MATRIX DECODER
Cross-Reference to Related Applications
This application claims the benefit of priority of United States Provisional
Application No. 61/010,896, filed January 11, 2008, hereby incorporated by
reference.
Field of the Invention
The invention relates to audio signal processing. More particularly, the
invention relates to an audio matrix decoder or decoding function or to a computer
program stored on a computer-readable medium executing the decoding function.
Although the decoder or decoding function is particularly useful for playback from a
portable player using a headphone or loudspeaker virtualizer, a matrix decoder or
decoding function according to aspects of the present invention is not limited to such
uses.
Summary of the Invention
In accordance with an aspect of the present invention, an audio matrix
decoding method receiving a stereo signal pair Lt, Rt, in which method the relative
amplitudes and polarities of the pair determine the reproduced direction of decoded
signals, comprises panning Lt and Rt to outputs associated with front directions in
response to a measure of the sum of Lt and Rt being greater than a measure of the
difference between Lt and Rt, and panning Lt and Rt to outputs associated with rear
directions in response to a measure of the sum of Lt and Rt being less than a measure
of the difference between Lt and Rt, and modifying Lt and Rt to shift the direction of
reproduced signals.
Modifying Lt and Rt to shift the direction of reproduced signals may shift
signals panned to outputs associated with rear directions. Modifying Lt and Rt to shift
the direction of reproduced signals shifts signals panned to outputs associated with
rear directions may shift signals away from the rear-center direction. Such shifting
away from the rear-center direction may be in the direction in which such signals have
the largest amplitude. Such shifting may progressively decrease for signals at
directions increasingly away from the rear-center-direction.
Modifying Lt and Rt to shift the direction of reproduced signals may also shift
signals panned to outputs associated with front directions. Such shifting of signals
panned to outputs associated with front directions may shift least signals at the front-
center direction and such shifting may progressively increase for signals at directions
increasingly away from the front-center direction.
The degree of shifting, whether to the front or to the rear may be based on a
measure of the difference between Lt and Rt.
The degree of shifting may change only when Lt and Rt are panned to outputs
associated with rear directions.
According to a further aspect of the present invention, in an audio matrix
decoding method receiving a stereo signal pair Lt, Rt, in which method the relative
amplitudes and polarities of the pair determine the reproduced direction of decoded
signals, a method comprises shifting the direction of outputs associated with front and
rear directions to the left or right, the direction of outputs associated with rear
directions being shifted to a greater degree than the direction of outputs associated
with front directions, wherein the shifting includes modifying the stereo signal pair Lt,
Rt by forming a difference signal of Lt and Rt signals, scaling the difference signal by
a bias gain factor, and summing the scaled difference signal to both Lt and Rt signals
to produce modified Lt and Rt signals such that the relative amplitudes and polarities
of the modified Lt and Rt pair determine the reproduced direction of decoded signals.
According to a further aspect of the present invention, a method for modifying
a stereo signal pair if, Rt before the signal pair is decoded by an audio matrix decoder
or decoding method, the relative amplitudes and polarities of the pair determining the
reproduced direction of decoded signals comprises modifying the stereo signal pair Lt,
Rt by forming a difference signal of Lt and Rt signals, scaling the difference signal by
a bias gain factor, and summing the scaled difference signal to both Lt and Rt signals
to produce modified Lt and Rt signals such that the relative amplitudes and polarities
of the modified Lt and Rt pair determine the reproduced direction of decoded signals.
Brief Description of the Drawings
FIG. 1 is a schematic functional block diagram showing an example of how Lt
and Rt signals may be panned or steered to front and rear directions in accordance
with aspects of the present invention.
FIG. 2 is a schematic functional block diagram showing an example of the
details of the "Front-Back Steering Determination" of FIG. 1.
FIG. 3 is a schematic functional block diagram showing art example how Lt
and Rt may be modified in accordance with aspects of the present invention.
FIG. 4 is a conceptual diagram useful in understanding an effect of modifying
the Lt and Rt signals in accordance with aspects of the present invention.
FIG. 5 is a schematic functional block diagram showing an example of how
the LR_bias control signal of FIG. 3 may be derived.
FIG. 6 is a schematic functional block diagram showing the overall
arrangement of the arrangements of FIGS. 1, 2, 3, and 5.
Description of the Invention
Front-back panning
The matrix decoder according to aspects of the present invention treats the Lt
and Rt signals applied to its inputs as a stereo signal pair, and it pans those signals to
the front (left, L and right, R) or to the back (left surround, Ls, and right surround,
Rs). Lt and Rt are panned to outputs associated with front directions in response to a
measure of the sum of Lt and Rt being greater than a measure of the difference
between Lt and Rt. Lt and Rt are panned to outputs associated with rear directions in
response to a measure of the sum of Lt and Rt being less than a measure of the
difference between Lt and Rt. The Front-Back panning may be achieved, for
example, as shown in FIG. 1. In this block diagram, the panF and panB signals are
slow-changing gain signals (not full bandwidth audio signals) that may vary, for
example, between 0 to 1. The panF and panB signals operate together (they are
complementary to each other) to effect a smooth crossfade between the L and R front
signals and the Ls and Rs back signals.
Referring to FIG. 1, the Lt input signal is applied to the L output via a
multiplier or multiplier function 2 and to the Ls output via a multiplier or multiplier
function 4. The Rt input signal is applied to the R output via a multiplier or multiplier
function 6 and to the Rs output via a multiplier or multiplier function 8. The gain of
each of the multipliers 2 and 6 are controlled by the panF gain signal; the gain of each
of the multipliers 4 and 8 are controlled by the panB gain signal. The Lt and Rt input
signals are also applied to a circuit or function ("Front-Back Steering Determination")
10 that generates the panF and panB signals. Details of the Front-Back Steering
Determination are shown in FIG. 2.
Subject to lime smoothing, as described below, when the "'Front-Back Steering
Determination" 10 detects out-of-phase audio but no in-phase audio in the Lt and Rt
input signals for a sufficient period of time, it sets panB=1.0 and ponF=0.0, thereby
directing, panning, or "steering" the Lt and Rt input signals only to the Ls and Rs
surround output channels (hard rear steering). Likewise, when there is in-phase audio
but no out-of-phase audio present in the input signal for a sufficient period of time,
the "Front-Back Steering Determination" 10 sets panB=0.0 and panF=1.0, thereby
steering the Lt and Rt input signals only to the front output channels, L and R (hard
front steering).
The arrangement in FIG. 2 generates, on an instantaneous basis, the difference
between the magnitudes of the sum and the difference of the input signals Lt and Rt (a
rapidly-varying waveform swinging both positively and negatively) and compares it
with a small threshold s (epsilon). This is accomplished by adder or adding function
12 that receives Lt and Rt to produce Lt + Rt at its output, adder or adding function 14
that subtracts Rt from Lt to produce Lt - Rt at its output, scalers or scaling functions i 6
and 18 that scale the amplitudes of if + Rt and Lt - Rt to produce "Front" and "Back"
signals F and B,

which signals F and B have their absolute values taken, shown at absolute value
devices or functions 20 and 22, and an adder or adding function 24 that subtracts the
absolute value of B from the absolute value of F and adds a small value epsilon.
Elements 12, 14, 16, 18, 20, 22 and 24 may be considered collectively as a
"Difference of Measures of Sum and Difference" device or function as shown in the
overall arrangement of FIG. 6.
The polarity of the result | F |- | B | + e is determined by a "Detect Polarity"
device or function 26. If negative, the answer is one value, for example minus 1, if
positive, another value, such as zero. Clearly, values other than minus 1 and zero
may be employed. The result is a two-valued waveform alternating between two
levels, minus 1 and 0, in this example. A low-pass filter or filtering function ("Low-
pass Filter") ("LPF") 28 is applied, resulting in a more slowly varying waveform FB
that may have any value in the range between or including the values of the two
levels, depending on the proportion of time that the square wave spends at each of the
levels. In response to real audio signals, the smoothed waveform produced by LPF 28
tends to remain near one or the other of the extremes. In effect, LPF 28 delivers a
short-term average of its input, having a time constant, for example, in the range of 5
to 100 milliseconds. Although a 40 millisecond time constant has been found to be
suitable, the value is not critical. LPF 28 may be implemented as a single-pole filler.
Still referring to the example of FIG. 2, having determined the intermediate
control signal FB, two complementary panning coefficients panF and panB may then
be obtained in any of a number of ways by a "Determine Panning Functions" device
or function 30. In principle, any of various commonly-used crossfade functions may
be employed, such as a linear ramp, log, Hanaing, Hamming and sine functions. It
will be appreciated that the actual formulae will vary depending on the output values
chosen for Detect Polarity 26.
If constant power panning is desired, the following formulae may be
employed:

Alternatively, if constant sound pressure is deemed preferable, or at least
acceptable, the following formulae may be employed:

Although equations 3 and 4 above provide constant power (the sum of the
squares of the panF and panB coefficients is one), constant power can be
approximated by employing the following formulae:

The values of each of panF and panB in the example of equations 7 and 8 can lie
anywhere between 0 and 1 and are complementary to each other, each tracing the path
of a parabola. The result is two coefficients or control signals with ranges between 0
and I, whose squares add approximately to 1.
If panF were consistently greater say than panB in any of the above sets of
formulae, which is the result, for example, when Lt and Rt are equal with the same
polarity, so that the input to the LPF 28 is 0 over a long time, the panning would steer
hard front (panF=1 and panB=0). If panF were consistently smaller than panB, which
is the result, for example, when Lt and Rt are equal but out of phase, so that the input
to the LPF would be -1 over a long time, the panning would steer hard back (panF=0
and panB=1). On real signals,as with the intermediate signal FB, panning tends to
remain either hard front or hard back. Thus, Lt and Rt are panned to outputs
associated with front directions in response to a measure of the sum of Lt and Rt being
greater than a measure of the difference between Lt and Rt, and Lt and Rt are panned
to outputs associated with rear directions in response to a measure of the sum of LtT
and Ri being less than a measure of the difference between Lt and Rt. When a
measure of the sum of Lt and Rt is the same as a measure of the difference between Lt
and Rt, Lt and Rt may be panned to outputs associated with front directions, although
this is not critical.
FIG. 2 provides an example of generating suitable panF and panB control
signals. Modifications of FIG. 2, for example as suggested above, may be employed.
Alternatively, other arrangements that provide smooth panning signals in response to
measures of the sum and difference of Lt and Rt may be employed.
Left-Right panning
Ideally, left-right panning is as follows:
When Lt, Rt is panned to the front (L, R), use less left-right
steering than is applied when Lt, Rt is panned to the rear, because the
Lt, Rt signal likely contains complete L, C, R signal components
already mixed into a stereo pair in a manner that is likely to provide a
good left-right soundfield when reproduced, including a phantom
center image.
When Lt, Rt is panned to the back (Ls, Rs), determine which
channel (Ls orRs) has the greater amplitude, and then modify the Lt,
Rt signals so that rear signals are shifted to the side in which such
signals have the largest amplitude. As explained further below, in an
implementation of the invention, such shifting may also have an effect,
albeit a lesser one, when Lt, Rt is panned to the front (L, R).
A common problem in many matrix decoders is the inability to work well for
the case where input signals are panned to the rear-center position. This is
particularly a problem when playback employs a headphone virtualizer or a
loudspeaker virtualizer. The rear-center position, for example, is encoded with Lt and
Rt out-of-phase with each other. Hence, when the Lt, Rt signals are panned to Ls, Rs,
rear-center signals appear in the Ls, Rs signals out-of-phase. A rear phantom image is
not formed well by such out-of-phase signals.
An aspect of the present invention is to shift Ls, Rs signals to the left or right,
thereby avoiding the rear-center phantom position that causes difficulty in imaging.
This may be achieved by performing a "shift" operation on the Lt, Rt signals, as
shown in FIG. 3 and as described below. The greatest shift may be applied to rear-
center signals and less shift for positions progressively away from rear center. The
least shift (or no shift) may be applied to front-center signals with a progressively
increasing shift for positions away from front-center. In other words shifting should
alter the rear-center the most and the front-center the least. By avoiding or
minimizing shift at the front-center position under all conditions, image location shifts
of voices (dialog), which are usually at the front-center, are avoided or minimized. In
principle, a shifting device or function in the manner of the example of FIG. 3 may be
employed so as to modify the Lt, Rt input to any two input matrix decoder or
decoding function in which the decoder or decoding function operation responds to
the relative amplitudes and polarities of Lt and Rt.
One suitable "shift" operation is shown in FIG. 3 in which an Lt-Rt difference
signal is generated. Then, a weighted amount of this difference signal is mixed back
into both Lt and Rt to produce Ltbiased and Rtbiased The control input (LR_Bias) may
take on a value of 1 a or -a, depending on whether the "shift" is intended to shift the
rear channels to the left or the right. LR bias may be determined, for example, as
shown in the example of FIG. 5. Alpha may have a value, for example, in die range
of 0.05 to 0.2. A value of 0.1 has been found to provide useful results.
Referring to the details of FIG. 3, Rt is subtracted from Lt in an adder or
adding device 32 to obtain Lt-Rt which is then scaled by LR_bias in a multiplier or
multiplying function 34. The scaled version of Lt-Rt is then summed with each of Lt
and Rt in respective adders or adding functions 36 and 38 to obtain Ltbiased and Rtbiased
Consider several examples of the operation of the shifting arrangement of FIG.
3 as follows.
For example, when LR Bias = +0.1 (indicating that the shift should be to the
left), one gets:

Continuing with this example (LR_Bias - +0.1), consider the case where the
Lt, Rt input signal is composed of a center-panned signal: Lt = Rt = C. In this case,
one has:
In this case, the Ltbiased and Rtbiased signals are the same as Lt, Rt. In other words, the
shift circuit does not modify the Lt, Rt signals when the input contains only front-
center panned audio.
In contrast, consider the case where the Lt, Rt input signal is composed of a
rear-center panned signal, S: Lt = S, Rt = -S. In this case, one gets:
In this case, the Ltbiased and Rtbiased signals are modified by the shift circuit or process,
such that Ltbiased has been boosted in amplitude, and Rtbiased has been reduced in
amplitude. Note that, if LR_Bias were set to -0.1 instead of -0.1, the amplitude shifts
would be reversed, with Rtbiased being boosted in level while Ltbiased is reduced.
Ideally, the shifting circuit or process operates so that the surround channels
are shifted to the left or right, and the front channels are similarly shifted but to a
lesser degree. An example of shifting to the left is shown in FIG. 4 in which the solid
line circle represents a matrix encoding circle, in which traditional L (left). C (center),
R (right), Ls (left surround), S (surround or rear surround), and Rs (right surround)
channel positions are shown. This circle has unity radius, reflecting the fact that each
channel has unity power. The dashed line circle shows the effect on the unit circle of
the shift operation. The shift away from the unit circle indicates that the power of
some signal directions has been boosted or attenuated. In particular, note that the
rear-center position S is shifted by the greatest amount with progressively less shifting
for directions farther and farther away from S with no shifting occurring at the front-
center position C.
An example of a way to determine a suitable LR_bias signal is shown in FIG.
5. The LR bias signal is based principally on LR, a short-term -averaged amplitude
difference between the Ltbiiased and Rtbiased signals. In other words, LR is an estimate of
Libiased versus Rtbiased. LR_Bias is calculated in "Determine Shifting" device or
function 40 in response to whether each of LR, FB (FIG. 2) is less than or greater
than a threshold, and in response to Lt - Rt. Such a calculation may be expressed in
programming pseudocode:
Alternatively. FB and LR may be multiplied and the bias determined by
whether the result is greater than a threshold. Such calculation may be expressed in
programming pseudocode:
The LR_bias signal may be determined as follows. First measure the relative
amplitude of the Ltbiased and Rtbiased signals. Intermediate signal, LR, an estimate of
Ltbiased versus Rtbiased. a short-term-averaged amplitude difference between the Ltbiased
and Rtbiased signals, may be determined as follows:

Note that a small positive offset, e (epsilon), is added to the denominator of the
fraction in equation 7, to ensure that no error occurs when Lt and Rt are both zero. In
order to estimate LR, one notes that the correct value of LR should result in ErrorsLR
being equal to zero:

One way to create the short-term smoothed value of LR is to increment or
decrement the instantaneous value of the amplitude difference between the Ltbiased and
Rtbiased signals (by a small increment, such as 2"'°), based on the value of Errorlr, as
follows:

In this way, the next value of LR (referred to as LR' in the equation above),
will move towards the correct value in a stair step manner.
The short-term smoothing or averaging (reflected in equations 1.5 and 1.6 as
"avg") is a result of the smoothing that results from the incremental steps that attempt
to reduce the LR error. The smoothing may have a time constant between about 5 and
100 milliseconds. Values of 20 and 40 milliseconds have been found to be useful. In
the implementation described, LR can Lake on values from -1 (indicating a hard left
pan) to +1 (indicating a hard right pan). LR may have an initial value of zero, thus
requiring 1024 increments for it to reach +1 or -1. Obviously. 2048 increments are
required for LR to go from hard left to hard right.
If implemented in a digital system, the increments and decrements may be
done at the audio bit rate (48 kHz, for example, when increments of 2-10 are
employed). In principle, the present invention may be implemented wholly or partly
in the analog domain.
Referring again to FIG. 5, Ltbiased and Rtbiased have their absolute values taken,
shown at absolute value devices or functions 42 and 44. An adder or adding function
46 adds the absolute value of Ztbiased and the absolute value Rtbiased to the small value
epsilon and applies the result to a multiplier or multiplier function 48 that also
receives a one-sample-delayed version of LR to produce the product of LR and the
sum of the absolute value of Ltbiased, the absolute value Rtbiased, and epsilon. An adder
or adding function 50 subtracts the absolute value of Rtbiased from the absolute value of
Ltbiased. The error signal (equation 8) is then obtained from the output of adder or
adding function 52. The error signal is applied to signum() device or function 54 that
produces +1 if the input is greater than zero, -1 if the input is less than zero, and 0 if
the input is zero (although some DSP implementations of such a function are
simplified, so that signum () may be +1 for an input that is greater than or equal to
zero, and -1 for negative input). The signum device or function 54 output is
multiplied by the 2-10 scaling factor in multiplier or multiplying function 56 and
summed with the one-sample-delayed version of LR (provided by delay device or
function 60) in adder or adding function 5S. Elements 42, 44, 46, 48, 50, 52, 54, 56.
59 and 60 may be considered collectively as a "Determine Short-Term Averaged
Difference" device or function as shown in the overall arrangement of FIG. 6.
Once the value of LR has been determined, the LR_Bias signal value is
updated in Determine Shifting 40 according to the pseudocode shown first above and
the following logical rules:
1. LR_Bias will always be equal to +a or -a, where a is in the range
of, for example, 0.05 to 0.2. In practice, a value of 0.1 has been found to
provide useful results.
2. The LR_Bias signal only flips between its two allowable values
when there is a zero-crossing in the Lt-Rt signal. This minimizes the
possibility that a change in LR_Bias will result in an audible click in the
output.
3. When the LR signal indicates that Ltbiased is greater in amplitude than
Rtbiased (when LR>0), and when the FB signal indicates that the Lt, Ri signals
should be panned towards the back by more than an appropriate threshold (for
example, when FB<-0.1), then set LR_Bias to +0.1 (when there is a zero-
crossing in the Lt-Rt difference signal). In other words, the value of LR Bias
is allowed to change when the Lt, Rt signals are panned to the back by more
than a threshold. However, the latest value of LR_Bias remains effective
whether or not the Lt, Rt signals are panned to the back or panned to the front.
4. When the LR signal indicates that Rtbiased is greater in amplitude than
Ltbiased (when LR<0), and when the FB signal indicates that the Lt, Rt signals
should be panned towards the back by more than a threshold (for example,
when FB<-0.1 as mentioned above), then set LR_Bias to -0.1 (when there is a
zero-crossing in the Lt-Ri difference signal). The manner in which LR = 0 is
handled is not critical. One possibility is that when LR = 0 do nothing (leave
LR_bias unchanged) or, alternatively, act as when LR>0 as described just
above in paragraph 3.
Note that the LR Bias signal is determined from the amplitudes of the Ltbiased
and Rtbiased signals, and the Ltbiased and Rtbiased signal are modified by the LR_Bias
signal, thus forming a feedback loop in the overall algorithm. This is a positive
feedback loop that makes the overall behavior bi-stable in nature. As a result, the
arrangement exhibits hysteresis. For example, when LR_Bias = +0.1, this causes the
shifting circuit to exaggerate the Ltbiased signal, boosting it proportionally in
comparison to the Rtbiased signal which will, in turn, increase the LR signal (pushing it
upwards in a positive direction). As a result, a much larger Rt signal (relative to Lt) is
required to flip LR_Bias back to -0.1. Such hysteresis ensures that the system is less
likely to exhibit rapid flipping back and forth in the LRBias signal, which might
otherwise be undesirable by causing audible artifacts such as image shifting.
Image shifting is also minimized by allowing LR_bias to change only when
the pan is to the rear. Image shifts are more noticeable when at the front. Also,
retaining the same shift when panning from rear to front and from front to rear avoids
image shifts when such pans occur. However, changes in LR_bias typically will
occur when a change in audio content occurs. Thus, a shift in image location is often
required at such a change and is desirable.
It will be noted that both the front-back panning and left-right panning employ
time constants. Although suggested values for such time constants has been given, it
wilt be understood that smoothing values are to a degree a matter of the designer's
taste and may be chosen by trial and error. In addition, desirable smoothing values
may vary depending on the audio content.
FIG. 6 shows the manner in which the above-described FIGS. 1, 2, 3 and 5 fit
together.
Implementation
Although in principle the invention may be practiced either in the analog or
digital domain (or some combination of the two), in practical embodiments of the
invention, audio signals arc represented by samples in blocks of data and processing is
done in the digital domain.
The invention may be implemented in hardware or software, or a combination
of both (e.g., programmable logic arrays). Unless otherwise specified, algorithms and
processes included as part of the invention are not inherently related to any particular
computer or other apparatus. In particular, various general-purpose machines may be
used with programs written in accordance with the teachings herein, or it may be
more convenient to consuuet more specialized apparatus (e.g., integrated circuits) to
perform the required method steps. Thus, the invention may be implemented in one
or more computer programs executing on one or more programmable computer
systems each comprising at least one processor, at least one data storage system
(including volatile and non-volatile memory and/or storage elements), at least one
input device or port, and at least one output device or port. Program code is applied
to input data to perform the functions described herein and generate output
information. The output information is applied to one or more output devices, in
known fashion.
Bach such program may be implemented in any desired computer language
(including machine, assembly, or high level procedural, logical, or object oriented
programming languages) to communicate with a computer system. In any case, the
language may be a compiled or interpreted language.
Each such computer program is preferably stored on or downloaded to a
storage media or device (e.g., solid state memory or media, or magnetic or optical
media) readable by a general or special purpose programmable computer, for
configuring and operating the computer when the storage media or device is read by
the computer system to perform the procedures described herein. The inventive
system may also be considered to be implemented as a computer-readable storage
medium, configured with a computer program, where the storage medium so
configured causes a computer system to operate in a specific and predefined maimer
to perform the functions described herein.
A number of embodiments of the invention have been described. Nevertheless, it will
be understood that various modifications may be made without departing from the
spirit and scope of the invention. For example, some of the steps described herein
may be order independent, and thus can be performed in an order different from that
described.
WE CLAIM:
1. An audio matrix decoding method receiving a stereo signal pair Lt, Rt, in
which method the relative amplitudes and polarities of the pair determine the
reproduced direction of decoded signals, comprising:
panning Lt and Rt to outputs associated with front directions in response to a
measure of the sum of Lt and Rt being greater than a measure of the difference
between Lt and Rt, and panning Lt and Rt to outputs associated with rear directions in
response to a measure of the sum of Lt and Rt being less than a measure of the
difference between Lt and Rt, and
modifying the stereo signal pair Lt and Rt to shift the direction of reproduced
signals by forming a difference signal of Lt and Rt signals, scaling said difference
signal by a bias gain factor, and summing said scaled difference signal to both Lt and
Rt signals to produce modified Lt and Rt signals such that the relative amplitudes and
polarities of the modified Lt and Rt pair determine the reproduced direction of
decoded signals.
2. A method according to claim 1 wherein modifying Lt and Rt to shift the
direction of reproduced signals shifts signals panned to outputs associated with rear
directions.
3. A method according to claim 2 wherein modifying Lt and Rt to shift the
direction of reproduced signals shifts signals panned to outputs associated with rear
directions so as to shift signals away from the rear-center direction.
4. A method according to claim 3 wherein signals panned to outputs
associated with rear directions are shifted to away from the rear-center direction in the
direction in which such signals have the largest amplitude.
5. A method according to claim 3 or claim 4 wherein the degree of shifting is
greatest for signals at the rear-center position, the shifting progressively decreasing
for signals at directions increasingly away from the rear-center direction.
6. A method according to any one of claims 2-5 wherein modifying Lt and Rt
to shift the direction of reproduced signals also shifts signals panned to outputs
associated with front directions.
7. A method according to claim 6 wherein modifying Lt and Rt to shift the
direction of reproduced signals shifts signals panned to outputs associated with front
directions so as to shift least signals at the front-center direction.
8. A method according to claim 7 wherein the degree of shifting is least for
signals at the front-center position, the shifting progressively increasing for signals at
directions increasingly away from the front-center direction.
9. A method according to any one of claims 1-8 wherein the degree of shifting
is based on a measure of the difference between Lt and Rt.
10. A method according to any one of claims 1-9 wherein the degree of
shifting changes only when Lt and Rt are panned to outputs associated with rear
directions.
11. In an audio matrix decoding method receiving a stereo signal pair Lt, Rt,
in which method the relative amplitudes and polarities of the pair determine the
reproduced direction of decoded signals, a method comprising:
shifting the direction of outputs associated with front and rear directions to the
left or right, the direction of outputs associated with rear directions being shifted to a
greater degree than the direction of outputs associated with front directions, wherein
said shifting includes modifying the stereo signal pair Lt. Rt by forming a difference
signal of Lt and Rt signals, scaling said difference signal by a bias gain factor, and
summing said scaled difference signal to both Lt and Rt signals to produce modified
Lt and Rt signals such that the relative amplitudes and polarities of the modified Lt
and Rt pair determine the reproduced direction of decoded signals.
12. A method for modifying a stereo signal pair Lt. Rt before the signal pair is
decoded by an audio matrix decoder or decoding method, the relative amplitudes and
polarities of the pair determining the reproduced direction of decoded signals,
comprising
modifying the stereo signal pair Lt, Rt by forming a difference signal of Lt and
Rt signals, scaling said difference signal by a bias gain factor, and summing said
scaled difference signal to both Lt and Rt signals to produce modified Lt and Rt
signals such that the relative amplitudes and polarities of the modified Lt and Rt pair
determine the reproduced direction of decoded signals.
13. An audio matrix decoding method receiving a stereo signal pair Lt, Rt, in
which method the relative amplitudes and polarities of the pair determine the
reproduced direction of decoded signals, comprising
panning Lt and Rt to outputs associated with front directions in response to a
measure of the sum of Lt and Rt being greater than a measure of the difference
between Lt and Rt. and panning Lt and Rt to outputs associated with rear directions in
response to a measure of the sum of Lt and Rt being less than a measure of the
difference between Lt and Rt, and
modifying Lt and Rt to shift the direction of reproduced signals, wherein said
modifying includes shifting the direction of outputs associated with front and rear
directions to the left or right, the direction of outputs associated with rear directions
being shifted to a greater degree than the direction of outputs associated with front
directions, wherein said shifting includes modifying the stereo signal pair Lt, Rt by
forming a difference signal of Lt and Rt signals, scaling said difference signal by a
bias gain factor, and summing said scaled difference signal to both Lt and Rt signals
to produce modified Lt and Rt signals such that the relative amplitudes and polarities
of the modified Lt and Rt pair determine the reproduced direction of decoded signals.
14. Apparatus adapted to perform the methods of any one of claims 1 through
13.
15. A computer program, stored on a computer-readable medium for causing
a computer to perform the methods of any one of claims 1 through 13.

This audio matrix surround decoder requires minimal digital processing, useful in portable applications, particularly
in playback from a portable player using a headphone or loudspeaker virtualizer. In one embodiment it pans inputs Lt and Rt to
outputs associated with front directions in response to a measure of the sum of Lt and Rt being greater than a measure of the difference
between Lt and Rt, and pans Lt and Rt to outputs associated with rear directions in response to a measure of the sum of Lt and Rt
being less than a measure of the difference between Lt and Rt, Lt and Rt are modified to shift the direction of reproduced signals.