Abstract:
An audio spatial environment engine is provided for converting between different formats of audio data. The audio spatial environment engine ( 100 ) allows for flexible conversion between N-channel data and M-channel data and conversion from M-channel data back to N′-channel data, where N, M, and N′ are integers and where N is not necessarily equal to N′. For example, such systems could be used for the transmission or storage of surround sound data across a network or infrastructure designed for stereo sound data. The audio spatial environment engine provides improved and flexible conversions between different spatial environments due to an advanced dynamic down-mixing unit ( 102 ) and a high-resolution frequency band up-mixing unit ( 104 ). The dynamic down-mixing unit includes an intelligent: analysis and correction loop ( 108, 110 ) capable of correcting for spectral, temporal, and spatial inaccuracies common to many down-mixing methods. The up-mixing unit utilizes the extraction and analysis of important inter-channel spatial cues across high-resolution frequency bands to derive the spatial placement of different frequency elements. The down-mixing and up-mixing units, when used individually or as a system, provide improved sound quality and spatial distinction.

Description:
RELATED APPLICATIONS  
       [0001]     This application claims priority to U.S. provisional application 60/622,922, filed Oct. 28, 2004, entitled “2-to-N Rendering;” U.S. patent application Ser. No. 10/975,841, filed Oct. 28, 2004, entitled “Audio Spatial Environment Engine;” U.S. patent application Ser. No. 11/261,100 (attorney docket 13646.0014), “Audio Spatial Environment Down-Mixer,” filed herewith; and U.S. patent application Ser. No. 11/262,029 (attorney docket 13646.0012), “Audio Spatial Environment Up-Mixer,” filed herewith, each of which are commonly owned and which are hereby incorporated by reference for all purposes. 
     
    
     FIELD OF THE INVENTION  
       [0002]     The present invention pertains to the field of audio data processing, and more particularly to a system and method for transforming between different formats of audio data.  
       BACKGROUND OF THE INVENTION  
       [0003]     Systems and methods for processing audio data are known in the art. Most of these systems and methods are used to process audio data for a known audio environment, such as a two-channel stereo environment, a four-channel quadraphonic environment, a five channel surround sound environment (also known as a 5.1 channel environment), or other suitable formats or environments.  
         [0004]     One problem posed by the increasing number of formats or environments is that audio data that is processed for optimal audio quality in a first environment is often not able to be readily used in a different audio environment. One example of this problem is the transmission or storage of surround sound data across a network or infrastructure designed for stereo sound data. As the infrastructure for stereo two-channel transmission or storage may not support the additional channels of audio data for a surround sound format, it is difficult or impossible to transmit or utilize surround sound format data with the existing infrastructure.  
       SUMMARY OF THE INVENTION  
       [0005]     In accordance with the present invention, a system and method for an audio spatial environment engine are provided that overcome known problems with converting between spatial audio environments.  
         [0006]     In particular, a system and method for an audio spatial environment engine are provided that allows conversion between N-channel data and M-channel data and conversion from M-channel data back to N′-channel data where N, M, and N′ are integers and where N is not necessarily equal to N′.  
         [0007]     In accordance with an exemplary embodiment of the present invention, an audio spatial environment engine for converting from an N channel audio system to an M channel audio system and back to an N′ channel audio system, where N, M, and N′ are integers and where N is not necessarily equal to N′, is provided. The audio spatial environment engine includes a dynamic down-mixer that receives N channels of audio data and converts the N channels of audio data to M channels of audio data. The audio spatial environment engine also includes an up-mixer that receives the M channels of audio data and converts the M channels of audio data to N′ channels of audio data, where N is not necessarily equal to N′. One exemplary application of this system is for the transmission or storage of surround sound data across a network or infrastructure designed for stereo sound data. The dynamic down-mixing unit converts the surround sound data to stereo sound data for transmission or storage, and the up-mixing unit restores the stereo sound data to surround sound data for playback, processing, or some other suitable use.  
         [0008]     The present invention provides many important technical advantages. One important technical advantage of the present invention is a system that provides improved and flexible conversions between different spatial environments due to an advanced dynamic down-mixing unit and a high-resolution frequency band up-mixing unit. The dynamic down-mixing unit includes an intelligent analysis and correction loop for correcting spectral, temporal, and spatial inaccuracies common to many down-mixing methods. The up-mixing unit utilizes the extraction and analysis of important inter-channel spatial cues across high-resolution frequency bands to derive the spatial placement of different frequency elements. The down-mixing and up-mixing units, when used either individually or as a system, provide improved sound quality and spatial distinction.  
         [0009]     Those skilled in the art will further appreciate the advantages and superior features of the invention together with other important aspects thereof on reading the detailed description that follows in conjunction with the drawings.  
     
    
     BRIEF DESCRIPTION OF THE DRAWINGS  
       [0010]      FIG. 1  is a diagram of a system for dynamic down-mixing with an analysis and correction loop in accordance with an exemplary embodiment of the present invention;  
         [0011]      FIG. 2  is a diagram of a system for down-mixing data from N channels to M channels in accordance with an exemplary embodiment of the present invention;  
         [0012]      FIG. 3  is a diagram of a system for down-mixing data from 5 channels to 2 channels in accordance with an exemplary embodiment of the present invention;  
         [0013]      FIG. 4  is a diagram of a sub-band vector calculation system in accordance with an exemplary embodiment of the present invention;  
         [0014]      FIG. 5  is a diagram of a sub-band correction system in accordance with an exemplary embodiment of the present invention;  
         [0015]      FIG. 6  is a diagram of a system for up-mixing data from M channels to N channels in accordance with an exemplary embodiment of the present invention;  
         [0016]      FIG. 7  is a diagram of a system for up-mixing data from 2 channels to 5 channels in accordance with an exemplary embodiment of the present invention;  
         [0017]      FIG. 8  is a diagram of a system for up-mixing data from 2 channels to 7 channels in accordance with an exemplary embodiment of the present invention;  
         [0018]      FIG. 9  is a diagram of a method for extracting inter-channel spatial cues and generating a spatial channel filter for frequency domain applications in accordance with an exemplary embodiment of the present invention;  
         [0019]      FIG. 10A  is a diagram of an exemplary left front channel filter map in accordance with an exemplary embodiment of the present invention;  
         [0020]      FIG. 10B  is a diagram of an exemplary right front channel filter map;  
         [0021]      FIG. 10C  is a diagram of an exemplary center channel filter map;  
         [0022]      FIG. 10D  is a diagram of an exemplary left surround channel filter map; and  
         [0023]      FIG. 10E  is a diagram of an exemplary right surround channel filter map.  
     
    
     DETAILED DESCRIPTION OF PREFERRED EMBODIMENTS  
       [0024]     In the description that follows, like parts are marked throughout the specification and drawings with the same reference numerals. The drawing figures might not be to scale and certain components can be shown in generalized or schematic form and identified by commercial designations in the interest of clarity and conciseness.  
         [0025]      FIG. 1  is a diagram of a system  100  for dynamic down-mixing from an N-channel audio format to an M-channel audio format with an analysis and correction loop in accordance with an exemplary embodiment of the present invention. System  100  uses 5.1 channel sound (i.e. N=5) and converts the 5.1 channel sound to stereo sound (i.e. M=2), but other suitable numbers of input and output channels can also or alternatively be used.  
         [0026]     The dynamic down-mix process of system  100  is implemented using reference down-mix  102 , reference up-mix  104 , sub-band vector calculation systems  106  and  108 , and sub-band correction system  110 . The analysis and correction loop is realized through reference up-mix  104 , which simulates an up-mix process, sub-band vector calculation systems  106  and  108 , which compute energy and position vectors per frequency band of the simulated up-mix and original signals, and sub-band correction system  110 , which compares the energy and position vectors of the simulated up-mix and original signals and modifies the inter-channel spatial cues of the down-mixed signal to correct for any inconsistencies.  
         [0027]     System  100  includes static reference down-mix  102 , which converts the received N-channel audio to M-channel audio. Static reference down-mix  102  receives the 5.1 sound channels left L(T), right R(T), center C(T), left surround LS(T), and right surround RS(T) and converts the 5.1 channel signals into stereo channel signals left watermark LW′ (T) and right watermark RW′ (T).  
         [0028]     The left watermark LW′ (T) and right watermark RW′ (T) stereo channel signals are subsequently provided to reference up-mix  104 , which converts the stereo sound channels into 5.1 sound channels. Reference up-mix  104  outputs the 5.1 sound channels left L′ (T), right R′ (T), center C′ (T), left surround LS′ (T), and right surround RS′ (T).  
         [0029]     The up-mixed 5.1 channel sound signals output from reference up-mix  104  are then provided to sub-band vector calculation system  106 . The output from sub-band vector calculation system  106  is the up-mixed energy and image position data for a plurality of frequency bands for the up-mixed 5.1 channel signals L′ (T), R′ (T), C′ (T), LS′ (T), and RS′ (T). Likewise, the original 5.1 channel sound signals are provided to sub-band vector calculation system  108 . The output from sub-band vector calculation system  108  is the source energy and image position data for a plurality of frequency bands for the original 5.1 channel signals L(T), R(T), C(T), LS(T), and RS(T). The energy and position vectors computed by sub-band vector calculation systems  106  and  108  consist of a total energy measurement and a 2-dimensional vector per frequency band which indicate the perceived intensity and source location for a given frequency element for a listener under ideal listening conditions. For example, an audio signal can be converted from the time domain to the frequency domain using an appropriate filter bank, such as a finite impulse response (FIR) filter bank, a quadrature mirror filter (QMF) bank, a discrete Fourier transform (DFT), a time-domain aliasing cancellation (TDAC) filter bank, or other suitable filter bank. The filter bank outputs are further processed to determine the total energy per frequency band and a normalized image position vector per frequency band.  
         [0030]     The energy and position vector values output from sub-band vector calculation systems  106  and  108  are provided to sub-band correction system  110 , which analyzes the source energy and position for the original 5.1 channel sound with the up-mixed energy and position for the 5.1 channel sound as it is generated from the left watermark LW′ (T) and right watermark RW′ (T) stereo channel signals. Differences between the source and up-mixed energy and position vectors are then identified and corrected per sub-band on the left watermark LW′ (T) and right watermark RW′ (T) signals producing LW(T) and RW(T) so as to provide a more accurate down-mixed stereo channel signal and more accurate 5.1 representation when the stereo channel signals are subsequently up-mixed. The corrected left watermark LW(T) and right watermark RW(T) signals are output for transmission, reception by a stereo receiver, reception by a receiver having up-mix functionality, or for other suitable uses.  
         [0031]     In operation, system  100  dynamically down-mixes 5.1 channel sound to stereo sound through an intelligent analysis and correction loop, which consists of simulation, analysis, and correction of the entire down-mix/up-mix system. This methodology is accomplished by generating a statically down-mixed stereo signal LW′ (T) and RW′ (T), simulating the subsequent up-mixed signals L′ (T), R′ (T), C′ (T), LS′ (T), and RS′ (T), and analyzing those signals with the original 5.1 channel signals to identify and correct any energy or position vector differences on a sub-band basis that could affect the quality of the left watermark LW′ (T) and right watermark RW′ (T) stereo signals or subsequently up-mixed surround channel signals. The sub-band correction processing which produces left watermark LW(T) and right watermark RW(T) stereo signals is performed such that when LW(T) and RW(T) are up-mixed, the 5.1 channel sound that results matches the original input 5.1 channel sound with improved accuracy. Likewise, additional processing can be performed so as to allow any suitable number of input channels to be converted into a suitable number of watermarked output channels, such as 7.1 channel sound to watermarked stereo, 7.1 channel sound to watermarked 5.1 channel sound, custom sound channels (such as for automobile sound systems or theaters) to stereo, or other suitable conversions.  
         [0032]      FIG. 2  is a diagram of a static reference down-mix  200  in accordance with an exemplary embodiment of the present invention. Static reference down-mix  200  can be used as reference down-mix  102  of  FIG. 1  or in other suitable manners.  
         [0033]     Reference down-mix  200  converts N channel audio to M channel audio, where N and M are integers and N is greater than M. Reference down-mix  200  receives input signals X 1 (T), X 2 (T), through X N (T). For each input channel i, the input signal X 1 (T) is provided to a Hilbert transform unit  202  through  206  which introduces a 90° phase shift of the signal. Other processing such as Hilbert filters or all-pass filter networks that achieve a 90° phase shift could also or alternately be used in place of the Hilbert transform unit. For each input channel i, the Hilbert transformed signal and the original input signal are then multiplied by a first stage of multipliers  208  through  218  with predetermined scaling constants C i11  and C i12 , respectively, where the first subscript represents the input channel number i, the second subscript represents the first stage of multipliers, and the third subscript represents the multiplier number per stage. The outputs of multipliers  208  through  218  are then summed by summers  220  through  224 , generating the fractional Hilbert signal X′ i (T). The fractional Hilbert signals X′ i (T) output from multipliers  220  through  224  have a variable amount of phase shift relative to the corresponding input signals X i (T). The amount of phase shift is dependent on the scaling constants C i11  and C i12 , where 0° phase shift is possible corresponding to C i11 =0 and C i12 =1, and ±90° phase shift is possible corresponding to C i11 =±1 and C i12 =0. Any intermediate amount of phase shift is possible with appropriate values of C i11  and C i12 .  
         [0034]     Each signal X′ i (T) for each input channel i is then multiplied by a second stage of multipliers  226  through  242  with predetermined scaling constant C i2j , where the first subscript represents the input channel number i, the second subscript represents the second stage of multipliers, and the third subscript represents the output channel number j. The outputs of multipliers  226  through  242  are then appropriately summed by summers  244  through  248  to generate the corresponding output signal Y j (T) for each output channel j. The scaling constants C i2j  for each input channel i and output channel j are determined by the spatial positions of each input channel i and output channel j. For example, scaling constants C i2j  for a left input channel i and right output channel j can be set near zero to preserve spatial distinction. Likewise, scaling constants C i2j  for a front input channel i and front output channel j can be set near one to preserve spatial placement.  
         [0035]     In operation, reference down-mix  200  combines N sound channels into M sound channels in a manner that allows the spatial relationships among the input signals to be arbitrarily managed and extracted when the output signals are received at a receiver. Furthermore, the combination of the N channel sound as shown generates M channel sound that is of acceptable quality to a listener listening in an M channel audio environment. Thus, reference down-mix  200  can be used to convert N channel sound to M channel sound that can be used with an M channel receiver, an N channel receiver with a suitable up-mixer, or other suitable receivers.  
         [0036]      FIG. 3  is a diagram of a static reference down-mix  300  in accordance with an exemplary embodiment of the present invention. As shown in  FIG. 3 , static reference down-mix  300  is an implementation of static reference down-mix  200  of  FIG. 2  which converts 5.1 channel time domain data into stereo channel time domain data. Static reference down-mix  300  can be used as reference down-mix  102  of  FIG. 1  or in other suitable manners.  
         [0037]     Reference down-mix  300  includes Hilbert transform  302 , which receives the left channel signal L(T) of the source 5.1 channel sound, and performs a Hilbert transform on the time signal. The Hilbert transform introduces a 90° phase shift of the signal, which is then multiplied by multiplier  310  with a predetermined scaling constant C L1 . Other processing such as Hilbert filters or all-pass filter networks that achieve a 90° phase shift could also or alternately be used in place of the Hilbert transform unit. The original left channel signal L(T) is multiplied by multiplier  312  with a predetermined scaling constant C L2 . The outputs of multipliers  310  and  312  are summed by summer  320  to generate fractional Hilbert signal L′ (T). Likewise, the right channel signal R(T) from the source 5.1 channel sound is processed by Hilbert transform  304  and multiplied by multiplier  314  with a predetermined scaling constant C R1 . The original right channel signal R(T) is multiplied by multiplier  316  with a predetermined scaling constant C R2 . The outputs of multipliers  314  and  316  are summed by summer  322  to generate fractional Hilbert signal R′ (T). The fractional Hilbert signals L′ (T) and R′ (T) output from multipliers  320  and  322  have a variable amount of phase shift relative to the corresponding input signals L(T) and R(T), respectively. The amount of phase shift is dependent on the scaling constants C L1 , C L2 , C R1 , and C R2 , where 0° phase shift is possible corresponding to C L1 =0 and C L2 =1 and C R1 =0 and C R2 =1, and ±90° phase shift is possible corresponding to C L1 =±1 and C L2 =0 and C R1 =±1 and C R2 =0. Any intermediate amount of phase shift is possible with appropriate values of C L1 , C L2 , C R1 , and C R2 . The center channel input from the source 5.1 channel sound is provided to multiplier  318  as fractional Hilbert signal C′ (T), implying that no phase shift is performed on the center channel input signal. Multiplier  318  multiplies C′ (T) with a predetermined scaling constant C 3 , such as an attenuation by three decibels. The outputs of summers  320  and  322  and multiplier  318  are appropriately summed into the left watermark channel LW′ (T) and the right watermark channel RW′ (T).  
         [0038]     The left surround channel LS(T) from the source 5.1 channel sound is provided to Hilbert transform  306 , and the right surround channel RS(T) from the source 5.1 channel sound is provided to Hilbert transform  308 . The outputs of Hilbert transforms  306  and  308  are fractional Hilbert signals LS′ (T) and RS′ (T), implying that a full 90° phase shift exists between the LS(T) and LS′ (T) signal pair and RS(T) and RS′ (T) signal pair. LS′ (T) is then multiplied by multipliers  324  and  326  with predetermined scaling constants C LS1  and C LS2 , respectively. Likewise, RS′ (T) is multiplied by multipliers  328  and  330  with predetermined scaling constants C RS1  and C RS2 , respectively. The outputs of multipliers  324  through  330  are appropriately provided to left watermark channel LW′ (T) and right watermark channel RW′ (T).  
         [0039]     Summer  332  receives the left channel output from summer  320 , the center channel output from multiplier  318 , the left surround channel output from multiplier  324 , and the right surround channel output from multiplier  328  and adds these signals to form the left watermark channel LW′ (T). Likewise, summer  334  receives the center channel output from multiplier  318 , the right channel output from summer  322 , the left surround channel output from multiplier  326 , and the right surround channel output from multiplier  330  and adds these signals to form the right watermark channel RW′ (T).  
         [0040]     In operation, reference down-mix  300  combines the source 5.1 sound channels in a manner that allows the spatial relationships among the 5.1 input channels to be maintained and extracted when the left watermark channel and right watermark channel stereo signals are received at a receiver. Furthermore, the combination of the 5.1 channel sound as shown generates stereo sound that is of acceptable quality to a listener using stereo receivers that do not perform a surround sound up-mix. Thus, reference down-mix  300  can be used to convert 5.1 channel sound to stereo sound that can be used with a stereo receiver, a 5.1 channel receiver with a suitable up-mixer, a 7.1 channel receiver with a suitable up-mixer, or other suitable receivers.  
         [0041]      FIG. 4  is a diagram of a sub-band vector calculation system  400  in accordance with an exemplary embodiment of the present invention. Sub-band vector calculation system  400  provides energy and position vector data for a plurality of frequency bands, and can be used as sub-band vector calculation systems  106  and  108  of  FIG. 1 . Although 5.1 channel sound is shown, other suitable channel configurations can be used.  
         [0042]     Sub-band vector calculation system  400  includes time-frequency analysis units  402  through  410 . The 5.1 time domain sound channels L(T), R(T), C(T), LS(T), and RS(T) are provided to time-frequency analysis units  402  through  410 , respectively, which convert the time domain signals into frequency domain signals. These time-frequency analysis units can be an appropriate filter bank, such as a finite impulse response (FIR) filter bank, a quadrature mirror filter (QMF) bank, a discrete Fourier transform (DFT), a time-domain aliasing cancellation (TDAC) filter bank, or other suitable filter bank. A magnitude or energy value per frequency band is output from time-frequency analysis units  402  through  410  for L(F), R(F), C(F), LS(F), and RS(F). These magnitude/energy values consist of a magnitude/energy measurement for each frequency band component of each corresponding channel. The magnitude/energy measurements are summed by summer  412 , which outputs T(F), where T(F) is the total energy of the input signals per frequency band. This value is then divided into each of the channel magnitude/energy values by division units  414  through  422 , to generate the corresponding normalized inter-channel level difference (ICLD) signals M L (F), M R (F), M C (F), M LS (F) and M RS (F), where these ICLD signals can be viewed as normalized sub-band energy estimates for each channel.  
         [0043]     The 5.1 channel sound is mapped to a normalized position vector as shown with exemplary locations on a 2-dimensional plane comprised of a lateral axis and a depth axis. As shown, the value of the location for (X LS , Y LS ) is assigned to the origin, the value of (X RS , Y RS ) is assigned to (0, 1), the value of (X L , Y L ) is assigned to (0, 1-C), where C is a value between 1 and 0 representative of the setback distance for the left and right speakers from the back of the room. Likewise, the value of (X R , Y R ) is (1, 1-C). Finally, the value for (X C , Y C ) is (0.5, 1). These coordinates are exemplary, and can be changed to reflect the actual normalized location or configuration of the speakers relative to each other, such as where the speaker coordinates differ based on the size of the room, the shape of the room or other factors. For example, where 7.1 sound or other suitable sound channel configurations are used, additional coordinate values can be provided that reflect the location of speakers around the room. Likewise, such speaker locations can be customized based on the actual distribution of speakers in an automobile, room, auditorium, arena, or as otherwise suitable.  
         [0044]     The estimated image position vector P(F) can be calculated per sub-band as set forth in the following vector equation: 
 
 P ( F )= M   L ( F )*( X   L   , Y   L )+ M   R ( F )*( X   R   , Y   R )+ M   C ( F )*( X   C   , Y   C )+ i. M   LS ( F )*( X   LS   , Y   LS )+ M   RS ( F )*( X   RS   , Y   RS ) 
 
         [0045]     Thus, for each frequency band, an output of total energy T(F) and a position vector P(F) are provided that are used to define the perceived intensity and position of the apparent frequency source for that frequency band. In this manner, the spatial image of a frequency component can be localized, such as for use with sub-band correction system  110  or for other suitable purposes.  
         [0046]      FIG. 5  is a diagram of a sub-band correction system in accordance with an exemplary embodiment of the present invention. The sub-band correction system can be used as sub-band correction system  110  of  FIG. 1  or for other suitable purposes. The sub-band correction system receives left watermark LW′ (T) and right watermark RW′ (T) stereo channel signals and performs energy and image correction on the watermarked signal to compensate for signal inaccuracies for each frequency band that may be created as a result of reference down-mixing or other suitable method. The sub-band correction system receives and utilizes for each sub-band the total energy signals of the source T SOURCE (F) and subsequent up-mixed signal T UMIX (F) and position vectors for the source P SOURCE (F) and subsequent up-mixed signal P UMIX (F), such as those generated by sub-band vector calculation systems  106  and  108  of  FIG. 1 . These total energy signals and position vectors are used to determine the appropriate corrections and compensations to perform.  
         [0047]     The sub-band correction system includes position correction system  500  and spectral energy correction system  502 . Position correction system  500  receives time domain signals for left watermark stereo channel LW′ (T) and right watermark stereo channel RW′ (T), which are converted by time-frequency analysis units  504  and  506 , respectively, from the time domain to the frequency domain. These time-frequency analysis units could be an appropriate filter bank, such as a finite impulse response (FIR) filter bank, a quadrature mirror filter (QMF) bank, a discrete Fourier transform (DFT), a time-domain aliasing cancellation (TDAC) filter bank, or other suitable filter bank.  
         [0048]     The output of time-frequency analysis units  504  and  506  are frequency domain sub-band signals LW′ (F) and RW′ (F). Relevant spatial cues of inter-channel level difference (ICLD) and inter-channel coherence (ICC) are modified per sub-band in the signals LW′ (F) and RW′ (F). For example, these cues could be modified through manipulation of the magnitude or energy of LW′ (F) and RW′ (F), shown as the absolute value of LW′ (F) and RW′ (F), and the phase angle of LW′ (F) and RW′ (F). Correction of the ICLD is performed through multiplication of the magnitude/energy value of LW′ (F) by multiplier  508  with the value generated by the following equation: 
 
[ X   MAX   −P   X,SOURCE ( F )]/[ X   MAX   −P   X,UMIX ( F )]
 
 where 
        X MAX =maximum X coordinate boundary     P X,SOURCE (F)=estimated sub-band X position coordinate from source vector     P X,UMIX (F)=estimated sub-band X position coordinate from subsequent up-mix vector 
 
 Likewise, the magnitude/energy for RW′ (F) is multiplied by multiplier  510  with the value generated by the following equation: 
 
[ P   X,SOURCE ( F )− X   MIN   ]/[P   X,UMIX ( F )− X   MIN ]
 
 where 
    X MIN =minimum X coordinate boundary        
 
         [0053]     Correction of the ICC is performed through addition of the phase angle for LW′ (F) by adder  512  with the value generated by the following equation: 
 
+/−π*[ P   Y,SOURCE ( F )− P   Y,UMIX ( F )]/[ Y   MAX   −Y   MIN ]
 
 where 
        P Y,SOURCE (F)=estimated sub-band Y position coordinate from source vector     P Y,UMIX (F)=estimated sub-band Y position coordinate from subsequent up-mix vector     Y MAX =maximum Y coordinate boundary     Y MIN =minimum Y coordinate boundary        
 
         [0058]     Likewise, the phase angle for RW′ (F) is added by adder  514  to the value generated by the following equation: 
 
−/+π*[ P   Y,SOURCE ( F )− P   Y,UMIX ( F )]/[ Y   MAX   −Y   MIN ]
 
 Note that the angular components added to LW′ (F) and RW′ (F) have equal value but opposite polarity, where the resultant polarities are determined by the leading phase angle between LW′ (F) and RW′ (F). 
 
         [0059]     The corrected LW′ (F) magnitude/energy and LW′ (F) phase angle are recombined to form the complex value LW(F) for each sub-band by adder  516  and are then converted by frequency-time synthesis unit  520  into a left watermark time domain signal LW(T). Likewise, the corrected RW′ (F) magnitude/energy and RW′ (F) phase angle are recombined to form the complex value RW(F) for each sub-band by adder  518  and are then converted by frequency-time synthesis unit  522  into a right watermark time domain signal RW(T). The frequency-time synthesis units  520  and  522  can be a suitable synthesis filter bank capable of converting the frequency domain signals back to time domain signals.  
         [0060]     As shown in this exemplary embodiment, the inter-channel spatial cues for each spectral component of the watermark left and right channel signals can be corrected using position correction  500  which appropriately modify the ICLD and ICC spatial cues.  
         [0061]     Spectral energy correction system  502  can be used to ensure that the total spectral balance of the down-mixed signal is consistent with the total spectral balance of the original 5.1 signal, thus compensating for spectral deviations caused by comb filtering for example. The left watermark time domain signal and right watermark time domain signals LW′ (T) and RW′ (T) are converted from the time domain to the frequency domain using time-frequency analysis units  524  and  526 , respectively. These time-frequency analysis units could be an appropriate filter bank, such as a finite impulse response (FIR) filter bank, a quadrature mirror filter (QMF) bank, a discrete Fourier transform (DFT), a time-domain aliasing cancellation (TDAC) filter bank, or other suitable filter bank. The output from time-frequency analysis units  524  and  526  is LW′ (F) and RW′ (F) frequency sub-band signals, which are multiplied by multipliers  528  and  530  by T SOURCE (F)/T UMIX (F), where 
 
 T   SOURCE ( F )=| L ( F )|+| R ( F )|+| C ( F )|+| LS ( F )|+| RS ( F )|
 
 T   UMIX ( F )=| L   UMIX ( F )|+| R   UMIX ( F )|+| C   UMIX ( F )|+| LS   UMIX ( F )|+| RS   UMIX ( F )|
 
         [0062]     The output from multipliers  528  and  530  are then converted by frequency-time synthesis units  532  and  534  back from the frequency domain to the time domain to generate LW(T) and RW(T). The frequency-time synthesis unit can be a suitable synthesis filter bank capable of converting the frequency domain signals back to time domain signals. In this manner, position and energy correction can be applied to the down-mixed stereo channel signals LW′ (T) and RW′ (T) so as to create a left and right watermark channel signal LW(T) and RW(T) that is faithful to the original 5.1 signal. LW(T) and RW(T) can be played back in stereo or up-mixed back into 5.1 channel or other suitable numbers of channels without significantly changing the spectral component position or energy of the arbitrary content elements present in the original 5.1 channel sound.  
         [0063]      FIG. 6  is a diagram of a system  600  for up-mixing data from M channels to N channels in accordance with an exemplary embodiment of the present invention. System  600  converts stereo time domain data into N channel time domain data.  
         [0064]     System  600  includes time-frequency analysis units  602  and  604 , filter generation unit  606 , smoothing unit  608 , and frequency-time synthesis units  634  through  638 . System  600  provides improved spatial distinction and stability in an up-mix process through a scalable frequency domain architecture, which allows for high resolution frequency band processing, and through a filter generation method which extracts and analyzes important inter-channel spatial cues per frequency band to derive the spatial placement of a frequency element in the up-mixed N channel signal.  
         [0065]     System  600  receives a left channel stereo signal L(T) and a right channel stereo signal R(T) at time-frequency analysis units  602  and  604 , which convert the time domain signals into frequency domain signals. These time-frequency analysis units could be an appropriate filter bank, such as a finite impulse response (FIR) filter bank, a quadrature mirror filter (QMF) bank, a discrete Fourier transform (DFT), a time-domain aliasing cancellation (TDAC) filter bank, or other suitable filter bank. The output from time-frequency analysis units  602  and  604  are a set of frequency domain values covering a sufficient frequency range of the human auditory system, such as a 0 to 20 kHz frequency range where the analysis filter bank sub-band bandwidths could be processed to approximate psycho-acoustic critical bands, equivalent rectangular bandwidths, or some other perceptual characterization. Likewise, other suitable numbers of frequency bands and ranges can be used.  
         [0066]     The outputs from time-frequency analysis units  602  and  604  are provided to filter generation unit  606 . In one exemplary embodiment, filter generation unit  606  can receive an external selection as to the number of channels that should be output for a given environment. For example, 4.1 sound channels where there are two front and two rear speakers can be selected, 5.1 sound systems where there are two front and two rear speakers and one front center speaker can be selected, 7.1 sound systems where there are two front, two side, two rear, and one front center speaker can be selected, or other suitable sound systems can be selected. Filter generation unit  606  extracts and analyzes inter-channel spatial cues such as inter-channel level difference (ICLD) and inter-channel coherence (ICC) on a frequency band basis. Those relevant spatial cues are then used as parameters to generate adaptive channel filters which control the spatial placement of a frequency band element in the up-mixed sound field. The channel filters are smoothed by smoothing unit  608  across both time and frequency to limit filter variability which could cause annoying fluctuation effects if allowed to vary too rapidly. In the exemplary embodiment shown in  FIG. 6 , the left and right channel L(F) and R(F) frequency domain signals are provided to filter generation unit  606  producing N channel filter signals H 1 (F), H 2 (F), through H N (F) which are provided to smoothing unit  608 .  
         [0067]     Smoothing unit  608  averages frequency domain components for each channel of the N channel filters across both the time and frequency dimensions. Smoothing across time and frequency helps to control rapid fluctuations in the channel filter signals, thus reducing jitter artifacts and instability that can be annoying to a listener. In one exemplary embodiment, time smoothing can be realized through the application of a first-order low-pass filter on each frequency band from the current frame and the corresponding frequency band from the previous frame. This has the effect of reducing the variability of each frequency band from frame to frame. In another exemplary embodiment, spectral smoothing can be performed across groups of frequency bins which are modeled to approximate the critical band spacing of the human auditory system. For example, if an analysis filter bank with uniformly spaced frequency bins is employed, different numbers of frequency bins can be grouped and averaged for different partitions of the frequency spectrum. For example, from zero to five kHz, five frequency bins can be averaged, from 5 kHz to 10 kHz, 7 frequency bins can be averaged, and from 10 kHz to 20 kHz, 9 frequency bins can be averaged, or other suitable numbers of frequency bins and bandwidth ranges can be selected. The smoothed values of H 1 (F), H 2 (F) through H N (F) are output from smoothing unit  608 .  
         [0068]     The source signals X 1 (F), X 2 (F), through X N (F) for each of the N output channels are generated as an adaptive combination of the M input channels. In the exemplary embodiment shown in  FIG. 6 , for a given output channel i, the channel source signal X i (F) output from summers  614 ,  620 , and  626  are generated as a sum of L(F) multiplied by the adaptive scaling signal G i (F) and R(F) multiplied by the adaptive scaling signal 1−G i (F). The adaptive scaling signals G i (F) used by multipliers  610 ,  612 ,  616 ,  618 ,  622 , and  624  are determined by the intended spatial position of the output channel i and a dynamic inter-channel coherence estimate of L(F) and R(F) per frequency band. Likewise, the polarity of the signals provided to summers  614 ,  620 , and  626  are determined by the intended spatial position of the output channel i. For example, adaptive scaling signals G i (F) and the polarities at summers  614 ,  620 , and  626  can be designed to provide L(F)+R(F) combinations for front center channels, L(F) for left channels, R(F) for right channels, and L(F)−R(F) combinations for rear channels as is common in traditional matrix up-mixing methods. The adaptive scaling signals G i (F) can further provide a way to dynamically adjust the correlation between output channel pairs, whether they are lateral or depth-wise channel pairs.  
         [0069]     The channel source signals X 1 (F), X 2 (F), through X N (F) are multiplied by the smoothed channel filters H 1 (F), H 2 (F), through H N (F) by multipliers  628  through  632 , respectively.  
         [0070]     The output from multipliers  628  through  632  is then converted from the frequency domain to the time domain by frequency-time synthesis units  634  through  638  to generate output channels Y 1 (T), Y 2 (T), through Y N (T). In this manner, the left and right stereo signals are up-mixed to N channel signals, where inter-channel spatial cues that naturally exist or that are intentionally encoded into the left and right stereo signals, such as by the down-mixing watermark process of  FIG. 1  or other suitable process, can be used to control the spatial placement of a frequency element within the N channel sound field produced by system  600 . Likewise, other suitable combinations of inputs and outputs can be used, such as stereo to 7.1 sound, 5.1 to 7.1 sound, or other suitable combinations.  
         [0071]      FIG. 7  is a diagram of a system  700  for up-mixing data from M channels to N channels in accordance with an exemplary embodiment of the present invention. System  700  converts stereo time domain data into 5.1 channel time domain data.  
         [0072]     System  700  includes time-frequency analysis units  702  and  704 , filter generation unit  706 , smoothing unit  708 , and frequency-time synthesis units  738  through  746 . System  700  provides improved spatial distinction and stability in an up-mix process through the use of a scalable frequency domain architecture which allows for high resolution frequency band processing, and through a filter generation method which extracts and analyzes important inter-channel spatial cues per frequency band to derive the spatial placement of a frequency element in the up-mixed 5.1 channel signal.  
         [0073]     System  700  receives a left channel stereo signal L(T) and a right channel stereo signal R(T) at time-frequency analysis units  702  and  704 , which convert the time domain signals into frequency domain signals. These time-frequency analysis units could be an appropriate filter bank, such as a finite impulse response (FIR) filter bank, a quadrature mirror filter (QMF) bank, a discrete Fourier transform (DFT), a time-domain aliasing cancellation (TDAC) filter bank, or other suitable filter bank. The output from time-frequency analysis units  702  and  704  are a set of frequency domain values covering a sufficient frequency range of the human auditory system, such as a 0 to 20 kHz frequency range where the analysis filter bank sub-band bandwidths could be processed to approximate psycho-acoustic critical bands, equivalent rectangular bandwidths, or some other perceptual characterization. Likewise, other suitable numbers of frequency bands and ranges can be used.  
         [0074]     The outputs from time-frequency analysis units  702  and  704  are provided to filter generation unit  706 . In one exemplary embodiment, filter generation unit  706  can receive an external selection as to the number of channels that should be output for a given environment, such as 4.1 sound channels where there are two front and two rear speakers can be selected, 5.1 sound systems where there are two front and two rear speakers and one front center speaker can be selected, 3.1 sound systems where there are two front and one front center speaker can be selected, or other suitable sound systems can be selected. Filter generation unit  706  extracts and analyzes inter-channel spatial cues such as inter-channel level difference (ICLD) and inter-channel coherence (ICC) on a frequency band basis. Those relevant spatial cues are then used as parameters to generate adaptive channel filters which control the spatial placement of a frequency band element in the up-mixed sound field. The channel filters are smoothed by smoothing unit  708  across both time and frequency to limit filter variability which could cause annoying fluctuation effects if allowed to vary too rapidly. In the exemplary embodiment shown in  FIG. 7 , the left and right channel L(F) and R(F) frequency domain signals are provided to filter generation unit  706  producing 5.1 channel filter signals H L (F), H R (F), H C (F), H LS (F), and H RS (F) which are provided to smoothing unit  708 .  
         [0075]     Smoothing unit  708  averages frequency domain components for each channel of the 5.1 channel filters across both the time and frequency dimensions. Smoothing across time and frequency helps to control rapid fluctuations in the channel filter signals, thus reducing jitter artifacts and instability that can be annoying to a listener. In one exemplary embodiment, time smoothing can be realized through the application of a first-order low-pass filter on each frequency band from the current frame and the corresponding frequency band from the previous frame. This has the effect of reducing the variability of each frequency band from frame to frame. In one exemplary embodiment, spectral smoothing can be performed across groups of frequency bins which are modeled to approximate the critical band spacing of the human auditory system. For example, if an analysis filter bank with uniformly spaced frequency bins is employed, different numbers of frequency bins can be grouped and averaged for different partitions of the frequency spectrum. In this exemplary embodiment, from zero to five kHz, five frequency bins can be averaged, from 5 kHz to 10 kHz, 7 frequency bins can be averaged, and from 10 kHz to 20 kHz, 9 frequency bins can be averaged, or other suitable numbers of frequency bins and bandwidth ranges can be selected. The smoothed values of H L (F), H R (F), H C (F), H LS (F), and H RS (F) are output from smoothing unit  708 .  
         [0076]     The source signals X L (F), X R (F), X C (F), X LS (F), and X RS (F) for each of the 5.1 output channels are generated as an adaptive combination of the stereo input channels. In the exemplary embodiment shown in  FIG. 7 , X L (F) is provided simply as L(F), implying that G L (F)=1 for all frequency bands. Likewise, X R (F) is provided simply as R(F), implying that G R (F)=0 for all frequency bands. X C (F) as output from summer  714  is computed as a sum of the signals L(F) multiplied by the adaptive scaling signal G C (F) with R(F) multiplied by the adaptive scaling signal 1−G C (F). X LS (F) as output from summer  720  is computed as a sum of the signals L(F) multiplied by the adaptive scaling signal G LS (F) with R(F) multiplied by the adaptive scaling signal 1−G LS (F) Likewise, X RS (F) as output from summer  726  is computed as a sum of the signals L(F) multiplied by the adaptive scaling signal G RS (F) with R(F) multiplied by the adaptive scaling signal 1−G RS (F). Notice that if G C (F)=0.5, G LS (F)=0.5, and G RS (F)=0.5 for all frequency bands, then the front center channel is sourced from an L(F)+R(F) combination and the surround channels are sourced from scaled L(F)−R(F) combinations as is common in traditional matrix up-mixing methods. The adaptive scaling signals G C (F), G LS (F), and G RS (F) can further provide a way to dynamically adjust the correlation between adjacent output channel pairs, whether they are lateral or depth-wise channel pairs. The channel source signals X L (F), X R (F), X C (F), X LS (F), and X RS (F) are multiplied by the smoothed channel filters H L (F), H R (F), H C (F), H LS (F), and H RS (F) by multipliers  728  through  736 , respectively.  
         [0077]     The output from multipliers  728  through  736  are then converted from the frequency domain to the time domain by frequency-time synthesis units  738  through  746  to generate output channels Y L (T), Y R (T), Y C (F), Y LS (F), and Y RS (T). In this manner, the left and right stereo signals are up-mixed to 5.1 channel signals, where inter-channel spatial cues that naturally exist or are intentionally encoded into the left and right stereo signals, such as by the down-mixing watermark process of  FIG. 1  or other suitable process, can be used to control the spatial placement of a frequency element within the 5.1 channel sound field produced by system  700 . Likewise, other suitable combinations of inputs and outputs can be used such as stereo to 4.1 sound, 4.1 to 5.1 sound, or other suitable combinations.  
         [0078]      FIG. 8  is a diagram of a system  800  for up-mixing data from M channels to N channels in accordance with an exemplary embodiment of the present invention. System  800  converts stereo time domain data into 7.1 channel time domain data.  
         [0079]     System  800  includes time-frequency analysis units  802  and  804 , filter generation unit  806 , smoothing unit  808 , and frequency-time synthesis units  854  through  866 . System  800  provides improved spatial distinction and stability in an up-mix process through a scalable frequency domain architecture, which allows for high resolution frequency band processing, and through a filter generation method which extracts and analyzes important inter-channel spatial cues per frequency band to derive the spatial placement of a frequency element in the up-mixed 7.1 channel signal.  
         [0080]     System  800  receives a left channel stereo signal L(T) and a right channel stereo signal R(T) at time-frequency analysis units  802  and  804 , which convert the time domain signals into frequency domain signals. These time-frequency analysis units could be an appropriate filter bank, such as a finite impulse response (FIR) filter bank, a quadrature mirror filter (QMF) bank, a discrete Fourier transform (DFT), a time-domain aliasing cancellation (TDAC) filter bank, or other suitable filter bank. The output from time-frequency analysis units  802  and  804  are a set of frequency domain values covering a sufficient frequency range of the human auditory system, such as a 0 to 20 kHz frequency range where the analysis filter bank sub-band bandwidths could be processed to approximate psycho-acoustic critical bands, equivalent rectangular bandwidths, or some other perceptual characterization. Likewise, other suitable numbers of frequency bands and ranges can be used.  
         [0081]     The outputs from time-frequency analysis units  802  and  804  are provided to filter generation unit  806 . In one exemplary embodiment, filter generation unit  806  can receive an external selection as to the number of channels that should be output for a given environment. For example, 4.1 sound channels where there are two front and two rear speakers can be selected, 5.1 sound systems where there are two front and two rear speakers and one front center speaker can be selected, 7.1 sound systems where there are two front, two side, two back, and one front center speaker can be selected, or other suitable sound systems can be selected. Filter generation unit  806  extracts and analyzes inter-channel spatial cues such as inter-channel level difference (ICLD) and inter-channel coherence (ICC) on a frequency band basis. Those relevant spatial cues are then used as parameters to generate adaptive channel filters which control the spatial placement of a frequency band element in the up-mixed sound field. The channel filters are smoothed by smoothing unit  808  across both time and frequency to limit filter variability which could cause annoying fluctuation effects if allowed to vary too rapidly. In the exemplary embodiment shown in  FIG. 8 , the left and right channel L(F) and R(F) frequency domain signals are provided to filter generation unit  806  producing 7.1 channel filter signals H L (F), H R (F), H C (F), H LS (F), H RS (F), H LB (F), and H RB (F) which are provided to smoothing unit  808 .  
         [0082]     Smoothing unit  808  averages frequency domain components for each channel of the 7.1 channel filters across both the time and frequency dimensions. Smoothing across time and frequency helps to control rapid fluctuations in the channel filter signals, thus reducing jitter artifacts and instability that can be annoying to a listener. In one exemplary embodiment, time smoothing can be realized through the application of a first-order low-pass filter on each frequency band from the current frame and the corresponding frequency band from the previous frame. This has the effect of reducing the variability of each frequency band from frame to frame. In one exemplary embodiment, spectral smoothing can be performed across groups of frequency bins which are modeled to approximate the critical band spacing of the human auditory system. For example, if an analysis filter bank with uniformly spaced frequency bins is employed, different numbers of frequency bins can be grouped and averaged for different partitions of the frequency spectrum. In this exemplary embodiment, from zero to five kHz, five frequency bins can be averaged, from 5 kHz to 10 kHz, 7 frequency bins can be averaged, and from 10 kHz to 20 kHz, 9 frequency bins can be averaged, or other suitable numbers of frequency bins and bandwidth ranges can be selected. The smoothed values of H L (F), H R (F), H C (F), H LS (F), H RS (F), H LB (F), and H RB (F) are output from smoothing unit  808 .  
         [0083]     The source signals X L (F), X R (F), X C (F), X LS (F), X RS (F), X LB (F), and X RB (F) for each of the 7.1 output channels are generated as an adaptive combination of the stereo input channels. In the exemplary embodiment shown in  FIG. 8 , X L (F) is provided simply as L(F), implying that G L (F)=1 for all frequency bands. Likewise, X R (F) is provided simply as R(F), implying that G R (F)=0 for all frequency bands. X C (F) as output from summer  814  is computed as a sum of the signals L(F) multiplied by the adaptive scaling signal G C (F) with R(F) multiplied by the adaptive scaling signal 1−G C (F). X LS (F) as output from summer  820  is computed as a sum of the signals L(F) multiplied by the adaptive scaling signal G LS (F) with R(F) multiplied by the adaptive scaling signal 1−G LS (F). Likewise, X RS (F) as output from summer  826  is computed as a sum of the signals L(F) multiplied by the adaptive scaling signal G RS (F) with R(F) multiplied by the adaptive scaling signal 1−G RS (F). Likewise, X LB (F) as output from summer  832  is computed as a sum of the signals L(F) multiplied by the adaptive scaling signal G LB (F) with R(F) multiplied by the adaptive scaling signal 1−G LB (F). Likewise, X RB (F) as output from summer  838  is computed as a sum of the signals L(F) multiplied by the adaptive scaling signal G RB (F) with R(F) multiplied by the adaptive scaling signal 1−G RB (F). Notice that if G C (F)=0.5, G LS (F)=0.5, G RS (F)=0.5, G LB (F)=0.5, and G RB (F)=0.5 for all frequency bands, then the front center channel is sourced from an L(F)+R(F) combination and the side and back channels are sourced from scaled L(F)−R(F) combinations as is common in traditional matrix up-mixing methods. The adaptive scaling signals G C (F), G LS (F), G RS (F), G LB (F), and G RB (F) can further provide a way to dynamically adjust the correlation between adjacent output channel pairs, whether they be lateral or depth-wise channel pairs. The channel source signals X L (F), X R (F), X C (F), X LS (F), X RS (F), X LB (F), and X RB (F) are multiplied by the smoothed channel filters H L (F), H R (F), H C (F), H LS (F), H RS (F), H LB (F), and H RB (F) by multipliers  840  through  852 , respectively.  
         [0084]     The output from multipliers  840  through  852  are then converted from the frequency domain to the time domain by frequency-time synthesis units  854  through  866  to generate output channels Y L (T), Y R (T), Y C (F), Y LS (F), Y RS (T), Y LB (T) and Y RB (T). In this manner, the left and right stereo signals are up-mixed to 7.1 channel signals, where inter-channel spatial cues that naturally exist or are intentionally encoded into the left and right stereo signals, such as by the down-mixing watermark process of  FIG. 1  or other suitable process, can be used to control the spatial placement of a frequency element within the 7.1 channel sound field produced by system  800 . Likewise, other suitable combinations of inputs and outputs can be used such as stereo to 5.1 sound, 5.1 to 7.1 sound, or other suitable combinations.  
         [0085]      FIG. 9  is a diagram of a system  900  for generating a filter for frequency domain applications in accordance with an exemplary embodiment of the present invention. The filter generation process employs frequency domain analysis and processing of an M channel input signal. Relevant inter-channel spatial cues are extracted for each frequency band of the M channel input signals, and a spatial position vector is generated for each frequency band. This spatial position vector is interpreted as the perceived source location for that frequency band for a listener under ideal listening conditions. Each channel filter is then generated such that the resulting spatial position for that frequency element in the up-mixed N channel output signal is reproduced consistently with the inter-channel cues. Estimates of the inter-channel level differences (ICLD&#39;s) and inter-channel coherence (ICC) are used as the inter-channel cues to create the spatial position vector.  
         [0086]     In the exemplary embodiment shown in system  900 , sub-band magnitude or energy components are used to estimate inter-channel level differences, and sub-band phase angle components are used to estimate inter-channel coherence. The left and right frequency domain inputs L(F) and R(F) are converted into a magnitude or energy component and phase angle component where the magnitude/energy component is provided to summer  902  which computes a total energy signal T(F) which is then used to normalize the magnitude/energy values of the left M L (F) and right channels M R (F) for each frequency band by dividers  904  and  906 , respectively. A normalized lateral coordinate signal LAT(F) is then computed from M L (F) and M R (F), where the normalized lateral coordinate for a frequency band is computed as: 
 
 LAT ( F )= M   L ( F )* X   MIN   +M   R ( F )* X   MAX  
 
         [0087]     Likewise, a normalized depth coordinate is computed from the phase angle components of the input as: 
 
 DEP ( F )= Y   MAX −0.5*( Y   MAX   −Y   MIN )*sqrt([COS( /   L ( F ))−COS( /   R ( F ))]ˆ2+[SIN( /   L ( F ))−SIN( /   R ( F ))]ˆ2 
 
         [0088]     The normalized depth coordinate is calculated essentially from a scaled and shifted distance measurement between the phase angle components  / L(F) and  / R(F). The value of DEP(F) approaches 1 as the phase angles  / L(F) and  / R(F) approach one another on the unit circle, and DEP(F) approaches 0 as the phase angles  / L(F) and  / R(F) approach opposite sides of the unit circle. For each frequency band, the normalized lateral coordinate and depth coordinate form a 2-dimensional vector (LAT(F), DEP(F)) which is input into a 2-dimensional channel map, such as those shown in the following  FIGS. 10A through 10E , to produce a filter value H i (F) for each channel i. These channel filters H i (F) for each channel i are output from the filter generation unit, such as filter generation unit  606  of  FIG. 6 , filter generation unit  706  of  FIG. 7 , and filter generation unit  806  of  FIG. 8 .  
         [0089]      FIG. 10A  is a diagram of a filter map for a left front signal in accordance with an exemplary embodiment of the present invention. In  FIG. 10A , filter map  1000  accepts a normalized lateral coordinate ranging from 0 to 1 and a normalized depth coordinate ranging from 0 to 1 and outputs a normalized filter value ranging from 0 to 1. Shades of gray are used to indicate variations in magnitude from a maximum of 1 to a minimum of 0, as shown by the scale on the right-hand side of filter map  1000 . For this exemplary left front filter map  1000 , normalized lateral and depth coordinates approaching (0, 1) would output the highest filter values approaching 1.0, whereas the coordinates ranging from approximately (0.6, Y) to (1.0, Y), where Y is a number between 0 and 1, would essentially output filter values of 0.  
         [0090]      FIG. 10B  is a diagram of exemplary right front filter map  1002 . Filter map  1002  accepts the same normalized lateral coordinates and normalized depth coordinates as filter map  1000 , but the output filter values favor the right front portion of the normalized layout.  
         [0091]      FIG. 10C  is a diagram of exemplary center filter map  1004 . In this exemplary embodiment, the maximum filter value for the center filter map  1004  occurs at the center of the normalized layout, with a significant drop off in magnitude as coordinates move away from the front center of the layout towards the rear of the layout.  
         [0092]      FIG. 10D  is a diagram of exemplary left surround filter map  1006 . In this exemplary embodiment, the maximum filter value for the left surround filter map  1006  occurs near the rear left coordinates of the normalized layout and drop in magnitude as coordinates move to the front and right sides of the layout.  
         [0093]      FIG. 10E  is a diagram of exemplary right surround filter map  1008 . In this exemplary embodiment, the maximum filter value for the right surround filter map  1008  occurs near the rear right coordinates of the normalized layout and drop in magnitude as coordinates move to the front and left sides of the layout.  
         [0094]     Likewise, if other speaker layouts or configurations are used, then existing filter maps can be modified and new filter maps corresponding to new speaker locations can be generated to reflect changes in the new listening environment. In one exemplary embodiment, a 7.1 system would include two additional filter maps with the left surround and right surround being moved upwards in the depth coordinate dimension and with the left back and right back locations having filter maps similar to filter maps  1006  and  1008 , respectively. The rate at which the filter factor drops off can be changed to accommodate different numbers of speakers.  
         [0095]     Although exemplary embodiments of a system and method of the present invention have been described in detail herein, those skilled in the art will also recognize that various substitutions and modifications can be made to the systems and methods without departing from the scope and spirit of the appended claims.