step  202 . The absolute value S r (k) of the sum of the spectral values is assessed in step  203  according to if the absolute value S r (k) is greater than a desired value A soll, r (k). The desired value A soll, r (k) is determined according to the following: 
         [0000]      A soll, r (k)=√{square root over (|r(k)| 2 +|c(k)| 2 )}{square root over (|r(k)| 2 +|c(k)| 2 )}
 
         [0029]    If the absolute value S r (k) is greater than A soll, r (k) then the value r′(k) of the right channel is determined in step  204  as: 
         [0000]      r′(k)=A soll, r (k)+(|r(k)+c(k)|−A soll, r (k))*n,
 
         [0000]    where n is a factor greater than 0.1 and less than 0.4. 
         [0030]    If the absolute value S r (k) is not greater than the desired value A soll, r (k), then the spectral value r′(k) of the right channel is determined according to step  205 , in which the spectral value r(k) is multiplied by a factor m r (k). The factor m r (k) is greater than 1 and is used to adapt the level, similar to the aforementioned factor n. The product m r (k)*r(k) is added to the spectral value c(k) of the centre channel (i.e., m r (k)*r(k)+c(k)). 
         [0031]    In the end, the level adapted signal r′(k) determined either according to m r (k)*r(k)+c(k) or A soll, r (k)+(lr(k)+c(k)l−A soll, r (k))*n, as discussed above, is then put through an inverse transformation, as shown in step  106 , to determine the second sum signal R′. 
         [0032]    In  FIG. 5 , Analysis and Correction  3  (block  300 ) is designed to carry out a first transformation of the input rear left signal Ls and the first sum signal L′ to spectral values, e.g. via FFTs, as shown in step  301 . The formed spectral values ls(k) and l′(k) are added in the sum function shown in step  302 . The absolute value S ls (k) of sum of the spectral values is assessed in step  303  according to if the absolute value S ls (k) is greater than a desired value A soll, ls (k). The desired value A soll, ls (k) is determined according to the following: 
         [0000]      A soll, ls (k)=√{square root over (|ls(k)| 2 +|l′(k)| 2 )}{square root over (|ls(k)| 2 +|l′(k)| 2 )}
 
         [0033]    If the absolute value S ls (k) is greater than A soll, ls (k), then the value l lRT  of the rear left channel is determined in step  304  as: 
         [0000]      l lRT (k)=A soll, ls (k)+(|ls(k)+l′(k)|−A soll, ls (k))*n,
 
         [0000]    where n is a factor greater than 0.1 and less than 0.4. 
         [0034]    If the absolute value S ls (k) is not greater than the desired value A soll, ls (k), then the spectral value l lRT  is determined according to step  305 , in which the spectral value l′(k) is multiplied by a factor m ls (k). The factor m ls (k) is greater than one and is used to adapt the level, similar to the aforementioned factor n. The product m ls (k)*l′(k) is added to the spectral value ls(k) of the rear left channel (i.e., m ls (k)*l′(k)+ls(k)). 
         [0035]    In the end, the level adapted signal determined either according to m ls (k)*l′(k)+ls(k) or A soll, ls (k)+(ll′(k)+ls(k)l−A soll, ls (k))*n, as discussed above, is then put through an inverse transformation, as shown in step  306 , to determine the third sum signal and therefore the left output signal L. 
         [0036]    In  FIG. 6 , Analysis and Correction  4  (block  400 ) is designed to carry out a first transformation of the input rear right signal Rs and the second sum signal R′ to spectral values, e.g. via FFTs, as shown in step  401 . The formed spectral values rs(k) and r′(k) are added in the sum function shown in step  402 . The absolute value S rs (k) of the sum of the spectral values is assessed in step  403  according to if the absolute value S rs (k) is greater than a desired value A soll, rs (k). The desired value (A soll, rs (k)) is determined according to the following: 
         [0000]      A soll, rs (k)=√{square root over (|rs(k)| 2 +|r′(k)| 2 )}{square root over (|rs(k)| 2 +|r′(k)| 2 )}
 
         [0037]    If the absolute value S rs (k) is greater than A soll, ls (k), then the value r lRT  of the rear right channel is determined in step  404  as: 
         [0000]      r lRT (k)=A soll, rs (k)+(|r′(k)+rs(k)|−A soll, rs (k))*n,
 
         [0000]    where n is a factor greater than 0.1 and less than 0.4. 
         [0038]    If the absolute value S rs (k) is not greater than the desired value A soll, rs (k), then the spectral value r lRT  is determined according to step 405, in which the spectral value r′(k) is multiplied by a factor m rs (k). The factor m rs (k) is greater than one and is used to adapt the level, similar to the aforementioned factor n. The product m rs (k)*r′(k) is added to the spectral value rs(k) of the rear right channel (i.e., m rs (k)*r′(k)+rs (k)). 
         [0039]    In the end the level adapted signal determined either according to m rs (k)*r′(k)+rs(k) or A soll, rs (k)+(lr′(k)+rs(k)l−A soll, rs (k))*n, as discussed above, is then put through an inverse transformation, as shown in step  406 , to determine the fourth sum signal and therefore the right output signal R.