Abstract:
A method for processing and transducing audio signals. An audio system has a first audio signal and a second audio signal that have amplitudes. A method for processing the audio signals includes dividing the first audio signal into a first spectral band signal and a second spectral band signal; scaling the first spectral band signal by a first scaling factor proportional to the amplitude of the second audio signal; and scaling the first spectral band signal by a second scaling factor to create a second signal portion. Other portions of the disclosure include application of the signal processing method to multichannel audio systems, and to audio systems having different combinations of directional loudspeakers, full range loudspeakers, and limited range loudspeakers.

Description:
CROSS-REFERENCE TO RELATED APPLICATIONS 
   Not applicable. 
   STATEMENT REGARDING FEDERALLY SPONSORED RESEARCH OR DEVELOPMENT 
   Not applicable. 
   The invention relates to audio signal processing in audio systems having multiple directional channels, such as so-called “surround systems,” and more particularly to audio signal processing that can adapt multiple directional channel systems to audio systems having fewer or more loudspeaker locations than the number of directional channels. 
   BACKGROUND OF THE INVENTION 
   For background, reference is made to surround sound systems and U.S. Pat. Nos. 5,809,153 and 5,870,484. It is an important object of the invention to provide an improved audio signal processing system for the processing of directional channels in a multi-channel audio system. 
   BRIEF SUMMARY OF THE INVENTION 
   According to the invention, an audio system has a first audio signal and a second audio signal having amplitudes. A method for processing the audio signals includes dividing the first audio signal into a first spectral band signal and a second spectral band signal; scaling the first spectral band signal by a first scaling factor to create a first signal portion, wherein the first scaling factor is proportional to the amplitude of the second audio signal; and scaling the first spectral band signal by a second scaling factor to create a second signal portion. 
   In another aspect of the invention. An audio system has a first audio signal, a second audio signal and a directional loudspeaker unit. A method for processing the audio signals includes electroacoustically directionally transducing the first audio signal to produce a first signal radiation pattern; electroacoustically directionally transducing the second audio signal to produce a second signal radiation pattern, wherein the first signal radiation pattern and the second signal radiation pattern are alternatively and user selectively similar or different. 
   In another aspect of the invention, An audio system has a first audio signal, a second audio signal, and a third audio signal that is substantially limited to a frequency range having a lower limit at a frequency that has a corresponding wavelength that approximates the dimensions of a human head. The audio system further includes a directional loudspeaker unit, and a loudspeaker unit, distinct from the directional loudspeaker unit. A method for processing the audio signals, includes electroacoustically directionally transducing by the directional loudspeaker unit the first audio signal to produced a first radiation pattern; electroacoustically directionally transducing by the directional loudspeaker unit the second audio signal to produce a second radiation pattern; and electroacoustically transducing by the distinct loudspeaker unit the third audio signal. 
   In another aspect of the invention, an audio system has a plurality of directional channels. A method for processing audio signals respectively corresponding to each of the plurality of channels includes dividing a first audio signal into a first audio signal first spectral band signal and a first audio signal second spectral band signal; scaling the first audio signal first spectral band signal by a first scaling factor to create a first audio signal first spectral band first portion signal; scaling the first spectral band signal by a second scaling factor to create a first audio signal first spectral band second portion signal; dividing a second audio signal into a second audio signal first spectral band signal and a second audio signal second spectral band signal; scaling the second audio signal first spectral band signal by a third scaling factor to create a second audio signal first spectral band first portion signal; and scaling the second audio signal first spectral band signal by a fourth scaling factor to create a second audio signal first spectral band second portion signal. 
   In another aspect of the invention, a method for processing an audio signal includes filtering the signal by a first filter that has a frequency response and time delay effect similar to the human head to produce a once filtered signal. The method further includes filtering the once filtered audio signal by a second filter, the second filter having a frequency response and time delay effect inverse to the frequency and time delay effect of a human head on a sound wave. 
   In another aspect of the invention, an audio system has a plurality of directional channels, a first audio signal and a second audio signal, the first and second audio signals representing adjacent directional channels on the same lateral side of a listener in a normal listening position. A method for processing the audio signals includes dividing the first audio signal into a first spectral band signal and a second spectral band signal; scaling the first spectral band signal by a first time varying calculated scaling factor to create a first signal portion; and scaling the first spectral band signal by a second time varying calculated scaling factor to create a second signal portion. 
   In still another aspect of the invention, an audio system has an audio signal, a first electroacoustical transducer designed and constructed to transduce sound waves in a frequency range having a lower limit, and a second electroacoustical transducer designed and constructed to transduce sound waves in a frequency range having a second transducer lower limit that is lower than the first transducer lower limit. A method for processing audio signals, includes dividing the audio signal into a first spectral band signal and a second spectral band signal; scaling the first spectral band signal by a first scaling factor to create a first portion signal; scaling the first spectral band signal by a second scaling factor to create a second portion signal; transmitting the first portion to the first electroacoustical transducer for transduction; and transmitting said second portion signal to said second electroacoustical transducer for transduction. 

   
     Other features, objects, and advantages will become apparent from the following detailed description, which refers to the following drawing in which: 
     BRIEF DESCRIPTION OF THE SEVERAL VIEWS OF THE DRAWING 
       FIGS. 1   a – 1   c  are diagrammatic views of configurations of loudspeaker units for use with the invention; 
       FIG. 2   a  is a block diagram of an audio signal processing system incorporating the invention; 
       FIGS. 2   b  and  2   c  are block diagrams of audio signal processing systems for creating directional channels in accordance with the invention; 
       FIGS. 3   a – 3   d  are block diagrams of alternate directional processors for use in the audio signal processing system of  FIG. 2   a;    
       FIG. 4  is a block diagram of some of the components of the directional processors of  FIGS. 3   a – 3   c;    
       FIG. 5  is a diagrammatic view of a configuration of loudspeakers helpful in explaining aspects of the invention; 
       FIG. 6  is of a configuration of loudspeaker units for use with another aspect of the invention; 
       FIG. 7  is a block diagram of an audio signal processing system incorporating another aspect of the invention; 
       FIG. 8  is a block diagram of a directional processor for use with the audio signal processing system of  FIG. 7 ; 
       FIG. 9  is a block diagram of an alternate directional processor for use with the audio signal processing system of  FIG. 7 ; 
       FIGS. 10   a – 10   c  are top diagrammatic views of some of the components of an audio system for describing another feature of the invention; and 
       FIG. 11  is a block diagram of a component of  FIGS. 3   a – 3   d for creating directional channels in accordance with the invention; 
   

   DETAILED DESCRIPTION 
   With reference now to the drawing and more particularly to  FIGS. 1   a – 1   c , there are shown top diagrammatic views of three configurations of surround sound audio loudspeaker units according to the invention. In  FIG. 1   a , two directional arrays each including two full range (as defined below in the discussion of  FIGS. 2   a – 2   c ) acoustical drivers are positioned in front of a listener  14 . A first array  10  including acoustical drivers  11  and  12  may be positioned to the listener&#39;s left and a second array  15 , including acoustical drivers  16  and  17  may be positioned to the listener&#39;s right. In  FIG. 1   b , two directional arrays each including two full range acoustical drivers are positioned in front of a listener  14 . A first array  10  including acoustical drivers  11  and  12  may be positioned to the listener&#39;s left and a second array  15 , including acoustical drivers  16  and  17  may be positioned to the listener&#39;s right. In addition, a first limited range (as defined below in the discussion of  FIGS. 2   a – 2   c ) acoustical driver  22  is positioned behind the listener, to the listener&#39;s left, and a second limited range acoustical driver  24  is positioned behind the listener to the listener&#39;s right. In  FIG. 1   c , two directional arrays each including two full range acoustical drivers are positioned in front of a listener  14 . A first array  10  including acoustical drivers  11  and  12  may be positioned to the listener&#39;s left and a second array  15 , including acoustical drivers  16  and  17  may be positioned to the to the listener&#39;s right. In addition, a first full range acoustical driver  28  is positioned behind the listener, to the listener&#39;s left, and a second limited range acoustical driver  30  is positioned behind the listener to the listener&#39;s right. Other surround sound loudspeaker systems may have loudspeaker units in additional locations, such as directly in front of listener  14 . Surround sound systems may radiate sound waves in a manner that the source of the sound may be perceived by the listener to be in a direction (for example direction X) relative to the listener at which there is no loudspeaker unit. Surround sound systems may further attempt to radiate sound waves in a manner such that the source of the sound may be perceived by the listener to be moving (for example in direction Y-Y′) relative to the viewer 
   Referring to  FIG. 2   a , there is shown a block diagram of an audio signal processing system for providing audio signals for the loudspeaker units of  FIGS. 1   a – 1   c . An audio signal source  32  is coupled to a decoder  34  which decodes the audio source from the audio signal source into a plurality of channels, in this case a low frequency effects (LFE) channel, and bass channel, and a number of directional channels, including a left surround (LS) channel, a left (L) channel, a left center (LC) channel, a right center (RC) channel, a right (R) channel, and a right surround (RS) channel. Other decoding systems may output a different set of channels. In some systems, the bass channel is not broken out separately from the directional channels, but instead remains combined with the directional channels. In other systems, there may be a single center (C) channel, instead of the RC and LC channels, or there may be a single surround channel. An audio system according to the invention may be used with any combination of directional channels, either by adapting the signal processing to the channels, or by decoding the directional channels to produce additional directional channels. One method of decoding a single C channel into an RC channel and an LC channel is shown in  FIG. 2   b . The C channel is split into an LC channel and an RC channel and the LC and the RC channel are scaled by a factor, such as 0.707. Similarly, a method of decoding a single S channel into an RS channel and an LS channel is shown in  FIG. 2   c . The S channel is split into an RS channel and an LS channel, and the RS channel and LS channel are scaled by a factor, such as 0.707. If the audio input signal has no surround channel or channels, there are several known methods for synthesizing surround channels from existing channels, or the system may be operated without surround sound. 
   Some surround sound systems have a separate low frequency unit for radiating low frequency spectral components and “satellite” loudspeaker units for radiating spectral components above the frequencies radiated by the low frequency units. Low frequency units are referred to by a number of names, including “subwoofers” “bass bins” and others. 
   In surround sound systems having both an LFE channel and a bass channel, the LFE and bass channels may be combined and radiated by the low frequency unit, as shown in  FIG. 2   a . In surround systems not having a combined bass channel, each directional channel, including the bass portion of each directional channel) may be radiated by separate directional loudspeaker units, with only the LFE radiated by the low frequency unit. Still other surround systems may have more than one low frequency unit, one for radiating bass frequencies and one for radiating the LFE channel. “Full range” as used herein, refers to audible spectral components having frequencies above those radiated by a low frequency unit. If an audio system has no low frequency unit, “full range” refers to the entire audible frequency spectrum. “Directional channel” as used herein is an audio channel that contains audio signals that are intended to be transduced to sound waves that appear to come from a specific direction. LFE channels and channels that have combined bass signals from two or more directional channels are not, for the purposes of this specification, considered directional channels. 
   The directional channels, LS, L, LC, RC, R, and RS are processed by directional processor  36  to produce output audio signals at output signal lines  38   a – 38   f  for the acoustical drivers of the audio system. The signals output by directional processor  36  and the low frequency unit signal in signal line  40  may then be further processed by system equalization (EQ) and dynamic range control circuitry  42 . (System EQ and dynamic range control circuitry is shown to illustrate the placement of elements typical to audio processing circuitry, but does not perform a function relevant to the invention. Therefore, system EQ and dynamic range control circuitry  42  are not shown in subsequent figures and its function will not be further described. Other audio processing elements, such as amplifiers that are not germane to the present invention are not shown or described). The directional channels are then transmitted to the acoustical drivers for transduction to sound waves. The signal line  38   a  designated “left front (LF) array driver A” is directed to acoustical driver  12  of array  10  (of  FIGS. 1   a – 1   c ); the signal line  38   b  designated “left front (LF) array driver B” is directed to acoustical driver  11  of array  10  (of  FIGS. 1   a – 1   c ); the signal line  38   c  designated “right front (RF) array driver A” is directed to acoustical driver  17  of array  15  (of  FIGS. 1   a – 1   c ); and the signal line  38   d  designated “right front (RF) array driver B” is directed to acoustical driver  16  of array  15  (of  FIGS. 1   a – 1   c ). The signal line  38   e  designated “left surround (LS) driver” is directed to limited range acoustical driver  22  of  FIG. 1   b  or acoustical driver  28  of  FIG. 1   c  as will be explained below, and the signal line  38   f  designated “right surround (RS) driver” is directed to acoustical driver  24  of  FIG. 1   b  or acoustical driver  30  of  FIG. 1   c , as will also be explained below. In some implementations, there is no output signal from LS output terminal  38   e  or RS output terminal  38   f  or both. In other implementations one or both of LS output terminal  38   e  or RS output terminal  38   f  may be absent entirely, as will be explained below. 
   Referring now to  FIGS. 3   a – 3   d , there are shown four block diagrams of audio directional processor  36  for use with surround sound loudspeaker systems as shown in  FIGS. 1   a – 1   c .  FIGS. 3   a – 3   d  show the portion of the directional processor for the LC, LS, and L channels. In each of the implementations, there is a mirror image for processing the RC, RS, and R channels. In  FIGS. 3   a – 3   d , like reference numerals refer to like elements performing like functions. 
     FIG. 3   a  shows the logical arrangement of directional processor  36  for a configuration having no rear speakers. In  FIG. 3   a , the L channel is coupled to presentation mode processor  102  and to level detector  44 . One output terminal  35  of presentation mode processor  102 , designated L′, is coupled to summer  47 . The operation of presentation mode processor  102  will be described below in the discussion of  FIG. 11 . LS channel is coupled to level detector  44  and frequency splitter  46 . Level detector  44  provides front/rear scaler  48 , front head related transfer function (HRTF) filters and rear HRTF filters with signal levels to facilitate the calculation of filter coefficients as will be described below. Frequency splitter  46  separates the signal into a first frequency band including signals below a threshold frequency and a second frequency band including signals above the threshold frequency. The threshold frequency is a frequency that corresponds to a wavelength that approximates dimensions of a human head. A convenient frequency is 2 kHz, which corresponds to a wavelength of about 6.8 inches. Hereinafter, the portion of the surround signal above the threshold frequency will be referred to as “high frequency surround signal” and the portion of the surround signal below the threshold frequency will be referred to as “low frequency surround signal.” The low frequency surround signal is input by signal path  43  to summer  54 , or alternatively to summer  47  as will be explained in the discussion of  FIG. 3   d . The high frequency surround signal is input by signal path  45  to front/rear scaler  48 , which splits the high frequency surround signal into a “front” portion and a “rear” portion in a manner that will be described below in the discussion of  FIG. 4 . The “front” portion of the high frequency surround signal is transmitted by signal line  49  to front head related transfer function (HRTF) filter  50 , where it is modified in a manner that will be described below in the discussion of  FIG. 4 . Modified front high frequency surround is then optionally delayed by five ms by delay  52  and input to summer  54 . “Rear” portion of the high frequency surround signal is transmitted by signal line  51  to rear HRTF filter  56 , where it is modified in a manner that will be described below in the discussion of  FIG. 4 . The modified rear portion is then optionally delayed by ten ms by delay  58 , and summed with front portion and low frequency surround signal at summer  54 . The summed front, rear, and low frequency surround portions are modified by front speaker placement compensator  60  (which will be further explained below following the discussion of  FIGS. 4 and 5 ) and input to summer  47 , so that at summer  47  the L channel, the low frequency surround, and the modified high frequency surround are summed. The output signal of summer  47  may then be adjusted by a left/right balance control represented by multiplier  57  and is then input subtractively through time delay  61  to summer  62  and additively to summer  58 . LC channel is coupled to presentation mode processor  102 . Output terminal  37 , designated LC′ of presentation mode processor  102  is coupled additively to summer  62  and subtractively through time delay  64  to summer  58 . Output signal of summer  58  is transmitted to acoustical driver  11  (of  FIGS. 1 and 2 ). Output signal of summer  62  is transmitted to acoustical driver  12  (of  FIGS. 1 and 2 ). Time delays  61  and  64  facilitate the directional radiation of the signals combined at summer  47 . If desired, the outputs of time delay  61  and  64  can be scaled by a factor such as 0.631 to improve directional radiation performance. Directional radiation using time delays is discussed in U.S. Pat. Nos. 5,809,153 and 5,870,484 and will be further discussed below. 
     FIG. 3   b  shows directional processor  36  for a configuration having a limited range rear speaker, that is, a speaker that is designed to radiate frequencies above the threshold frequency. In the circuitry of  FIG. 3   b , summer  54  of  FIG. 3   a  is not present. Instead, front HRTF filters and optional five ms delay are coupled through front speaker placement compensator  60  to summer  47  and rear HRTF filters and optional ten ms delay are coupled to rear speaker placement compensator  66 , which is in turn coupled to limited range acoustical driver  22  of  FIGS. 1 and 2 . 
     FIG. 3   c  shows directional processor  36  for a configuration having a full range rear speaker, that is, a speaker that is designed to radiate the full audible spectrum of frequencies above the frequencies radiated by a low frequency unit. The circuitry of  FIG. 3   c  is similar to the circuitry of  FIG. 3   b , but low frequency surround signal output of frequency splitter  46  is summed with output signal of rear HRTF filter and optional ten ms delay  58  at summer  70 , which is output to full-range acoustical driver  28 . 
     FIG. 3   d  shows directional processor  36  that can be used with no rear speaker, with a limited-range rear speaker, or with a full range rear speaker.  FIG. 3   d  includes a switch  68  and summer  69  arranged so that with switch  68  in a closed position, the low frequency surround signal is directed to summer  70 . With switch  68  in an open position, the low frequency is directed to summer  47  for radiation from the front speaker array.  FIG. 3   d  further includes a switch  72  and summer  73 , arranged so that with switch  72  in an open position, the output signal from summer  70  is directed to rear speaker placement compensator  66  for radiation from a rear speaker. With switch  72  in a closed position, the output signal from summer  70  is directed to summer  54 . With switch  72  in an open position and  68  in an open position, the circuitry of  FIG. 3   d  becomes the circuitry of  FIG. 3   b . With switch  72  in an open position and switch  68  in a closed position, the circuitry of  FIG. 3   d  becomes the circuitry of  FIG. 3   c . With switch  72  in a closed position and switch  68  in a closed position, the circuitry of  FIG. 3   d  (since the effect of the signal on line  43  being coupled to summer  54  as in the embodiment of  FIG. 3   d  is functionally equivalent to the signal on line  43  being directly connected to summer  54  as in the embodiment of  FIG. 3   a ) becomes the circuitry of  FIG. 3   a . With switch  72  in a closed position and switch  68  in an open position, the circuitry of  FIG. 3   d  becomes the circuitry of  FIG. 3   a , with the low frequency surround signal directed to summer  47 . 
   In operation, switch  72  is set to the open position when there is a rear speaker and to the closed position when there is no rear speaker. Switch  68  is set to the open position for a limited range rear speaker and to the closed position for a full range rear speaker. Logically if switch  72  is set to the closed position, the position of switch  68  should be irrelevant. It was stated in the preceding paragraph that that if switch  72  is in the closed position, the low frequency surround signal may be summed with the high frequency surround signal before or after the front speaker placement compensator depending on the position of switch  68 . However, as will be explained below in the discussion of  FIG. 4 , the front and rear speaker placement compensators have little effect on frequencies below the threshold frequency, so it does not matter whether the low frequency surround is summed with the high frequency surround before or after the front speaker placement compensator. Alternatively, switches  68  and  72  could be linked so that if switch  72  is in the closed position, switch  68  would automatically be set to the open or closed position as desired. 
   In an exemplary embodiment, the directional processor  36  is implemented as digital signal processors (DSPs) executing instructions with digital-to-analog and analog-to-digital converters as necessary. In other embodiments, the directional processor  36  may be implemented as a combination of DSPs, analog circuit elements, and digital-to-analog and analog-to-digital converters as necessary. 
     FIG. 4 , shows the frequency splitter  46 , the front/rear scaler  48 , the front HRTF filter  50  and the rear HRTF filter  56  of  FIGS. 3   a – 3   c  in greater detail. Frequency splitter  46  is implemented as a high pass filter  74  and a summer  76 . High pass filter  74  and summer  76  are arranged so that high pass filtered LS channel is combined subtractively with the LS channel signal so that the low frequency surround is output on line  43 . The high pass filter  74  is directly coupled to signal line  45 , so that the high frequency surround is output on signal line  45 . Front/rear scaler is implemented as a summer  78  and a multiplier  80 . Multiplier  80  scales the signal by a factor that is related to the relative amplitudes of the signals in the LS channel and the L channel. In the embodiment of  FIG. 4 , the factor is 
                    LS   _                   LS   _          +          L   _              .         
Summer  78  and multiplier  80  are arranged so that scaled signal is combined subtractively with the unscaled signal and output on signal line  49  so that the signal on signal line  49  is the input signal scaled by
 
             (     1   -            LS   _                   LS   _          +          L   _                )     .         
Multiplier is directly coupled to signal line  51  so that the signal on the signal line  51  is the input signal scaled by
 
                    LS   _                   LS   _          +          L   _              .         
It can be seen that if |  LS | approaches zero, the portion of the input signal that is directed to signal line  49  approaches one and the portion of the signal that is directed to signal line  51  approaches zero. Similarly if |  LS | is much greater than |  L |, the portion of the input signal that is directed to signal line  49  approaches zero and the portion of the input signal that is directed to signal line  51  approaches one. If |  LS | and |  L | are approximately equal, then the portion of the input signal that is directed to signal line  49  is approximately equal to the portion of the input signal that is directed to signal line  51 . The effect of the front/rear scaler is to orient the apparent source of a sound relative to the listener. If |  L | is greater than |  LS |, a greater portion of the high frequency surround signal will be directed to the front speaker unit, and the apparent source of the sound is toward the front. If |  LS | is greater than |  L |, a greater portion of the high frequency surround signal will be directed to the rear speaker unit (or in the absence of a rear speaker unit, be processed so that it will appear to come from the rear) and the apparent source of the sound is toward the rear. If |  LS | and |  L | are relatively equal, then an approximately equal portion of the high frequency surround signal will be directed to the front and rear loudspeaker units, and the apparent source of the sound is to the side. The values |  L | and |  LS | are made available to multiplier  80  by level detectors  44  of  FIGS. 3   a – 3   d . Scaling factors
 
                  LS   _                   LS   _          +          L   _                  
and
 
           (     1   -            LS   _                   LS   _          +          L   _                )         
may be calculated as often as practical. In one implementation, the scaling factors are recalculated at five millisecond intervals.
 
   Front HRTF filter  50  may be implemented as, in order in series, a multiplier  82 , a first filter  84  representing the frequency shading effect of the head (hereinafter the head shading filter), a second filter  86  representing the diffraction path delay of the head (hereinafter the head diffraction path delay filter), a third filter  88  representing the diffraction path delay of the pinna (hereinafter the pinna diffraction path delay filter), and a summer  90 . Summer  90  sums the output signal from pinna diffraction path delay filter  88  with the output of head diffraction path delay filter  86 , the output of head frequency shading filter  84 , and the unmultiplied input signal of front HRTF filter  50 . Rear HRTF filter  56  may be implemented as, in order in series, multiplier  82 , head frequency shading filter  84 , pinna diffraction path delay filter  88 , head diffraction path delay  86 , and a fourth filter  92  representing the frequency shading effect of the rear surface of the pinna (hereinafter the pinna rear frequency shading filter), and a summer  94 . Summer  94  sums the output of pinna rear frequency shading filter  92 , output of head diffraction path delay filter  86 , pinna diffraction path delay filter  88 , and the unmultiplied input signal of the rear HRTF filter  56 . In one implementation, the signal from head diffraction path delay  86  to summer  94  is scaled by a factor of 0.5 and the signal from pinna rear frequency shading filter  92  to summer  94  is scaled by a factor of two. 
   Head frequency shading filter  84  is implemented as a first order high pass filter with a single real pole at −2.7 kHz; head diffraction path delay filter  86  is implemented as a fourth order all-pass network with four real poles at −3.27 kHz and four real zeros at 3.27 kHz; pinna diffraction delay filter  88  is implemented as a fourth order all-pass network with four real poles at −7.7 kHz and four real zeros at 7.7 kHz; and pinna rear frequency shading filter  92  is implemented as a first order high pass filter with a single real pole at −7.7 kHz. Multiplier  82  scales the input signal by a factor of 
             Y       (     Y   -          LS   _            )     +     (     Y   -          L   _            )     +   Y       ,         
where Y is the larger of |  L | and |  LS |. The values |  L | and |  LS | are made available to multiplier  80  by level detectors  44  of  FIGS. 3   a – 3   d . “Pinna” as used herein refers to the auricle portion of the external ear as shown on p. 1367  Gray&#39;s Anatomy,  38 th    Edition , Churchill Livingston 1995. “Pinna rear” or “rear surface of the pinna” as used herein, refers to the anterior surface or the external ear, or the external ear as viewed in the direction of the arrow in Appendix 1. The pinna is an acoustic surface for sounds from all directions, while the rear pinna is an acoustic surface only for sounds from directions ranging from the side to the rear.
 
   Filters having characteristics other than those described above (including a filter having a flat frequency response, such as a direct electrical connection) may be used in place of the filter arrangements shown in  FIG. 4  and described in the accompanying portion of the disclosure. 
     FIG. 5  illustrates the purpose of the front speaker placement compensator  60  and the rear speaker placement compensator  66  of  FIGS. 3   a – 3   d . Front speaker placement compensator is implemented as a filter or series of filters that has an effect that is inverse to the front HRTF filter  50  when front HRTF filter  50  acts upon a signal that radiated from a first specific angle. Similarly, the rear speaker placement compensator is implemented as a filter or series of filters that has an effect that is inverse to the rear HRTF filter  56  when rear HRTF filter  56  acts upon a signal that radiated from a second specific angle. 
     FIG. 5  shows for explanation purposes a sound system according to the configuration of  FIG. 3   b , with desired apparent source of a sound is at point Z, which is oriented at an angle θ relative to a listener  14 . All angles in  FIG. 5  lie in a horizontal plane which includes the entrances to the ear canals of listener  14 . The reference line for the angles is a line passing through the points that are equidistant from the entrances to the ear canals of listener  14 . Angles are measured counter-clockwise from the front of the listener  14 . Placement of the apparent source of the sound at point Z is accomplished in part by the front/rear scaler  48  of  FIGS. 3   a – 3   c  and  FIG. 4 . Front/rear scaler directs more of the high frequency surround signal to the front array  10  than to the rear speaker unit, so that the apparent source of the sound is somewhat forward. Placement of the apparent source of the sound at point Z is further accomplished by the front and rear HRTF filters  50  and  56  (of  FIGS. 3   a – 3   d ) respectively. Front and rear HRTF filters  50  and  56  alter the audio signals so that when the signals are transduced to sound waves by front array  10  and limited range acoustical driver  22 , the sound waves will have the frequency content and phase relationships as if the sound waves had originated at point Z and had been modified by the head  96  and pinna  98  of listener  14 . However, when the sound waves are actually transduced by front array  10  and rear limited range acoustical driver  22 , the frequency content and the phase relationships of the sound waves will be modified by the physical head  96  and pinna  98  of listener  14 , so that in effect the sound waves that reach the ear canal have the frequency content and phase relationships that have been twice modified by the head and pinna of the listener over angle φ 1 . Front speaker placement compensator  60  modifies the audio signal so that when it is transduced by front array  10 , the sound waves will not have the change in frequency content and phase relationships attributable to the angle φ 1 , leaving in the audio signal the change in frequency and phase relationships attributable to the difference between angle θ and angle φ 1 . Then, when the sound waves are transduced by front array  10  and modified by the head and pinna of the listener, the sound waves that reach the ear canal will have the frequency content and phase relationships as a sound from a source at angle θ. Similarly, the rear speaker placement compensator  66  modifies the audio signal so that when it is transduced by rear limited range acoustical driver  22 , the sound waves will not have the change in frequency content and phase relationships attributable to the angle φ 2 , leaving the change in frequency and phase relationships attributable to the difference between angle θ and angle φ 2 . Then, when the sound is transduced by rear limited range acoustical driver  22 , the sound waves that reach the ear canal will have the same frequency content and phase relationships as a sound from a source at angle θ. If the speaker configuration is the configuration of  FIG. 3   a  the same explanation applies. However the configuration having the limited range rear speaker was chosen to illustrate that the front and rear HRTF filters  50  and  56  and the front and rear speaker placement compensators  60  and  66 , all have little effect below frequencies having corresponding wavelengths that approximate the dimensions of the head, for example 2 kHz. In one embodiment, the angles φ 1  and φ 2  are measured and input into audio system so that speaker placement compensators  60  and  66  calculate using the precise angle. One technique for measuring angles φ 1  and φ 2  is to physically measure them. In a second embodiment, speaker placement compensators are set to pre-selected typical values of angles φ 1  and φ 2  (for example 30 degrees and 150 degrees). This second embodiment gives acceptable results, but does not require actual measurement of the speaker placement angles and may require somewhat less complex computing in speaker placement compensators  60  and  66 . 
   Speaker placement compensators  60  and  66  may be implemented as filters having the inverse effect as front and rear HRTF filters, respectively, evaluated for the selected values of angles φ 1  and φ 2 , by using values derived from the relationships 
               ϕ   1     =         arcsin   ⁡     [     1   -     [       Y   -          LS   _          +   Y   -          L   _            Y     ]       ]       ⁢           ⁢   and   ⁢           ⁢     ϕ   2       =     arcsin   ⁡     [     1   -     [       Y   -          LS   _          +   Y   -          L   _            Y     ]       ]           ,         
respectively.
 
   If some filter arrangement other than the filter arrangement of  FIG. 4  is used for the front HRTF filter  50  and the rear HRTF filter  56 , the front speaker placement compensator  60  and the rear speaker placement compensator  66  may be modified accordingly. If HRTF filters  50  and  56  have a flat frequency response, the front speaker placement compensator  60  and rear speaker placement compensator  66  may be replaced by a filter having a flat frequency response (such as a direct electrical connection). 
   Referring now to  FIG. 6 , there is shown an example of two more acoustical loudspeaker configurations for illustrating another feature of the invention. In  FIG. 6 , there is an acoustical driver array  10 , similar to the acoustical driver array  10  of  FIGS. 1   a – 1   c , placed at a point displaced by 30 degrees from listener  14 . In addition, there are limited range acoustical drivers, similar to the limited range acoustical drivers  22  of  FIGS. 1   a – 1   c , at 60 degrees, 90 degrees, 120 degrees, and 150 degrees OR full range acoustical drivers  28  similar to the full range acoustical drivers  28  of  FIGS. 1   a – 1   c . The limited range acoustical drivers are designated  22 - 60 ,  22 - 90 ,  22 - 120 , and  22 - 150 , respectively, to indicate the angular position of the limited range acoustical driver. The alternate full range acoustical drivers are designated  28 - 60 ,  28 - 90 ,  28 - 120 , and  28 - 150 , respectively, to indicate the angular position of the limited range acoustical driver. All angles in  FIG. 6  lie in the horizontal plane that includes the entrances to the ear canal of listener  14 . The reference line for the angles is a line passing through the points that are equidistant from the entrances to the listener&#39;s ear canals. The angles for the acoustical driver units on the left of listener  14  are measured counterclockwise from the reference line in front of the listener. The angles for the acoustical driver units on the right of listener  14  are measured clockwise from the reference line in front of the listener. There may also be other acoustical driver units, such as a center channel acoustical driver unit or a low frequency unit, which are not shown in this view. 
     FIG. 7  shows a block diagram of an audio signal processing system for providing audio signals for the loudspeaker units of  FIG. 6 . An audio signal source  32  is coupled to a decoder  34  which decodes the audio source from the audio signal source into a plurality of channels, in this case a low frequency effects (LFE) channel, and bass channel, and a number of directional channels, including a left (L) channel, a left center (LC) channel, and further including a number of left channels, L 60 , L 90 , L 120 , and LS in which the numerical indicator corresponds to the angular displacement, in degrees, of the channel relative to the listener. There are corresponding right channels, RC, R, R 60 , R 90 , R 120  and RS. The remainder of the discussion will focus on the left channels, since the right channels can be processed in a similar manner to the left channels. The left channel signals are processed by directional processor  36  to produce output signals for low frequency (LF) array driver  12  on signal line  38   a , for LF array driver  11  on signal line  38   b , for driver  22 - 60 L or driver  28 - 60 L on signal line  39   a , for driver  22 - 90 L or driver  28 - 90 L on signal line  39   b , for driver  22 - 120 L or  28 - 120 L on signal line  39   c , and for driver  22 - 150 L or driver  28 - 150 L on signal line  39   d . As with the embodiment of  FIG. 2   a , the outputs on the signal lines are processed by system EQ and dynamic range controller  42 . 
   In an exemplary embodiment, the directional processor  36  is implemented as digital signal processors (DSPs) executing instructions with digital to analog and analog-to-digital converters as necessary. In other embodiments, the directional processor  36  may be implemented as a combination of DSPs, analog circuit elements, and digital to analog and analog-to-digital converters as necessary. 
     FIG. 8  shows a block diagram of the directional processor  36  of  FIG. 7 , for an implementation with limited range side and rear acoustical drivers. The directional processor has inputs for five left directional channels. The five directional channels can be created from an audio signal processing system having two channels, a left (L) channel designed, for example, to be radiated at 30 degrees) and a left surround (LS) channel, designed, for example to be radiated at 150 degrees). The L and LS channels can be decoded according the teachings of U.S. patent application Ser. No. 08/796,285, incorporated herein by reference, to produce channel L 90  (intended to be radiated at 90 degrees). Channels L and L 90  and channels L 90  and LS can then be decoded to produce channels L 60  and L 120 , respectively. The invention will work equally well with fewer directional channels or more directional channels. The audio signal processing system of  FIG. 7  has several elements that are similar to elements of the system of  FIGS. 3   a – 3   d  and perform similar functions to the corresponding elements of  FIGS. 3   a – 3   d . The similar elements use similar reference numerals. Some elements of  FIGS. 3   a – 3   d  that are not germane to the invention (such as multiplier  57 ) are not shown in  FIG. 8 . A mirror image audio processing system could be created to process right directional channels corresponding to the left directional channels. 
   Referring now to  FIG. 8 , the input terminals for channels L 60 , L 90 , L 120 , and LS are coupled to level detector  44  for making measurements for the scalers and HRTF filters. The input terminal for channel L is coupled to presentation mode processor  102 . Output terminal  35  designated L′ of presentation mode processor  102  is coupled to summer  47 . The input terminal for channel LC is coupled to presentation mode processor  102 . Output terminal  37  of presentation mode processor  102  designated LC′ is coupled subtractively to summer  58  through time delay  58  and additively to summer  62 . The audio signal in channel L 60  is split by frequency splitter  46 a into a low frequency (LF) portion and a high frequency (HF) portion. LF portion is input to summer  47 . HF portion of the audio signal in channel L 60  is input to front/rear scaler  48   a , (similar to the front/rear scaler  48  of  FIGS. 3   a – 3   d  and  4 ), using the values |  L | and |  L 60   | respectively for the values |  L | and |  LS | in the discussion of  FIG. 4 . Front/rear scaler  48   a  separates the HF portion of the audio signal in channel L 60  into a “front” portion and a “rear” portion. Front portion of the HF portion of the audio signal in channel L 60  is processed by front HRTF filter  50   a  (similar to the front HRTF filter  50  of  FIGS. 3   a – 3   d  and  4 ), using the values |  L | and |  L 60   | respectively for the values |  L | and |  LS | in the discussion of  FIG. 4 , and speaker placement compensator  60   a , (similar to the speaker placement compensator  60  of  FIGS. 3   a – 3   d  and  4 ), calculated for 30 degrees, and input to summer  47 . Rear portion of the audio signal in channel L 60  is processed by front HRTF filter  50   b  (similar to the front HRTF filter  50  of  FIGS. 3   a   3   d  and  4 ), using the values |  L | and |  L 60   | respectively for the values |  L | and |  LS | in the discussion of  FIG. 4 ) and speaker placement compensator  60   a , similar to the speaker placement compensator  60  of  FIGS. 3   a – 3   d  and  4 , calculated for 60 degrees, and input to summer  100 - 60 . 
   The audio signal in channel L 90  is split by frequency splitter  46   b  into a low frequency (LF) portion and a high frequency (HF) portion. LF portion is input to summer  47 . HF portion of the audio signal in channel L 90  is input to front/rear scaler  48   b , similar to the front/rear scaler  48  of  FIGS. 3   a – 3   d  and  4 , using the values |  L 60   | and |  L 90   | respectively for the values |  L | and |  LS | in the discussion of  FIG. 4 . Front/rear scaler  48   b  separates the HF portion of the audio signal in channel L 90  into a “front” portion and a “rear” portion. Front portion of the HF portion of the audio signal in channel L 90  is processed by front HRTF filter  50   c  (similar to the front HRTF filter of  FIGS. 3   a – 3   d  and  4 ), using the values |  L 60   | and |  L 90   | respectively for the values |  L | and |  LS | in the discussion of  FIG. 4 ), and speaker placement compensator  60   b , calculated for 60 degrees, and input to summer  100 - 60 . Rear portion of the audio signal in channel L 60  is processed by front HRTF filter  50   d  (similar to the front HRTF filter of  FIGS. 3   a – 3   d  and  4 ), using the values |  L 60   | and |  L 90   | respectively for the values |  L | and discussion of  FIG. 4 , and speaker placement compensator  60   d , (similar to the speaker placement compensator  60  of  FIGS. 3   a – 3   d  and  4 ), calculated for 90 degrees, and input to summer  100 - 90 . 
   The audio signal in channel L 120  is split by frequency splitter  46   c  into a low frequency (LF) portion and a high frequency (HF) portion. LF portion is input to summer  47 . HF portion of the audio signal in channel L 120  is input to front/rear scaler  48 c, (similar to the front/rear scaler  48  of  FIGS. 3   a – 3   d  and  4 ), using the values |  L 90   | and |  L 120   | respectively for the values |  L | and |  LS | in the discussion of  FIG. 4 . Front/rear scaler  48 c separates the HF portion of the audio signal in channel L 120  into a “front” portion and a “rear” portion. Front portion of the HF portion of the audio signal in channel L 120  is processed by front HRTF filter  50   e  (similar to the front HRTF filter  50  of  FIGS. 3   a – 3   d  and  4 , using the values |  L 90   | and |  L 120   | respectively for the values |  L | and |  LS | in the discussion of  FIG. 4  and speaker placement compensator  60   e  (similar to the speaker placement compensator  60  of  FIGS. 3   a – 3   d  and  4 ), calculated for 90 degrees, and input to summer  100 - 90 . Rear portion of the audio signal in channel |  L 90   | is processed by rear HRTF filter  56   a  (similar to the rear HRTF filter  56  of  FIGS. 3   a – 3   d  and  4 ), using the values |  L 90   | and |  L 120   | respectively for the values |  L | and |  LS |, and speaker placement compensator  60   f  (similar to the speaker placement compensator  60  of  FIGS. 3   a – 3   d  and  4 ), calculated for 120 degrees, and input to summer  100 - 120 . 
   The audio signal in channel LS is split by frequency splitter  46   d  into a low frequency (LF) portion and a high frequency (HF) portion. LF portion is input to summer  47 . HF portion of the audio signal in channel LS is input to front/rear scaler  48   d , (similar to the front/rear scaler  48  of  FIGS. 3   a   3   d  and  4 ), using the values |  L 120   | and |  LS | respectively for the values |  L | and |  LS | in the discussion of  FIG. 4 . Front/rear scaler  48   d  separates the HF portion of the audio signal in channel LS into a “front” portion and a “rear” portion. Front portion of the HF portion of the audio signal in channel LS is processed by rear HRTF filter  56   b  (similar to the rear HRTF filter  56  of  FIGS. 3   a – 3   d  and  4 ), using the values |  L 120   | and |  LS | respectively for the values |  L | and |  LS | in the discussion of  FIG. 4 , and speaker placement compensator  60   fg  (similar to the speaker placement compensator  60  of  FIGS. 3   a – 3   d  and  4 ), calculated for 120 degrees, and input to summer  100 - 120 . Rear portion of the audio signal in channel LS is processed by rear HRTF filter  56   c  (similar to the rear HRTF filter  56  of  FIGS. 3   a – 3   d  and  4 ), and speaker placement compensator  60   h  (similar to the speaker placement compensator  60  of  FIGS. 3   a – 3   d  and  4 ), calculated for 150 degrees. 
   The output signal of summer  47  is transmitted additively to summer  58  and subtractively through time delay  61  to summer  62 . The output signal of summer  58  is transmitted to full range acoustical driver  11  (of speaker array  10 ) for transduction to sound waves. The output signal of summer  62  is transmitted to full range acoustical driver  12  for transduction to sound waves. Time delay  61  facilitates the directional radiation of the signals combined at summer  47 . Output signals of summers  100 - 60 ,  100 - 90 ,  100 - 120 , and of speaker placement compensator  60   h  are transmitted to limited range acoustical drivers  22 - 60 ,  22 - 90 ,  22 - 120 , and  22 - 150 , respectively, for transduction to sound waves. 
     FIG. 9  shows the directional processor of  FIG. 7  for an implementation having full range side and rear acoustical drivers. The implementation of  FIG. 9  has the same input channels as the implementation of  FIG. 7 . The invention will work with fewer directional channels or more directional channels. The audio signal processing system of  FIG. 7  has several elements that are similar to elements of the system of  FIGS. 3   a – 3   d  and perform similar functions to the corresponding elements of  FIGS. 3   a – 3   d . The similar elements use similar reference numerals. A mirror image audio processing system could be created to process right directional channels corresponding to the left directional channels. 
     FIG. 9  is similar to  FIG. 8 , except for the following. The low frequency (LF) signal line from frequency splitter  46 a is coupled to summer  100 - 60  instead of summer  47 ; the LF signal line from frequency splitter  46   b  is coupled to summer  100 - 90  instead of summer  47 ; the LF signal line from frequency splitter  46   c  is coupled to summer  100 - 120  instead of summer  47 ; the LF signal line from frequency splitter  46   d  is coupled to summer  100 - 150  instead of summer  47 ; and the output of speaker placement compensator  60   h  is coupled to a summer  100 - 150 . Output signals of summers  100 - 60 ,  100 - 90 ,  100 - 120 , and  100 - 150  are transmitted to full range acoustical drivers  28 - 60 ,  28 - 90 ,  28 - 120 , and  28 - 150 , respectively, for transduction to sound waves. 
   Referring now to  FIGS. 10   a – 10   c , there are shown three top diagrammatic views of some of the components of an audio system for describing another feature of the invention. As described in patents such as U.S. Pat. Nos. 5,809,153 and 5,870,484, arrays of acoustical drivers and signal processing techniques can be designed to radiate sound waves directionally. By radiating the same sound wave from two acoustical drivers subtractively (functionally equivalent to out of phase) and time-delayed, a radiation pattern can be created in which the acoustic output is greatest along one axis (hereinafter the primary axis) and in which the acoustic output is minimized in another direction (hereinafter the null axis). In  FIGS. 10   a – 10   c , an array  10 , including acoustical drivers  11  and  12  is arranged as in an audio system shown in  FIGS. 1   a – 1   c ,  2   a , and  FIGS. 3   a – 3   d . The parameters of time delay  64  of  FIGS. 3   a – 3   d  are set such that a signal that is transmitted undelayed to acoustical driver  12  and delayed to acoustical driver  11  and transduced results in a radiation pattern that has a primary axis in a direction  104  generally toward a listener  14  in a typical listening position, a null axis in a direction  106  generally away from listener  14  in a typical listening position, and a radiation pattern  105  as indicated in solid line. The parameters of time delay  61  of  FIGS. 3   a – 3   d  are set such that a signal that is transmitted undelayed to acoustical driver  11  and delayed to acoustical driver  12  and transduced results in a radiation pattern that has a primary axis in direction  106  generally away from a listener  14  in a typical listening position, a null axis in direction  104  generally toward listener  14  in a typical listening position, and a radiation pattern  107  as indicated in dashed line. In  FIG. 10   a , the audio signal in channel LC is processed and radiated such that the radiation pattern has a primary axis in direction  104  and a null axis in direction  106  and the audio signal in channels L and LS are processed and radiated such that they have a primary axis in direction  106 . In  FIG. 1   b , the audio signal in channels L and LC are processed and radiated such that the radiation patterns have a primary axis in direction  104  and a null axis in direction  106 , and the audio signal in channel LS is processed and radiated such that it has a primary axis in direction  106  and a null axis in direction  104 . In  FIG. 10   c , the audio signals in channels L, LC, and LS are processed and radiated such that they all have primary axes in direction  106  and null axes in direction  104 . Hereinafter, the combination of radiation patterns, primary axes, and null axes will referred to as “presentation modes.” Generally, the presentation mode of  FIG. 10   a  is preferable when the audio system is used as a part of a home theater system, in which is desirable to have a strong center acoustic image and a “spacious” feel to the directional channels. The presentation mode of  FIG. 10   b  may be preferable when the audio system is used to play music, when center image is not so important. The presentation mode of  FIG. 10   c  may be preferable if the audio system is placed in a situation in which the array  10  must be placed very close to a center line (that is when the angle φ 1  of  FIG. 5  is small). As with several of the previous figures, there may be mirror image audio system for processing the right side directional channels. 
   Referring now to  FIG. 11 , there is shown presentation mode processor  102  (of  FIGS. 3   a – 3   c ,  8 , and  9 ) in more detail. Channel L input is connected additively to summer  108  and to the one side of switch  110 . Other side of switch  110  is connected additively to summer  112  and subtractively to summer  108 . Channel LC is connected additively to summer  112  which is connected additively to summer  116  and to one side of switch  118 . Other side of switch  118  is connected additively to summer  114  and subtractively to summer  116 . Summer  114  is connected to terminal  35 , designated L′. Summer  116  is connected to terminal  37 , designated LC′. Depending on whether switches  110  and  118  are in the open or closed position, the signal at output terminal  35  (designated L′) may be the signal that was input from channel L, the combined input signals from channels L and LC, or no signal. Depending on whether switches  110  and  118  are in the open or closed position, the signal at output terminal  37  (designated LC′) may be the signal that was input from channel LC, the combined input signals from channels L and LC, or no signal. 
   Referring now to any of  FIGS. 3   a – 3   c , the output signal of terminal  35  is summed with the low frequency portion of the surround channel at summer  47 , and is transmitted to summer  58 , which is coupled to acoustical driver  11 , and through time delay  61  to summer  62 , which is coupled to acoustical driver  12 . The output signal of terminal  37  is coupled to summer  62  and through time delay  64  to summer  58 . Thus the output of terminal  35  is summed with the low frequency (LF) portion of the left surround (LS) signal and transmitted undelayed to acoustical driver  11  and delayed to acoustical driver  12 . The output of terminal  37  is transmitted undelayed to acoustical driver  12  and delayed to acoustical driver  11 . As taught above in the discussion of  FIGS. 10   a – 10   c , the parameters of time delay  64  may be set so that an audio signal that is transmitted undelayed to acoustical driver  12  and delayed to acoustical driver  11  and transduced results in an radiation pattern that has a primary axis in direction  104  of  FIGS. 10   a – 10   b . Similarly, the discussion of  FIGS. 10   a – 10   c  teaches that the parameters of time delay  61  may be set so that an audio signal that is transmitted undelayed to acoustical driver  11  and delayed to acoustical driver  12  and transduced results in radiation pattern that has a primary axis in direction  106  of  FIGS. 10   a – 10   b . Therefore, by setting the switches  110  and  118  of presentation mode processor  102  to the “closed” or “open” position, it is possible for a user to achieve the presentation modes of  FIGS. 10   a – 10   c . The table below the circuit of  FIG. 11  shows the effect of the various combinations of “open” and “closed” positions of switches  110  and  118 . For each of the four combinations, the table shows which of channels L and LC are output on the output terminals designated L′ and LC′ (terminals  35  and  37 , respectively), which channels when radiated have a radiation pattern that has a primary axis in direction  104  and a null axis in direction  106  and which have a primary axis in direction  106  and a null axis in direction  104 , and which of  FIGS. 10   a – 10   c  are achieved by the combination of switch settings. In the implementation of  FIGS. 3   a – 3   c ,  10 , and  11 , the low frequency portion of surround channel LS is always radiated with the primary axis in direction  106 . Also, if switch  118  is in the closed position, the radiation pattern of  FIG. 10   c  results, regardless of the position of switch  110 . 
   In the implementations of  FIGS. 8 and 9 , the presentation mode processor  102  has the same effect on input channels L and LC and the signals on the output terminals  35  and  37  (designated L′ and LC′, respectively). 
   It is evident that those skilled in the art may now make numerous modifications of and departures from the specific apparatus and techniques herein disclosed without departing from the inventive concepts. Consequently, the invention is to be construed as embracing each and every novel feature and novel combination of features herein disclosed and limited only by the spirit and scope of the appended claims.