Dolby Surround Pro Logic Decoder Principles Of

Dolby Surround Pro Logic Decoder Principles Of
Dolby Surround Pro Logic Decoder
Principles Of Operation
by Roger Dressler
To many people, the term surround implies that something new has been added to a stereo audio
signal something requiring more than two speakers for reproduction. While this is true, it should
also be realized that stereo itself is not confined to just two channels of sound; it can mean three,
four, six, or any number so desired. Somewhere along the way, home stereo became
synonymous with two channels, while at the movie theatre several multi-channel stereo formats
appeared–and disappeared–in the 1950's and 60's.
By the latter 1970's, Dolby Stereo was established as a stereophonic reproduction system having
from three to as many as six channels of sound to enhance the action and drama of theatrical
presentations in ways only approached by two-channel systems. The most obvious feature of
Dolby Stereo is that an additional channel of sound is reproduced along the sides and back of the
theatre to "surround" the audience with sound.
Dolby Laboratories then devised a simple method to emulate the overall effect of Dolby Stereo in
a home environment by recovering these extra surround sound effects. Introduced in 1982, Dolby
Surround products today enjoy widespread popularity, with millions sold worldwide and the pace
accelerating due to technical breakthroughs in home audio/video delivery formats, especially
stereo television and satellite.
The advanced Dolby Pro Logic Surround system appeared in consumer products in the fall of
1987, quickly gaining favor with home theatre enthusiasts for its improved spatial articulation and
expansive listening area. This leap in performance, however, came at a price; the decoder
circuitry was significantly more complex, in effect limiting Pro Logic technology to high-end A/V
Just one year later came the breakthrough needed to make Pro Logic decoding as economically
attractive as it is sonically: a custom integrated circuit was developed, consolidating a significant
number of processing circuits into one package [everything shown in Fig. 10 below and more].
Since that time several more analog and digital integrated circuits have become available,
providing high performance multi-channel sound in a wide range of cost-effective home systems.
This paper is intended to help the reader understand how Dolby Pro Logic works. If you are
familiar with the general principles of Dolby Surround, you may wish to scan ahead to Part 2 for
the discussion of Pro Logic decoding. If you are new to the subject of Dolby Surround, it is
suggested that you read the background material in Part 1.
Dolby analog movies and Dolby Surround video and television programs include an additional
sonic dimension over conventional stereo productions. They are made using a Dolby MP (Motion
Picture) Matrix encoder, which encodes four channels of audio into a standard two-channel
format, suitable for recording and transmission in the same manner as regular stereo programs.
To recapture the dimensional properties brought by the additional channels, a Dolby Surround
decoder is used. In the theatre, a professional decoder is part of the Dolby Stereo cinema
processor used to play 35 mm stereo optical prints. The decoder recovers the left, center, and
right signals for playback over three front speakers, and extracts the surround signal for
distribution over an array of speakers wrapped around the sides and back of the theatre. (These
same speakers may also be driven from four discrete tracks on 70 mm Dolby Stereo magnetic
prints, but in this case no decoder is needed.)
Home viewing of movies on video has become extremely popular, and with the advent of hi-fi
stereo VCR's, stereo television and laser discs, the audio side of the video presentation has
improved considerably, inviting the use of full-range sound reproduction. The ability to deliver
high quality audio in these formats made it easy to bring matrix-encoded surround soundtracks
into the home as well, thus establishing the foundation for Dolby Surround.
1.1. The Dolby MP Matrix
One of the original goals of the MP Matrix was to enable Dolby Stereo soundtracks to be
successfully played in theatres equipped for mono or two-channel stereo sound. This allowed
movies to be distributed in a single optical format, and has furthermore resulted in complete
compatibility with home video media (without requiring separate mixes). Since the three front
channels of the MP Matrix are assembled in virtually the same way as a conventional stereo mix–
left in left, center equally in left and right, and right in right–playing a Dolby Stereo mix over two
speakers reproduces the entire encoded soundtrack. There is only one thing missing: the
surround signal is not reproduced in its proper spatial perspective. When the first home decoder
was developed in 1982, its goal was to recover this missing spatial dimension.
Before we discuss decoders, however, it is necessary to see how the MP Matrix encoder works.
Referring to the conceptual diagram in Fig. 1 (below) the encoder accepts four separate input
signals, left, center, right, and surround (L, C, R, S), and creates two final outputs, left-total and
right-total (Lt and Rt).
The L and R inputs go straight to the Lt and Rt outputs without modification. The C input is
divided equally to Lt and Rt with a 3 dB level reduction (to maintain constant acoustic power in
the mix). The S input is also divided equally between Lt and Rt, but it first undergoes three
additional processing steps:
Frequency bandlimiting from 100 Hz to 7 kHz.
Encoding with a modified form of Dolby B-type noise reduction.
Plus and minus 90-degree phase shifts are applied to create a 180 degree phase
differential between the signal components feeding Lt and Rt.
It is clear that there is no loss of separation between the left and right signals; they remain
completely independent. Not so obvious is that there is also no theoretical loss of separation
between the center and surround signals. Since the surround signal is recovered by taking the
difference between Lt and Rt, the identical center channel components in Lt and Rt will exactly
cancel each other in the surround output. Likewise, since the center channel is derived from the
sum of Lt and Rt, the equal and opposite surround channel components will cancel each other in
the center output.
The ability for this cancellation technique to maintain high separation between center and
surround signals requires that the amplitude and phase characteristics of the two transmission
channels be as close as possible. For instance, if the center channel components in Lt are not
identical to the ones in Rt as a result of a mismatch in channel balance, center information will
come out of the surround channel in the form of unwanted crosstalk.
1.2. The Dolby Surround Decoder
This leads us to the original Dolby Surround decoder. The block diagram in Fig. 2 (below) shows
how the decoder works. The Lt input signal passes unmodified and becomes the left output. The
Rt input signal likewise becomes the right output. Lt and Rt also carry the center signal, so it will
be heard as a "phantom" image between the left and right speakers, and sounds mixed anywhere
across the stereo soundstage will be presented in their proper perspective.
The L-R stage in the decoder will detect the surround signal by taking the difference of Lt and Rt,
then passing it through a 7 kHz low-pass filter, a delay line, and complementary Dolby noise
reduction. The surround signal will also be reproduced by the left and right speakers, but it will be
heard out-of-phase, which will diffuse the image.
Since the heart of the decoding process is a simple L-R difference amplifier, it is referred to
generically as a "passive" decoder. This is to distinguish it from decoders using active processes
to enhance separation, which are consequently known as "active" decoders.
1.3. Separation Maps
A plot, or map, can be drawn to represent the measured signal separation between any pair of
channels in a matrix encode-decode process. The term "map" is used because of its similarity to
a compass; it uses a circle divided into 360 degrees and has four cardinal points, in this case,
four signal channels. Since the use of the MP Matrix encoder is assumed, the differences in
mapped values will be a function of the decoding process itself. The map in Figs. 4 and 5 (below)
are able to show how the separation between the opposite channels (left to right; center to
surround) compares with the separation between the adjacent channels (left to center; right to
surround, etc.).
It should be realized that even though the electrical signals are isometrically distributed in the
transmission medium as depicted by Fig. 4 (below), the physical arrangement of the speakers in
the listening room compresses three of the four channels across the front, and spreads out the
remaining channel around the sides and rear (see Fig. 3 below) . This concentrates fully half of
the spatial resolution of the system to only about one-fourth of the area of coverage, enabling a
high degree of sound placement accuracy to complement on-screen images.
The map in Fig. 4 (below) shows the measured separation of a four-output passive decoder,
which gives 3 dB of separation between any pair of adjacent channels. However, most passive
decoders omit the center output, depending on the main left and right speakers to create a
"phantom" center signal, which is actually the ideal arrangement for the front channels. To see
why this is so, the system will be analyzed assuming that the listener is positioned equidistant
from the left and right speakers such that monophonic (center channel) signals create a focused
phantom image mid way between them.
The map in Fig. 5 (below) shows the perceived separation for each of the four cardinal input
signals. Given a left-only signal, the sound comes from the left and surround speakers together.
An analogous result occurs with a right-only signal. With a center-only signal, the sound comes
from the left and right speakers but is perceived as coming only from the phantom center position.
Therefore, considering just the front channels, the passive decoder can reproduce left, center,
and right input signals with no perceived loss of separation. This should not be surprising–the
identical psychoacoustic principle is the basis for all two-channel stereophonic reproduction, and
provides the motivation for audiophiles to carefully arrange their listening room seating squarely
in front of the speakers.
Additional techniques described below are employed to increase the perceived separation
between the front and surround channels.
1.4. Controlling the Effects of Channel Crosstalk
The fact that some of the surround signal will continue to come from the left and right channels is
actually of little consequence. One reason is that sounds are expected to come predominantly
from the screen direction, since that is where we see the action. Another reason is that signals
assigned to the surround track usually are not associated with specific source locations. We
might see a lightning bolt on screen, but we'll hear the thunder, rain, and wind all around.
The ability of the surround channel to project its sonic image into the listening room does not rely
on perfect signal isolation. However, that does not mean it is acceptable to allow crosstalk
between the front and rear channels to exist unimpeded. It has been shown that the effects of
crosstalk from the front channels–especially from dialogue in the center channel leaking into the
surround speakers–represents the greatest potential for detracting from the presentation. By
combining the effects of time delay, limited frequency bandwidth, and complementary noise
reduction in the remainder of the surround decoding chain, we can invoke the psychoacoustic
principles of the Haas effect, spectral modification, and signal masking, which together act to
mitigate the effects of such leakage.
Time delay ensures that any front channel sounds that happen to leak from the surround
speakers will arrive at the listener just after the front channel sounds. This will help
prevent the leakage from pulling the sound image away from the screen. Such leakage
may be caused by phase or amplitude differences present in the input signals caused by
azimuth misalignment or frequency response errors, or simple balance errors in the
stereo program.
The 7 kHz low-pass filter is used for several reasons; the main one is that for a given
azimuth error between the two audio channels, the leakage signal magnitude will
increase with frequency, making separation at the high frequencies much more difficult to
achieve. Dialogue sibilance, for example, could become quite strong and distracting from
the surrounds without use of the filter. Reducing the high-frequency content also has the
effect of making the surround speakers sound further away and more difficult to localize,
two characteristics which benefit the persons seated closest to the surround speakers.
Modified Dolby B-type NR is used to reduce noise as well as front channel signal
leakage. The amount of processing was relaxed from 10 dB to 5 dB to prevent the
encoded surround signal from significantly altering the nature of the left and right channel
The shaded line in Fig. 5 (below) represents that the perceived front-to-back separation is
subjectively improved by the techniques just described. Together with the use of a phantomcenter image, the passive decoder is quite capable of producing the intended spatial effects.
Some passive decoders do offer an output for a center speaker intended to improve dialogue
imaging for off-axis viewers. However, this benefit comes at the expense of a narrowed
soundstage since the "passive" center output includes not just the center signal but the left and
right signals as well.
From the above, it is clear that psychoacoustics plays a significant role in the success of Dolby
Surround. This equally true of the Pro Logic process, as discussed next.
Today, A/V systems have taken on new dimensions; televisions with 27- to 35-inch screens are
popular, with a move toward even larger screens and 16:9 "home theatre" aspect ratios
underway. Rear projection sets from 40 to 60 inches are becoming mainstream products, as are
projection systems with 6- to 12-foot screens. Larger screens coupled with increased video
resolution bring the home viewer more of the movie theatre experience, and benefit greatly from
expanded sonic dimensions to balance the presentation. These factors led to Dolby's introduction
of Pro Logic, the second generation in Dolby Surround decoding technology.
Pro Logic is an active process designed to enhance sound localization through the use of highseparation decoding techniques. The system is a direct descendant of the one used in Dolby
Stereo cinema processors, and features a center channel speaker along with the left, right, and
surround outputs.
2.1. Concept of Active Decoding–Directional Enhancement
Passive decoders, as noted earlier, use a simple differential stage to extract the surround signal
from the left and right input signals. The decoders maintain high channel separation across the
front, but localization is proper only within the particular seating area where a phantom-center
image is effective. Furthermore, even with the effects of surround channel processing, it is not
possible to obtain a total degree of isolation from front to back since the surround speakers
reproduce any difference information in the Lt/Rt composite. It is due to these factors that passive
decoders are limited in their ability to place sounds with ultimate precision for all viewing
Directional enhancement is a term referring to any technique that attempts to remove the effects
of the matrix system crosstalk by manipulating the output signals of the decoder. Its goal is to
create sharply focused sound images and to recreate directional cues covering a wide listening
area. An active decoder can most easily be thought of as a passive decoder followed by an
enhancement circuit. To illustrate the concept, we will first describe the simplest technique for
directional enhancement–gain riding. Fig. 6 (below) shows how the gain of each output is varied
with a voltage-controlled amplifier (VCA).
Consider the case where a single sound–dialogue–is present in the center channel (Lt=Rt). The
passive decoding map in Fig. 4 (below) shows that the center output will carry the dialogue, but
so will the left and right outputs only 3 dB down. With control circuits, the decoder can decide to
eliminate the unwanted dialogue leakage by turning down the gain of the left and right speakers,
leaving only the desired center output audible. The same procedure may also be used to isolate
the left channel output by turning down the center and surround outputs when only an Lt input
signal is present. In fact, for an input signal encoded at any position in the 360-degree circle, the
separation can be completely restored with this simple method to equal that of a discrete four
channel system.
2.2. Problems with Gain Riding
Unfortunately, soundtracks are rarely composed of single, isolated sound elements. Consider
how the described gain-riding decoder would work when the dialogue is mixed together with
stereo background music. We would like to hear the music from the left and right speakers and
the dialogue from the center speaker. First, let's examine the passive decoder outputs before
separation enhancement. Just as before, dialogue comes from the center, with dialogue leakage
in the left and right. The stereo music comes from left and right, with music leakage in center
(L+R) and in surround (L-R).
Since dialogue is dominant in the mix, it controls the gain-riding circuit to reduce the dialogue
leakage in the left and right outputs. But this action also cuts off the stereo music, leaving only the
monaural sum in the center channel and the difference signal in the surround channel. When the
dialogue stops, the control circuit will restore the gain in the left and right channel outputs,
allowing the music to be heard in stereo once again. The hope is that when dialogue is present, it
will mask the fact that the music has been mostly removed.
Not only does this fail to work, but consider what happens during the transition when the dialogue
starts and stops: the music coming from the left and right speakers goes off and then back on.
This is quite audible, resulting in "pumping" of the non-dominant sounds (the music) in response
to the dominant sounds (the dialogue). Such gain-riding techniques were used during the
quadraphonic era in some SQ decoders having "logic enhancement" circuitry, and were
considered aesthetically inadequate for musical reproduction because they gave the sound an
unstable character and caused the soundstage to wander. Whatever measurable benefits the
gain-riding decoder made on the test bench were lost in the realities of the listening room.
2.3. The Cancellation Concept
Another way of eliminating dialogue leakage in the left and right outputs is shown in Fig. 7
(below). By taking the right output signal, inverting its polarity and blending it into the left output,
the center signal components–being equal and opposite–cancel each other completely, thus
eliminating the dialogue leakage in the left output. This is the basic cancellation principle, which
can be implemented in several ways depending on the final performance goals of the system.
In this example, though, the music did not emerge unaltered from the left channel after the center
signal was removed; the inverted right signal was mixed with it. The right output will also receive
the inverted left channel signal in the process of canceling its center leakage components. This is
an unavoidable consequence of the cancellation process. Furthermore, there will continue to be
some music from both channels mixed together (L+R) in the center channel output. However, the
fact that the music signal has been redistributed spatially does not mean that its overall level has
been affected, as occurred unavoidably in the gain-riding system. With careful design, it is
possible to maintain constant signal power for all components of the soundtrack as they are
redistributed to the various speakers.
2.4. Constant-Power Concept
As the decoder operates, the dominant sounds are focused to the main point of origin within the
360-degree range, while the non-dominant sounds are redistributed among the remaining
speakers. The principle here is that the dominant sounds will temporarily limit the listener's ability
to detect a change in directionality of the non-dominant sounds. Since no overall change in
loudness occurs for any of the signal components, no pumping or signal modulation will be
The concept of one signal masking another is well documented and effectively used in Dolby
noise reduction systems. In these systems, the dominant signal (the music on the tape recording)
is some 20 to 40 dB greater than the tape noise level in the range where the NR circuit is
dynamically active, but even so, the noise must be handled carefully to prevent audible "noise
modulation." In movie soundtracks, however, the level difference between concurrent sound
elements is often less than 20 dB, and there may be little or no difference in level among them for
sustained periods. Therefore, dominant sounds are much less effective as a tool for masking
level changes in non-dominant sounds, since the non-dominant sounds can be plainly heard.
In the example, we assumed the dialogue was much louder than the background music, thus
calling for maximum action from the decoder. More frequently, however, the differences in sound
levels are not this great. As the sounds become closer in level, each one begins to mask the
crosstalk of the other, concealing the fact that separation is not perfect. When directional
enhancement is then applied, less of it is needed to improve the sense of localization, so less
redistribution of the non-dominant signals takes place.
It may also be desirable to suspend directional enhancement altogether, with the decoder
becoming passive at certain times. For instance, the sounds of rain or wind, which are so
involving on a subliminal level, are intended to come from all speakers simultaneously. This
requires no sound source localization, and hence, no directional enhancement.
By maintaining constant signal power and by applying directional enhancement only as needed to
preserve good localization, we can rely upon a modest degree of masking to be sufficient to cover
the fact that the non-dominant signals have been directionally redistributed.
2.5. Nature of Signal Dominance
A dominant sound is simply that–the sound that is most prominent in the mix at any given instant
in time. It is necessary to be able to sense when a dominant sound occurs because it will have
the greatest influence on the perception of "discreteness" or the effective separation of the
The highest degree of dominance occurs when all sounds are placed in one location. Remember
that in a passive decoder such a signal will be reproduced from its intended output location and,
undesirably, from its adjacent channels. There will be no other sounds present to help mask the
leakage. The condition of a purely dominant signal thus exposes the leakage most clearly, but
this is also a condition where the side-effects of directional enhancement can be most easily and
completely suppressed.
At the other end of the scale, sounds with similar intensities tend to confuse the listener's ability to
pinpoint their individual locations, thus needing little or no directional enhancement. However,
while two different sounds may seem to have the same average loudness, it is likely that, on an
instantaneous basis, one of them will be dominant over the other and that the dominance will
continuously alternate between them. Depending on the peak-to-average ratios of the sounds
over time, it may or may not be necessary to provide directional enhancement.
This suggests that the decoder must include two additional characteristics in order to work
effectively. The decoder first must be fast enough to provide enhancement on an instantaneous
basis between two or more encoded positions when the signal peaks are prominent enough to be
heard as individual events; in effect, time-division multiplexing its action among several individual
sounds occurring in rapid succession. Even though the decoder is essentially providing
directional enhancement for sounds at only one position at a time, all of them are perceived as
being separate from each other. The second characteristic is the ability to sense when the
relative dominance falls below the point where it is no longer necessary for the decoder to provide
any substantial directional enhancement, but, were it to continue its rapid operation, might create
an indistinct or audibly "nervous" soundfield.
For these reasons, the Pro Logic decoder has been designed to sense the level of dominance in
the soundtrack. When dominance is at very low values, the decoder stays in a "slow" yet fully
operative mode to give a stable feeling to the soundfield image. Above a certain point, the
decoder can ideally shift to a high speed mode to quickly process each individual sound element.
2.6. Detecting Soundtrack Dominance
Since it is the relative level of one sound to another which determines the perception of
separation, it is desirable to have sensing circuits that ignore absolute signal level in favor of
being responsive to the difference in level between two signals (the equivalent of taking their
ratio). Electrically, this is no simple task, but, by taking the logarithm of each signal and
subtracting one from the other we, we can obtain a measure of relative dominance.
The resultant control voltage–in this logarithmic form–closely follows the way loudness is
perceived. Consequently, the final process provides separation enhancement directly
corresponding with the amount needed to prevent crosstalk from becoming audible, and
proportional to the ability of the dominant sound to mask the spatial redistribution of the nondominant signals.
2.7. Sensing Direction of Dominance
Knowing which signal is dominant includes knowing the encoded position, or angle, of the signal.
It is in this direction that enhancement must take place, and may encompass any point in a 360degree circle, not just one of the four cardinal positions.
By definition, dominance can only occur in one place at any instant in time; it cannot exist in two
places simultaneously, since their equality of magnitude would mean that neither is dominant.
(These two signals may together constitute a single dominant quantity, however.) Therefore, it is
sufficient to be able to detect a single direction of soundfield dominance, no matter how rapidly
the soundfield changes. With two independent, orthogonal signal pairs available in the encoded
soundtrack (the left/right pair and the center/surround pair), it is possible to identify any point on
an X-Y coordinate plane within a given boundary area.
In Fig. 8 (below), the left/right pair rests on the x-axis, and the center/surround pair is on the yaxis. Resolving the magnitudes of the signals along each axis and converting from rectangular to
polar coordinates then gives the necessary information. Dominance is now shown as a vector
quantity; the angle represents the encoded angle of the dominant sound, and the magnitude
represents its relative dominance.
2.8. Pro Logic Adaptive Matrix
Just as the L-R differential stage is the heart of a passive decoder, the adaptive matrix is the
heart of a Pro Logic decoder. Two main signals go in (Lt and Rt), and four resultant signals
emerge (L, C, R, and S). Compare the block diagram of the Pro Logic decoder in Fig. 9 (below)
with the passive decoder in Fig. 2 (below) . Except for the matrix stage (and center output), they
are virtually the same.
To summarize, Pro Logic operates by continuously monitoring the encoded soundtrack,
evaluating the inherent soundfield dominance, and applying enhancement in the same direction
and in proportion to that dominance. To see how the circuit works, we will examine the adaptive
matrix block diagram in Fig. 10 (below) .
Notice that the adaptive matrix employs two parallel paths: a relatively direct audio path (L and R
inputs go straight to the combining networks) and a complex control path. Most of the decoder's
electronics are used to condition and analyze the input signals rather than to actually process the
audio itself.
As you may guess, the main order of business is to generate the signal dominance vector. The
first step in this process is to condition the incoming signals to prevent decoder errors. This is
done by bandpass filtering the Lt and Rt signals to strip off strong low-frequency signals which do
not provide directional cues, and to attenuate high frequencies that may contain uncertain phase
or amplitude characteristics.
The next step is to determine the magnitudes of the two orthogonal signal pairs, which is done by
first full-wave rectifying each cardinal signal, then subjecting the resulting DC voltages–in pairs–to
log conversion, and finally taking their difference. Two independent control signals are now
available; one represents dominance along the left/right axis, the other along the center/surround
axis. Even though there are only two control voltages at this point, they are of dual polarity, or
bipolar. For instance, when the left/right voltage deflects upward, the dominance is to the left;
when it goes downward, the dominance is to the right. At the midpoint, no dominance is indicated.
Each of these control voltages is evaluated continuously to determine if their relative dominance
exceeds a certain threshold point. If either one does, the control circuit switches to the fast mode
of operation.
A polarity splitter resolves the two bipolar dominance signals into four unipolar control voltages,
called EL, ER, EC, and ES. They now represent the soundtrack dominance in electrical terms
embodying psychoacoustic properties, so are ready to be applied to the signal-canceling VCA
stages. Since there are two input channels (Lt and Rt) and four control voltages, eight VCAs are
used to generate eight variable sub-terms. When added with the Lt and Rt inputs, ten individual
terms are available. To construct a decoded output signal, portions of each of the ten terms are
added or subtracted with a predetermined weighting factor in the combining networks. Selection
of the appropriate magnitudes and polarities for the forty summed components gives the desired
directional enhancement and non-dominant signal redistribution, all the while maintaining
constant acoustic power for the signal components.
2.9. Channel Separation Specifications
In this era of digital audio systems where channel separation may be on the order of 90 dB, it is
useful to try to understand just how much separation is necessary in a surround system. The
matrix encode/decode transmission path is not intended to convey unrelated signals, such as
dialogue in four languages, each meant to be heard by itself. Rather, it will carry sound elements
expressly assembled to be heard together, plus enough information to reproduce a coherent and
controllable soundfield. These are very different requirements.
On the one hand, any degree of channel crosstalk will result in just that–dialogue becoming
audible from one channel while listening to another, something difficult or impossible to ignore.
On the other hand, a surround system is only required to provide enough apparent separation to
create unambiguous directional cues. Furthermore, the crosstalk in a surround system is always
correlated with the main signal of interest. For example, the crosstalk from center channel
dialogue is just this same dialogue leaking from the adjacent channels; it is not some signal
unrelated to the program.
Another point to remember is that the ability for correlated crosstalk to cause a shift in apparent
sound source location diminishes as the angle between the speaker locations reduces. If a sound
is intended to come from one speaker, but it is also reproduced in another speaker placed right
next to it, the sound will still seem to be coming from the correct location. Only as the second
speaker is moved away from the first will the image shift away from the intended position.
From this, it may be concluded that it is important to provide more separation between the
opposing speaker locations rather than the between the adjacent positions, since the greater
spacing increases the possibility for the leakage to disrupt the soundfield. It has already been
shown that the MP Matrix has much higher separation between the opposite channels than the
adjacent channels. The critical surround channel, which is diametrically positioned relative to the
three front channels, has additional signal processing to give an extra degree of separation to
maintain a forward-focused soundfield.
In order to get a feel for how much separation is needed to prevent the correlated crosstalk in a
surround system from disrupting image placement, you can try the following experiment with a
two-speaker stereo system. Select any signal source, music or dialogue, and play it in monaural
from both speakers. While seated in the normal listening position, have someone turn the balance
control completely to the left, so that the right speaker is off. Then, while the control is rotated
slowly toward the center position, note the point where you first hear the right speaker come on,
or where you can detect that the image has moved away from the left speaker. If you then unplug
the left speaker, you will hear how loud the right speaker had to be to make this happen.
Depending on the angle between the speakers and the listener's acuity, the right speaker will be
only about 10 to 20 dB lower than the left one, substantially less than what is needed to prevent
unrelated signals from interfering with each other.
The results of the above experiment correlate with the separation needed between the left and
right channels of a surround system; somewhat less is needed between the center and the left or
right channels since the distance between them is only half as much. The separation map for a
typical Dolby Pro Logic decoder is provided in Fig. 11 (below) , showing that about 30 dB of
separation is available between any pair of channels. It should be noted that the results of the
above experiment represent the separation needed for a purely dominant sound source. The map
figures similarly represent separation measured with a purely dominant, steady-state test tone
The map of a theoretical Dolby Pro Logic decoder is shown in Fig. 12 (below) , which, under
these same test conditions, shows that the worst-case adjacent channel separation reaches 37
dB, and the opposite channel separation is not limited at all. Due to the precision and repeatability
of digital signal processing, these separation figures can be approached in DSP-based Pro Logic
decoders, but as was pointed out earlier, there is no improvement in perceived sound quality
brought by these sometimes dramatic, measurable differences.
2.10. Spectral Partitioning
It is possible to assign certain portions of the audio signal to different speakers or locations
without detrimental side-effects. One popular technique is to use a single woofer to carry the bass
for both channels of a stereo system. Even though the two satellite speakers do not reproduce
the bass, so long as it is present in the room, we are satisfied that the system is reproducing the
entire frequency spectrum. A relatively small center channel speaker can be made to sound much
larger than it actually is by using this same technique, either with a separate woofer or simply by
diverting the bass to full-range left and right speakers. Since it is of fundamental importance to
include the center speaker, a bass-splitting function is included in all Pro Logic decoders.
This principle also applies to the surround speakers. With the bass being well reproduced in the
listening room by the front speakers, there is little value in having the surround speakers attempt
to duplicate it. Furthermore, since the front three speakers are also reproducing the complete
high-frequency spectrum, it will not be readily apparent that the surround speakers are rolling off
above 7 kHz. Only if the surround speakers were playing a wideband signal in total isolation
would their limited frequency coverage be exposed–something that is rarely, if ever, done.
The Dolby Pro Logic Surround decoder system is the final link to bringing soundtracks with
greatly expanded spatial capabilities to the consumer. Due to the parameters of the Dolby MP
Matrix, the encoded soundtracks are fully compatible with two-channel stereo reproduction, and
also give outstanding results with monophonic reproduction. With advent of Pro Logic decoding,
conventional two-channel media can now offer several levels of performance to the consumer, all
from a single version of the soundtrack. The MP Matrix thus utilizes two-channel media more
effectively than any other system, and the Pro Logic decoder provides the means to extract multidimensional properties with an accuracy previously unattainable in consumer equipment.
Time delay is often used to create an echo effect, which can help give music reproduction a
feeling of greater spaciousness. However, that is not why it is used in Dolby Surround decoders.
The real reason is to improve the sense of clarity and directionality of front channel sounds. This
is done by taking advantage of the "Haas" or precedence effect, which enables the main frontal
sound to arrive at your ears before the surround sounds. The time delay stage compensates for
the travel time of sound through the air, which takes about 1 millisecond per foot distance. By
knowing how far the listening position is from the front and the surround speakers, it is possible to
adjust the time delay for optimal results.
There are generally two kinds of time delay available in Dolby Surround products: adjustable, or
unadjustable (fixed delay). As would be expected, the adjustable delay allows a wider range of
distances to be used than the fixed delay.
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