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BACKGROUND OF THE INVENTION
The present invention relates to the processing of information signals and,
more particularly, to techniques for efficiently encoding audio signals,
including signals representative of voice and music.
A significant amount of effort has been directed in recent years to
so-called perceptual audio coding, or PAC. In accordance with this
technique, each of a succession of time domain blocks of an audio signal
is coded in the frequency domain. Specifically, the frequency domain
representation of each block is divided into coder bands, each of which is
individually coded, based on psycho-acoustic criteria, in such a way that
the audio signal is significantly "compressed," meaning that the number of
bits required to represent the audio signal is significantly less than
would be the case if the audio signal were represented in a more
simplistic digital format, such as in the form of PCM words.
When the audio signal comprises two or more input channels, such as the
left and right channels of stereophonic (stereo) music, the
above-described perceptual coding is carried out on a like number of
so-called matrixed channels. In the most straightforward implementation,
each matrixed channel is directly derived from a respective input channel.
Thus in the stereo music case, for example, this would mean that the
perceptual coding codes the frequency domain representation of the left
stereo input channel over time, denoted herein as "L", and, separately,
the frequency domain representation of the right stereo input channel over
time, denoted herein as "R". However, further compression can be achieved
when the input channels are highly correlated with one another--as,
indeed, is almost always the case with stereo music channels--by switching
the coding carried out for each coder band between two coding modes in
which different sets of matrixed channels are used. In one of the modes,
the set of two matrixed channels simply comprises the input channels L and
R. In the other mode, the set of two matrixed channels comprises S=(L+R)/2
and D=(L-R)/2. The S and D channels are referred to as sum/difference
channels. This technique is taught in U.S. patent application Ser. No.
07/844,804 entitled "Method and Apparatus for Coding Audio Signals Based
on a Perceptual Model" filed Mar. 2, 1992, allowed Aug. 11, 1993, now U.S.
Pat. No. 5,285,498 issued Feb. 8, 1994, to J. D. Johnston, hereinafter
referred to as "the Johnston patent," and hereby incorporated by
reference.
A problem arises in PAC systems when the PAC coder output bit rate over
some period of time is sufficiently higher than average that the capacity
of the system to buffer those bits threatens to be completely used up.
This situation is referred to as "bit reservoir depletion" and, if not
dealt with, can lead to the loss of PAC-encoded information and, thus,
distortion in the decoded signal. The prior art deals with this problem by
increasing, by a multiplicative factor, the so-called noise thresholds
used to encode at least a particular coder band. These thresholds
determine the coarseness or fineness with which the PAC data within a band
are quantized. Thus increasing the thresholds lowers the number of bits
that are used to represent the matrixed channels and therefore ameliorates
the bit reservoir depletion problem.
SUMMARY OF THE INVENTION
While the above-described prior art approach is effective in dealing with
bit reservoir depletion, it can lead to severe artifacts in the decoded
signal since the mechanism for increasing the thresholds is deterministic
across the coder bands rather than being, as we have realized is
advantageous, based on psycho-acoustic considerations.
The present invention is directed to an approach to the bit reservoir
depletion issue which takes psycho-acoustic considerations into account.
Specifically, when the level of bit depletion reaches a predetermined
level, any threshold specified by the PAC encoding for a particular coder
band which is less than a particular lower bound defined for that band is
increased so as to be equal to that lower bound. The lower bound, in turn,
is a particular percentage of a "global masking threshold" determined for
each coder band.
If at a subsequent time it is observed that the bit reservoir reaches an
even more severe state of depletion, a higher percentage of the global
masking threshold is established as the lower bound, causing a further
increase in various ones of the thresholds. If, ultimately, the lower
bound needs to be established at 100 percent of the full global masking
threshold and bit reservoir depletion still continues, each of the
thresholds is further increased to a value given by a constant >1 times
the relevant global masking threshold value. Once the bit reservoir begins
to rebuild, lower and lower percentages of the global masking threshold
are used for determining the noise threshold lower bounds until,
ultimately, the thresholds are returned to their standard values.
In preferred embodiments, the value of the global masking threshold for
each coder band is taken to be the maximum of all of the noise thresholds
applicable to that band minus a safety margin. The safety margin, in turn,
is taken to be the so-called frequency dependent binaural masking level
difference, or MLD, as defined in the Johnston patent, plus a constant 4-5
dB.
BRIEF DESCRIPTION OF THE DRAWING
FIG. 1 is a block diagram of a system in which the present invention is
illustratively implemented;
FIG. 2 is a block diagram of .the perceptual audio coder used in the system
of FIG. 1;
FIG. 3 is a flowchart of a process performed within the perceptual audio
coder; the invention;
FIGS. 4-6 illustrate the principles of the present invention;
FIG. 7 shows the format of the PAC-encoded audio; and
FIG. 8 is a block diagram of the perceptual audio decoder used in the
system of FIG. 1.
DETAILED DESCRIPTION
To simplify the present disclosure, the following patents, patent
applications and publications are hereby incorporated by reference in the
present disclosure as if fully set forth herein: U.S. Pat. No. 5,040,217,
issued Aug. 13, 1991 by K. Brandenburg et al; U.S. patent application Ser.
No. 07/292,598, entitled Perceptual Coding of Audio Signals, filed Dec.
30, 1988; J. D. Johnston, Transform Coding of Audio Signals Using
Perceptual Noise Criteria, IEEE Journal on Selected Areas in
Communications, Vol. 6, No. 2 (February 1988); International Patent
Application (PCT) WO 88/01811, filed Mar. 10, 1988; U.S. patent
application Ser. No. 07/491,373, entitled Hybrid Perceptual Coding, filed
Mar. 9, 1990, Brandenburg et al, Aspec: Adaptive Spectral Entropy Coding
of High Quality Music Signals, AES 90th Convention (1991); Johnston, J.,
Estimation of Perceptual Entropy Using Noise Masking Criteria, ICASSP,
(1988); J. D. Johnston, Perceptual Transform Coding of Wideband Stereo
Signals, ICASSP (1989); E. F. Schroeder and J. J. Platte, "`MSC`: Stereo
Audio Coding with CD-Quality and 256 kBIT/SEC," IEEE Trans. on Consumer
Electronics, Vol. CE-33, No. 4, November 1987; and Johnston, Transform
Coding of Audio Signals Using Noise Criteria, Vol. 6, No. 2, IEEE J.S.C.A.
(February 1988).
For clarity of explanation, the illustrative embodiment of the present
invention is presented as comprising individual functional blocks
(including functional blocks labeled as "processors"). The functions these
blocks represent may be provided through the use of either shared or
dedicated hardware, including, but not limited to, hardware capable of
executing software. Illustrative embodiments may comprise digital signal
processor (DSP) hardware, and software performing the operations discussed
below. Very large scale integration (VLSI) hardware embodiments of the
present invention, as well as hybrid DSP/VLSI embodiments, may also be
provided.
FIG. 1 is an overall block diagram of a system in which the present
invention is implemented. In FIG. 1, an analog audio signal on lead 101 is
fed into a preprocessor 102 where it is sampled (typically at 48 KHz) and
converted into a 16-bit-per-sample digital pulse code modulation ("PCM")
signal on lead 103 in standard fashion. The PCM signal is fed into a
perceptual audio coder ("PAC") 104 which compresses the PCM signal and
outputs the compressed PAC signal on lead 105 to either a communications
channel or a storage medium 106. The latter may be, for example, a
magnetic tape, compact disc or other storage medium. From the
communications channel or the storage medium the compressed PAC-encoded
signal on lead 107 is fed into a perceptual audio decoder 108 which
decompresses the compressed PAC-encoded signal and outputs a PCM signal on
lead 109 which is a digital representation of the original analog signal.
From the perceptual audio decoder, the PCM signal on lead 108 is fed into
a post-processor 110 which creates an analog representation.
An illustrative embodiment of the perceptual audio coder 104 is shown in
block diagram form in FIG. 2. The perceptual audio coder of FIG. 2 may
advantageously be viewed as comprising an analysis filter bank 202, a
perceptual model processor 204, a composite coder 205, a
quantizer/rate-loop processor 206 and an entropy encoder 208.
The structure and operation of the various components of perceptual audio
coder 104 are generally similar to the structure and operation of like
components in FIG. 2 of the Johnston patent when processing stereo
signals, and thus will not be described in detail herein except to the
extent necessary for an explication of the present invention. This will
include a description of composite coder 205, for which no explicit
counterpart is shown in FIG. 2 of the Johnston patent (although its
functionality--relating to the switching between coding modes--is
implicitly carried out in the Johnston patent within analysis filter bank
202).
Turning, then, to FIG. 2 hereof, the analog audio input signal on lead 103
is illustratively a five-channel signal comprising, in the time domain,
PCM samples of a set of input channels, those being left, right and center
front channels and left surround and right surround back channels, denoted
l(t), r(t), c(t), ls(t) and rs(t), respectively. Analysis filter bank 202
receives those samples and divides them into time domain blocks. More
specifically, filter bank 202 switches between two window lengths for the
blocks--a "short" window comprising 128 time samples and a "long" window
comprising 1024 time samples. For each block, filter bank 202 performs a
Modified Discrete Cosine Transform on each of the five channels separately
to provide a frequency domain representation of each channel for the block
in question. The frequency domain representation of each channel includes
1024 uniformly spaced frequency spectrum lines divided into 49 coder bands
for the long-window-length blocks and 128 uniformly spaced frequency
spectrum lines divided into 14 coder bands for the short-window-length
blocks. The frequency domain representations of the left, right, center,
left surround and right surround input channels are denoted in the FIG. as
L(f), R(f), C(f), LS(f) and RS(f), but, for convenience, will be
hereinafter denoted as L, R, C, LS and RS.
The outputs of analysis filter bank 202 are applied to composite coder 205
which in a manner described in detail below produces five matrixed
channels M.sub.1 (f)-M.sub.5 (f) which are applied to quantizer/rate loop
processor 206. The latter encodes the matrixed channels by a) generating
binary data representing the signed magnitude of each of the frequency
spectrum lines of each matrixed channel for a given block and b)
formatting that data along with other information needed by the decoder.
That data includes various pieces of "housekeeping" data as described in
the Johnston patent and also hereinbelow, as well as data including, for
example, an indication of what coding mode was used to encode each coder
band in a given block, as described below, as well as the values of
certain prediction coefficients also described below.
The output of quantizer/rate loop processor 206 is applied to entropy
encoder 208. The latter operates in conjunction with the former to achieve
yet further compression.
As noted above, the output of composite coder 205 comprises five matrixed
channels. Composite coder 205 has a number of coding modes each
characterized by a different set of matrixed channels, the different
coding modes being invoked individually for each coder band in a manner to
be described. Some of the matrixed channels in some of the modes are the
input channels L, R, C, LS and RS. Others of of the matrixed channels are
the so-called sum/difference channels S=(L+R)/2, D=(L-R)/2, SS=(LS+RS)/2
and SD=(LS-RS)/2.
Moreover, at least one of the modes includes at least one matrixed channel
given by an input channel or sum/difference channel from which has been
subtracted a prediction of itself. In the present illustrative embodiment,
six modes are provided for coding the front channels. Three, involving the
front input channels L, R and C directly, are
##EQU1##
The other three, involving the front sum/difference channels S and D, are
##EQU2##
Eight modes are provided for the back channels. Four, involving the back
channels LS and RS directly are
##EQU3##
The other four, involving the back sum/difference channels SS and DS are
##EQU4##
In the above coding modes, the terms that are subtracted from the input and
sum/difference channels L, R, C, S, D, LS, RS SS and DS are, indeed, the
predicted values thereof. More specifically, as can be seen, the
prediction for a particular input channel or sum/difference channel is
derived from at least one other, "predicting" channel. For the front
channels, C is illustratively used as a predicting channel for L, R, S and
D, while L and R are used jointly as predicting channels for C. For the
back channels, all three front channels are illustratively used as
predicting channels for LS, RS, SS and SD, either by themselves or in
various combinations. The " " over the predicting channels, e.g., C,
denotes that the predictions are based on the encoded values of the
predicting channels--generated by quantizer/rate loop processor 206 and
fed back to composite coder 205 over lead 216--rather than their actual
values. The reasoning behind this is as follows: Firstly, the encoded
predicting channel serves as virtually as good a predictor as the
unencoded channel so there is no harm done by doing this. It is noted,
however, that the predicted value of each encoded channel has to be added
back in the decoder. That is, L is recovered from the transmitted
L-.alpha.C by adding .alpha.C to that which is received. It is only C, and
not C, that is available in the decoder. We are thus able to add back to
the received coded channel in the decoder that which was subtracted from
it in the coder. If the unencoded predicting channel were used in the
coder, a quantization noise artifact would, disadvantageously, be
introduced in the decoder.
The scalar prediction coefficients .alpha., .beta., etc may be computed
using a variety of different criteria. In the simplest implementation,
these may all be set to "1" or some other constant less than "1". Indeed,
some of those coefficients could be set to zero if the prediction of which
they are a part is expected to be very small. This would be the case, for
example, for the the prediction of a difference matrixed channel, such as
D or DS, because those matrixed channels are, themselves, expected to have
near-zero values.
In a slightly more complex system, the prediction coefficients may be
computed for each block (but only one set of coefficients being computed
for all coder bands), using a minimum mean squared error (mmse) or a
perceptually weighted mmse criterion, and transmitted as part of the
bitstream that is output by quantizer/rate loop processor 206 (the
composite coding decision still being independent for each of the coder
bands). The reason that the coefficients are denoted with a " ", e.g.,
.alpha., is that if their values are computed, rather than being
constants, those values are necessarily coded, i.e., quantized, when
stored or transmitted in digital form, the " " notation being indicative
of this. In a yet more complex system, the prediction coefficients are
estimated separately for each of the coder bands. Such a scheme is
attractive when prediction gain is low because of time delays between
different channels. However, transmitting prediction coefficients for each
of the coder bands can be quite expensive. Fortunately, experiments
suggest that coefficients from the previous block may be used with a
relatively small loss in prediction gain. The prediction coefficients in
such a system can therefore be computed in a backward fashion from the
decoded values of the previous block both at the encoder and the decoder.
As seen from (1), (2), (3) and (4) above, sum/difference channels are
formed only between pairs of front channels or pairs of back channels.
More complex combinations, or "basis transformations",--involving perhaps
three or more input channels and/or involving, perhaps, sums and
differences between front and back channels--might prove advantageous from
a compression point of view. Precluding such more complex basis
transformations, however, is preferred because we then ensure that
quantization noise masking--which is a driving principle of perceptual
coding--is effective not only in a five-speaker listening room environment
but also in the case of a so-called stereo downmix of the five input
channels into two channels for headphone reproduction, for example.
Moreover, the more complex basis transformations will usually require the
use of lower noise thresholds (discussed below) for encoding, thereby
reducing the amount of compression that is achieved.
We now address the questions of a) how the encoding of the five matrixed
channels of any particular coding mode is carried out, and b) how the
decision is made as to which mode is to be used in order to encode a
particular coder band for a given block.
In particular, the Johnston patent describes how, for each block, the
perceptual model processor generates for each coder band a noise threshold
for each matrixed channel, that threshold being a critical parameter for
the quantizer/rate loop processor in its encoding of the respective
matrixed channel. For each coder band for a two-channel system, then, four
noise thresholds are available, one each for L, R, S and D. The noise
thresholds are supplied by perceptual model processor 204 to composite
coder 205 via lead 214. When L and R are the encoded channels for a
particular coder band during one coding mode of the Johnston patent, their
respective thresholds are used to encode those channels. Similarly, when S
and D are the encoded channels during the other coding mode, their
respective thresholds are used to encode those channels.
In the present illustrative embodiment, nine thresholds per coder band are
available. They are the thresholds corresponding to the five input
channels L, R, C, LS and RS and the four sum/difference channels S, D, SS
and DS. Obviously, when the selected coding mode includes an input channel
or sum/difference channel as one of the matrixed channels, the
corresponding threshold is used. Additionally, the threshold associated
with a particular input or sum/difference channel is to be used to encode
each predicted version of that channel. For example, the threshold for L
is also used for the matrixed channel L-.alpha.C. The manner in which the
nine thresholds are generated for each coder band during each block is a
straightforward application of the techniques described in the Johnston
patent for the generation of its four thresholds and thus further
elucidation on this point is not needed.
The manner in which a particular coding mode is selected is illustrated by
the flowchart of FIG. 3. The flowchart represents the processing carried
out in this regard for the front channels and for a particular coder band.
Similar processing is carried out individually for both the front and back
channels for each of the coder bands. Specifically, the items in ›! relate
to the processing carried out for the back channels.
Initially, the aforementioned nine thresholds are generated by perceptual
model processor 204 (block 301). A decision is then made in composite
coder 205 as to whether the coding mode for the front channels should be
of a type that involves input channels or sum/difference channels, i.e.,
one of modes (1) or one of modes (2). One criterion that can be employed
in making this decision is to compare the thresholds for L and R (block
303). If they differ by more than a predetermined amount, e.g., 2 dB,
input channel coding is used i.e., one of the three modes in (1) (block
311). If they do not differ by more than the predetermined amount, one
approach, which is not implemented here is to immediately choose
sum/difference coding i.e., one of the three modes in (2). Here, however,
a more sophisticated approach is used. In particular, we recognize that
the use of sum/difference matrixed channels is desirable when L and R are
highly correlated not only a) because a high degree of compression will be
achieved in that case, but also b) because it will control so-called noise
localization. Noise localization control can also be achieved, however, by
suitably lowering the thresholds for L and R and, as it turns out,
encoding L and R with those lowered thresholds sometimes requires fewer
bits than encoding S and D. Thus one can use, for example, a "perceptual
entropy" criterion as taught in the prior art to determine which
approach--coding L and R with lowered thresholds or encoding S and D--will
require the fewer number of bits. Once the coding mode has been narrowed
to being from (1) or (2) (block 307), the particular coding mode to be
used is selected simply by, again, using the aforementioned perceptual
entropy criterion to identify the mode that requires the smallest number
of bits (block 309 or block 314). A similar process is carried out
vis-a-vis the back channels and an indication of which coding mode was
used for both the front and back channels is stored or transmitted along
with the encoded channels themselves.
The present invention relates to the generation of the thresholds for the
five input channels and the four sum/difference channels. The invention,
in particular, uses a so-called global masking threshold to take advantage
of the masking ability of the signal component in the matrixed channel
whose signal component is the strongest to mask the noise in the other
matrixed channels.
In order to explain the invention, we begin by considering that in coder
104, as in the coders known in the prior art, a so-called "bit reservoir"
is maintained which is basically a count of excess channel capacity
measured in bits. That count, in essence, is a measure of the difference
between a) the number of bit transmission slots available in the past at
the average output bit rate and b) the number of bits that were actually
encoded. The maximum size of the count in the bit reservoir depends on the
amount of buffering (latency) allowed in the system. This unused capacity
can be used to deal with the fact that future blocks may require a
higher-than-average number of bits to represent the content of those
blocks. Thus a relatively constant output bit rate can be supported even
though the number of encoded bits generated for each block varies from
block to block and will, in general, be sometimes higher than that rate
(on a per/second averaged basis) and sometimes lower. Typically, the bit
reservoir capacity is five times the average bit rate per block.
If the bit reservoir reaches its maximum capacity, this is an indication
that the bit requirement has been consistently lower than the average
output bit rate--so much so that it is advantageous to use up such excess
capacity by lowering the noise thresholds, thereby performing a finer
quantization. This is advantageous in that it provides better quality
reproduction. It is, however, optional, the alternative being to simply
send mark, or other non-information-bearing bits.
A technique is, however, definitely required to deal with the case when the
bit reservoir becomes empty because at that point, unless remedial steps
are taken, the buffering capacity of the system will have been all used up
and encoded bits will begin to be lost. The prior art deals with this
problem by switching to a coarser quantization once a) the bit reservoir
is depleted, and b) in addition, the bit requirement for the current block
exceeds the average output bit rate per block at that time.
Implementationally, this is effected by iteratively multiplying each of
the noise thresholds by a series of increasing constant values greater
than 1 and recalculating the bit requirement until such time as the bit
requirement for the current block can be accommodated. Once the bit
requirements of future blocks goes below the average rate, the normal
threshold values can again be used and the bit reservoir will begin to
replenish.
While this prior art approach is effective in dealing with the bit
reservoir depletion issue, it can lead to severe artifacts in the decoded
signal since the mechanism for increasing the thresholds is deterministic
across the coder bands rather than being, as we have realized is
advantageous, based on psycho-acoustic considerations.
In accordance with the present invention, an approach to the bit reservoir
depletion which takes psycho-acoustic considerations into account is used.
That approach establishes a "global masking threshold" for each coder band
which is constant across all five matrixed channels. The value of the
global masking threshold is computed as described below. For the present
it suffices to indicate that it represents the maximum noise level in any
of the channels that will not be noticed by a listener in a listening room
environment.
The manner in which the global masking threshold is used to control the
noise threshold values in the event of imminent bit reservoir depletion is
illustrated in FIGS. 4-6. FIG. 4, in particular, shows the threshold level
for, illustratively, the first four coder bands of L, as well as the
global masking threshold established for each of the bands. (A similar
representation can be made for each of the other channels.) It is assumed
that at the point in time represented by FIG. 4, the bit reservoir is just
below a level of 80% depletion. Thus the standard threshold values are
used. As soon as the depletion level reaches 80%, however, a lower bound
for each of the thresholds of, illustratively, 50% of the global masking
threshold is employed so that any threshold which is less than that lower
bound is increased so as to be equal to it. Since a different global
masking threshold is established for each coder band and the noise
threshold is different for each band, this means that some of the
thresholds will be increased more than others and some may not be
increased at all. This is illustrated in FIG. 5 where it can be seen that
the threshold for coder bands 1 and 3 have not been increased while those
for coder bands 2 and 4 have. This approach is advantageous in that
although additional noise is introduced as a consequence of increasing the
thresholds, such additional noise is introduced in coder bands in which
the possibility of it being detected by the listener is the lowest. The
operative mechanism here is explained at a more opportune point
hereinbelow.
If at a subsequent time it is observed that the bit reservoir reaches an
even more severe state of depletion, a higher percentage of the global
masking threshold, e.g. 75%, is established as the lower bound, causing a
further increase in various ones of the thresholds. If, ultimately, the
lower bound needs to be established at the full global masking threshold
and bit reservoir depletion still continues, each of the thresholds is
further increased to a value given by a constant >1 times the relevant
global masking threshold value, thereby continuing to implement
psycho-acoustic considerations in the setting of the thresholds-at least
to some extent. This is illustrated in FIG. 6. Once the bit reservoir
begins to rebuild, lower and lower percentages of the global masking
threshold come into effect for determining the noise threshold lower
bounds until, at the 80% point, the thresholds return to their standard
values.
The actual value of the global masking threshold for each coder band is
taken to be the maximum of the five input channel thresholds for that band
minus a safety margin. The safety margin, in turn, is taken to be the
frequency dependent binaural masking level difference, or MLD, as defined
in the Johnston patent, plus a constant 4-5 dB.
Given that the value of the global masking threshold is determined in this
way, we are now in a position to understand why the approach described
above is effective. What is happening here is that the technique takes
advantage of the masking ability of the signal component in the matrixed
channel whose signal component is the strongest to mask the noise in the
other matrixed channels.
Them are at least two reasons why only a percentage of the global masking
threshold is initially used to establish the noise threshold lower bounds,
rather than the full value (unless ultimately needed as described above).
One reason is that if a full global threshold is used, then the noise may
not be fully masked for all listeners in a listening room
environment--particularly listeners who are close to the speakers. The
other is that the probability that noise in the aforementioned stereo
downmixed channels will not be fully masked increases with increasing
percentage of the global masking threshold used to establish the
aforementioned lower bounds.
FIG. 7 shows a portion of a storage medium 700--illustratively magnetic
tape--on which PAC-encoded data is stored and from which it can be
subsequently read, decoded and presented to a listener as discussed above
in connection with FIG. 1. The data is stored in frames . . . , F.sub.i-1,
F.sub.i, F.sub.i+1, . . . , each corresponding to one block of the input
signal. The frames are stored sequentially on the storage medium and
conform to a predetermined format that is very similar to that taught in
the prior art for two-channel PAC. By way of example, the entirety of
frame F.sub.i is explicitly shown in FIG. 7. It has the following fields:
sync word 701 defining the start of the frame; channels flag 702
indicating the number of channels in the input signal--in this example,
five; window type flag 704 indicating whether the frame represents a long
or short window; coding mode flags 706 which indicate which of the
fourteen encoding modes was used to encode the block in question in each
of the coder bands, flags 706 being represented in Huffman coded form
using a predefined codebook; dc values 709 specifying a respective dc
value for each of the five matrixed channels; predictor coefficient flag
710 whose value, if "1", indicates that the prediction coefficients are
explicitly specified within the frame, in which case they are provided in
prediction coefficient field 713 and, if "0", indicates either that all
the prediction coefficients have the value 1.0 or that they are to be
computed in the decoder in the manner described hereinabove, the choice
between these two options being predetermined when the system is designed;
and PAC data fields 714-718 each containing the encoded data for a
respective one of the (in this case) five matrixed channels, M.sub.1 (f) .
. . M.sub.5 (f). As in prior art, two-channel PAC systems, the PAC-encoded
data in each one of fields 714-718 is, in turn, represented in
Huffman-coded form. Various Huffman code codebooks are used to encode the
data for the various coder bands within each channel and therefore the
data in each of fields 714-718 also includes information identifying which
codebooks were used to encode which coder bands of the matrixed channel in
question.
It will be appreciated that the format shown in FIG. 7 could equally as
well be used for storing the PAC frames in other types of storage media,
such as compact disc, optical disc, semiconductor memory, etc.
FIG. 8 is an illustrative embodiment of PAC decoder 109 of FIG. 1. The
incoming bit stream, formatted as a sequence of frames as shown in FIG. 7,
is parsed into its various components by bit stream parser 801. Although
not shown explicitly in FIG. 8, parser 801 not only presents the
PAC-encoded data to the next functional block in the decoder--entropy
decoder 804--but also supplies the various flags and other values
described above in connection with FIG. 7 to each of the various
components of the decoder that need them. (In similar fashion, it will be
appreciated that although not explicitly shown in in FIG. 2, the flags and
other values that may have been generated by various components of encoder
104, to the extent that they are not generated within quantizer/rate loop
processor 106 itself, are supplied thereto so as to be able to be
incorporated into the format of each frame.)
Continuing with the processing carried out by decoder 109, entropy decoder
804 performs the inverse function of entropy coder 208 and provides as its
outputs the five matrixed channels M.sub.1 (f) . . . M.sub.5 (f).
Dematrixer 807 recovers from the matrixed channels the frequency domain
input channels L, R, C, LS and RS, which are denoted, as in FIG. 2, as
L(f), R(f), C(f), LS(f) and RS(f). The latter are then processed by
Inverse Modified Discrete Cosine Transform (Inverse MDCT) processor 808 to
produce the five time domain channels l(t), r(t), c(t), ls(t) and rs(t).
The processing performed by each of the elements shown in FIG. 8 as just
described can be implemented straightforwardly and thus needs not be
described in further detail. Indeed that processing is very similar to
that carried out in the prior art for two-channel PAC.
The foregoing merely illustrates the principles of the present invention.
Those skilled in the art will be able to devise numerous alternative
arrangements which, although not explicitly shown or described herein,
embody those principles and are thus within the spirit and scope of the
invention.
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