Note: Descriptions are shown in the official language in which they were submitted.
` W094/16538 PCT~S93/~688
Z1~3~62
SOUND IMAGE MANIPULATION
APPARATUS AND 1~nOD FOR 80UND IMAGE ENHAN~
R~OuND OF THE ~Nv~.-lON
This invention is directed to an automatic sound
image enhancement method and apparatus wherein the electronic
signal which corresponds to the audio signal is electronically
treated by amplitude and phase control t~ produce a perception
of enhancements to music and sounds. The invention preferably
operates on stereophonically recorded music (post production
enhancement) or in recording and mixing stereophonic
recordings (production). The invention may also be used in
connection with enhancing monophonic or monaural sound sources
to synthesize a stereo-like effect or to locate such sources
to positions beyond those normally found in the streo sound
stage.
Sound is vibration in an elastic medium, and
acoustic energy is the additional energy in the medium
produced by the sound. Sound in the medium is propagated by
compression and refraction of the energy in the medium. The
medium oscillates, but the sound travels. A single cycle is
a complete single excursion of the medium, and the frequency
is the number of cycles per unit time. Wavelength is the
distance between wave peaks, and the amplitude of motion
(related to energy) is the oscillatory displacement. In
fluids, the unobstructed wave front spherically expands.
Hearing is the principal response of a human subject
to sound. The ear, its mechanism and nerves receive and
transmit the hearing impulse to the brain which receives it,
compares it to memory, analyzes it, and translates the impulse
into a concept which evokes a mental response. The final step
in the process is called listening and takes place in the
brain; the ear is only a receiver. Thus, hearing is objective
and listening is subjective. Since the method and apparatus
of this invention is for the automatic stereophonic image
enhancement for human listening, the listening process is in
perceptions of hearing. This patent describes the perceptions
WO94/16538 PCT~S93/12~8
21S3062 2
of human subjects. Because a subject has two ears, laterally
spaced from each other, the sound at each eardrum is nearly
always different. Some of the differences are due to the
level, amplitude or energy, while others are due to timing or
phase differences. Each ear sends a different signal to the
brain, and the brain analyzes and compares both of the signals
and extracts information from them, including information in
determining the apparent position and size of the source, and
acoustic space surrounding the listener.
The first sound heard from a source is the direct
sound which comes by line-of-sight from the source. The direct
sound arrives unchanged and uncluttered, and lasts only as
long as the source emits it. The direct sound is received at
the ear with a frequency response (tonal quality) which is
relatively true to the sound produced by the source because
it is subject only to losses in the fluid medium (air). The
important transient characteristics such as timbre, especially
in the higher registers, are conveyed by direct sound. The
integral differences at each eardrum are found in time,
amplitude and spectral differences. The physical spacing of
the ears causes one ear to hear after the other, except for
sound originating from a source on the median plane between
the ears. The time delayed difference is a function of the
direction from which the sound arrives, and the delay is up
to about 0.8 millisecond. The 0.8 millisecond time delay is
about equal to the period of 1 cycle at 1,110 Hz. Above this
frequency, the acoustic wavelength of arriving sounds becomes
smaller than the ear-to-ear spacing, and the interaural time
difference-decE-eases-in-significance so that it is useful only
below about 1,400 Hz to locate the direction of the sound. The
difference in amplitude between the sound arriving at the two
ears results principally from the defracting and shadowing
effect of the head and external ear pinna. These effects are
greater above 400 Hz and become the source of information the
brain interprets to determine the direction of the source for
higher frequencies. Other clues to elevation and direction of
the sound derive from our practice of turning our head during
the sound direction evaluation process. This changes the
relative amplitude and time difference to provide further data
W094/16538 21 5~ ~2 PCT~S93/~688
for mental processing to evaluate direction. Both processes
are frequency dependent, but it has been shown that the time
difference is more useful wi~h transient portions of sound
while both are used for evaluation of the source direction of
~ 5 continuous signals. ~~
In human listening, memory plays an important role
in the evaluation of sound. The brain compares the interaural
temporal difference, interaural amplitude difference,
interaural spectral difference, as well as the precedence
effect, and temporal fusion, to be described later, with
memories of the same ~actors. The brain is constantly
comparing present perceptions with stored impression so that
those signals which are currently being received are compared
with memory to provide a conception of the surrounding
activity. In listening, the combination of the sound as
perceived and the memory of similar events, together, produce
a mental image of an aural conceptual geometrical framework
around us associated with the sources of sound to become thus
a conceptual image space. In the conceptual image space, what
is real and what seems to be real are the same. The present
system and apparatus is directed toward generating a
conceptual image space which seems to be real but, from an
objective evaluation, is an illusion.
In an apparatus where there are two, spaced
loudspeaker sound sources in front of the observer, with the
observer centered between them, the production of
substantially the same sound from each speaker, in-phase and
of the same amplitude, will present to the observer a virtual
sound image midway between the two speakers. Since the sound
source is in-phase, this virtual sound image will be called
a "homophasic image". By changing the relative amplitude, the
- homophasic image can be moved to any point between the two
speakers. In conventional professional processing of sound
signals, this moving action is called "panning" and is
controlled by a pan pot (panoramic potentiometer).
An equally convincing virtual sound image can be
heard if the polarity is reversed on one of the signals sent
to one of the same two loudspeakers. This results in an 180
degree phase shift for the sound from that speaker reaching
W0 ~/16538 ! PCT~S93/12
2 153 06~ 4
the ears. For simplification, the first 0 degree retarded
phase-shifted signal from the left speaker first reaches the
left ear and later the right çar, simultaneously the second
180 degree retarded phased-shifter signal from the right
speaker first reaches the right ear and later the left ear,
providing information to the ear-brain mechanism which
manifests a virtual sound image to the rear of the ~center
point of the listener's head. This virtual image is the
"antiphasic" image. Since it is a virtual image created by
mental process, the position is different for different
listeners. Most listeners hear the antiphasic image as
external and to the rear of the skull. The antiphasic image
does not manifest itself as a point source, but is diffused
and forms the rear boundary of the listener's conceptual image
space. By changing the phase relationship and/or amplitude of
various frequencies of the left and right signals, virtual
images can be generated along an arc or semicircle from the
back of the observer's head toward the left or right speakers.
Another factor which influences the perception of
sound is the "precedence effect" wherein the first sound to
be heard takes command of the ear-brain mechanism, and sound
arriving up to 50 milliseconds later seems to arrive as part
of and from the same direction as the original sound. By
delaying the signal sent to one speaker, as compared to the
other, the apparent direction of the source can be changed.
As part of the precedence effect, the apparent source
direction is operative through signal delay for up to 30
milliseconds. The effect is dependent upon the transient
~ characteristics of the signal.
An intrinsic part of the precedence effect, yet an
identifiably separate phenomenon, is known as "temporal
fusion" which fuses together the direct and delayed sounds.
The ear-brain mechanism blends together two or more very
similar sounds arriving at nearly the same time. After the
first sound is heard, the brain suppresses similar sounds
arriving within about the next 30 milliseconds. It is this
phenomenon which keeps the direct sound and room reverberation
all together as one pleasing and natural perception of live
listening. Since the directional hearinq mechanism works on
WO94tl6~8 21 53 ~62 PCT~S93/12688
the direct sound, the source of that sound can be localized
even though it is closely followed by multiple waves coming
from different directions.
The walls of the room are reflection surfaces from
which the direct sound reflects to form complex reflections.
The first reflection to reach the listener is known as a first
order reflection; the second, as second order, etc. An
acoustic image is formed which can be considered as coming
from a virtual source situated on the continuation of a line
linking the listener with the point of reflection. This is
true of all reflection orders. If we generate signals which
produce virtual images, boundaries are perceived by the
listener. This is a phenomenon of conditioned memory. The
position of the boundary image can be expanded by amplitude
and phase changes within the signal generating the virtual
images. The apparent boundary images broaden the perceived
space.
Audio information affecting the capability of the
ear-brain mechanism to judge location, size, range, scale,
reverberation, spatial identity, spatial impression and
ambiance can be extracted from the difference between the left
and right source. Modification of this information through
frequency shaping and linear delay is necessary to produce the
perception of phantom image boundaries when this information
is mixed back with the original stereo signal at the
antiphasic image position.
SUNMARY OF THE lNv~.~lON
The common practice of the recording-industry, for
producing a stereo signal, is to use two or more microphones
near the sound source. These microphones, no matter how many
= are used, are always electrically polarized in-phase. When the
program source is produced under these conditions (which are
industry standard), the apparatus described herein generates
a "synthetic" conditioning signal for establishment of a third
point with its own time domain. This derivation is called
synthetic because there is a separation, alternation and
regrouping to form the new whole.
To further help establish a point with a separate
W094116~8 2 15 3 0 6 2 PCT~S93/~688
time domain, a third microphone may be used to define the
location of the third point in relation to the stereo pair.
Contrary to the normal procedure of adding the output of a
third microphone to the left and right side of the stereo
microphone pair, the third microphone is added to the left
stereo pair and subtracted from the right stereo pair. This
arrangement provides a two-channel stereo signal which is
composed of a left signal, a right signal, and a recoverable
signal which has its source at a related but separate position
in the acoustic space being recorded. This is called organic
derivation and it compares to the synthetic situation
discussed above, where the ratios are proportional to the left
minus the right tfrom which it was derived) but is based on
its own time reference, which is, as will be seen, related to
the spacing between the three microphones. The timing between
the organic conditioning signal is contingent upon the
position of the original sound source with respect to the
three microphones. The information derived more closely
approximates the natural model than that of the synthetically
derived conditioning signal.
Control over either the organic or synthetic
situations, the processing thereof, and the generation of a
conditioning signal therefrom will produce an expanded
listening experience.
All sources of sound recorded with two or more
microphones in synthetic or organic situations contain the
original directional cues. When acted upon by the apparatus
of this invention, a portion of the original directional cues
are isolated, modified, r~constituted and added, in the form
of a conditioning signal, to the original forming a new whole.
The new whole is in part original and in part synthetic. The
control of the original-to-synthetic ratio is under the
direction of the operator via two operating modes:
(1) Space, in which the ratio is constant.
Synthetic is directly proportional to the original
and, therefore, enhancement depends upon the amount
of original information present in the stereo
program material.
(2) Auto Space, in which the ratio is
~- W094/16538 ~1 S306~ PCT~S93/~688
electrically varied. Synthetic is inversely
proportional to the original and, therefore, the
enhancement is held at a constant average
regardless of program material.
When a stereo recording is reproduced
monophonically, it is said to be compatible if the overall
musical balance does not change. The dimensionality of the
stereo recording will disappear when reproduced monophonically
but the inner-instrumental balance should remain stable when
L+R (i.e., the left plus right sources-have been combined to
monophonic sound, also called L=R).
The compatibility problem arises because monophonic
or the L+R signal broadcast in a conventional stereo broadcast
does not contain the total information present in the left and
right sources. When combined as such, it contains only the
information of similarity in vectorial proportion. The
differential information is lost. It is possible for the
differential signal to contain as much identity about the
musical content of a source as does the summation signal.
Since differential information will be lost in left
plus right combining, directional elements should comprise
most of the differential signal. Directional information will
be of little use in monophonic reproduction and its loss will
be of no consequence with respect to musical balance.
Therefore, additional dimensional or spatial producing
elements must be introduced in such a way that their removal
in L+R combining will not destroy the musical balance
established in the original stereophonic production.
Insertion of the conditioning signal at the
antiphasic image position produces enhancement to and
generation of increased spatial density in the stereo mode but
is completely lost in the mono mode where the directional
information will be unused. Information which can be lost in
the mono mode without upsetting the inner-instrument musical
balance includes clues relating to size, location, range and
ambience but not original source information.
To accomplish this, directional information is
obtained exclusively from the very source which is lost in the
monophonic mode, namely, left signal minus right signal.
WO94/16538 2 1 ~ 3 0 6 2 PCT~S93/~6~
Whether in the synthetic or organic model derivation
of a conditioning signal, subtracting the left signal from the
right signal and reinserting i~t at the antiphasic position
will not challenge mono/stereo compatibility, providing that
the level of conditioning signal does not cause the total RMS
difference energy to exceed the total RMS summation energy at
the output.
In order to aid in the understanding of this
invention, it can be stated in essentially summary form that
it is directed to a stereophonic image enhancement system and
apparatus wherein a conditioning signal is provided and
introduced into electronic signals which are to be reproduced
through two spaced loudspeakers so that the perceived sound
frame between the two loudspeakers is an open field which at
least extends toward the listener from the plane between the
loudspeakers and may include the perception of boundaries
which originate to the side of the listener. The conditioning
signal may be organic, if the original sound source is
approximately miked, or it may be derived from the left and
right channel stereo signals.
In one aspect, the present invention provides an
automatic stereophonic image enhancement system and apparatus
wherein two channel stereophonic sound is reproduced with
signals therein which generate a third image point with which
boundary image planes can be perceived within the listening
experience resulting in an extended conceptual image space for
the listener.
In another aspect, the present invention provides
- a stereophonic image enhancement system which includes
automatic apparatus for introducing the desired density of
conditioning signal regardless of program content into the
electronic signal which will be reproduced through the two
spaced speakers.
It is another objective to provide an automatic
stereophonic image enhancement system and apparatus wherein
the inner-instrumental musical balance remains stable when
heard in monophonic or stereophonic modes of reproduction.
It is another objective to provide a monophonically
compatible automatic stereophonic image enhancement system and
~ W094tl6538 2 ~ ~ 3 0 6 2 PCT~S931~8
apparatus wherein the operator can be readily trained to
employ the system and apparatus to achieve desirable
recordings with enhanced conceptual image space.
The features of the present invention which are
believed to be novel are set forth with particularity in the
appended claims. The present invention, both as to its
organization and manner of operation, together with further
objects and advantages thereof, may be best understood by
reference to the following description, taken in conjunction
with the accompanying drawings.
BRIEF DESCRIPTION OF THE DRAWINGS
Figure 1 is a perspective view of listener facing
two spaced loudspeakers, and showing the outline of an
enclosure.
Figure 2 is a schematic plan view of the perception
of a sound frame which includes a synthetic conditioning
signal which is included in the signals to the speakers.
Figure 3 is a schematic plan view of the perceived
open field sound frame where an organic conditioning signal
is introduced into the signal supplied to the speakers.
Figure 4 is a schematic plan view of the open field
sound frame, as perceived from the listener's point of view,
as affected by various changes within the conditioning signal.
Figure 5 is a schematic plan view of a sound source
and microphone placements which will organically produce a
conditioning signal.
Figure 6 is a simplified schematic diagram of a
circuit which combines the organically-derived conditioning
signal with the left and right channel signals.
Figures 7(a) and 7(b) form a schematic electrical
diagram of the automatic stereophonic image enhancement system
and apparatus in accordance with this invention.
Figure 8 is a schematic electrical diagram of an
alternate circuit therefore.
Figure 9 is a front view of the control panel for
the apparatus of Figure 8.
Figures lO(a) and lO(b) form a digital logic diagram
of a digital embodiment of the invention.
WO94/16~8 2 1 5 3 0 6 2 PCT~S93/12~8
Figure 11 is a front view of a joystick box, a
control box, and a interconnecting data cable 420 which can
be used to house the embodiment of the invention described
with reference to Figures 12(a), 12(b), 13(a)-13(f), 14(a) and
14(b)-
Figures 12(a) through 12(d) form a schematic diagram
--of--an embodiment of the invention wherein joysticks may be
used to move a sound around in a perceived sound field.
Figures 13(a)-13(f) are graphical representations
of the control outputs which are generated by the joysticks
and associated circuitry and applied to voltage controlled
amplifiers of Figures 12(a)-12(d).
Figures 14(a) and 14(b) form a digital sound
processor logic diagram similar to that of Figures 10(a) and
10(b), but adapted for use as the digital sound processor 450
in Figures 12(a)-12(d)
Figure 15 is a schematic diagram of an embodiment
of the invention which is adapted for use in consumer quality
audio electronics apparatus, of the type which may be used in
the home, etc.
Figure 16 is a block diagram of an embodiment of the
invention adapted for use in consumer-quality audio electronic
apparatus, which embodiment includes an automatic control
circuit for controlling the amount of spatial enhancement
2S which the circuit generates.
Figures 17A and 17B may be joined to form a
schematic diagram corresponding to the block diagram of Figure
16.
Figure-18 is a block diagram of another embodiment
of the invention, which block diagram includes an integrated
circuit implementing the circuitry of Figure 15 or of Figures
16, 17a and 17b.
Figure 19 is a block diagram of another embodiment,
similar to the embodiment of Figure 18, but providing for
multiple inputs.
BRIEF DESCRIPTION OF THE TABLES
Tables A through F set forth the data which is
graphically presented in Figures 13(a)-(f), respectively.
~ WO94/16538 215 ~ ~ 62 PCT~S93/~688
~ 11
Tables G through X set forth additional data.
DET~T~n DESCRIPTION OF EMBODINENTS OF THE lN~ ON
Figure 1 illustrates the usual physical arrangement
of loudspeakers for monitoring of sound. It should be
understood that in the recording industry sound is "monitored"
during all stages of production. It is "reproduced" when
production is completed and the product is in the market
place. At that point and on, what is being reproduced is the
production. Several embodiments of the invention are
disclosed. Some embodiments are intended for use during sound
production, while one embodiment is intended for use during
sound reproduction, in the house, for example. Embodiments of
the invention include the system and apparatus illustrated in
a first embodiment in Figures 5 and 6, a second embodiment 10
in Figure 7, a third embodiment 202 in Figure 8, a fourth
embodiment of Figures 10(a) and 10(b), a fifth and presently
preferred embodiment (for professional studio use) in Figures
11, 12(a), 12(b), 13(a)-13(f), 14(a) and 14(b). These
embodiments may be employed in record, compact disc,
mini-disc, cassette, motion picture, video and broadcast
production, to enhance the perception of sound by human
subjects, i.e. listeners. Another and sixth embodiment, which
is disclosed with reference to Figure 15, may be used in a
consumer quality stereo sound apparatus found in a home
environment, for example.
During monitoring of sound for sound production, the
two loudspeakers 12 and 14 are of suitable quality with
enclosures to produce the desired fidelity. They are laterally
spaced, and the listener 16 faces them and is positioned
substantially upon a normal plane which bisects the line
between the speakers 12 and 14. Usually, the listener is
enclosed in a room, shown in phantom lines, with the
loudspeakers. During reproduction, the two loudspeakers may
be of any quality. The loudspeaker and listener location is
relatively unimportant. During monitoring, the effect is one
of many separate parts being blended together. Hence,
monitoring requires a standard listening position for
evaluating consistency, whereas during reproduction, the
W094tl6538 215 3 0 6 2 PCT~S931~688
12 ~
effect has become one with the whole sound and can be
perceived from any general location.
Since several embodiments of the apparatus are
designed as a production tool, the loudspeakers 12 and 14
should be considered monitors being fed from an electronic
system which includes the sound production enhancement
apparatus of this invention. The electronic system may be a
professional recording console, multi-track or two-track
analogue, or digital recording device, with a stereophonic
two-channel output designated for recording or broadcasting.
The sound source material may be a live performance or it may
be recorded material in a combination of the foregoing.
Figure 2 illustrates the speakers 12 and 14 as being
enclosed in what is perceived as a closed field sound frame
24 (without the lower curved lines 17 and 26) which is
conventional for ordinary stereophonic production. By varying
the amplitude between the speakers 12 and 14, the apparent
source can be located anywhere within the sound frame 24, that
is, between the speakers. When a synthetic conditioning
signal is reinserted at the antiphasic image position 34,
amplitude and time ratios 17 are manifested between the three
points 12, 14 and 34. Because the antiphasic point 34 is the
interdependent product of the left point 12 and the right
point 14, the natural model is approached by a synthetic
construction, but never fully realized. The result is open
field sound from 26. Listener 16 perceives the open field 26.
Figure 3 illustrates open field sound frame 28 which
is perceived by listener 16 when a conditioning signal
derived, as in Figure 2, is supplied and introduced as part
of the signal to speakers 12 and 14, but has as its source an
organic situation. The density of spatial information is
represented by the curved lines 17 in Figure 2 and is
represented by the curved lines 19 in Figure 3. It is apparent
that the density of spatial information is greater in Figure
3 because the three points which produced the original
conditioning signal are not electrically interdependent but
are acoustically interactive; information more closely
reflecting the natural model is supplied to the ear-brain
mechanism of listener 16.
~ WO94/16~8 2 1~ 30 ~ PCT~S93/126~
13
Figure 4 illustrates the various factors which are
sensed by the listener 16 in accordance with the stereophonic
image enhancement systems of this invention. The two speakers
12 and 14 produce the closed field sound frame 24 when the
speakers are fed with homophasic signals. Homophasic image
position 30 is illustrated, and the position can be shifted
left and right in the frame 24 by control of the relative
amplitude of the speakers 12 and 14. The speakers 12 and 14
produce left and right real images, and a typical hard point
image 32 is located on the line between the speakers because
it is on a direct line between the real images produced by the
two real speakers. As described above, the hard point source
image can be shifted between the left and right speakers.
The antiphasic image position 34 is produced by
speakers 12 and 14 and may be perceived as a source location
behind the listener's head 16 at 34 under test or laboratory
demonstrations. Under normal apparatus operating conditions,
source 34 is not perceived separately but, through temporal
fusion, is the means by which an open filed sound frame is
perceived. Position 34 is a perceived source, but is not a
real source. There is no need for a speaker at position 34.
Rather, by controlling the relationship between the antiphasic
image position and one or both of the real images all produced
by speakers 12 and 14, the image source can be located on a
line between one of the real images and the antiphasic image
position 34. Since the antiphasic image position 34 is a
perceived source, but is not a real source, the point between
it and speakers 12 and 14 is considered a soft point source
image. Such a soft point source image is shown at point 36.
Open field sound frame is thus produced and provides the
perception of virtual space boundaries 40, 42, 44 or 46 (not
on line), depending on the conditioning signal's phase
relationship to the original source. The perceived distance
for the virtual space boundaries 40, 42, 44 and 46 from the
closest hard point is from 2 to 30 feet (approximately 1-10
meters), depending on the dimension control setting of Figure
5 and the distance between speakers 12 and 14.
Figure 18 is a schematic diagram of an Eighth
embodiment of the invention, which embodiment modifies the
WO94/16~8 2 15 3 0 6 2 PCT~S93t~688
14
embodiment described in reference to Figure 15.
Figure 19 is a schematic diagram of a Ninth
embodiment of the invention, ~which is similar to the Eighth
embodiment, but which includes panning pots connected to the
inputs of the circuitry such as via a conventional recording
counsel.
First Embodiment
Figure 5 is a schematic diagram of a sound source
which is to be recorded or amplified. Three microphones L, R
and C are shown located in front of the source. The L (left)
and R (right) microphones are approximately equally spaced
from the source on its left and right sides. The C
(conditioning) microphone is located further spaced from the
source and approximately equally spaced from the L and R
microphones.
The signal from the C microphone is adjusted in gain
and then is added (at adder A, for example) and subtracted (at
subtractor S, for example) from the stereo signals L, R as
shown in Figure 6. The resulting signal processed outputs PL
and PR, when amplified and applied to speakers 12 and 14
(Figure 1), will produce an expanded sound image as described
with reference to Figures 3 and 4. By adjusting the gain of
conditioning signal, C, the amount of expansion which occurs
can be controlled easily. In this embodiment, the conditioning
signal, C, is produced organically, that is, by a microphone
array pickup as shown in Figure 5 and connected as shown in
Figure 6. There exist many previous stereo recordings for
whi~h there was no microphone at location C connected as shown
in Figure 6, and thus there would seem to be no simple way of
recreating the effect described above. However, as will be
seen, the conditioning signal can be created synthetically,
and introduced into the left and right channel signals, when
(1) the sound source is mixed-down from a prerecorded tape in
a recording studio, for example, (2) the sound is broadcast,
or (3) when prerecorded sound is received or reproduced in a
home environment. The conditioning signal is delayed time-wise
and filtered compared to the signals from microphones L and
R due to the placement of microphone C.
WOg4/16~8 21 S3 ~2 PCT~S93/~688
Second Embodiment
Now considering an embodiment of the apparatus 100
which produces the conditioning signal synthetically, the left
input lines 48 and 49 and right input lines 50 and 51 are
received from musical signal sources. The system and apparatus
10 is described in this embodiment as being a system which
introduces the conditioning signal before the two-channel
recording and, thus, is a professional audio laboratory and
apparatus. Thus, the left and right inputs 48, 49, 50 and 51
may be the product of a live source or a mixdown from a
multiple channel tape produced by the live recording, or it
may be a computer generated source, or a mixture of same. The
inputs of the apparatus 48, 49, 50 and 51 addresses the output
of the recording console's "quad bus" or "4-track bus". Each
position on the recording console can supply each and every
bus of the quad bus with a variable or "panned" signal
representing a particular position. Two channels 49, 51 of the
quad bus are meant for use as stereo or the front half of
quadraphonic sound; the other two channels, 48, 50, are for
the rear half of quadraphonic sound. Normally, each position
or input of a modern recording console has a panning control
to place the sound of the input between left, right, front or
back via the quad bus. A recording console may have any number
of inputs or positions which are combined into the quad bus
as four separate outputs. The left front quad bus channel
address apparatus input 49; the right front quad bus channel
addresses apparatus input 51; the left rear quad bus channel
addresses apparatus input 48; and, the right rear quad bus
channel address apparatus input 50. Alternate insertion of the
apparatus of Figure ~ is possible in the absence of a quad bus
by using the stereo bus plus two effect buses. Left front
input 49 (unprocessed) is connected to amplifier 52. Left rear
input 48 (to processed) is connected to amplifier 54. Right
rear input 50 (to processed) is connected to amplifier 56.
Right front input 51 (unprocessed) is connected to amplifier
58. The outputs of amplifiers 52 and 58 are respectively
connected to adders 60 and 62, respectively, so that
amplifiers 52 and 58 effectively bypass the enhancement system
100. The use of the quad bus allows the apparatus to address
W094tl6~8 2 1 5 3 0 6 2 PCT~S93/~
16
its function to each input of a live session or each track of
recorded multi-track information, separately. This means that,
in production, the operator/engineer can determine the space
density of each track rather than settling for an overall
space density. This additional degree of creative latitude is
unique to this apparatus and sets it apart as a production
tool.
The amplified left and right signals in lines 68 and
70 are both connected to summing amplifier 72 and differencing
amplifier 74. The output in line 76 iSr thus, L+R, but the
amplifier 72 also serves to invert the output so that it
appears as -(L-R). These sum and difference signals in lines
76 and 78 are added together in adder 60 and generate the left
program with a conditioning signal CL which adds additional
spatial effects to the left channel. The signal in line 78
also goes through invertor 80 to produce in line 82 the (L-R)
signal. Lines 76 and 82 are introduced into adder 62 to
generate in its output line 84 the right program with
conditioning signal CR. The output lines 79 and 84 from adders
60 and 62 go to the balanced-output amplifiers 86 and 88 for
the left output and 90 and 92 for the right output. The output
amplifiers are preferably differential amplifiers operating
as a left pair and a right pair, with one of each pair
operating in inverse polarity with the other half of each pair
25 for balanced line output.
The conditioning signals CL and CR are similar to
conditioning signal C of Figure 6, but are synthetically
produced. Also, they have somewhat different frequency
filtering which tends to broaden the rear sound images,
30 particularly the antiphasic position 34 (Figure 4).
Conditioning signals CL and CR derived from the difference
signal -(L-R) in line 78 at the output of differencing
amplifier 74. The difference signal in line 78 passes through
high pass filter 94 which has a slope of about 18 decibels per
35 octave and a cutoff frequency of about 300 HZ to prevent comb
filtering effects at lower frequencies. The filtered signal
preferably, but not necessarily, passes through delay 96 with
an adjustable and selectable delay as manually input from
manual control 98, which is called "the Dimension Control".
WO g4/16~8 2 I S3 ~ ~2 17 PCT~S93/~6~
The output of the delay 96 goes to voltage controlled
amplifier (VCA) 102 which provides level control. The DC
control voltage in line 104 wpich controls voltage control
amplifier 102, is supplied by potentiometer 106, in the Manual
Mode, or by the hereinafter described control circuit in the
Automatic Mode. Potentiometer 106 provides a DC voltage
divided down from a DC source 107. It functions as a "Space
Control" and it effectively controls the amount of expansion
of the sound perceived by a listener, i.e., it controls the
amount of the conditioning signal which is added and
subtracted from the left and right channel signals.
The output from voltage controlled amplifier 102 in
line 108 is preferably connected via left equalizer 110 and
right equalizer 112 for proper equalization and phasing for
the individual left and right channels, which tends to broaden
the rear image. The illustrated equalizers 110 and 112 are of
the resonant type (although they could be any type) with a
mid-band boost of 2 db at a left channel center frequency in
equalizer 110 of about 1.5 kilohertz and a right channel
frequency in equalizer 112 of about 3 kilohertz. After passing
through the equalization circuits, the left conditioning
signal _CL occurs in line 114 and the right conditioning
signal _CR occurs in line 116. The left conditioning signal
_CL is added in adder 60. The right conditioning signal in
25 line 116 is connected to invertor 80 where the conditioning
signal _CR is added to the difference signal -(L-R) and the
sum is added to the sum signal to result in the right signal
minus right conditioning signal on line 84 and left signal
plus left conditi~ning signal on line 79.
The automatic control circuit generally indicated
at 118 monitors the output signal in line 79 and 84 and
regulates the amount of conditioning signal to keep a
Lissajous figure generated on an X-Y oscilloscope, connected
to the outputs, relatively constant. The Lissajous figure is
a figure displayed on the CRT of an oscilloscope when the two
outputs are connected to the sweep and amplitude drives of the
oscilloscope. When the Lissajous figure is fairly round, the
energy ratio between the sum and difference of the two outputs
is substantially equal (a desirable characteristic). Lines 84
W094116~8 215 3 0 62 PCT~S93tl2688
18
and 79 are respectively connected to the inputs of
differencing amplifier 120 and adding amplifier 122. The
outputs are respectively rectif~ied, and rectifiers 124 and 126
provide signals in line 128 and 130. The signal in lines 128
and 130 are, thus, the full wave rectified sum and difference
signals of the apparatus output respectively out of subtractor
120 and adder 122.
Lines 128 and 130 are connected to filter 132 and
134 which have adjustable rise and fall ballistics. Selector
switch 136 selects between the manual and automatic control
of the control voltage in line 104 to voltage controlled
amplifier 102. The manual position of selector switch 136 is
shown in Figure 7(a), and the use of the space expansion
control potentiometer 106 has been previously described. There
are several individual switches controlled by selector switch
136, as indicated in Figure 7(a). When the space control
switch is switched to the other, automatic position, the
outputs of filters 132 and 134 in lines 138 and 140,
respectively, are processed and are employed to control
voltage control amplifier 102.
When space control selector switch 136 is in the
automatic position, the output of error amplifier 142 is
connected through gate 144 to control the voltage in line 104.
The error amplifier 142 has inputs directly from line 138 and
from 140 through switch segment 146 and back through line 148.
The filtered sum signal in line 140 is connected through the
space expansion potentiometer 106 so that it can be used to
reduce the apparent level of the output sum information to
- error amplifier 142 to force the error amplifier 142 to reduce
the sum/difference ratio.
Comparator 150 is connected to receive the filtered
sum and difference information in lines 138 and 140.
Comparator 150 provides an output into gate line 152 when
space control selector switch 136 is in the automatic mode and
when a monophonic signal is present at inputs 48 and 50. This
occurs, for example, when an announcer speaks between music
material. When comparator 150 senses monophonic material, gate
line 152 turns off gate 144 to shut down voltage controlled
amplifier 102 to stop the conditioning signal. This is done
- W094/1~8 21 S 30 62 PCT~S931~688
to avoid excessive increase in stereo noise, from random phase
and amplitude changes, while the input program material is
fully balanced. The automatic control circuit 118 cannot
distinguish between unwanted noise and desired program
material containing difference information. Therefore, a
threshold ratio is established between the sum and difference
information in lines 138 and 140 by control of the input
potentiometer into comparator 150. The comparator 150 and gate
144 thus avoid the addition of false space information in a
conditioning signal which, in reality, would be response to
difference-noise in the two channels. The comparator 150 thus
requires a specific threshold ratio between the sum and
difference information, under which the gate 144 is turned off
and over which the gate 144 is turned on.
Clipping circuit 153, see the center left of Figure
7(a), is provided to present a signal when the system is
almost in a clipping situation and another signal when
clipping is present. "Clipping" is a rapid increase in
distortion caused by dynamic peaks in the program material
being limited by the static limit imposed by the power supply
voltage in the circuit. Lines 154 and 156 which are the inputs
of amplifiers 52 and 58, are connected, along with lines 68,
70, 79 and 84, each through their own diode to bus 158. Bus
158 is connected through a resistance to input 160 of
comparator 162. A negative constant voltage source is
connected through another resistor to the input 160, and the
comparator 162 is also connected to ground. By management of
the two resistors, the comparator 162 has an input when bus
158 reaches a particular level. When that level is rea~hed,
output signal 164, such as a signal light, is actuated. Bus
158 is similarly connected through a resistor to the input 166
of comparator 168. The negative voltage source is connected
through another resistor to input 166, and the resistance
values are adjusted so that comparator 168 has an input when
clipping is taking place. Latching circuit 170 is actuated
when clipping has taken place to illuminate the two signal
lights 172 and 174. Those lights stay illuminated until reset
176 is actuated.
In the cutting of V-groove stereo records, a
W094/16~8 PCT~S93/12688
21~306~ 20
difference signal results in vertical motion. Vertical motion
is the most difficult to track in playback. Therefore, large
signals which produce too much v~ertical motion when referenced
to lateral motion are usually avoided. It can be considered
saturation of the cutting function. Not exceeding the
saturation point is extremely important in proper disk
cutting. In FM broadcasting, similar restraints still apply,
since governmental regulatory bodies tend to require that the
difference signal be kept less than the L+R signal. Therefore,
a detection circuit 178 is shown in the lower right corner of
Figure 7(b). The rectified sum and difference signals in lines
130 and 128 are connected to peak followers 180 and 182. The
peaks generated by peak followers 180 and 182 are connected
to comparators 184 and 186. Comparator 184 gives an output
pulse whenever the difference peak envelope becomes greater
than the sum peak envelope, within plus or minus 3 dB. The
level controls at the outputs of the peak followers 180 and
182 allow an adjustment in the plus or minus 6 dB difference
for different applications. Comparator 186 has an output when
sum/difference peak ratio approaches the trigger point of
comparator amplifier 184 within about 2 dB, and lights signal
light 188 on the front panel, illustrated in Figure 7 (b), as
a visual warning of approaching L-R overload. This is
accomplished by reducing the apparent level of the sum
2 5 envelope by about 2 dB with the potentiometer connecting
comparator 186 to ground. The output of comparator amplifier
184 feeds a latching circuit 190 which activates light 195 and
which holds until reset by switch 192. When the latching
circuit is active, it activates driving circuit 194 which
lights panel lights 196 and 197 and, after a time delay, rings
audible alarm 198. At the same time, driving circuit 194
energizes line 199 which cuts off gate 144 to withhold the
signal to amplifier 102 which controls the conditioning
signal. Actuation of gate 144 removes the conditioning signal
from line 108, but permits the normal stereo signal to
continue through the circuit.
Third Embodiment
A third embodiment of the system and apparatus of
~ WO94/16~8 PCT~S931126~
2I 530 ~2 21
this invention is shown in Figure 8 and is generally indicated
at 200. For reasons already stated with respect to the system
and apparatus loO of Figures 7(a) and (b), the left front quad
bus channel address unprocessed input 49 which is connected
to amplifier 204; the left rear quad bus channel address
processed input 48 which is connected to amplifier 206; the
right rear quad bus channel address processed input 50 which
is connected to amplifier 212; and, the right front quad bus
channel address unprocessed input 51 which is connected to
amplifier 214. Amplifiers 204, 206, 212 and 214 are inverting
and provide signals in lines 208, 210, 216 and 218,
respectively. Both lines 208 and 210 are connected to summing
amplifier 220, while both lines 216 and 218 are connected to
summing amplifier 222. Lines 210 and 216 carry -L and -R
signals.
The conditioning signals CR and _CL are derived by
connecting differencing amplifier 224 to both lines 210 and
216. The resulting difference signal, -(R-L), is filtered in
high pass filter 226, similar to filter 94 in Figure 7(a), and
the result is subject to selected delay in delay circuit 228.
The delay time is controlled from the front panel, as will be
described with respect to Figure 9. The output from delay 228
goes through voltage controlled amplifier 230 which has an
output signal, -C, in line 232, which is supplied to both
non-inverting equalizer 234 and inverting equalizer 236. Those
equalizers respectively have conditioning signal outputs _CL
and +CR which are connected to the inverting summing
amplifiers 220 and 222. The left conditioning signal _CL is
added (and inverted) with the o~iginal left signal at
amplifier 220 to form L+CL, and the right conditioning signal
+CR is effectively subtracted from the original right signal
at invertor amplifier 222 to form R_CR. The outputs from
amplifiers 220 and 222, in lines 238 and 240, respectively,
are preferably and respectively connected to balanced left
amplifiers 242 and 244 and balanced riqht amplifiers 246 and
248, in the manner described with respect to amplifiers 86
through 92 of Figure 7(b). It may be useful to connect the
various points in the circuit of Figure 8 to the clipping and
L-R overload warning circuits 153 and 178 in the same manner
W094/16538 PCT~S93/~ -
2153062 22
as previously described with reference to Figure 7(b).
Alternatively, VCA 230 may be manually controlled by a
potentiometer and DC supply compination, such as potentiometer
106 and supply 107. The difference between the two embodiments
S of the system in Figures 7(a), 7(b) and 8 lies in the way the
original left and right signals are routed. In Figures 7(a)
and (b), the left and right signals are added and subtracted.
This sum and difference information is then re-added and
re-subtracted to reconstruct the original left and right
signals. In the circuit of Figure 8, the original left and
right signals are not mixed together. They remain independent
of each other from input to output.
The enhancement system may be automatic with
self-controlling features in the apparatus so that the
stereophonic image enhancement can be achieved without
continual adjustment of the system and apparatus.
Alternatively, manual control may be used, if desired.
The foregoing description of the invention, as it
has been described with reference to the detailed circuitry
shown in Figures 7(a), 7(b) and 8, has been basically in
analog terms with the various elements of the circuitry being
either analog devices or devices which could be either analog
or digital. For example, the delay line devices 96 (Figure
7(a)) and 228 (Figure 8) are more likely to be implemented
using digital components than by using analog components.
Thus, an analog to digital converter might be used immediately
prior to a linear digital delay line 96, 228 whose output can
than be converted to analog using a digital to analog
converter.
Alternatively, and preferably for the professional
equipment, predominately digital implementations of the
invention are quite practicable, as will be seen in the
following embodiments.
Fourth Embodiment
Turning now to Figures lO(a) and lO(b), they form
a digital logic diagram of a digital embodiment of the
invention which is conceptually somewhat similar to the
analog, or mostly analog, embodiment of Figures 7(a) and (b).
W094/16~8 21 530 ~2 PCT~S93/~8
23
In Figures lO(a) and (b), data transmission lines are shown
in solid lines, while control lines are shown in dashed lines.
Left and right audio channel information is supplied
in multiplexed digital format an input 302. Clock information
is also supplied at an input 304 to a formatter 306-which
separates the left channel information from the right channel
information. Preferably, formatter 306 de-multiplexes the
digital data which can be supplied in different multiplexed
synch schemes. For example, a first scheme might assume that
the data is being transmitted via a Crystal Semiconductor
model CS8402 chip for AES-EBU, S-PDIF inputs, or a second
scheme might assume that the digital data comes from an analog
to digital converter such as a Crystal Semiconductor model
CS5328 chip. The I/0 mode input 305 preferably advises the
formatter 306 at the front end and the formatter 370 at the
rear end of the type of de-multiplexing and multiplexing
schemes required for the chips upstream and downstream of the
circuitry shown in Figures lO(a) and (b). Those skilled in the
art will appreciate that other multiplexing and
de-multiplexing schemes can be used or that the left and right
channel data could be transmitted in parallel, i.e.,
non-multiplexed data paths.
The left channel digital audio data appears on line
308 while the right channel digital audio data appears on line
309. This data is subtracted from each other at a subtractor
324 to form R-L data. The R-L data is supplied to a switch 329
and may be filtered though a high-pass filter 326, and a low
pass filter 327 and is subjected to digital time delay at
device 328. The signal is filtered by filter 310 having a
narrow band pass preferably centered at 500 Hz with 6
dB/octave slopes on either side of its center frequency.
Those skilled in the art will appreciate that filter 326, 327
and 310 are represented as they might be, i.e., as separate
filters, in an analog embodiment. In a digital embodiment,
the functions of filters 326, 327 and 310 are preferably
implemented in a digital signal processor and they, along with
delay 328, be preformed in different sequences and the
functions can be combined as a matter of design choice. If
desired, filters 326, 327 may be eliminated.
W094116~8 ~ 1~ 3 ~ 6 2 PCT~S93/~688 --
24
Switch 329 is controlled by a C-mode control 303
which effectively controls the position of switch 329, which
is shown in Figures lO(a) and~ (b), in its C-mode position,
that is, where the filters 326, 327 and 310 and the time delay
328 are bypassed. The C-mode is preferably used when the
apparatus is used with live sources, such as might be
encountered during a concert or a theatrical performance, and
a C microphone input source (Figures 5 and 6) is available,
so that the C signal then need not be synthetically produced.
The R-L data is preferably subjected to the filtering and time
delay to generate the conditioning signal C when the invention
is used to mixdown a recorded performance from a multi-track
tape deck, for example.
The output from switch 329 is supplied to a variable
gain digital circuit 330 which is functionally similar to the
voltage controlled amplifier 102 shown in Figure 7(b). A mute
control input can be used to reduce the gain at gain control
330 very quickly, if desired. The output of variable gain
digital circuit 330 is applied to an adder 320 and to a
subtractor 332 so that the control signal C is added and
subtracted from the left audio data and right audio data on
lines 379 and 384, respectively. That data is then multiplexed
at formatter 370 and output in digital form at serial output
390.
The variable gain circuitry 330, which can be
implemented rather easily in the digital domain by shifting
bits, for example, is controlled either from a manual source
or an automatic source, much like the voltaqe controlled
- amplifier 102 of Figure 7(b). In the manual position of switch
367 shown in Figure lO(b), the gain through circuitry 330 is
controlled by a "space control" input 362 which is
conceptually similar to the space control potentiometer 106
shown in Figure 7(a) and the potentiometer shown Figure 6. In
the automatic position of switch 367, the gain in circuitry
330 is automatically controlled in a manner similar to that
of Figures 7(a) and (b). In Figures lO(a) and (b), the data
on lines 379 and 384 are summed at a summer 342 and, at the
same time, subtracted at subtractor 340. The outputs are
respectively applied to high-pass filters 346 and 344, whose
W094/1~8 21 ~306^2 PCT~S93112688
outputs are in turn applied to root mean square (RMS)
detectors 350 and 348, respectively. Detector 348 outputs a
log difference signal, while d~etector 350 outputs a log sum
signal. The value of the log difference signal from detector
348 can be controlled from the "Space In" input 362 at adder
352, in the automatic mode, so that the "Space In" value
offsets the output of the log difference detector, either:
(1) 00 for a difference level 12 dB below the sum
level;
(2) 80 for a difference level equal to the sum
level; and
(3) FF for a difference level 12 dB over the sum
level.
The output of adder 352 and the log sum output from
detector 350 are applied to a comparator 354, which is
conceptually similar to the comparator 150 of Figure 7(a). The
output of comparator 354 is applied to a rate limiter 356
which preferably limits the rate at which the output from
comparator 354 limits the rate of gain change of circuit 330
to approximately 8 dB per second.
Those skilled in the art will appreciate that the
circuitry shown in Figures lO(a) and (b), instead of
implementing it in discrete digital circuitry, preferably can
be implemented by programming a digital signal processor chip,
such as the model DSP 56001 chip manufactured by Motorolla,
by known means.
The automatic control circuitry 378 is also shown
in Figures lO(a) and (b). When switch 367 is in its automatic
position, the automatic control circuitry 378 effectively
controls the amount of spatial effect added by the invention
depending upon the amount of spatial material initially in the
left and right audio. That is to say, if the left and right
audio data being input into the circuitry has high spatial
impressions already, the amount of spatial effect added by the
present invention is less than if the incoming material has
less spatial impression information in it originally. The
control circuitry 378 also helps to keep the envelope of the
L-R signal less than the envelope of the L+R signal. That can
W094/16~8 2 15 3 0 6 2 PCT~S93/12~ -
26
be important for FM and television broadcasting where
governmental agencies, such as the FCC in the United States,
often prefer that the broadcast~ L-R signal be no greater than
the L+R signal. Thus, the embodiment of the invention
disclosed with respect to Figures lO(a) and (b), is
particularly useful in connection with the broadcast industry
where the spatial effects added by the circuitry can be
automatically controlled without the need for constant manual
input. It also should be emphasized that the present invention
is completely mono-compatible, that is to say, when the
present invention is used to enhance the spatial effects in
either a radio FM broadcast or a Television sound FM
broadcast, those receivers which are not equipped with stereo
decoding circuitry, do not produce any undesirable effects in
their reproduction of the L+R signal due the spatial effects
which are added by the present invention to the L-R signal
being broadcast.
The R/L equalization on line 312 controls the amount
of boost provided by filter. That boost is currently set in
the range of 0 to +8 dB and more preferably at +4 dB. The
center frequency of filter 310 is preferably preferably set
at 500 Hz, but it has been determined filter 310 may have
center frequencies in the range of 300 Hz to 3 kHz.
The WARP In input to time delay 328 adjusts the time
delay. The time delay is preferably set at zero delay for
audio reproduction, 1.0 mSec for broadcasting applications,
4-6 mSec for mechanical record cutting, and up to 8 mSec for
cinematic production applications.
Fifth Embodiment
While the automatic mode version of the present
invention can be very useful in broadcasting, the manual mode
of operation of the present invention will be very important
for the recording industry and for the production of theater,
concerts and the like, that is, in those applications in which
large multichannel sound mixing panels are currently used.
Such audio equipment usually has a reasonable number of audio
inputs, or audio channels, each of which are essentially mono.
The sound recording engineer has control of not only the
W094l16538 ~3 ~ ~2 PCT~S931~8
_ 27
levels of each one of the channels but also, in the prior art,
uses a pan control to control how much of the mono signal
coming into the sound board goes into the left channel and how
much goes into the right channel. Additionally, the engineer
can control the amount of the signal going to the rear left
and rear right channels on a quad bus audio board.
Turning again to Figure 4, the pan control of the
prior art permits a sound source point image 32 to be located
anywhere on the line between the left and right speakers 12
and 14 depending on the position of the pan control. For that
simple reason, stereo recording was a large improvement over
the mono recordings of forty years ago. Just imagine, however,
the even greater effect which can impart to a listener 16 if
the image point can be moved anywhere: not only between the
two speakers, but to the left of the left speaker or to the
right of the right speaker, to the foreground position (such
as point 36) shown in Figure 4, or even to a point behind the
listener such as the antiphasic image position 34 shown in
Figure 4. The present invention provides audio engineers with
such capabilities. Instead of using two pan controls such as
can be found on a quad deck, the audio engineer will be
provided with a joystick by which he or she will be able to
move the sound image both left and right and front and back
at the same time. The joystick can be kept in a given position
during the course of an audio recording session, a theatrical
or concert production, or alternatively, the position of the
joystick can be changed during such recording sessions or
performances. That is to say, the image position of the sound
can be moved with respect to a listener 16 to the left and
right and forward and back, as desired. If desired, the
effective position of the joystick can be controlled by a MIDI
interface.
Initially, in connection with the audio recording
and mix down industries, the present invention will likely be
packaged as an add-on device which can be used with
conventional audio mixing boards. In the future, however, the
present invention will likely find its way into the audio
mixing board itself, the joystick controls (discussed above)
being substituted for the linear pan control of present
WO ~116~8 2 1 ~ 3 0 6 2 PCT~S93/~688
28
technology audio mixing boards.
Figure 11 shows the outward configuration of audio
components using the present invention which can be used with
conventional audio mixing boards known today. As shown in
Figure 11, the device has twenty-four mono inputs and
twenty-four joysticks, one for each input. Preferably, the
equipment comprises a control box 400 and a number of joystick
boxes 410 which are coupled to the control box 400 by a data
line 420. The joystick box 410 (shown in Figure 11) has eight
joysticks associated with it and is arranged so that it can
be daisy-chained with other joystick boxes 410, coupling with
the control box 400 by data cable 420 in a serial fashion.
Instead of having only eight joysticks in joystick box 410,
the joystick box 410 could have all twenty-four joysticks, one
for each channel, and, moreover, the number of joysticks and
channels can be varied as a matter of design choice. At
present it is preferred to package the invention as shown,
with eight joysticks in one joystick box 410. In due time,
however, it is believed that this invention will work its way
into the audio console itself, wherein the joysticks will
replace the panning controls presently found on audio
consoles.
This embodiment of the invention has enhanced
processed, left and right outputs 430 and 432 wherein all the
inputs have been processed left and right, front and back,
according to the position of the respective joysticks 415.
These outputs can be seen on control box 400. Unprocessed
outputs are also preferably provided in the form of a direct
- left 434, a direct right 436, a direct front 438 and a direct
back 440 output, which are useful in some applications where
the mixing panel is used downstream of the control box, and
the audio engineer then has the ability to mix processed left
and/or right outputs, with unprocessed outputs, when desired.
Figures 12(a)-12(d) form a schematic diagram of the
invention, particularly as it may be used with respect to the
joystick embodiment. Turning now to Figures 12(a)-(d),
twenty-four inputs are shown at numerals 405-1 through 405-24.
Each input 405 is coupled to an input control circuit 404,
each associated with an input 405. Since, in this embodiment,
W094116~8 2 1 ~ 3 0 62 29 PCT~S93/~8
there are twenty-four inputs 405, there are twenty-four input
control circuits 404-1 through 404-24. However, only one of
which, namely 404-1, is shown in detail, it being understood
that the others, namely 404-2 through 404-24, are preferably
of the same design as 404-1. The input control circuitry 404
is responsive to the position of its associated joystick for
the purpose of distributing the incoming signal at input 405,
on to bus 408. Each joystick provides conventional X and Y dc
Voltage signals indicative of the position of the joystick
which signal's are converted to digital data, the data being
used to address six look-up tables, a look-up table being
associated with each of the voltage controlled amplifiers
(VCA's) 407 which comprise an input circuit 404. The value in
the table for a particular X and Y coordinates of the joystick
indicate the gain of its associated VCA 407. The digital
output of the look-up table is converted to an analog signal
for its associated VCA 407. Each VCA 407 has a gain between
unity and zero, depending on the value of the analog control
voltage signal. Thus, from 0% to 100% of the signal being
input at input 405-1, for example, finds its way on to the
various lines forming bus 408 depending upon the position of
joystick 415-1. Similarly, input 405-2 has its input
distributed amongst the various lines making up bus 408,
depending upon the position of its joystick 415-2. The same
thing is true for the remaining inputs and remaining
joysticks. Also, as will be seen, the distribution of the
signals is controlled somewhat by the position of a switch
409, whose function will be discussed in due course.
The currently preferred values in the look-up tables
are tabulated in Tables G-X. The data in Tables G-X
correspond to the action of VCA's 407L, 407F, 407R, 407BL,
407M, and 407BR, assuming that the position of the joystick
is resolved to 5 bits in its x-axis and to 5 bits in its y-
axis. As such, the position of the joystick can be resolved
to one of 32 positions along an x-axis and to one of 32
positions along a y-axis. Hence, each Table has 32 by 32
entries, corresponding to the possible position of the
joystick. In practice, the x and y position information is
preferably resolved to greater precision (for example, to 320
wo g4,l6~8 2 1 S 3 0 6 2 PCT~S93l~6~
x 320) and the data points are interpolated for those x and
y coordinate positions between the data points set forth in
the Tables. With this data, the joysticks can move the sound
source to the front, back, to the right or left by moving the
joystick to a corresponding position. The data in Tables A-F,
and graphically depicted in Figures 13(a)-(f), are
conceptually similar but the full left and full right
positions are in the left and right quadrants of the joystick.
These Tables and Figures show the percentage of the signal
input at an input 405 which finds its way onto the various
lines comprising bus 408, where the various signals on each
line of the bus from different input circuits 404 are summed
together. The Tables G-X show the percentages for various
positions of a joystick 415 as it is moved left and right and
front and rear. Table G, which is associated with VCA 407L,
indicates that VCA 407L outputs 100% of the inputted signal
when its associated joystick is moved to the position maximum
left and maximum front. The outputted signal from VCA 407L
drops to under 20% of the inputted signal when the joystick
is moved to its maximum right, maximum back (or rear)
position. Other positions of the joystick cause VCA 407L to
output the indicated percentage of the inputted signal at 405.
VCA 407L, receives a control voltage input VCX-L for
controlling the amount of the input signal at 405 which finds
its way onto bus 408L. Similarly, VCA 407R controls the amount
of input signal at 405 which finds its way onto line 408R. The
same thing is true for VCA's 407F, 407R, etc. The voltage
control amplifiers 407 in the remaining input circuits 404-2
through 404-24, are also coupled to bus 408 in a similar
fashion and, thus, the current supplied by the voltage control
amplifiers 407 are summed onto that bus structure. Thus, the
various input signals 405-1 through 405-24 are steered, or
mapped, onto the appropriate line of bus 408 depending upon
the position of the respective joysticks 415-1 through 415-24.
The signals on lines comprising bus 408 are then converted
back into voltages by summing amplifiers 409, each of which
is identified by subscript letter or letters corresponding to
the line of bus with which they are coupled. The outputs of
summing amplifiers 409L, 409R, and 409F are applied directly
W094/16538 PCT~S93t~6~
2153062 31
to three of the four direct outputs, 434, 436 and 438,
respectively. The direct back output 440 is summed from the
output of the summing amplifiers 409CDL, 409CDR, 409EL and
409ER.
Tables G-X are the preferred tables when the
invention is used for cinema production.
Before going deeper into the description, it might
be helpful to the reader to explain some of the terminology,
particularly the subscripts which are being used in this
description. The reader has probably noted that the letter "L"
is associated with the left channel, the letter "R" with the
right channel, the letter "F" with front and the letter "M"
with mono-compatibility. The letters "BL" mean back left and
the letters "BR" mean back right. The perceived sound
locations for L, R, F, BL and BR are shown in Figure 2, for
example. The letter "C" is associated with the C-mode of
operation, which was briefly discussed with reference to
Figures lO(a) and (b). There is also a D-mode of operation and
an E-mode of operation in the embodiment of the invention now
being described. The mode of each input 405 is controllable
from a controller 410. See, for example, Figure 11 where for
each joystick 415 there is a mode switch 411 which can be
repeatedly pushed to change from mode C, to mode D, to mode
EL, to mode ER, and then back to mode C. In mode C and D,
switches 409L and 409R are in the position shown in Figure
12(c). Switch 409L changes position when in mode EL, while
switch 409R changes position when in mode ER Light emitting
diodes (LED's) 412L and 412R of Figure 11, report the mode in
which the controller is for each channel. For example, LED's
412L and 412R may both be amber while in mode C, may both be
green while in mode D, while in mode EL the left LED (412L)
would be preferably red while the right LED (412R) would be
off, and an opposite convention when in mode ER
Mode C is preferably used for live microphone array
recording of instruments, groups, ensembles, choirs, sound
effects and natural sounds, where a microphone array can be
placed at the locations shown in Figure 5. Mode D is a
directional mode which places a mono-source to any desired
location within the listeners conceptual image space, shown
WO94116~8 2 1 5 3 0 6 2 32 PCT~S93/~6~
in Figure 4, for example. Applications on mode D are in
multi-track mix-down, commercial program production, dialogue
and sound effects, and concert~sound reinforcement.
Mode E expands a stereo source and, therefore, each
input is associated with either a left channel (mode EL), or
a right channel (mode ER) of the stereo source. This mode can
be used to simulate stereo from mono-sources and allows
placement within the listener's conceptual image space, as
previously discussed. Its applications are the same as for
mode D.
Returning to Figures 12(a) and (b), the output from
summing amplifiers 409CDL and 409CDR, correspond to the back
left and back right signals for the C and D-modes. The signals
are applied to a stereo analog-to-digital converter 412CD
which multiplexes its output onto line 414CD. Similarly,
stereo analog-to-digital converter 412E takes the E-mode back
left and E-mode back right analog data, and converts it to
multiplexed digital data on line 414E. The digital data on
lines 414CD and 414E are applied to digital sound processors
(DSP's) 450 which will be subsequently described with
reference to Figures 14(a) and (b). The audio processors may
be identical, and may receive external data for the purpose
of determining whether they operate in the C-mode, D-mode or
E-mode, as will be described. The programming of the digital
sound processor (DSP) 450 can be done at a mask level or it
can be programmed in a manner known in the art by a
microprocessor attached to a port on DSP 450 which
microprocessor then downloads data stored in E proms or ROM's
into the DSP 45e during initial power-up of the apparatus. The
current preference is to use model 56001 DSP's manufactured
by Motorolla. In practicing the present invention, it is
preferred to download the programming into the DSP 450 chips
using a microprocessor, since that makes it easier to
implement design changes should that become necessary. In due
course, it will be preferred to use mask level programming
since that should make the device more economical to produce.
In any event, the programming emulates the digital logic shown
in Figures 14(a) and (b). The outputs from the DSP 450 chips
are again converted back to analog signals by stereo digital
WO94/16~8 2 1 ~ 3 0 62 ` PCT~S93/~688
_ 33
to analog converters 418CD and 418E. The outputs of stereo
digital to analog converters 418CD and 418E are summed along
with outputs from the mono co,mpatibility channel, the front
channel 409F, the right channel 409R and the left channel
409L, through summing resistors 419, before being applied to
summing amplifiers 425L and 425R and thence to processed
stereo outputs 430 and 432. The summing resistors 419 all
preferably have the same value. The mono compatibility signal
from summing amplifier 409M is applied to a low and high-pass
equalization circuit which preferably has a low q typically
q on the order of .2 or .3, centered around 1,700 Hz.
Equalization circuit 422 typically has a 6 dB loss at 1,700
Hz.
In the D-mode, processed directional enhancement
information, i.e., the conditioning signal C, is added (and
subtracted) to the output channels. This information is band
pass filtered by filters 456 and 457, for example, so that it
peaks in the mid-range. If the enhanced left and right signals
are summed together to form a L+R mono signal, this can show
up as a notch in the spectrum in that mid-range area. To
counteract that effect, the mono compatibility signal is
preferably used which has a notch which is the antithesis of
the mid-range notch and which, in effect, balances the output
spectrum of a L+R mono signal. When the joystick is in the
center, equal amounts of the conditioning signals go to the
left and right channels and when those channels are summed to
form the R+L signal, the conditioning signal is effectively
canceled out since it was originally added to one channel and
subtracted from the other channel. So, with a back-centered
joystick, some mono compatibility signal is needed, and can
be seen in Table E and in Figure 13(e), for example, where the
VCX-M input to VCA 407M goes to approximately -5 dB (60%) when
the joystick is centered between left and right, but pulled
all the way towards the back. It should be understood by the
reader that spatial enhancement and mono compatibility of the
perceived conceptual image space and collapsed sound field is
achieved within a surprisingly very narrow difference range
of a few dB. This is the nature of human hearing with respect
to the psychoacoustic phenomena toward which this invention
W094/16538 PCT~S93112~8
~ 21 ~3 0 62 34
is directed.
Turning now to Figures 14(a) and 14(b), these
figures form a sound processor~ logic diagram similar to data
Figures lO(a) and (b), but with a number of changes, the most
important of which follows:
(1) There is no need for an automatic control
circuit 378, as shown in Figures lO(a) and (b) since, in this
embodiment, the amount of expansion (the amount of spatial
effects which are added) is controlled manually by the
position of the joystick 415. There is also no variable gain
circuit such as 330 (Figure lO(b)) since, the amount of gain
is controlled by the position of the joystick 415 as it, in
turn, controls the gain of the various VCA's 407(Figures 12(a)
and (c)).
(2) The embodiment of Figures lO(a) and (b) operated
in either a C-mode or a non-C expansion mode (which is
identified as mode E in Figures 12(a), 12(b), 14(a) and
14(b)). The embodiment of Figures 12(a), 12(b), 14(a) and
14(b) also include another mode (mode D) which, as will be
seen, causes certain changes to be made to the audio processor
logic of Figures 14(a) and (b) compared to the audio processor
logic of Figures lO(a) and (b). Referring again to Figures
14(a) and (b), the incoming serial data which was multiplexed
onto line 414, is de-multiplexed by formatter 451. Preferably,
the stereo A to D converters 412 (see Figure 12(d)) are
Crystal Semiconductor model CS5328 chips, while the stereo D
to A converters 418 (see Figure 12(d)) are Crystal
Semiconductor model CS4328 chips and, therefore,-formatters
451 and 470 would be set up to de-multiplex and multiplex the
left and right digital channel information in a matter
appropriate for those chips. The left and right digital data
is separated onto buses 452 and 453, and is communicated to,
for example, a subtractor 454, to produce a R-L signal. The
R-L signal passes through the low pass and high pass filters
456 and 457 and the time delay circuit 458, when the circuit
is connected in the E-mode as depicted by switch 455 (which
is controlled by an E-mode control signal). When in the
D-mode, switches 455 take the other position shown in
W094/16~8 21 ~ ~ 0 62 PCT~S93/126~
schematic diagram and, therefore, the left channel digital
data on line 452 is passed through the top set of high pass
and low pass filters 456 and 457 and the time delay circuit
458, while the right channel digital data on line 453 is
directed through the lower set of high pass and low pass
filter 456 and 457 and time delay circuit 458. There is no
need to control the amplitude of the signal from the time
delay circuits 458, as was done in the embodiment of Figures
lO(a) and (b), because of the fact that the amplitude the
signals are being controlled at the input control circuits 404
of Figures 12(a) and (c) and the amount of pr~cessing is being
controlled at the input by the position of joysticks 415 (see
Figure 11). The outputs of time delay circuits 458 are applied
to respective left and right channel equalization circuits
460. The output of the left equalization circuit 460L is
applied via a switch 462 to an input of formatter 470. The
output of the right equalization circuit 460R is applied via
a switch 462 and an invertor 465 to an input of formatter 470.
As previously indicated, formatter 470 multiplexes the signals
received at its inputs onto serial output line 416.
Time delay circuits 458 preferably add a time delay
of 0.2 millisecond. It is to be appreciated that the DSPs'
450 and their associated A to D and D to A converters 412E,
412CD, 418E and 418CD have inherent delays of about 0.8
millesecond. Thus, the total delay produced by the inherent
delay of the circuit devices and the added delay in time delay
circuits 458 total about 1 millisecond compared to the left
and right analog signals from amplifiers 409L and 409R.
Switches 462 are shown in the C-mode position, which
has been previously described. When in the D-mode or the
E-mode, the switches 462 change position so as to communicate
the outputs of the equalizers 460 to the formatter and
invertor 465, as opposed to communicating the unfiltered
signals on lines 452 and 453 which is done when in the C-mode.
The inversion which occurs in the right channel information
by invertor 465, is effectively done by subtractor 332 in the
embodiment of Figures lO(a) and (b). It is to be recalled that
subtractor 332 subtracts the right channel conditioning
information CR (from equalizer 312), from the right channel
W094/16~8 PCT~Sg31~688
21 $30~2 36
audio data. In the embodiment of Figures 14(a) and (b), the
right channel conditioning signal is inverted by invertor 465.
It is then communicated via formatter 470 and the stereo
digital to analog converter 418 (see Figure 12(d)) onto a
summing bus, where it is summed through resistors 419, along
with the right channel information from summing amp 409R, into
an input of summing amp 425R. The left channel conditioning
signal CL is communicated, without inversion, via formatter
470 and the stereo digital to analog converter 418 (see Figure
12(d)) onto a summing bus where it is summed through resistors
419, along with the left channel information from summing amp
409L, into an input of summing amp 425L.
The invention has been described with respect to
both analog and digital implementations, and with respect to
several modes of operation. The broadcast mode, mode B, uses
a feedback loop to control the amount of processing being
added to stereo signals. In the C, D and E-modes, the amount
of processing being added is controlled manually. In the final
embodiment disclosed, the amount of processing is input
controlled by joystick. In the C-mode, the conditioning signal
which is added and subtracted from the left and right channel
data, undergoes little or no processing. Indeed, no processing
is required if the conditioning signal is organically produced
by the location of microphone "C" in Figure 5. In the mode C
operation described with reference to Figures 10(a), 10(b),
12(a), 12(b), 14(a) and 14(b), the conditioning signal
bypasses the high pass/low pass filters and the time delay
circuitry. On the other hand, in the D and E-modes, the
conditio~ing signals are synthesized by the high pass/low pass
filter and, preferably, the time delay. In the E-mode it is
a R-L signal which is subjected to filtering, whereas in the
D-mode, the left and right signals are independently subjected
to filtering, for the purpose of generating the conditioning
signal C.
As can be seen by reference to Figures 10(a), 10(b),
14(a) and 14(b), the amount of time delay is controllable.
Indeed, some practicing the instant invention may do away with
time delay altogether. However, time delay is preferably
inserted to de-correlate the signal exiting the filters from
W094/16~8 21 5 3 0 6 2 PCT~S931~688
37
the left and right channel information to help ensure
monocompatibility. Unfortunately, comb filtering effects can
be encountered, but these seem to be subjectively minimized
by filters 456, 457, and 460. In order to minimize such
effects in the B, D and E-modes of operation, it is preferred
to use a time delay circuit such as 328 (Figures lO(a) and
(b)) or 458 (Figures 14(a) and (b)). In the organic mode
(Figure 6), the time delay is organically present due to the
placement of microphone "C" further from the sound source than
microphones "L" and "R".
The present invention can be used to add spatial
effects to sound for both the purposes of recording,
broadcasting, or a public performance. If the spatial effects
of the invention are used, for example, in audio processing
at the time of mixing down a multi-track recording to stereo
for the purposes of release of tapes, records or digital
discs, when the tape, record or digital disc is played back
on conventional stereo equipment, the enhanced spatial effects
will be perceived by the listener. Thus, there is no need for
additional spatial treatment of the sound after it has been
broadcast or after it has been mixed down and recorded for
public distribution on tapes, records, digital discs, etc.
That is to say, there is no need for the addition of spatial
effects at the receiving apparatus or on a home stereo set.
The spatial effects will be perceived by the listener whether
they are broadcast or whether they are heard from a
prerecorded tape, record or digital disc, so long as the
present invention was used in the mixdown or in the broadcast
process.
The present invention is also mono compatible. That
is to say, if a person listens to a L=R signal, for example,
the output at 430 and 432, no artifacts of the process will
be perceived by the listener. This is important for
television, FM and AM stereo broadcast as the greater populace
will continue to listen to mono signals for some time to come.
The present invention, while adding spatial expansion to the
stereo signals, does not induce artifacts of the process in
the L+R signal.
Digital delay devices can delay any frequency for
W094/16538 21 5 3 ~ ~ 2 38 PCT~S93/126~
any time length. Linear digital delays delay all frequencies
by the same duration. Group digital delays can delay different
groups of frequencies by differ~ent durations.
The present invention preferably uses linear digital
delay devices because the effect works using those devices and
because they are less expensive than are group devices.
However, group devices may be used, if desired.
The previously described embodiments, and
particularly the embodiments of Figures 7 through 14(a) and
(b) will be quite useful in the professional audio industry
in the various applications previously mentioned. However,
those embodiments tend to be too complex for convenient use
in consumer electronics equipment such as might be used in the
home. Thus, there is a need for embodiment which may be
conveniently used in consumer quality electronics equipment,
and which preferably can be embodied in an easily manufactured
chip. Such an embodiment is disclosed with reference to Figure
15.
Sixth Embodiment
Figure 15 is a schematic diagram of a sixth
embodiment of the invention, which embodiment can be
relatively easily implemented using a single semiconductor
chip and which may be used in consumer quality electronics
equipment, including stereo reproduction devices, television
receivers, stereo radios and personal computers, for example,
to enhance stereophonically recorded or broadcast music and
sounds.
In Figure 15, the circuit 500 has two inputs, 501
and 502 for the left and right audio channels found within a
typical consumer quality electronic apparatus. The signals at
input 501 are communicated to two operational amplifiers,
namely amplifiers 504 and 505. The signals at input 502 are
communicated to two operational amplifiers, namely 504 and
506. The left and right channels are subtracted from each
other at amplifier 504 which produces an output L-R. That
output is communicated to a potentiometer 503 which
communicates a portion (depending upon the position of
potentiometer 503) of the L-R signal back through a band pass
WO ~116538 t ~30 62 39 PCT~S93/~6~
filter 507 formed by a conventional capacitor and resistor
network.
Filter 507, in addition to band-passing the output
from amplifier 504, also adds some frequency dependent
time-delay (phase delay) to that signal, which is subsequently
applied to an input of amplifier 508. The output of amplifier
508 is the conditioning signal, C, which appears on line 509.
The conditioning signal, C, i5 added to the left channel
information at amplifier 505 and is subtracted from the right
channel information at amplifier 506 and thence output as
spatially enhanced left and right audio channels 511 and 512,
respectively.
Filter 507 preferably has a center frequency of 500
Hz with 6 dB/octave slopes. As previously mentioned, it has
been determined that the center frequency can fall within the
range of about 300 Hz to 3,000 Hz.
Outputs 511 and 512 may then be conveyed to the
inputs of the power amplifier of the consumer quality audio
apparatus and thence to loudspeakers, in the usual fashion.
The listener controls the amount of enhancement added by
adjusting potentiometer 503. If the wiper of potentiometer 503
is put to the ground side, then the stereo audio program will
be heard with its usual un-enhanced sound. However, as the
wiper of potentiometer 503 is adjusted to communicate more and
more of the L-R signal to the band pass filter 507, more and
more additional spatially processed stereo is perceived by the
listener. For example, if the listener happens to be watching
a sporting contest on television which is broadcast with
stereo sound, by adjusting potentiometer 503, the listener
will start to perceive that he or she is actually sitting in
the stadium where the sporting contest is occurring due to
the additional spatial effects which are perceived and
interpreted by the listener.
If the signals at input is mono (i.e., R=L), then
no artifacts of the present process will be perceived by the
listener. If enhancement of a monaural signal (i.e., L=R) is
desiredj then that may be obtained by the eighth embodiment,
which will be subsequently described.
The circuitry of Figure 15 is shown with essentially
WOg4/16538 PCT~S93t~688
2153 62 40
discreet components with the exception of the amplifiers,
which are preferably National Semiconductor model LM837
devices. However, those skille~d in t~e art, will appreciate
that all (or most) of the circuit 500 can be reduced to a
single silicon chip, if desired. Those skilled in the art will
also appreciate that capacitors C1 and C2 in band pass filter
507 will tend to be rather large if implemented on the chip
and, therefore, it may be desirable to provide appropriate
pin-outs from the chip for those devices and to use discreet
devices for capacitors C1 and C2. That is basically a matter
of design choice.
8eventh Embodiment
The sixth embodiment of the invention, which was
described with reference to Figure 15, may be used in consumer
quality electronics equipment. However, as has been made
clear by discussion relative to the earlier described
embodiments, the present invention can also be used
professionally when recording music (and other audio material)
and/or when broadcasting music (and other audio material).
Thus, the present invention can be used to increase the
spatial image of music (or other recorded material) before or
after being recorded on disc, or before or after being
transmitted by a broadcaster or just before being heard by a
listener. It is preferable, however, that when music or other
sounds are spatially enhanced in accordance with the present
invention, that the material not be overly enhanced.
Prev~ously described embodiments of the present invention
include an automatic control circuit 118 which regulates the
amount of the conditioning signal generated in order to keep
the energy ratio between the sum and difference of the two
spatially enhanced outputs substantially equal. In this
regard, the second and third embodiments of the invention
include control circuit 118 which effectively controls the
amount of expansion which occurs.
The seventh embodiment of the invention is a
modified version of the sixth embodiment and includes an
automatic control circuit 518 for automatically controlling
WO94/16~8 PCT~S93/~688
2t 5~062 41
the amount of spatial expansion which occurs. The seventh
embodiment is described with reference to Figures 16, 17A and
17B and is also intended to be used in consumer quality
electronics equipment as is the case with the sixth
embodiment. However, since the stereo signals being supplied
to the circuit may already be spatially enhanced (the signals
can be spatially enhanced during production or during
broadcasting, for example), the seventh embodiment includes
the automatic control circuit 518 to limit the amount of
spatial energy added by the circuit of this seventh
embodiment.
Figure 16 is a block diagram and Figures 17A and
17B may be joined together to form a schematic diagram. The
seventh embodiment is quite similar to the sixth embodiment,
and, therefore, common reference numerals are used for common
elements. Indeed, the biggest change is the addition of the
aforementioned automatic control circuit 518 which controls
the amount of spatial enhancement which the circuit generates.
Another change is the provision of a stereo synthesis mode of
operation. If the music or other audio material occurring at
the inputs 501 and 502 already has a high degree of spatial
energy because the music or other audio material has
previously been processed in accordance with the present
invention before it was received by the circuit of Figures 16,
17A and 17B, then it is not desirable to add further spatial
enhancement in this circuit. However, if the inputs 501 and
502 receive stereo music or other audio stereo material which
has not been previously spatially enhanced, then the circuitry
of Figures 17A and 17B should add the desired spatial
enhancement. Thus, the control system 518 of the present
invention acts to control the amount of the spatial
enhancement added. If the incoming stereo music or sound is
already spatially enhanced, little or no additional spatial
enhancement is provided. If the incoming music or sounds have
not been previously spatially enhanced, then the control
system permits the spatial enhancement to occur. If the
incoming material is monaural, the stereo sounds may be
synthesized.
The control system 518 of the present embodiment is
W094/16538 2 ~ 5 3 0 6 2 PCT~S93/126~
42
conceptually similar to the automatic control circuitry 118
previously described with reference to Figures 7A, 7B and 8,
but it is nevertheless described here with respect to this
presently preferred consumer electronics embodiment of the
invention.
In the sixth embodiment, the amount of spatial
enhancement which occurs is controlled by using a
potentiometer 503, which controls the amplitude of the
conditioning signal, C, which appears on line 509. In this
seventh embodiment, instead of using a manual potentiometer
503, the magnitude of the conditioning signal C is controlled
by a voltage controlled amplifier 503' which is responsive to
a control input on line 510. The voltage controlled amplifier
503' is preferably a model 2151 device sold by That
Corporation or its equivalent. The conditioning signal is
output on the line 560, which output can be utilized to drive
an ambience or surround speaker, often located to the rear of
the listener.
The outputs on lines 511 and 512 are sampled and
added in circuitry 522 and subtracted in circuitry 520 to form
sum and difference signals on lines 530 and 528, respectively.
These sum and difference signals are applied to inputs of RMS
detectors 524 and 526. The RMS detectors are preferably model
2252 devices currently manufactured by That Corporation. The
output of the RMS detectors are applied to a comparator 550
whose output is coupled to a current source 551 (Figure 16).
The output of the current source is applied via a diode 552
to an amplifier 544. In Figures 17A and 17B, the current
source 551 and amplifier 550 are provided by a single device
which is called an operational transconductance amplifier 550,
551, which serves as a current source. When connected as
shown in Figures 17A and 17B, it can source up to 10 microamps
of current.
Potentiometer 534 whose wiper is connected to one
side of a resistor-capacitor network 553 allows the user to
control the amount of spatial enhancement which they desire
the circuit to produce. Resistor-capacitor network 553
controls the rate at which the spatial enhancement can be
changed by the output of current generator 551. The current
W094/16~8 PCT~S931~688
- 2~ 530 62 43
flowing through diode 552 alters the voltage across
resistor-capacitor network 553 which is fed to amplifier 554.
The automatic control circuit 518 functions to limit the
amount of spatial energy which the circuit can add to the
signals on the left and right channels, that is to say, it
does not allow the user to overly spatially enhance the music
material. The user can, however, use less spatial
enhancement, if they so desire, by adjusting potentiometer
534. The output from the resistor-capacitor network 553 is
applied via a switch 581 to a high impedance buffer amplifier
554 whose output goes to the control input of the voltage
controlled amplifier 503'. Resistor-capacitor network 553 in
combination with the current source 551, controls the
ballistics of the circuitry, that is, the number of decibels
per second change invoked through voltage controlled amplifier
503'.
In addition to adding spatial enhancement to stereo
material, the present invention can also be used to synthesize
stereo when the music or other sounds inputted at inputs 501
and 502 is monaural material. In order to synthesize stereo
from monaural information, a signal on line 580 causes
switches 581, 582 and 583 to change position from that shown
in the drawing. The input to buffer amplifier 554 is then a
bias voltage which is preferably provided by a voltage divider
network 584. In this stereo synthesis mode, differential
amplifier 504 has one of its inputs grounded via switch 582
and the other input continues to receive monaural information,
which is assumed to be applied to both inputs 501 and 502.
In the figures, the left input to differential amplifier 504
is shown as being grounded via switch 582, but it could be the
other input, if desired. Additionally, switch 583 adjusts the
gain of amplifier 502 in order to keep the output channels in
subjective balance in the stereo synthesis mode.
The various semiconductor devices shown in the
schematic diagram of Figures 17A and 17B have previously been
implemented as bipolar semiconductor devices. It is believed
that all of those devices, plus most of the resistors and
capacitors, can be implemented together on a large single
bipolar semiconductor chip. Of course, some of the components
W094/16~8 2 l5 3a6~ PCT~Sg31126
44
including potentiometers and relatively large capacitors are
best implemented off the chip and those elements are therefore
shown within dotted lines on Figure 16. Additionally,
assuming that those elements not enclosed within dotted lines
on Figure 16 are implemented on a single chip, then the single
chip would have pin-outs as indicated by the numerals in small
boxes, which numerals run between 1 and 18. Those skilled in
the art will appreciate, of course, that such a single chip
IC can be very conveniently packaged.
With respect to the parts -which would not be
implemented on a single chip, switch 530 mutes or turns off
the conditioning signal, so that no spatial enhancement is
added by circuitry 518 when switch 530 is opened. Switch 530
can be a manual switch as shown, or it can be an electrically
operated switch, such as a transistor. The signal on line 580
might be controlled by the stereo detection circuitry of a
conventional radio, for example, to change the positions of
switches 581, 582 and 583, when no stereo signal is present,
to cause stereo to be synthesized. When monaural information
is only available at inputs 501 and 502, this circuit should
desirably not try to spatially enhance the signal in the
manner as done with a stereo signal, i.e., without changing
the signal on line 580 since the circuitry enhances the
difference information which, in terms of monaural
information, is noise.
Capacitors 516 and 517 form high pass filter that
limit the action of the automatic control circuit 518 to those
frequencies above the extremely low bass, i.e., above 100 Hz.
- This is desirable because no significant spatial information
exists below 100 Hz.
As in the case of the embodiment of Figure 15, the
band pass filter 507 comprises relatively large capacitors
which are preferably implemented off the chip, given their
sizes. Band pass filter 507 is preferably centered at 500 Hz.
Capacitors 516 and 517, which couple outputs 511 and 512 to
the differencing 520 and adding 522 circuit arrangements, are
also preferably implemented off-chip given their size. Also,
RMS detectors 524 and 526 have some relatively large
capacitors associated with them which are similarly preferably
` W094/16538 215 3 0 ~ 2 PCT~S931~688
_ 45
implemented off-chip. The distortion trim pot 585 for VCA
503' is also implemented off-chip. However, a large number
of the components can be implemented on a single chip, and
therefore, this embodiment of the present invention can be
implemented very conveniently and inexpensively for consumer
grade stereo audio equipment.
Fi~hth Embodiment
The sixth embodiment of the invention, which was
described with reference to Figure 15, is monaural compatible
in that if a monaural signal (i.e. R=L) is applied at its
inputs 501 and 502, no artifacts of the present sound imaging
enhancement method will be perceived by the listener. That
is to say, when a stereo signal input is supplied at the
inputs S01, 502, image enhancement occurs, but if a monaural
signal is applied at those inputs, no image enhancement
occurs. However, the embodiment of Figure 15 (or of Figures
16, 17a and 17b) can be used to enhance monaural information
if it is (or they are) modified as shown in Figure 18.
In Figure 18, it is assumed that the circuitry of
Figure 15 (or of Figures 16, 17a and 17b) has been implemented
as a chip 802, with the exception of the potentiometer 503
shown in Figure 15 (potentiometer 534 of Figures 16 or 17a).
Thus, inputs 822 and 824 correspond to the inputs +L and +R
which appear at inputs 501 and 502. Two potentiometers 804
and 806 are coupled between inputs 814, 816 and inputs 822,
824. These potentiometers are preferably wired and gained for
counter operation, i.e., the resistance of one increases as
the other decreases with movement of the gained wip~rs.
Potentiometer 503 appears as potentiometer 812 in Figure 18.
Additional potentiometers 808 and 810 have been connected at
the outputs 812 and 828 (which correspond to outputs 511 and
512 in the environment of Figure 15). These potentiometers
are also gained for counter operation as explained above with
reference to potentiometers 804 and 806.
The action of the manipulation apparatus and method
depends upon the existence of a difference between the input
signals applied at inputs 822 and 824 (which correspond to +L
and +R on Figure 15). The difference may be spectral and/or
W094116538 3 0 6 2 46 PCT~S931~688
temporal. By spectral differences, it is meant that the
distribution of energy over the sound spectrum differs between
the left and right signals. By temporal differences, it is
meant that--the synchronization of the two input signals may
be offset in time with respect to the period of each signal.
It is this combination of spectral and temporal
differences, resulting from the nature of live stereophonic
recording and multi-track stereophonic recording, that cause
the generation of the conditioning signal, C, which has been
previously described. The conditioning signaL appears, for
example, on line 509 of the embodiment of Figure 15. The
conditioning signal can cause the sound emanating from two
loudspeakers, located to the left and right front of the
listener and coupled via amplifiers to outputs 818 and 820,
to actually appear, to the listener, to originate from behind
the listener, or at other locations in the listener's audio
space, depending upon the content of the conditioning signal,
C. A monophonic signal, when applied to the left and right
channels, has equal spectral and temporal quality, even under
broad-band conditions. Thus, in that event, both inputs 822
and 824 receive the same signal. In that circumstance, and
independently of the position of potentiometer 812, the
enhancement effect of the described manipulation system is at
its minimum and approaches zero, provided that the resistance
added by potentiometers 804 and 806 is the same in both the
left and right inputs.
Differences Due to Broad Spectral Imbalance
However, by changing the resistance values of
potentiometers 804 and 806, so that the resistances are no
longer equal, then the signals appearing at inputs 822 and 824
are no longer the same signal. In effect, a broad-band
spectral difference is realized at those two inputs. This
difference will generate the conditioning signal, C, at line
509 inside device 802. It should be noted, however, that the
conditioning signal is presented at outputs 826 and 828 at the
anti-phasing position, independent of the difference in broad-
band spectral content of inputs 822 and 824 by virtue of the
~ W094/16538 2 1 S 3 0 6 2 PCT~S93/12688
_ 47
relative position of controls 804 and 806. Inside the device
802, the signals at inputs 822 and 824 are also routed
directly to outputs 826 and 828, respectively. Thus, a broad-
band spectral difference is realized by changing the value of
potentiometer 804 compared to 806. This results not only in
a conditioning signal, C, being presented at outputs 826 and
828 with equal intensity and opposite polarity, but also
produces a difference in signal intensity at outputs 826 and
828 proportional to the difference represented at inputs 822
and 824 by virtue of the change in the resistances of
potentiometers 804 and 806. Therefore, when potentiometer 804
is counter-rotated compared potentiometer 806, for example,
several things occur simultaneously.
Assume a monophonic signal is equally applied to
inputs 814 and 816. Further assume that potentiometers 804
and 806 are adjusted so as to provide an equal signal to input
822 and input 824. Under such conditions the listener will
hear a phantom image midway and forward between the two
loudspeakers. This is the usual monophonic effect.
Assume a monophonic signal is equally applied to
inputs 814 and 816. Further assume potentiometers 804 and 806
are adjusted so as to provide a larger signal at input 822 and
a smaller signal at input 824. In that event the listener
will hear (i) a louder signal from the left loudspeaker and
a softer signal from the right loudspeaker by virtue of that
part of the circuit that directly routes input signals to the
outputs, and the listener will simultaneously hear (i) a sound
at the antiphasic position produced by the left and right
loudspeakers by virtue of that part of the circuit that routes
half of the conditioning signal, C, inverted, to the right
loudspeaker. Since this is a synchronistic situation, the
superposition principle, the precedence effect and temporal
fusion all act together so as to blend these two distinctly
produced signals into one homogeneous signal which upon being
transduced into sound over loudspeakers will, within the
listening experience, cause the listener to experience a
virtual image to "appear" beyond and to the outside of the
left loudspeaker's physical location.
The extent to which potentiometers 804 and 806 are
WO94/16~8 PCT~S93/~8
21S3062 48
counter rotated will move the virtual image, described above,
to image along an arc or semicircle extending from the
physical location of the left loudspeaker to the back of the
listener's head.
If potentiometers 804 and 806 of 800 are counter
rotated in the opposite direction so that the right signal is
greater than the left signal at the inputs 822, 824, an
inverse situation from that described above will occur. Thus,
the circuit of Figure 18 can be said to be symmetrical in its
broad-band spectral imbalanced imaging abilities.
To further illustrate the imaging characteristi-cs
of circuit of Figure 18, assume a monophonic (i.e., monaural)
signal is supplied to inputs 814 and 816 at equal intensity.
Further assume potentiometer 804 is adjusted to be fully OFF
and potentiometer 806 is adjusted to be fully ON. Under such
circumstance, the signal at input 822 will be routed directly
to output 826 and reproduced over the left loudspeaker.
Simultaneously a conditioning signal, C, will be generated by
the circuitry and will be added to the left output signal 826
and inverted at the right output 828. Since no right signal
is present at input 824 no output signal will be directly
routed through to it. Only the inverted conditioning signal
will be present at output 828. Thus the left loudspeaker will
reproduce the left input signal and half of the conditioning
signal while the right loudspeaker will reproduce the other
half of the conditioning signal, which, by definition, is
inverted relative to the left signal. Under this condition,
and depending upon the intensity of the conditioning signal,
- as adjusted by B12, the listener will hear the sound imaged
at a point approximately 140 degrees from a center point
(straight forward) of zero degrees.
Differences Due To Selective Spectral Imbalance
Assume a monophonic signal of a broad-band white
noise is split with equal signals being applied to the inputs
of an external equalizer device, and the output of each
equalizer is connected to inputs 814 and 816. Under this
case, it is possible to selectively adjust each equalizer so
as to send peaked portions of the sound spectrum to inputs 814
~094/16~8 21 S3 0 62 PCT~S93l~6~
49
or 816. This arrangement can be said to provide a spectral
difference.
For purposes of explanation, further assume that the
equalizer connected to input 814 is peaked at 1000 Hz and that
the equalizer connected to input 816 is flat or non-peaked.
With the conditioning signal control potentiometer 812, set
at a normal level, and loudspeakers connected to the outputs
818 and 820 via amplifiers, a listener positioned between the
two loudspeakers will hear all broad-band frequencies at a
mid-point between the two loudspeakers and frequencies in the
1000 Hz band image beyond the left loudspeaker at
approximately 100 degrees left from center.
In explanation, input 814 receives the peaked signal
and input 816 the non-peaked input, both originating as broad-
band noise. Since the difference between the two inputsoccurs only in the 1000 Hz range, the side-chain of the
circuit (which side chain generates the conditioning signal,
C), controlled by potentiometer 812, will contain only this
narrow band of frequencies clustered about the 1000 Hz band.
Output 818 will contain the peaked signal as it is passed
through the circuit by its internal circuitry. Output 820
will contain both the inverted signal from the peaked
equalizer and the non-inverted signal directly from input 816
as routed through the circuit. Note that the peaked frequency
band is made to produce a signal at outputs 818 and 820 of
equal intensity but opposite polarity. Since the intensity
is equal, the image produced by such an arrangement is always
at the antiphasic position. The fact that the experience of
the listener is to hear the peaked frequency band to the left
of the left loudspeaker and not to the back of the head can
be best understood through a study of the phenomenon of
Temporal Fusion.
If the two equalizers used in the above example are
connected in a opposite manner so that the right signal is
peaked and the left signal is flat, the outputs of the
circuits will display an inverse situation. The listener will
experience an equal but opposite listening experience with the
peaked frequency band imaging to beyond the right loudspeaker.
Thus the circuit 800 can be said to be symmetrical with
w094l16538 2 1 5 3 0 6 2 PCT~S93/12688
respect to its differences due to selective spectral
imbalance.
All spectral differences between inputs 814 and 816
behave in similar fashion to the above explanation. Multiple
narrow-band differences, simultaneously formed on either side
of the stereo pair, are amalgamated into a coherent sound
field through the action of the human brain and the excitation
of Temporal Fusion.
Differences Due to Broad-Band Temporal Imbalance
By introducing a broad-band time delay difference
between inputs 822 and 824 a temporal imbalance then exists.
This imbalance will generate the conditioning signal, C, at
812, but notice that the conditioning signal is presented at
outputs 826 and 828 at the antiphasic position independent of
the temporal displacement at inputs 822 and 824.
Further, the signals at inputs 822 and 824 are also
routed directly to outputs 826 and 828, respectively, as shown
in Figure 15. Thus a broad-band temporal difference realized
by delaying signals presented at input 822 and not at input
824 results not only in a conditioning signal, C, at 812 being
presented at outputs 826 and 828 with equal intensity and
opposite polarity, but it also realizes a difference in time
of the signal at outputs 826 and 828 proportional to the time
difference presented at inputs 822 and 824.
Assume a monophonic signal is equally applied to
inputs 814 and 816. Further assume that potentiometers 804
and 806 are adjusted so as to provide an equal signal to input
822 and input 824. ~nder such conditions the listener will
hear a phantom image midway and forward between the two
loudspeakers.
Assume a monophonic signal is equally applied to
inputs 814 and 816 but that the signal applied to input 814
is delayed with respect to the signal applied to input 816.
Further assume potentiometers 804 and 806 are equally
adjusted. Under this condition the listener will hear (i) a
louder signal from the right loudspeaker and a softer signal
from the left loudspeaker by virtue of that part of the
circuit that directly routes input signals to the output, and
wo 94/16~8 2 1 5 3 0 6 2 PCT~S93/12688
51
the listener will simultaneously hear (ii) a sound at the
antiphasic position produced by the left and right
loudspeakers by virtue of that part of the circuit that routes
half of the conditioning signal, C, inverted, to the right
loudspeaker. Since this is a synchronistic situation, the
superposition principle, temporal fusion, and especially the
precedence effect all act together so as to blend these two
distinctly produced signals into one homogeneous signal which
upon being transduced into sound over loudspeakers will within
the listening experience, cause the listener to experience a
virtual image to "appear" beyond and to the outside of the
left loudspeaker's physical location.
The extent to which an adjustable delay is
discontinuous (up to about 50 milliseconds) will move the
virtual image, described above, to image along an arc or
semicircle extending from the physical location of the left
loudspeaker to the back of the listener's head.
If input 816 is delayed with respect to input 814,
an inverse situation from that described above will occur.
Thus the circuit of Figure 18 can be said to be symmetrical
in its broad-band temporal imbalanced imaging abilities.
Differences Due to Selective Temporal Imbalance
Assume a monophonic signal of broad-band white noise
is split with equal signals applied to the inputs of two
external equalizer devices, and the output of each equalizer
connected to a delay device, and the output of each delay
device being connected to the inputs of a second set of
equalizers, the outputs of the second set of equalizers being
connected to inputs 814 and 816. In this arrangement each
equalizer may be adjusted so as to send peaked portions of the
sound spectrum to be delayed, and after delay mixed back into
the original broad-band white noise signal through the second
equalizer set with a dip at 1000 Hz which is in opposition
with the first equalizer's peak but adjusted so as to produce
an equal intensity across the sound spectrum except with a
portion of the spectrum at 1000 Hz delayed with respect to the
other portions of the spectrum. This arrangement can be said
to provide a selective temporal imbalance.
WO94/16~8 2 15 3 0 6 2 PCT~S93/~6~
52
For purposes of explanation let us further assume
thatthe equalizer-delay-equalizer arrangementdescribed above
is connected to inputs 814 and 816. Let us further assume
that equalizer-delay-equalizer connected to input 814 is
peaked at 1000 Hz and that the equalizer-delay-equalizer
connected to input 816 is flat or non-peaked. With the
conditioning signal control potentiometer 812 set at a normal
level, and loudspeakers connected to the outputs 818 and 820
(via amplifiers), a listener positioned between the two
loudspeakers will hear all broad-band frequencies in the 1000
Hz band image beyond the right loudspeaker at approximately
100 degrees right from center.
In explanation, input 814 receives broad-band white
noise with the 1000 Hz band delayed signal and input 816 the
non-delayed input of broad-band white noise. Since the time-
difference between the two inputs occurs only in the 1000 Hz
range, the side-chain of 802, controlled by potentiometer 812,
will contain only this narrow band of frequencies clustered
about the 1000 Hz band. Output 818 will contain the delayed
signal as it is passed through the circuit. Output 820 will
contain both the inverted signal from the delayed equalizer
and the non-inverted signal from input 816 as routed through
the circuit. Note that the delayed frequency band is made to
produce a signal at outputs 818 and 820 of equal intensity but
of opposite polarity. Since the intensity is equal, the image
produced by such an arrangement is always at the antiphasic
position. The fact that the experience of the listener is to
hear the delayed frequency band to the right of the right
loudspeaker and not to the back of the head can be best
understood through a study of the estimable phenomenon of
Temporal Fusion.
If the two equalizer-delay-equalizer arrangements
used in the above example are connected in an opposite manner
so that the right signal is peak-delayed and the left signal
is flat, the outputs will display an inverse situation. The
listener will experience an equal but opposite listening
experience with the peaked and delayed frequency band imaging
to beyond the left loudspeaker. Thus the circuit can be said
to be symmetrical with respect to its differences due to
W094ll6~8 2 1 5 3 ~ 6 2 PCT~S93/~688
53
selective temporal imbalance.
All temporal differences of 50 millisecond or less
between inputs 814 and 816 behave in similar fashion to the
above explanation. Multiple narrow-band temporal differences,
simultaneously formed on either side of the stereo pair, are
amalgamated into a coherent sound field through the action of
the human brain and the excitation of Temporal Fusion.
Ninth Embodiment
Multiple Inputs in Professional Recording
The manipulation system and apparatus is not limited
to one panoramic potentiometer (panpot) control as shown in
Figure 18 at numbers 804 and 806. A useful variation of the
Eighth Embodiment is shown in Figure 19, and it comprises a
ninth embodiment of the invention. This embodiment involves
the use of multiple inputs. Buses 830 and 832 are extensions
from inputs 822 and 824 of the circuit of Figures 15 and 18
and said buses accommodate a plurality of panpots. In Figure
19, inputs 814A, 814B, 814C, 814D and 814E are all connected
to the left side of the respective panpots shown which in turn
are connected to bus 830. Inputs 816A, 816B, 816C, 816D and
816E are all connected to the right side of respective panpots
shown which in turn are connected to bus 832.
Signals inputted to any inputs 'A' through 'E' will
exhibit an influence on the manipulation system and apparatus
in a manner identical to signals inputted to inputs 814 and
816 as previously disclosed. Thus, connecting the
manipulation system and apparatus to a modern recording
console in such a way so as to direct the signals form the
console's combined panning buses to the inputs 814 and 816
will, in effect, extend the use of the circuit of Figures 15
and 18 to every panpot on the console. In such an arrangement
the output of the circuit can be redirected back into the
console's recording buses or to two isolated inputs that are
in turn recombined with other isolated inputs to be combined
into the line output of the console for purposes of recording
on an analog or digital, magnetic or optical recorder.
Monophonic Applications in Professional Recording
W094/16538 215 3 0 6 ~ PCT~S93/~688
54
In professional recording situations, the engineer
has control over the selective and broad-band spectral and/or
temporal content of the various instrumental elements
comprising a musical production. By varying the selective or
broad-band spectral and/or temporal content of specific
elements, in the manner descried above using the manipulation
system and apparatus presented, the engineer can exhibit a
keen degree of control over the image position of those
elements; an image position which includes as its field of
control that portion of the sound field which extends beyond
the physical location of the two loudspeakers and to a point
of at least 140 degrees from center in both directions.
Furthermore the engineer can move any element in seamless
progression along an arc extending from mid-point between the
two loudspeakers to a point of at least 140 degrees from the
center of the two loudspeakers to either the left or the right
of the two stereo loudspeakers.
Any number of monophonic signals may be
simultaneously inputted into any number of inputs under the
above arrangement, and each signal will be treated by the
manipulation system and apparatus as if independent, that is
to say, the manipulation of one signal will not influence the
treatment of any other inputted signals.
Stereophonic Applications in Professional Recording
Within the above arrangement but not limited to it;
if a stereo signal is inputted into any two inputs of a
console such as 814A and 816C and the panner of 814A is
rotated to the left (counter clock-wise) and the panner of
816C is rotated to the right (clock-wise), the action of the
circuit of Figures 15 and 18 will be the same as if the stereo
signal had been inputted to inputs 822 and 824 directly, as
described in Embodiment Eight (or Embodiment Six for that
matter).
Any number of stereo pairs may be simultaneously
inputted into any two inputs under the above arrangement, and
each stereo signal will be treated by the manipulation system
and apparatus as if independent.
WO94/16538 PCT~S93/12688
- 21 S3 062 55
Stereo and Mono Applications in Professional Recording
Further, any number of stereo pairs and mono inputs
may be simultaneously inputted into any combination of inputs
under the above arrangement, and each input, be it mono or
stereo, will be treated by the manipulation system and
apparatus as if independent.
By connecting the manipulation system and apparatus
to a modern recording console and operating this invention in
conjunction with the operation of a modern recording console,
it can be said that the production advantages afforded by this
invention can be retro-fitted to any console of common design
through the use and application of these circuits.
This invention has been described with reference to
a number of embodiments, and it is clear that it is
susceptible to numerous modifications, modes and embodiments
within the ability of those skilled in the art and without the
exercise of the inventive faculty. By way of example, the
conditioning signal in all embodiments is shown as being added
to the left channel and subtracted from the right channel.
This convention may be reversed, if desired, although it is
believed that the electronics industry will follow the
convention described herein for consistency. Other
modifications are well within the skill of those skilled in
this art. Accordingly, this invention is not limited to the
disclosed embodiments, except as required by the appended
claims.
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