U.S. patent number 3,632,886 [Application Number 04/888,440] was granted by the patent office on 1972-01-04 for quadrasonic sound system.
Invention is credited to Peter Scheiber.
United States Patent |
3,632,886 |
Scheiber |
January 4, 1972 |
QUADRASONIC SOUND SYSTEM
Abstract
A stereophonic sound system is disclosed utilizing a two-channel
transmission path yet capable of locating virtual sound sources at
any point on a circle around a listener. The two-channel
transmission path may consist of conventional stereophonic channels
such as records, tapes, broadcasting channels, etc. The recording
or transmitting means (as the case may be) of the invention
provides two audio signals which may comprise preselected
combinations of four (for example) directional inputs. One channel
may include the first input plus a signal proportional to the sum
of the second and fourth inputs, while the second channel consists
of the third input plus a signal proportional to the difference
between the second and fourth inputs. The sound reproducing means
couples these signals and various combinations thereof to four (for
example) loudspeakers which may be arranged on the circumference of
a circle around the listener. The first speaker may be responsive
to the signal on one channel, the next adjacent speaker is
responsive to the sum of the signals on the two channels, the third
successive speaker is responsive to the second channel, and the
last speaker is responsive to the difference between the signals on
the two channels. Means are disclosed for controlling the gain in
the signal paths of the voltages coupled to the various speakers
relative to the other speakers to increase the audio separation
between adjacent speakers and thus enhance the directional
effect.
Inventors: |
Scheiber; Peter (Peekskill,
NY) |
Family
ID: |
25393177 |
Appl.
No.: |
04/888,440 |
Filed: |
December 29, 1969 |
Related U.S. Patent Documents
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Application
Number |
Filing Date |
Patent Number |
Issue Date |
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853822 |
Aug 28, 1969 |
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967103 |
Jan 11, 1968 |
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Current U.S.
Class: |
381/22; 381/23;
369/89 |
Current CPC
Class: |
H04S
3/02 (20130101) |
Current International
Class: |
H04S
3/00 (20060101); H04S 3/02 (20060101); H04h
005/00 () |
Field of
Search: |
;179/1G,1GP,1GA,1.4ST,1.1TD,15ST,15MM,15BC ;325/36
;350/118,119 |
References Cited
[Referenced By]
U.S. Patent Documents
Other References
"Three Channel Stereo Playback of Two Tracks Derived From Three
Microphones" P. W. Klipsch IRE Transactions March-April 1959, p.
34-36.
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Primary Examiner: Claffy; Kathleen
Assistant Examiner: D'Amico; Tom
Parent Case Text
The present invention relates to audio systems and, in particular,
to a stereophonic sound system which is capable of locating virtual
sound sources at any point on a full circle around a listener.
Claims
1. For use with a stereophonic sound system wherein a receiver
means couples audio data to at least four separate speakers and
wherein audio signals are transmitted to said receiver means on
only two channels A and B, a recording or transmitting network,
comprising
means for coupling a first signal desired to be coupled to a first
speaker to at least one of said channels with a preselected
amplitude and polarity,
means for coupling a second signal desired to be coupled to a
second speaker to at least the other of said channels with a
preselected amplitude and polarity,
means for coupling a third signal desired to be coupled to a third
speaker to both A and B channels with the same polarity in both
channels and preselected amplitudes in said channels, and
means for coupling a fourth signal desired to be coupled to a
fourth speaker to both A and B channels with opposite polarities in
the two
2. A recording or transmitting network according to claim 1,
further including means for reducing the amplitude of said third
and fourth signals relative to said first and second signals,
respectively, in said A
3. A recording or transmitting network according to claim 2,
wherein said amplitude reducing means reduces the amplitude of said
third and fourth
4. A recording or transmitting network according to claim 3,
wherein said amplitude reducing means reduces the amplitude of said
third and fourth
5. A recording or transmitting network according to claim 1,
including means for superimposing a control tone on at least one of
said channels, said control signal being dependent upon the
relative signal strengths of
6. A recording or transmitting network according to claim 5,
including means for superimposing a control tone on each of said A
and B channels, each of said control tones being dependent upon the
ratio of the total signal power in two of the channels to the total
power in all four
7. For use with a stereophonic sound system wherein a receiver
means couples audio data to at least four separate speakers and
wherein audio signals are transmitted to said receiver means on two
channels A and B, a recording or transmitting network,
comprising:
at least four acoustical-electrical transducer means for producing,
respectively, at least four audio signals to be transmitted,
means for coupling a first signal desired to be coupled to a first
speaker to at least one of said channels with a preselected
amplitude and polarity,
means for coupling a second signal desired to be coupled to a
second speaker to at least the other of said channels with a
preselected amplitude and polarity,
means for coupling a third signal desired to be coupled to a third
speaker to both A and B channels with the same polarity in both
channels and preselected amplitudes in said channels, and
means for coupling a fourth signal desired to be coupled to a
fourth speaker to both A and B channels with opposite polarities in
the two
8. For use in a sound reproducing system capable of locating
virtual sound sources at essentially any point on a circle
surrounding a listener position, wherein audio information
corresponding to four separate directional inputs is transmitted on
a two-channel transmission path and wherein there are provided at
least four loudspeakers adapted to be arranged around a listener,
the combination comprising:
means for coupling the signal on at least one of said channels to a
first of said speakers,
means for coupling the signals on said first and second channels to
a second speaker with the same polarity and a preselected amplitude
relationship,
means for coupling the signal on at least the second of said
channels to a third speaker, and
means for coupling the signals on said two channels to a fourth
speaker
9. The combination according to claim 8, wherein said first and
third speakers define a diagonal intersecting a diagonal defined by
said second
10. The combination according to claim 8, wherein the signals fed
to said second and fourth speakers are reduced in amplitude
relative to the
11. For use in a sound reproducing system capable of locating
virtual sound sources at essentially any point on a circle
surrounding a listener position, wherein audio information
corresponding to four separate directional inputs is transmitted on
a two-channel transmission path and wherein there are provided at
least four loudspeakers adapted to be arranged around a listener,
the combination comprising:
means for coupling a signal on at least one of said channels to a
first of said speakers,
means for coupling the signal on at least the second of said
channels to a third speaker,
means for coupling the signals on said first and second channels to
a second speaker with the same polarity and with their amplitudes
reduced by a factor of approximately 0.707 relative to the
amplitude of the signals coupled to said first and third speakers,
and
means for coupling the signals on said two channels to a fourth
speaker with opposite polarities and with their amplitudes reduced
by a factor of approximately 0.707 relative to the amplitude of the
signals coupled to
12. The combination according to claim 10, wherein each of said
coupling means includes variable gain amplifier means, said
combination further including gain control means for varying the
gain of said variable gain amplifier means so as to increase
separation between adjacent speakers.
13. The combination according to claim 12, wherein the gains of the
variable gain amplifier means associated with the first and third
speakers are maintained equal and the gains of the variable gain
amplifier means
14. The combination according to claim 12, wherein said gain
control means includes means for computing the instantaneous value
of the ratio of the signal amplitude in one of said channels to the
signal amplitude in the
15. The combination according to claim 12, wherein said gain
control means comprises means for comparing the envelopes of the
signals in said
16. The combination according to claim 13, wherein the gain of said
variable gain amplifiers associated with said first and third
speakers is minimum and the gain associated with said second and
fourth speakers is maximum when said ratio is equal to one, and
wherein the gain of the variable gain amplifiers associated with
said first and third speakers is a maximum and the gain of the
amplifiers associated with said second and fourth speakers a
minimum when there is no signal in either one of said
17. The combination according to claim 16, wherein said gain
control means varies the gain of said variable gain amplifiers so
as to maintain the total power coupled to said speakers at a
substantially constant level.
18. The combination according to claim 12, wherein said gain
control means includes means responsive to a modulated tone on at
least one of said
19. The combination according to claim 12, wherein said gain
control means includes means responsive to modulated control tones
on each of said
20. The combination according to claim 12, including further
variable gain amplifier means and gain control means, said further
gain control means adapted to control the gain of said further
variable gain amplifiers and responsive to voltages dependent upon
(a) the power differential between the inputs and outputs of the
first-named variable gain amplifiers associated with the first and
third speakers and (b) the power differential between the inputs
and outputs of the first named variable
21. A stereophonic sound system wherein audio signals are
transmitted on only two channels A and B, in combination, an
encoder and a decoder,
said encoder comprising
means for coupling a first signal desired to be coupled to a first
speaker to at least the A channel,
means for coupling a second signal desired to be coupled to a
second speaker to at least the B channel,
means for coupling a third signal desired to be coupled to a third
speaker to both A and B channels with the same polarity in both
channels, and
means for coupling a fourth signal desired to be coupled to a
fourth speaker to both A and B channels with opposite polarity in
both channels,
said decoder comprising
means for coupling the signal on one of said channels to a first of
said speakers,
means for coupling the signals on said first and second channels to
a second speaker with the same polarity,
means for coupling the signal on the second of said channels to a
third speaker, and
means for coupling the signals on said two channels to a fourth
speaker
22. A method of recording or transmitting at least four directional
audio information signals on two channels comprising
forming at least two composite signals each of which includes at
least three of said audio information signals in preselected
amplitude and phase relationships, at least one of said audio
information signals having a different phase in each of said
composite signals, said amplitude and phase relationships being
such that four combinations of said two composite signals with
preselected phase and amplitude relationships will yield four
separate signals in which respective ones of said four
23. A method of recording or transmitting at least four directional
audio information signals according to claim 22, wherein at least
one of said audio information signals has the same relative phase
in each of said composite signals and at least one of said audio
information signals has
24. A method of recording or transmitting at least four directional
audio information signals according to claim 22, wherein at least
some of said
25. A method of substantially reproducing at least four individual
directional audio information signals which are contained in two
information channels, wherein the signal in each of said channels
comprises a combination of at least three of said directional
signals with preselected phase and amplitude relationships,
comprising
deriving said four output signals from the signals in said two
channels, at least one pair of said output signals being derived by
combining the signals in said channels with preselected phase and
amplitude relationships wherein different phase relationships are
used to derive each signal of said pair of signals, the remaining
output signals, if any, being derived from the signals in one of
said channels, whereby a different desired one of said four
directional signals is predominant in
26. A method of substantially reproducing at least four individual
directional audio information signals according to claim 25,
further including the step of controlling the gain of at least some
of said four output signals depending on a preselected comparison
of the signals in
27. A method of substantially reproducing at least four individual
directional audio information signals according to claim 25,
wherein at least one of said output signals is derived by combining
the signals in said two channels with a zero phase shift and at
least one other of said four output signals is derived by combining
said signals in said two channels with one of said signals in said
two channels being shifted in phase by 180.degree..
Description
PRIOR ART
Commercial stereophonic systems usually include two separate sound
channels. This helps to recreate actual listening conditions by
effectively increasing the region from which sound sources can
emanate. Where two loudspeakers are used, a virtual sound source
may be located at either of the two speakers or any point in
between. In this respect, a two-channel or binaural system is an
improvement over a monaural system which utilizes a single speaker
and is therefore only capable of locating a sound source at that
speaker.
It has been proposed to use a third speaker positioned between a
pair of outer speakers with the third speaker being fed by a simple
additive combination of the signals on the two stereo channels. To
further improve the audio effect, it is known to record a third
channel of audio information on the two stereophonic channels with
equal amplitude and same polarity so that when the signals in the
two stereo channels are added at the reproducing end, a third
channel signal is formed which can locate a source directly at the
third speaker. However, even this third channel cannot locate a
sound source behind the listener (assuming proper placement of the
third channel speaker) because the relatively equal signals
appearing in the two existing channels produce a sound "image"
between them resulting in a conflict as to the direction of the
source.
The logical extension of a binaural system would include third and
fourth channels with speakers positioned in front of and behind the
listener and such systems (known as quadrasonic systems) are
already commercially available in (at least) four channel tape
systems. Where a four-channel system is used, it is possible to
locate a source of sound at any point on a full circle around the
listener. This ability to locate sound sources behind a listener
has a significant audio effect and compares with binaural systems
in much the same way a binaural system compares with a monaural
system.
Where a full four-channel system is used, there are no special
problems involved in feeding audio signals to the respective
speakers to create the desired effect. For example, four
microphones may be used at the pickup end, corresponding to four
different directions, with the output of each microphone coupled by
a separate transmission path (e.g., a tape channel) to suitably
placed speakers on the circumference of a circle and corresponding
to the four microphones. However, as indicated previously,
virtually all stereophonic sound systems presently in use include
only two signal transmission paths. For example, conventional
stereophonic records include grooves wherein the surfaces are
inclined at 45 degrees with respect to the record face, each
surface serving as the equivalent of a separate "transmission
path." FM multiplex broadcasting techniques approved by the U.S.
Federal Communications Commission and currently in widespread use,
provide for the transmission of only two separate signals and,
obviously, most multiplex receivers presently used are capable of
receiving only these two separate channels. Most tape recording
equipment in use today records and reproduces only two channels and
could not readily be modified to record and/or play back four
separate audio signals, as presently required for full-circle
sound.
BRIEF DESCRIPTION OF THE INVENTION
The present invention provides a stereophonic system which is
capable of locating a sound source at any place on a full circle
around a listener wherein the information is transmitted (or
recorded) on only two separate transmission paths. The invention
may be used with three speakers, but, preferably, four separate
speakers are used.
In accordance with the invention, a signal is transmitted on one or
both channels with a selected relative polarity or phase difference
(and, preferably, gain also) depending upon the desired direction
of the source represented by that signal. For example, the system
may employ four separate microphones each of which is adapted to
respond to sound sources from a preselected quadrant of the circle
around the microphones. The signals from two oppositely directed
microphones are connected directly to the respective stereo
channels. Additive and subtractive combinations of the other two
microphones are also coupled to these two channels so that when the
signals are replayed (or received) with suitable combining, the
input to any microphone will be effectively "located" or replayed
at a corresponding speaker. The invention is fully compatible with
existing binaural systems, and produces two outputs (at the
recording or transmitting end) which can be replayed over existing
binaural equipment.
At the receiver or playback end, the four speakers are fed
successively with (1) the signal on the first channel, (2) the sum
of the signals on the two channels, (3) the signal on the second
channel, and (4) the difference of the signals on the two channels.
As explained in detail in the following specification, this permits
location of a virtual sound source at any point on a full circle
around the listener.
The invention also contemplates the use of special gain control
techniques both of a basic and more sophisticated form for the
purpose of increasing the signal separation between any speaker and
the two adjacent speakers. Such techniques may serve to increase
the directional effect and thus enhance the audio effect.
In the following specification and claims, the term "transmission
path" is used to include both recording and actual transmission
(e.g., FM multiplex) inasmuch as the invention is not dependent
upon the actual physical or electrical nature of the two separate
signals. Reference to locating sound on a circle around a listener
means the ability to locate virtual sound sources in front of,
behind, or to the sides of a listener but is not intended to limit
or define the precise placement of the speakers. The benefits of
the invention can best be provided where at least one of the
speakers is located behind the listener and the speakers are
arranged in fact on the circumference of a circle, with the
listener located at the center. Different arrangements of the
speakers may vary the directional effect achieved by this invention
yet remain within its scope.
The invention is described in detail with reference to the attached
drawings, wherein:
FIG. 1 is a block diagram of a system incorporating the principles
of the invention;
FIGS. 2A, B, C and D are explanatory diagrams used to illustrate
the operation of the system of FIG. 1;
FIG. 3 is a circuit diagram of a preferred embodiment of an encoder
used with the invention;
FIG. 4 is a circuit diagram of a preferred embodiment of a decoder
used with the invention;
FIG. 5 is a block diagram of a gain control circuit intended for
use with the system of FIG. 1;
FIG. 6 is a block diagram of a second gain control circuit intended
for use with the system of FIG. 1;
FIGS. 7 and 8 comprise block diagrams of the transmitting and
receiving ends of a further gain control circuit;
FIG. 9 is a block diagram of still a further embodiment of a gain
control circuit;
FIG. 10 is a block diagram of yet a further gain control embodiment
which can provide the desired results apart from the basic system
of FIG. 1;
FIG. 11 is a detailed block diagram of the control voltage
generator of FIG. 10;
FIG. 12 is a schematic diagram of the gain control circuit of FIG.
10; and
FIGS. 13 and 14 are schematic diagrams of the control voltage
generator of FIG. 11; and
FIG. 15 is a diagrammatic illustration of a conventional
45.degree..times.45.degree. record groove for explanatory
purposes.
The preferred embodiment of the invention is described below for
use with four separate speakers arranged on the circumference of a
circle around a listening position, so that they are capable of
locating a virtual sound source at any point around the listener.
For purposes of this description, two speakers will be considered
to be arranged directly in front of and behind the listener, with
the other two speakers arranged to the left and right. In
explaining the operation of the system, the letters F and X are
used with numerals to indicate parts of the system corresponding to
the front and rear speakers, respectively. The letters L and R are
used with numerals to represent parts corresponding to the left and
right speakers, respectively. The two transmission paths on which
the audio signals are conveyed are represented by the letters A and
B, it again being emphasized that such transmission paths may (and
frequently will) comprise recording media.
Referring now to FIG. 1, four microphones 12L, 12F, 12R and 12X are
diagrammatically shown as being arranged to receive acoustical
energy arriving from any direction around the microphones. For
purposes of explanation, each of the microphones 12 may be
considered as a directional device capable of receiving sound from
any direction within a 90.degree. arc so that four microphones can
be arranged to cover a full circle. The receptivity pattern of each
of the microphones 12 may be represented by the dashed lines
13F,R,X,L. In practice, of course, the receptivity patterns of the
respective microphones may overlap each other (which is required to
locate sound sources between adjacent speakers) or include regions
of relatively weak reception, but consideration of this simplified
situation is helpful in appreciating the principles by which the
invention can effectively transmit four audio signals on only two
transmission paths.
In FIG. 1, the four loudspeakers are shown at 14F, 14R, 14X and
14L. These loudspeakers are shown around a listener's position
(represented by a chair 16). If the electrical outputs of the
microphones 12F, 12R, 12X and 12L were conveyed (by any
conventional technique) on four channels directly to the respective
speakers 14F, 14R, 14X and 14L, the speakers would reproduce the
directional audio information received by the microphones. In this
way, it would be possible to locate a virtual sound source in any
quadrant of a circle around the listening position 16. This ability
to locate sounds behind and to the respective sides of a listener
is the principal way in which a quadrasonic system differs from
conventional binaural stereophonic systems.
As mentioned above, because of practical considerations it is not
feasible in all cases to convey the audio signals directly from the
microphones 12 to the respective loudspeakers 14. Because of the
great number of binaural stereophonic systems presently in use, it
is highly desirable to convey these four distinct audio signals to
the four loudspeakers using only the two conventional binaural
channels or transmission paths indicated as A and B in FIG. 1.
For this purpose, and pursuant to the invention, an encoder 18 at
the recording or transmitting station combines the signals from the
four microphones into two separate audio signals which can be
conveyed by the separate transmission paths A and B. For example,
if a stereophonic record were being made, the encoder 18 would be
used to generate the two signals which would be recorded in the
standard 45.degree..times.45.degree. record groove. In the case of
a live broadcast or other direct transmission system, the encoder
18 would be used to generate the A and B signals which would
ultimately be broadcast or otherwise transmitted.
The encoder serves to feed each of the four input signals to one or
both of the two channels A and B with an amplitude and polarity
which determine the relative amplitude of that signal at the
various speakers and thus the effective location of the sound
source relative to the listener. In this sense, directional
information is encoded. In the preferred embodiment the audio
signal in transmission path A includes the sum of the output from
microphone 12L and the outputs from microphones 12F and 12X, with
the amplitude of the F and X signals being attenuated (decreased in
amplitude) by a preselected constant, for example, 0.707. The audio
signal applied to the transmission path B comprises the output from
microphone 12R plus a signal equal to the difference between the
front and rear signals, also preferably attenuated by the same
preselected constant, i.e., 0.707.
At the playback or receiving station, a decoder 20 receives the A
and B signals and couples these signals in preselected combinations
to the loudspeakers 14F, 14R, 14X and 14L. As indicated in FIG. 1,
the signal fed to loudspeaker 14L is the A signal alone. The sum of
the A and B signals is fed to the front speaker 14F after being
attenuated by a factor 0.707. The B signal alone is fed to the
right loudspeaker 14R, and a signal representative of the
difference of the audio signals on A and B is fed to the rear
speaker 14X after being attenuated by a factor of 0.707. As
explained below, this combination of the four signals (preferably
with the indicated gain control) permits location of a virtual
sound source at any of the four loudspeakers 14 (or any point on a
circle defined by the speakers).
The theory of operation of the system illustrated in FIG. 1 is now
explained with reference to the diagrams of FIGS. 2A, B, C and D.
In the following, the letters F, R, X and L alone are used to
represent the respective voltage outputs from microphone 12F, R, X
and L.
Consider the relative amplitudes of the respective microphone
outputs F, R, X and L on the transmission paths A and B to be as
indicated in FIG. 1. Assume a sound source directly in front of
microphone 12F so that audio signal F has a relative output of one,
while the L, X and R signals are zero. With the gain criteria
established in FIG. 1, the signal on transmission path A will be
equal to 0.707F. The signal on channel B also will be equal to
0.707F. Accordingly, the signal (0.707 [A+B]) coupled to speaker
14F will be equal to F; the signal coupled to speaker 14L(A) will
be equal to 0.707F; the signal coupled to speaker 14R(B) will be
equal to 0.707F; and the signal (0.707[A-B]) coupled to speaker 14X
will be equal to zero. This means that the audio power outputs of
the left and right speakers 14L and 14R (proportional to
[0.707F].sup.2) will be three decibels down from that produced by
the front speaker 14F (proportional to F.sup.2), while no sound
energy at all will be emitted from speaker 12X. As a result, a
listener located at position 16 will consider the apparent location
of the sound source to be the speaker 12F.
FIG. 2B is a similar diagram for the case where the sound source is
directly to the right so that only microphone 12R produces an
electrical output. In this case, the signal on transmission path B
is equal to R, while there is no signal on transmission path A.
Consequently, the signal at speaker 14R is equal to R, the signals
coupled to the adjacent front and rear speakers 12F and 12X have a
relative amplitude of 0.707R (so that the audio output is down 3
db.) and no sound is produced at speaker 14L.
When the sound source is in the rear so that only microphone 12X
picks up a signal, as illustrated schematically in FIG. 2C, the
signals in both paths A and B will be equal to 0.707X. However, the
polarity of the two signals will be opposite. Thus, the A signal
may be considered equal to 0.707X and the B signal equal to
-0.707X. As a result, the A+B signal fed to speaker 14F will be
equal to zero and the A-B signal at the rear speaker 12X
(attenuated by 0.707) will be equal to X. The signals fed to the
adjacent left and right speakers 14L and 14R will be 0.707X (i.e.,
audio output down 3 db.) and of opposite polarity and the virtual
sound source will in this instance appear to be the rear speaker
14X.
By the same logic, where the source is coupled only to the left
microphone 12L as illustrated in FIG. 2D, the sound source will be
located at the left loudspeaker 14L, with the sound from the
adjacent front and rear speakers 14F and 14X being down 3db. (and
of opposite polarity), and no signal at the right speaker 14R.
The principles, as explained with reference to FIGS. 1, and 2A, B,
C and D, are applicable regardless of where the actual sound source
is located. By a similar analysis, it can be shown that a sound
source can be located at any point on the circumference of a circle
around the listener by effectively creating a sound "null" at the
location opposite the desired source location. The fact that the
audio output of two speakers is reduced for sounds which should
appear to be coming from the speaker between those two speakers
also aids in creating this directional effect.
Obviously, the speakers can be positioned in any desired way (for
example, two front speakers and two rear speakers) to provide the
same or similar effects. Moreover, it is unnecessary that the
microphones be directional or that they be arranged to respond to
sounds coming from the full circumference of a circle around the
microphones. Indeed, it is expected that in many cases, the
microphones will be arranged with respect to a musical group so
that each will respond to a different instrument or section to
create an artificial effect during playback by causing the music to
"surround" the listener. These and other techniques are standard
with respect to four-channel stereophonic systems and are not a
part of this invention. The invention provides a way to recreate
virtual sound sources at any location with respect to a listener
wherein the audio signals are transferred on only two separate
transmission paths. As such, the invention may be used to simulate
actual listening conditions or to create artificial effects.
Any number of microphones can be used, including one, to achieve
the desired effect. For example, if the output of a single
microphone were coupled directly to channel A or channel B, an
audio output would appear at loudspeaker 14L or 14R, respectively.
If a signal of equal amplitude and polarity from a single
microphone were coupled to the A and B channels, the sound source
would be located at the loudspeaker 14F. If audio signals of equal
amplitude but opposite polarity were applied to the A and B
channels, the sound source would be located at the rear speaker
14X. Creating a continuous transition between these conditions
would give the effect of a moving sound source.
The principles of the invention may also be used in a three-speaker
system. Thus, consider two speakers 14L and 14R located in front
(left and right) of the listener and a third speaker 14X directly
behind the listener position. The signal on channel A may be equal
to L -0.5R +0.5X and the signal on channel B equal to R -0.5L
+0.5X. The A and B signals would be coupled directly to the left
and right speakers, respectively, with the sum of the A and B
signals being coupled to the rear speaker X so that partial
cancellation of the R and L signals would occur at the rear speaker
thus effectively locating the X channel audio source at the rear
speaker.
The foregoing principles may be applied to combine only two signals
(e.g., L +0.707X and R -0.707X) on each of the transmission paths A
and B so that when the amplitudes of the respective signals are
properly combined with suitable consideration for polarity (e.g.,
A, B and 0.707[A-B] ) a desired microphone output will predominate
at a corresponding speaker. In this case, the rear signal X will
appear in all three speakers with opposite polarity at the left and
right speakers 14L and 14R. This will not prevent the desired
directional effect since the combined outputs of two speakers fed
by out of phase voltages results in a virtual sound source without
a definitive location. Hence, in the case of an X signal alone, the
source can be located at the rear speaker 14X.
FIG. 3 illustrates the specific details of the encoding circuit.
The input signal from microphone 12L is applied through a first
resistor 50 to the negative input terminal of a first operational
amplifier 52. Similarly, the input signal from microphone 12R is
applied through a second resistor 54 to the negative input terminal
of a second operational amplifier 56. The input signal from
microphone 12F is applied equally to the negative input terminals
of both operational amplifiers 52 and 56 through resistors 66 and
68 respectively.
The input signal from the microphone 12X is applied through a
resistor 70 to the negative input terminal of operational amplifier
52 and through a resistor 72 to the positive input terminal of
operational amplifier 56. The effect of this arrangement is that
the input signal from microphone 12X is applied to coded channels A
and B with opposite polarity and equal amplitude.
Coded channel A is obtained through a small isolating resistor 58
at the output of operational amplifier 52 and coded channel B is
obtained through a similar resistor 60 at the output of operational
amplifier 56. Both operational amplifiers 52 and 56 are provided
with resistive feedback loops connecting their output terminal and
their negative input terminal. These feedback paths include
resistors 62 and 64 respectively.
The operational amplifiers used in the encoder are well known
devices and are represented in a standard way. Each includes
negative and positive input terminals and a feedback resistor as
indicated. The signal levels on the negative inputs are independent
of the resistors in the other negative inputs and each appears at
the output with an amplitude equal to its input level times the
ratio of its feedback resistor (e.g. 62) to the input resistor
(e.g. 70) (assuming input channel source impedance is low). All
negative inputs are combined at the output with the same polarity.
The signal on the positive input is subtracted from the signals on
the negative inputs at the amplifier output, but is dependent to an
extent on the resistors in the negative input channels. Hence, a
compensating resistor 86 is provided in the positive input to
provide the precise amplitude relationship desired.
The relative amplitudes of the four input signals from microphones
12F, R, L, and X are adjusted by selection of the values of the
resistors in the circuit. Thus, the level at which the signal from
microphone 12X is applied to coded channel A is determined by the
ratio of the resistance of resistor 62 to the resistance of
resistor 70. The level at which the signal from microphone 12X is
applied to coded channel B is determined by relationships in the
resistances of resistors 68, 54, 72, 86 and 64.
In a preferred embodiment of the invention, as explained above, the
amplitude level at which the signals from microphones 12F and 12X
are applied to coded channels A and B is reduced (relative to the
gain of the L and R signals) by a factor of approximately
0.707.
Additionally, it may be desirable to weight the frequency response
of the signals from microphone 12X for reasons to be described
hereinafter. This may be accomplished by closing switches 74 and
76, thereby bringing into the circuit, in parallel with resistors
70 and 72, the series combinations of capacitor 78 with resistor
82, and capacitor 80 with resistor 84, respectively. The positive
input terminal of operational amplifier 56 is connected through an
appropriate resistor 86 to ground, and the positive input terminal
of operational amplifier 52 is grounded. The series combination of
capacitor 80 and resistor 84 and the series combination of
capacitor 78 and resistor 82 can be selected to have the effect of
emphasizing the high frequency content of the signals in the rear
channel.
In a preferred embodiment of the invention, the values of
capacitors 74 and 76 and the resistance of resistors 82 and 84 are
selected so that the components of the signal in the rear channel
having a frequency over approximately 3,000 Hz. are emphasized
approximately 6 db. over those components below 3,000 Hz. The
preemphasis introduced by the closing of switches 74 and 76 will be
reversed at the reproduction end of the system at a point after the
signal to be applied to speaker 14X has been derived from the coded
signals A and B. This use of preemphasis and deemphasis serves to
minimize transmission from the left and right channels into the
rear output channel.
FIG. 4 shows a simple decoding circuit for use in conjunction with
the system of FIG. 1. The signal in coded channel A is applied
directly to speaker 14L and the signal in coded channel B is
applied directly to speaker 14R. The signal for speaker 14F is
derived by taking the sum of the Signals in coded channels A and B,
added in phase and with a selected relative decrease in amplitude.
The signal coupled to speaker 14X is derived from the difference of
the signals in coded signal channels A and B at equal amplitudes
with a selected relative decrease in amplitude.
Specifically, in the circuit of FIG. 4, the signal in coded channel
A is applied at the primary winding 88 of a first transformer and
the coded signal in coded channel B is applied at the primary
winding 92 of a second transformer. The signal for speaker 14L is
taken across primary winding 88 while speaker 14R is connected
directly across primary winding 92. Both transformers are provided
with first and second secondary windings; the first transformer
having secondary windings 96 and 98 and the second transformer
having secondary windings 100 and 102. The signal for speaker 14F
is taken across secondary windings 96 and 100 connected together in
series addition as shown. The signal for speaker 14X is taken
across secondary windings 98 and 102 which are connected together
in series opposition so that it equals the difference between the
signals in channels A and B. The turns ratio between the primary
and secondary windings is selected so that the voltage at each of
the four secondaries will be 0.707 times the voltage at the
primary.
If preemphasis is employed in the rear or X channel of the encoder,
deemphasis should be employed in the corresponding X channel of the
decoder. This may be accomplished by employing a simple circuit
functionally similar to that introduced by the closing of switches
74 and 76 in FIG. 2, but having the opposite effect. Any of such
circuits, well known in the art, may be used.
The invention as described in FIGS. 1 through 4 is a relatively
simple device which, by virtue of the preferred selection of
amplitude ratios, provides three-decibel separation between
adjacent speakers and complete separation between opposite
speakers. It is obvious that numerous circuits other than those
illustrated in FIGS. 3 and 4 can be used to provide the functions
of the encoder and decoder. Operational amplifiers or transformers
or combinations thereof may be used with either or both devices to
provide the required functions. It is possible for the decoder to
be located preceding or following the power amplifiers. In the
latter case, only two power amplifiers are required for all four
channels which means that existing stereo systems can be adapted
merely by adding the decoder and two speakers at the receiving or
playback station.
The three-decibel separation between adjacent speakers provided by
the preferred embodiment of the invention provides the desired
result, i.e., location of a virtual sound source at any place on a
circle around a listener. However, to further enhance the effect,
it may be desired to increase the separation between adjacent
speakers beyond 3db. This can be accomplished in a number of ways,
some of which are of particular utility with the invention and, as
such, may be considered to be an improvement over the basic system
of FIG. 1. Four such improvements are described below with
reference to FIGS. 5, 6, 7, 8 and 9.
FIG. 5 is a block diagram of a simplified form of a gain control
circuit which varies the gain of any pair of "diagonal" channels
(e.g., left--right) with respect to the other diagonal channels.
Hereinafter, by diagonal channels is meant the combined left and
right channels or the combined front and rear channels as defined
with reference to FIG. 1.
Since, in accordance with the invention, an input from any given
microphone will appear in three adjacent speaker channels with
maximum gain in the speaker channel corresponding to the
microphone, the directional effect can be enhanced by decreasing
the gain of the two speakers on either side of the desired speaker.
In the system of FIG. 1 these two speakers must always be either
the front and rear speakers 14F and 14X, or the left and right
speakers 14L and 14R, i.e., the speakers of a diagonal pair of
channels. It can also be shown that where the absolute value of the
A signal is equal to the absolute value of the B signal (i.e., the
waveforms are identical except for possibly opposite polarity), the
sound source should either be located at the front or rear speaker.
Thus, when this condition exists it is desirable that the gain of
the A+B (front) and A-B (rear) channels be maximum relative to the
gain of the A (left) and B (right) channels. Similarly, when either
the A or B channel signal is zero or the waveforms in A and B are
unrelated, the sound source should be located at the left or right
speaker, in which case the gain of the A+B and A-B channels should
be minimum relative to the gain of the A and B channels.
The foregoing shows that the separation between the output channels
of FIG. 1 (and thus the directional characteristics of the audio
output) can be improved by simultaneously varying the gain in each
pair of two diagonal channels alternatively to controlling the gain
in each of the individual channels. It is also necessary that the
gain in one pair of diagonal channels be accompanied by an
appropriate decrease in the gain of the other diagonal channels.
Otherwise, an increase in gain (for example) to enhance the
directional characteristic of a signal would result in a volume
change of the audio output as a function of direction. However, by
simultaneously decreasing the gain in one pair of diagonal channels
while increasing the gain in the other diagonal channels it is
possible to maintain the total power at the speakers constant, and
separation between adjacent speakers can be improved without
changing the total volume of the system. These functions are
performed by the systems illustrated in FIG. 5.
In FIG. 5, the decoder output is shown at the left. The four
signals A, B, 0.707(A+B) and 0.707(A-B) are coupled to respective
variable gain amplifiers 110L, R, F, and X, which provide the
signals for driving the four speakers as indicated. The A and B
channels are also coupled directly through high-pass filters 112A
and 112B to absolute value circuits 114A and 114B. The absolute
value circuits 114 may comprise full-wave rectifiers the outputs of
which are of the same polarity. These signals, representing the
absolute value of the A and B signals, are fed to respective
logarithmic amplifiers 116A and 116B, which are well-known devices,
providing output voltages equal to the logarithm of the applied
input voltage. The outputs of these amplifiers 116A and 116B are
coupled to the negative and positive inputs, respectively, of an
operational amplifier 118 which subtracts the two signals providing
an output equal to log A -log B (i.e., log A/B ). This signal is
then fed through another absolute value circuit 120 and an
averaging network 122 (an integrating circuit) to a gain control
generator 123 which controls the gain of the two pairs of variable
gain amplifiers 110L, 110R and 110F, 110X as a function of the
output of amplifier 118.
As indicated above, where the absolute values of the A and B
signals are equal, the gain of amplifiers 110F and 110X should be
maximum and the gain of amplifiers 110L and 110R a minimum. When
this condition exists, the output from the operational amplifier
118 will be equal to zero and the gain of amplifiers 110F and 110X
should be a maximum (e.g., unity) while the gain of amplifiers 110L
and 110R is a minimum (e.g., zero). At the other extreme, where
either the B or A signal is equal to zero or the waveforms in A and
B are unrelated, the output of the amplifier 118 will be a maximum
(theoretically infinite but limited in practice to a definite
value, for example, 9 volts). This maximum voltage causes the gain
control generator 123 to provide output voltages which maximize the
gain of amplifiers 110L and 110R and minimize the gain of
amplifiers 110F and 110X.
For conditions between those described above, the gains of the
respective pairs of amplifiers 110 will be appropriately controlled
by generator 123. Mathematically, it can be shown that the curve of
the required gain as a function of log A-log B (the voltage output
of 118) approximates a square root curve to yield constant total
acoustical power output, with the gains in the respective diagonal
channels being equal (for example, to 0.707) when the ratio of A to
B (or B to A) is about 2.4. The following equations may be used to
determine the gain control voltages from
where T is the time constant of the averaging circuit 122, and K is
a constant.
The purpose of the high-pass filters 112A and 112B is to prevent
the passage of low-frequency signals which might otherwise appear
on the inputs to the variable gain amplifiers 110 and possibly
modulate the amplifier inputs. The averaging circuit 122 should
respond to changes in the output of the amplifier 118 quickly
enough so that the ear does not notice the delay, but not so fast
as to pass the actual wave form to the amplifiers 110. As an
example, a 20 millisecond charging rate has been found satisfactory
for practical purposes.
To stabilize the action of the gain control circuits of the decoder
(of FIG. 5) it may be desirable to mix slightly the A and B signals
at the encoder output (or the left and right inputs to the encoder)
to minimize excursions of the A/B signal ratio. Conversely, to
prevent the log of this ratio from going to zero, constant phase
differences may be introduced between the respective front (rear)
signals applied to the A and B channels at the encoder. This has
the effect of restricting gain control action to a relatively
narrow range so that the gain control action will not be audible at
the speakers. Such mixing can be done in proportions which will
accomplish the desired result without materially degrading the
audio characteristics. Where extreme channel separation is
required, this technique would not be used.
As noted previously, there are many different ways of controlling
the gain in the respective channels to provide the desired
directional enhancement at the speakers. The embodiment illustrated
and described with reference to FIG. 5 is a relatively inexpensive
way of providing the desired gain control of the respective pairs
of speakers. An equivalent circuit could compare the logs of A and
B and the logs of A and -B, take the absolute value of these two
signals, average them and use the combined value as a control
signal for gain control. This equivalent circuit would accomplish
the same results, but would probably require more components and
thus be more costly.
Of course, it is not necessary that the gain of diagonal channels
be concurrently controlled to improve the directional
characteristics of the audio output. Various well-known means may
be provided in each of the output channels (A, B, A+B and A-B) to
increase or decrease the gain of the signal, depending upon its
relationship to a preselected level. For example, FIG. 6
illustrates a circuit which may be placed in each of the four
output channels to control the gain of these channels
independently. It includes a series connected device 124 having a
high negative thermal resistance coefficient and a device 125
having a high positive thermal resistance coefficient across the
channel. If the signal from the decoder increases above a value
which may be nominally selected to provide a gain of 1, the
resultant temperature increase of devices 124 and 125 will cause a
simultaneous decrease in resistance of device 124 and increase in
resistance of device 125. This impedance change will cause an
increase in gain at the channel output. Similarly, reduced input
signals will be attenuated by increasingly greater amounts as the
signal decreases from the preselected nominal value. In this way,
those channels which carry the predominant signal will be provided
with an increase in gain relative to the other channels. This will
enhance the desired signals while attenuating the adjacent channel
signals thus improving the separation between adjacent
channels.
By way of example, device 134 may comprise a doped silicon or other
semiconductor element and device 125 may be a common incandescent
lamp. If desired, a capacitor may be placed in series with the
parallel device 125 so that gain control is essentially provided
only for high-frequency components. Numerous other well-known
devices also may be used in place of the devices 124 and 125 or
both of them.
FIGS. 7 and 8 illustrate a further embodiment of a gain control
system employing the basic principles of the invention wherein
subsonic control tones are impressed upon the A and B channels for
the purpose of controlling the gain of the two pairs of diagonal
channels from the decoder 20. In describing the operation of FIGS.
7 and 8, the microphones 12, speakers 14, encoder 18 and decoder 20
perform the same function as the corresponding parts of FIG. 1 and
therefore are not described further. Ultrasonic, or other, tones
can also be used.
To facilitate an understanding of this embodiment it is convenient
to refer to power ratios rather than voltage ratios as previously.
The power which can be derived from a given signal is directly
proportional to the square of the voltage level of that signal.
The signal recording means is illustrated in FIG. 7. The outputs of
the front and rear microphones 12F and 12X are sensed and coupled
to a power-adding circuit 130, while a similar power-adding circuit
131 sums the power outputs from microphones 12L and 12R. These two
power-adding circuits are devices which produce output voltages
directly proportional to the total power which can be derived from
the applied input voltages. Their output voltages are then summed
in an adding circuit 132, the output of which is thus proportional
to the total power in the four input channels.
The output of summing circuit 130 is also coupled to a multiplier
132 which doubles this signal, thus providing a voltage output
equal to twice the power in the front and rear input channels. The
ratio of these two quantities (twice the power in the front-rear
channels to total power in all four channels) is then computed by a
ratio circuit 134 which, in turn, causes respective A and B
modulators 136 and 138 to modulate a 20-cycle (or other subsonic)
tone from oscillator 140. Ratio circuit 134 may be any of a number
of well-known circuits and, for example, may produce a direct
output voltage having an amplitude proportional to the ratio of the
applied input voltages and a polarity indicative of whether the
measured ratio is greater or less than one.
Modulators 136 and 138 may be adapted to amplitude-modulate the
20-cycle tone from oscillator 140 with respect to a preselected
level, depending upon the magnitude and polarity of the applied
control voltage from the ratio circuit 134. When the A modulator
136 provides a tone of increased amplitude, the B modulator 138
should be providing a tone of proportionately decreased amplitude.
These modulated tones are then added to the A and B outputs of
encoder 18 to provide the signals which are to be conveyed by the
two-channel transmission path and which, in this particular
embodiment, are indicated as A' and B'.
The receiving end of the system is illustrated in FIG. 8. Two high
pass filters 142 and 144 are used to separate the audio control
tones from the A and B audio signals on the A' and B' channels.
These A and B signals from the filters 142 and 144 are coupled to
decoder 20 to provide the four output channels described above with
respect to FIG. 1.
The control tones from the filters 142 and 144 are coupled to a
gain control generator 146 which controls the gain of variable gain
amplifiers 148R, F, L, and X to increase the gain of one pair of
diagonal channels while appropriately decreasing the gain of the
other pair of diagonal channels. From the preceding discussion of
FIG. 7, it follows that the amplitudes of the control tones will
each be equal to twice the desired power in their corresponding
diagonal input channels divided by the total power in the system.
Each of these signals varies from a value of zero to two and their
sum should always equal one. Accordingly, since the desired power
ratios (i.e., the power ratios at the microphones) are directly
represented in the control tone signals it is a simple matter for
gain control generator 146 to utilize these known ratios to control
the gain of amplifiers 148R, L and 148X, F to recreate the same
ratios at the outputs of the amplifiers 148. This will necessarily
enhance the desired signals while deemphasizing these signals which
are not in their corresponding channels. The total power will also
not be varied due to directionality changes. Generator 146 also
serves as a normalizer to maintain the total gain of the four
channels such that the sum of the power in the respective channels
is maintained equal to a constant. This prevents unwanted changes
in the amplitude of the control tone from affecting the volume of
the outputs from the respective loudspeakers.
The embodiment of FIGS. 7 and 8 is capable of providing substantial
separation between any two adjacent channels. However, in its basic
form, this embodiment is not capable of recreating the exact power
ratio at the output speakers that existed at the input from the
microphones. Obviously, this would be desirable since if it could
be accomplished it would recreate the power relationship existing
at the microphones. As explained with reference to FIG. 9, a
further modification is possible which enables the exact recreation
of the power ratio which existed on the input channels.
After encoding and decoding some signal "spreading" occurs at the
speaker due to each input appearing at half power in each of the
outputs adjacent to the desired output (as well as at full power in
the desired output). That is, the outputs will contain spurious
signals in addition to wanted signals.
The following consequences follow:
1. The spurious signal power is equal to the wanted signal
power.
2. The sum of the powers of the signals in one pair of diagonal
channels is equal to that of the other pair of diagonal
channels.
3. The spurious powers in the two outputs of one pair of diagonal
channels are equal.
One step in bringing output powers to a level corresponding to the
respective input powers is to compute the necessary change in the
ratio of the total power in the right-left diagonal channels to the
power in the front-rear diagonal channels from equality (as exists
in the decoder output) to arrive at the proper ratio. The diagonal
pairs input power relation is available from the subaudible
frequency code signals or approximately from the analyzer circuits
described herein.
Simply increasing the power gain for one diagonal pair while
decreasing it for the other would adequately adjust the power
relation between the two pairs of diagonal channels. It would not
assure the proper relation within a diagonal pair (e.g., relation
of P.sub.A to P.sub.B).
Note however that if one considers that half of the total power (A,
B, A+B, A-B) is spurious power, and if the adjustment of power in
diagonal pairs of outputs is such as to halve the total power, then
the reduction (in absolute terms) in power in each output of a
diagonal pair should be equal. This is true because the origin of
spurious power is such that it is fed equally into each output of a
diagonal pair. Putting this another way, one should maintain the
power difference between A and B (and between (A-B) and (A+B))
while reducing overall power by an amount (one-half) corresponding
to the level of spurious signal power.
In order to restore the decoder output powers (P.sub.A, P.sub.B,
P.sub.A+B, P.sub.A-B) to relative values corresponding to encoder
inputs the adjusted powers Q.sub.A, Q.sub.B, Q.sub.A+B, Q.sub.A-B
should be as follows:
Q.sub. A =1/2([1+C.sub. LR ]P.sub. A -[1-C.sub. LR ]P.sub. B)
q.sub. b =1/2([1+c.sub. lr ]p.sub. b -[1-c.sub. lr] p.sub. a)
q.sub. a+b =1/2([1+c.sub. fr ]p.sub. a+b -[1-c.sub. fr] p.sub.
a-b)
q.sub. a-b =1/2([1+c.sub. fr ]p.sub. a-b -[1-c.sub. fr] p.sub.
a+b)
c.sub. lr and C.sub. FX are the code signal levels representing the
part of total input power in the left-right and front-rear pairs of
diagonal channels, respectively. C.sub. LR +C.sub. FX =1.
The gain G.sub.A for the A channel amplifier to produce the power
Q.sub.A is obviously
G.sub. A =(Q.sub. A /P.sub. A)=1/2([1+C.sub. LR ]-[1-C.sub. LR
](P.sub. B /P.sub. A)
Electronic analog computation systems implementing the above
formulas may be used to set each channel gain precisely at the
value to restore the output power to a relative value corresponding
to the corresponding input power. In practice this need only be
approximated to achieve excellent results. Various types of such
approximations are utilized in the systems herein disclosed.
Obviously, the gains of all channels may be multiplied by any
common factor without disturbing the relative power levels.
There are many circuits which can be used to achieve this
objective. One embodiment is illustrated in block diagram form in
FIG. 9, which shows only that portion of the circuit which would be
used for the right and left speakers 14R and 14L. This particular
embodiment incorporates portions of the circuit of FIG. 8 and such
portions are numbered accordingly. The control circuit for the
front and rear channels (A+B and A-B) will be the same as that
illustrated in FIG. 9, and is not illustrated merely for purposes
of convenience.
It is recalled that the outputs of the amplifiers 148 comprise two
pairs of signals (left-right and front-rear) which are directly
proportional in power to the power signals provided by the input
microphone pairs 12F, 12X and 12R, 12L. The left and right channel
signals are represented in FIG. 9 by the letters A" and B". The
inputs to the amplifiers 148R and 148L are the A and B signals
prior to any modification. In FIG. 9, signals proportional to the
power in and out of the respective amplifiers are summed and a
difference signal generated to control the gain of two additional
variable gain amplifiers which in turn feed the speakers.
Suitable voltage squaring circuits 150R and 150L are shown
responsive to the A and B signals for producing direct voltages
which are proportional to the power to be derived from the signals.
These outputs are fed to a summing circuit 151 which produces a
signal proportional to the sum of these power signals, i.e.,
P.sub.A +P.sub.B.
Similar voltage squaring circuits 152R and 152L produce signals
proportional to the power to be derived from the outputs of the
amplifiers 148R and 148L represented as P.sub.A" and P.sub.B", and
these latter signals are summed in a second summing circuit
154.
The difference between these two power sums is determined by a
difference circuit 156 which produces a signal proportional to this
power difference as indicated on the drawing.
As in the preceding circuits, amplifiers 151, 154 and 156 may
comprise conventional operational amplifiers. In practice, the
functions of the three illustrated amplifiers may be accomplished
by a single operational amplifier in an obvious way.
A further operational amplifier 156 compares the magnitude of the
P.sub.A and P.sub.B signals and determines which of the two
channels should be increased and which decreased. In each case, the
change in gain is dependent upon the output of the amplifier 156
but this output does not indicate which of the two signals should
be increased and which decreased. However, it can be shown that
when P.sub.A is greater than P.sub.B the signal level and the A
channel should be increased while the B signal is decreased, and
vice versa. Accordingly, a gain control circuit 158 is responsive
to the outputs of amplifiers 156 and 157 and produces two signals
which are coupled to two additional variable gain amplifiers 160R
and 160L in the A and B channels, respectively. The gain control
circuit 158 will increase the gain of one of amplifiers 160R while
equivalently decreasing the gain of the other amplifier 160,
depending upon the polarity of the output of amplifier 157 and the
amplitude of the output of amplifier 156. The gain control voltages
are such as to cause the respective outputs of amplifiers 160R and
160L, which are fed to the speakers 14R and 14L, to have the same
power ratio as existed at the outputs of the microphones 12R and
12L.
As noted previously, the present invention is compatible, and has
utility, with all present commercial binaural stereophonic systems.
All that is required is that the two encoded channels derived
pursuant to the principles of FIG. 1 be recorded or transmitted, as
the case may be, on the two available channels. In the case of
stereophonic records where the two channels are recorded in
respective grooves disposed at 45 degrees with respect to the
horizontal, the desired output channels may be derived from
suitable resolution of the recorded signals in these channels
alone. For example, the left and right channels (i.e., A and B)
would each be derived from components of groove modulation (i.e.,
stylus motion) at 45.degree. from the surface of the record; the
front or A+B channel would be derived from components parallel to
the record surface (actually equal to 0.707A+0.707B); and the rear
or A-B channel would be derived from components perpendicular to
the record surface. These components are shown in FIG. 15 as
indicated thereon.
Where a stereo record has been recorded in accordance with the
invention, the record itself will differ from prior art records in
that the A+B signal can contain frequencies which are not present
in the A-B signal. In a prior art binaural stereo record, the A+B
and A-B channels necessarily will contain the same frequencies.
Because of the special form of encoding provided by the invention,
where there is complete separation between the front and rear
channels, as explained previously, the rear signals will be
cancelled in the front channel (A+B) and the front frequencies will
be cancelled in the rear channel (A-B).
No effort has been made in describing the basic invention and
preferred embodiments of various improvements upon the basic
invention to consider every possible method and apparatus for
practicing the invention. The following is a description of an
early form of the circuit intended to be used as a decoder. In the
equations below the term "log" is used to designate logarithmic
functions which have a polarity as determined by the polarity of
the antilog.
In this embodiment of the invention, the gain associated with each
speaker is determined by a combination of a gain control element
serially connected in the respective channel, and a gain control
voltage generator whose output is coupled to the gain control
element. The audio signal in each playback channel (A, B, A+B, or
A-B) passes through the respective gain control element. Then,
according to the output signal of the control voltage generator,
the signal in the gain control element is either enhanced or
attenuated. When the output signal of the control voltage generator
is at a maximum, the output of the gain control element is at a
maximum, and vice versa.
The gain control elements 203 and 204 are thus controlled by an
output voltage V.sub.LR produced by the control voltage generator
210. The gain control element 208 is controlled by an output
voltage V.sub.F produced by the control voltage generator 212, and
the gain control element 216 is controlled by an output voltage
V.sub.X produced by the control voltage generator 218. If desired,
separate control voltage generators may be respectively coupled to
gain control elements 203 and 204 rather than just one (generator
210).
The expressions for each of the control voltages V.sub.LR, V.sub.F,
and V.sub.X are dictated by design considerations of the various
control voltage generators which produce these expressions, as well
as by the specific phase, waveform, and level cues present in the
original signals A and B which are to activate the respective
speakers.
For example, the desired acoustical reproduction requires that the
gain associated with the speakers of the left and right channels
increases as the ratio of the intensity levels of the signals A and
B diverges from unity, or their waveforms become increasingly
dissimilar. To achieve this result, the control voltage V.sub.LR
applied to the gain control elements 203 and 204 may be represented
by one of various expressions below. In the following three
equations A & B are absolute values.
V.sub. LR =k (A-B/A+B) ; V.sub. LR =k log (A-B/A+B) ; V.sub. LR =k
log (a/b) .
In accordance with this embodiment, however, the control voltage
generator 210 comprises analog circuitry which furnishes a control
voltage
V.sub. LR = log(A/B) + log-(A/B) -k log(envA/envB) -V.sub. SAT,
where the term "env" indicates the instantaneous value of the
intensity of the envelope voltage, independent of phase and
polarity, obtained by full wave rectification and smoothing of the
particular signal A or B. This control voltage V.sub.LR then
increases as the loudness level associated with either the A or B
signals becomes stronger with respect to the other, or their
waveforms become increasingly dissimilar. A constant voltage
V.sub.SAT assures a positive value for the control voltage.
The gain in the front channel, on the other hand, is to increase as
the ratio of the intensity levels of each of the signals A and B
approaches unity (L=0), and as their waveforms become similar and
in phase. Accordingly, the expression for the control voltage
produced by the control voltage generator 212 may be
V.sub. F =V.sub. SAT - log (A/B) .
The gain associated with the rear channel is to be activated
increasingly as the intensity level ratio of the signals approaches
unity (L=0), and as their waveforms become similar and of opposite
polarity. Accordingly, the control voltage generator 218 may
provide an output expression
V.sub. X =V.sub. SAT - log (-A/B) .
The construction of the control voltage generators 210, 212, and
218 is based upon analog circuits providing these control voltages
V.sub.LR, V.sub.F, and V.sub.X, respectively. For example, in order
to sense loudness level differences, circuit means for obtaining
the log ratio of the intensity levels of the signals A and B are
required. These circuits may be incorporated into a single
structure as depicted in FIGS. 11, 12 and 13. Accordingly, the
signals A and B are fed into the log ratio unit 220, the output of
log ratio unit 220 being coupled to the absolute value unit 221,
its output being coupled to the averaging filter unit 222. The
output of the averaging filter is then directed through an inverter
235, whose output along with the voltage V.sub.FSAT is applied to
the input of the adder 232. The output voltage V.sub. F =V.sub. SAT
- log (A/B) is then applied to one input terminal of the gain
control element 208 in the front channel.
Similarly, the control voltage V.sub.X is provided by applying the
signal voltage B and the inverted signal voltage A (the output of
the inverter 239) to the log ratio means 223. The output of the
means 223 is coupled to the absolute value means 224, which output
is coupled to the averaging filter 225. The output signal of the
filter 225 is applied to the input of the inverter 236, the
resulting inverted output being applied along with the constant
voltage V.sub.XSTAT to the adder 233. The output voltage V.sub. X
=V.sub. SAT - log (-A/B) is then applied to one input terminal of
the gain control element 216 in the rear channel X.
The control voltage generator 210 comprises the absolute value
circuit means 226 and 230, averaging filters 227 and 231, log ratio
means 228, absolute value means 229, inverters 237 and 238, and
adder 234. The output of the filters 222 and 225 are respectively
coupled along with the outputs of the inverters 238 and 237 to the
input of the adder 234. The output voltage V.sub. LR = log (A/B) +
log-(A/B) -k log (envA/envB) -V.sub. SAT is then applied to an
input terminal of the gain control elements 203 and 204.
The specific circuitry constituting the various blocks of FIG. 11
is illustrated in FIGS. 13 and 14. Accordingly, the log ratio means
220 and 223 comprise operational amplifiers AR.sub.2, AR.sub.3,
AR.sub.4, AR.sub.5, and AR.sub.10, transistors Q.sub.1 -Q.sub.6,
capacitors C.sub.1 -C.sub.3 and C.sub.14 -C.sub.17, and resistors
R.sub.9 -R.sub.11, R.sub.16 -R.sub.18, R.sub.23 -R.sub.26, and
R.sub.35 -R.sub.38. The absolute value means 221 and 224 comprise
operational amplifiers AR.sub.6, AR.sub.7, AR.sub.11, and
AR.sub.12, diodes D.sub.5 -D.sub.8, resistors R.sub.43, R.sub.44,
R.sub.47 -R.sub.50, R.sub.55, R.sub.56, R.sub.59, and R.sub.60, and
capacitors C.sub.25 and C.sub.26.
The averaging filters 222 and 225 comprise operational amplifiers
AR.sub.8 and AR.sub.13, capacitors C.sub.27 -C.sub.38, and
resistors R.sub.62 -R.sub.65, R.sub.67 -R.sub.74, R.sub.76, and
R.sub.77. The inverters 235 and 236 shown in block form in FIG. 11
are unnecessary when negative voltages are utilized for V.sub.FSAT
and V.sub.XSAT applied through resistors R.sub.82 and R.sub.85 to
the negative terminals of the amplifiers AR.sub.5 and AR.sub.14, as
illustrated in FIG. 13.
The inverter 239 comprises operational amplifier AR.sub.1, and
resistors R.sub.2 and R. The adder 232 comprises operational
amplifier AR.sub.9, resistors R.sub.93, and R.sub.94, and
capacitors C.sub.40 and C.sub.41. The adder 233 comprises
operational amplifier AR.sub.14, capacitors C.sub.42 and C.sub.43,
and resistors R.sub.95 and R.sub.96.
As illustrated by FIG. 14, the absolute value means 226 comprises
operational amplifiers AR.sub.15 and AR.sub.16, resistors R.sub.3,
R.sub.5, R.sub.6, R.sub.12, and R.sub.14, diodes D.sub.1 and
D.sub.2, and capacitor C.sub.4. Absolute value means 230 comprises
AR.sub.19 and AR.sub.20, resistors R.sub.1, R.sub.7, R.sub.8,
R.sub.13, and R.sub.15, diodes D.sub.3 and D.sub.4, and capacitor
C.sub.5.
Averaging filter 227 comprises operational amplifier AR.sub.17,
resistors R.sub.19, R.sub.27, R.sub.33, R.sub.39, R.sub.41,
R.sub.28, R.sub.21, and R.sub.31, and capacitors C.sub.6, C.sub.8,
C.sub.10, C.sub.12, C.sub.18, and C.sub.19. Averaging filter 231
comprises operational amplifier AR.sub.21, resistors R.sub.20,
R.sub.29, R.sub.34, R.sub.40, R.sub.42, R.sub.22, R.sub.30, and
R.sub.32, and capacitors C.sub.7, C.sub.9, C.sub.11, C.sub.13,
C.sub.20, and C.sub.45.
Log ratio means 228 comprises operational amplifier AR.sub.18,
AR.sub.22, AR.sub.23, transistors Q.sub.7 -Q.sub.10, resistors
R.sub.45 and R.sub.46, R.sub.51 -R.sub.54, R.sub.57, and R.sub.58,
and capacitors C.sub.21 -C.sub.24. Absolute value means 229
comprises operational amplifiers AR.sub.24 and AR.sub.25, resistors
R.sub.61, R.sub.66, R.sub.75, R.sub.78, and R.sub.79, and diodes
D.sub.9 and D.sub.10. Adder 234 comprises operational amplifier
AR.sub.26, resistors R.sub.88 -R.sub.91, R.sub.97, and capacitor
C.sub.44.
It is to be noted from FIG. 13 that the outputs of the operational
amplifiers AR.sub.8 and AR.sub.13 are actually negative, and are
applied to the positive terminals of the adder amplifier AR.sub.9
and AR.sub.14. This fact, plus the fact that negative supply
voltages (-15 volts) are utilized for V.sub.SAT, enables the
omission of the inverters 235 and 236. In like manner, the use of a
positive supply voltage for F.sub.SAT (FIG. 9) applied to the
negative input of amplifier AR.sub.26, and the application of the
positive voltage of the output of the amplifier AR.sub.25 obviates
the need for inverters 237 and 238. Negative outputs at the Z and Y
terminals shown in FIG. 13 are then applied to the negative input
terminal of amplifier AR.sub.26 as shown in FIG. 14.
Neither the expressions for V.sub.LR, V.sub.F, or V.sub.X nor the
specific circuitry illustrated in FIGS. 11, 13 and 14 for providing
these functions is critical to the present invention. Accordingly
various other circuitry providing the desired result may be
utilized to control the gain of the respective speakers. Even if
the control voltage generators are employed in conjunction with the
gain control elements to provide this gain control function, a
different spatial arrangement of the speakers, a different number
of channels, or mere economy may dictate entirely different
circuitry for the generators.
As a specific feature of the present invention, each of the gain
control elements 203, 204, 208, and 216 comprise circuitry as
illustrated in FIG. 12. The heart of the circuitry includes a
semiconductor diode light source D.sub.201 and a photo-responsive
resistor PR.sub.201. The light emitting diode D.sub.201 may be of
gallium arsenide, for example.
As the current through the diode D.sub.201 increases, an increasing
amount of light is thereby emitted which reaches the photoresistor
PR.sub.201. This results in a corresponding decrease in the
resistance of the photoresistor, and current flowing through the
photoresistor PR.sub.201 is thereby enhanced. Any decrease in
current through the light emitting diode then similarly results in
a decrease of current flowing through the resistor PR.sub.201.
Accordingly, current representing the audio signal, for example,
the left channel, passes through the photoresistor PR.sub.201 and
thereon through the emitter follower amplifier arrangement
including Q.sub.203 and Q.sub.204. The control voltage, V.sub.LR
for example, is applied to the other input terminal. As V.sub.LR
increases in response to the various level, waveform, and/or phase
cues, as described above, this causes the corresponding enhancement
of the signal current through the photoresistor, and consequently
the enhancement of the output voltage V.sub.OUT. This increased
voltage V.sub.OUT thus increases the gain associated with the
loudspeaker in that channel, in this instance loudspeaker 14L. As
the control voltage V.sub.LR decreases, the opposite effect occurs.
In like manner the gain of the other transducers may be
controlled.
Although not illustrated in the drawing, it may be desirable to add
a series resistor and capacitor (RC circuit) in parallel with the
photoresistor PR.sub.201 so that both gain and frequency response
may be simultaneously controlled. Thus, the frequency response may
be narrowed for low levels of current, and turntable rumble and 60
cycle noise may be eliminated at these low levels.
The circuits described with reference to FIGS. 12, 13 and 14 may be
constructed using conventional circuit components having desired
values. The circuits may be of conventional discrete components or
may utilize integrated circuits of monolithic or hybrid
construction. As an example, however, of typical component values,
the circuits illustrated in these figures were constructed
employing components having the following values:
OPERATIONAL AMPLIFIERS AR.sub.1, AR.sub.5 -AR.sub.17, AR.sub.19
-AR.sub.21, P85AU(Philbrick-Nexus) AR.sub.23 -AR.sub.26 AR.sub.2
-AR.sub.4, AR.sub.13, AR.sub.22 P25AU TRANSISTORS Q.sub.1, Q.sub.3,
Q.sub.5, Q.sub.7, Q.sub.9, 1/2-PL1/P (Philbrick-Nexus) Q.sub.2,
Q.sub.4, Q.sub.6, Q.sub.8, Q.sub.10 1/2-PL1/N Q.sub.201, Q.sub.203
2N3707 Q.sub.202, Q.sub.204 2N3704 DIODES D.sub.1 -D.sub.10 1N914
D.sub.201 TIXL09 RESISTORS R.sub.1, R.sub.3, R.sub.5 -R.sub.8,
R.sub.14, R.sub.15, R.sub.43 200k R.sub.44, R.sub.47 -R.sub.50,
R.sub.59 -R.sub.61, R.sub.66 R.sub.78, R.sub.79 R.sub.9 -R.sub.11,
R.sub.23 -R.sub.26, R.sub.45, R.sub.46 10k R.sub.52, R.sub.54
R.sub.12, R.sub.13, R.sub.55, R.sub.56, R.sub.75, R.sub.82 100k
R.sub.83, R.sub.85, R.sub.86, R.sub.88 -R.sub.91, R.sub.93
-R.sub.97 R.sub.202 R.sub.2, R.sub.4 22k R.sub.16 -R.sub.18,
R.sub.51, R.sub.53 4.7k R.sub.19, R.sub.20, R.sub.33, R.sub.34,
R.sub.62, R.sub.63 56k R.sub.73, R.sub.74 R.sub.21, R.sub.22,
R.sub.27, R.sub.29, R.sub.31, R.sub.32 220k R.sub.64, R.sub.65,
R.sub.67, R.sub.69, R.sub.71, R.sub.72 R.sub.28, R.sub.30,
R.sub.68, R.sub.70 860k R.sub.35 -R.sub.38, R.sub.57, R.sub.58,
R.sub.205, R.sub.206 180k R.sub.39 -R.sub.42, R.sub.76, R.sub.77,
R.sub.80, R.sub.81 156k R.sub.203 470k R.sub.204, R.sub.207 180
R.sub.209 330 R.sub.208 1.8k R.sub.84, R.sub.87, R.sub.92 25k
linear pot. R.sub.201 10k linear pot. CAPACITORS C.sub.1 -C.sub.3,
C.sub.21, C.sub.22 6.8pf. C.sub.4, C.sub.5, C.sub.14 -C.sub.17,
C.sub.23 -C.sub.26 100pf. C.sub.39 -C.sub.44 C.sub.6, C.sub.7,
C.sub.12, C.sub.13, C.sub.27, C.sub.28, C.sub.33, c.sub.34 0.03mf.
C.sub.8 -C.sub.11, C.sub.29 -C.sub.32 0.12mf. C.sub.18, C.sub.20,
C.sub.35, C.sub.37 0.012mf. C.sub.19, C.sub.36, C.sub.38 0.05mf.
C.sub.201 10mf. C.sub.202 120mf. PHOTORESISTOR PR.sub.201 CL3AL
(Clairex)
In the gain control embodiment of the invention shown in FIG. 5,
where the channel gain is controlled as a function of the
instantaneous value of the A and B signals, an error can occur if
different input signals of approximately equal amplitudes are
coupled to the encoder by the front and rear microphones 12F and
12X. In this case, assuming no other input, the A signal will
differ from the B signal which, as explained previously, will cause
the gain in the left and right output channels to be maximized
while the gain in the front and rear channels is minimized.
Analogously, where the logs of the envelopes are compared (as in
the embodiment of FIGS. 10-14), the appearance of different signals
of approximately equal amplitudes on the left and right microphone
inputs will cause an error in the output by minimizing the gain of
the left and right speakers while maximizing the gain of the front
and rear speakers. From a practical viewpoint, this condition in
most cases does not affect the utility of the invention
particularly where there is suitable control of the audio material
being recorded (or transmitted). However, if this error should
prove to be a problem, it is conceivable that two systems such as
shown in FIGS. 5 and 11 could be combined by deriving both control
signals simultaneously and averaging these control signals to
provide the final gain control. One way to effectively "combine"
the two systems would be to include an averaging circuit (such as
221 and 222) in series after each of the absolute value circuits
116A, 116B of FIG. 2, with an adjustable resistance in shunt around
the added components. When the averaging circuit is shorted out
(the resistance is set to zero), the instantaneous ratio of A/B is
sensed as in FIG. 5. At infinite resistance the circuit senses the
ratio of envelope A to envelope B (FIG. 11). At intermediate
resistance, combinations of the two are sensed.
The signal source of the present invention having three or more
signals encoded in two is also compatible with and will playback
through conventional monaural and two channel stereophonic speaker
arrangements. Various modifications of the disclosed embodiments
may also be employed while still utilizing the principles of the
present invention. For example, while the reproduction or playback
apparatus has particular applicability when a two channel signal
source is used, additional playback channels may also be coupled to
a three or more channel signal source, the speakers associated with
each channel being activated by cues present in one or all of the
three or more source signals.
Additional channels may be coupled to the stereo signal source,
their respective gains being determined by other defined
relationships in the signals A and B. Thus, a fifth or sixth
channel may be added, for example, the gain associated with the
left and right channels increasing responsive to dissimilar
waveforms and unequal loudness levels of signals A and B; the gain
associated with the fifth and sixth channels increasing with
similar waveforms and unequal levels. The loudspeakers in these
channels may be added at any point around the listener, thereby
enabling the location of the virtual sound source at additional
points around a 360.degree. circle, or even in another plane
perpendicular to that of the four channels.
For purpose of convenience reference has been made herein to
polarity differences which may be considered as phase differences
of 180.degree.. This is preferred because of the relative ease of
"inverting" a signal to obtain a reversal of polarity. The
principles of the invention may also be employed with other phase
relationships by using known devices which are capable of providing
substantially constant phase shifts for all frequencies in the
audio band of interest. Instead of polarity or phase relationships,
it may be possible to utilize time delays for encoding the
respective channels with complementary time delays in the
decoder.
Various other modifications may be made to the above described
embodiments by one ordinarily skilled in the art without departing
from the spirit and scope of the invention.
* * * * *