 |
Claims  |
|
|
We claim:
1. A method of perceptually encoding an audio signal comprising a set of
input channels, the method comprising the steps of
generating a set of matrixed channels in response to said input channels,
and
perceptually encoding said matrixed channels, said perceptually encoding
step including selecting noise threshold values based upon noise masking
criteria and using said noise threshold values to control the coarseness
of quantizing said audio signal during said encoding;
at least an individual one of the matrixed channels of said set of matrixed
channels being a function of a) an individual one of said input channels
or the sum of, or the difference between, two of them, and b) a prediction
of a).
2. A method of perceptually encoding an audio signal comprising a set of
input channels, the method comprising the steps of
generating a set of matrixed channels in response to said input channels.
perceptually encoding said matrixed channels, at least an individual one of
the matrixed channels of said set of matrixed channels being a function of
a) an individual one of said input channels or the sum of, or the
difference between, two of them, and b) a prediction of a), and
alternatively applying to an output a) said perceptually encoded matrixed
channels and b) a perceptual encoding of said input channels.
3. A method of perceptually encoding an audio signal comprising a set of
input channels, the method comprising the steps of
generating a set of matrixed channels in response to said input channels.
perceptually encoding said matrixed channels, at least an individual one of
the matrixed channels of said set of matrixed channels being a function of
a) an individual one of said input channels or the sum of, or the
difference between, two of them, and b) a prediction of a), and
alternatively applying to a communication channel or a storage medium a)
said perceptually encoded matrixed channels and b) a perceptual encoding
of said input channels.
4. A method of perceptually encoding an audio signal comprising a set of
input channels, the method comprising the steps of
generating a set of matrixed channels in response to said input channels,
and
perceptually encoding said matrixed channels, at least an individual one of
the matrixed channels of said set of matrixed channels being a function of
a) an individual one of said input channels or the sum of, or the
difference between, two of them, and b) a prediction of a),
wherein in said audio signal, said input channels are represented in the
frequency domain and wherein said generating step comprises the step of
generating said set of matrixed channels from said input channels in such
a way that each of said matrixed channels is represented, for each of a
series of time domain blocks, by frequency spectrum lines.
5. The invention of claim 4 wherein said frequency spectrum lines are
divided into a plurality of coder bands and wherein the magnitudes of the
frequency spectrum lines of each coder band are represented by values that
are quantized as a function of a noise threshold associated with that
coder band.
6. Apparatus for processing an audio signal comprising a set of input
channels, said apparatus comprising
means for perceptually encoding one of a predetermined plurality of sets of
matrixed channels generated in response to said input channels, said
perceptually encoding means including selecting noise threshold values
based upon noise masking criteria and using said noise threshold values to
control the coarseness of quantizing said audio signal during said
encoding, the channels of an individual one of said sets of matrixed
channels being said input channels and the channels of others of said sets
of matrixed channels including at least an individual one sum of, or the
difference between, two of them, and b) a prediction of a), and
means for applying the perceptually encoded matrixed channels to a selected
one of a) a communications channel, and b) a storage medium,
7. Apparatus for processing an audio signal comprising a set of input
channels, said apparatus comprising
means for perceptually encoding one of a predetermined plurality of sets of
matrixed channels generated in response to said input channels, the
channels of an individual one of said sets of matrixed channels being said
input channels and the channels of others of said sets of matrixed
channels including at least an individual one sum of, or the difference
between, two of them, and b) a prediction of a), and
means for applying the perceptually encoded matrixed channels to a selected
one of a) a communications channel, and b) a storage medium,
wherein in said audio signal, said input channels are represented for each
of a series of time domain blocks, by frequency spectrum lines divided
into a plurality of coder bands, the magnitudes of the frequency spectrum
lines of each coder band being represented by values that are quantized as
a function of a noise threshold associated with that coder band.
8. A method for processing a perceptually encoded audio signal, said
perceptually encoded audio signal having been generated by generating a
set of matrixed channels in response to a set of input channels;
perceptually encoding said matrixed channels wherein said perceptually
encoding includes selecting noise threshold values based upon noise
masking criteria and using said noise threshold values to control the
coarseness of quantizing said audio signal during said encoding; and
applying said perceptually encoded matrixed channels to a communication
channel or a storage medium, said set of matrixed channels comprising a
selected one of i) said input channels, and ii) a set of matrixed channels
in which at least an individual one of the matrixed channels is a function
of a) an individual one of said input channels or the sum of; or the
difference between, two of them, and b) a prediction of a),
said method comprising the steps of
receiving said perceptually encoded matrixed channels from said
communications channel or storage medium,
decoding the received perceptually encoded matrixed channels, and
recovering said input channels from the decoded matrixed channels.
9. A method for processing a perceptually encoded audio signal, said
perceptually encoded audio signal having been generated by generating a
set of mat fixed channels in response to a set of input channels;
perceptually encoding said matrixed channels; and applying said
perceptually encoded matrixed channels to a communication channel or a
storage medium, said set of matrixed channels comprising a selected one of
i) said input channels, and ii) a set of matrixed channels in which at
least an individual one of the matrixed channels is a function of a) an
individual one of said input channels or the sum of, or the difference
between, two of them, and b) a prediction of a),
said method comprising the steps of
receiving said perceptually encoded matrixed channels from said
communications channel or storage medium,
decoding the received perceptually encoded matrixed channels, and
recovering said input channels from the decoded matrixed channels and
determining how said set of matrixed channels was generated in response to
said set of input channels.
10. Apparatus for processing a perceptually encoded audio signal, said
perceptually encoded audio signal having been generated by generating a
set of matrixed channels in response to a set of input channels;
perceptually encoding said matrixed channels wherein said perceptually
encoding includes selecting noise threshold values based upon noise
masking criteria and using said noise threshold values to control the
coarseness of quantizing said audio signal during said encoding; and
applying the perceptually encoded matrixed channels to a communication
channel or a storage medium; at least an individual one of the matrixed
channels of said set of matrixed channels being a function of a) an
individual one of said input channels or the sum of, or the difference
between, two of them, and b) a prediction of a),
said apparatus comprising
means for receiving said perceptually encoded matrixed channels from said
communications channel or storage medium,
means for decoding the received perceptually encoded matrixed channels, and
means for recovering said input channels from the decoded matrixed
channels.
11. Apparatus for processing a perceptually encoded audio signal, said
perceptually encoded audio signal having been generated by generating a
set of matrixed channels in response to a set of input channels;
perceptually encoding said matrixed channels; and applying the
perceptually encoded matrixed channels to a communication channel or a
storage medium; at least an individual one of the matrixed channels of
said set of matrixed channels being a function of a) an individual one of
said input channels or the sum of, or the difference between, two of them,
and b) a prediction of a),
said apparatus comprising
means for receiving said perceptually encoded matrixed channels from said
communications channel or storage medium,
means for decoding the received perceptually encoded matrixed channels, and
means for recovering said input channels from the decoded matrixed channels
and determining how said set of matrixed channels was generated in
response to said set of input channels.
12. Apparatus in which is stored information representing a perceptually
encoded audio signal comprising a set of input channels, said perceptually
encoded audio signal having been generated by the steps of
generating a set of matrixed channels in response to said input channels,
and
perceptually encoding said matrixed channels, said perceptually encoding
step including selecting noise threshold values based upon noise masking
criteria and using said noise threshold values to control the coarseness
of quantizing said audio signal during said encoding,
at least an individual one of the matrixed channels of said set of matrixed
channels being a function of a) an individual one of said input channels
or the sum of, or the difference between, two of them, and b) a prediction
of a).
13. Apparatus in which is stored information representing a perceptually
encoded audio signal comprising a set of input channels, said perceptually
encoded audio signal having been generated by the steps of
generating a set of matrixed channels in response to said input channels,
and
perceptually encoding said input channels alternatively with said matrixed
channels,
at least an individual one of the matrixed channels of said set of matrixed
channels being a function,of a) an individual one of said input channels
or the sum of, or the difference between, two of them, and b) a prediction
of a).
14. The invention of claim 1 or claim 6 or claim 8 or claim 10 or claim 12
wherein said individual one of said matrixed channels is a function of the
difference between a) and b).
15. The invention of claim 2 wherein said set of input channels includes
left, right and center channels.
16. The invention of claim 156 wherein a first one of said matrixed
channels is a function of said left channel and a prediction of said left
channel; wherein a second one of said matrixed channels is a function of
said right channel and a prediction of said right channel; and wherein a
third one of said matrixed channels is said center channel; each said
prediction being a function of said center channel.
17. The invention of claim 16 wherein each said prediction is a function of
a perceptually encoded version of said center channel.
18. The invention of claim 16 wherein each said prediction is a function of
the product of a) a perceptually encoded version of said center channel,
with b) a respective prediction coefficient.
19. The invention of claim 15 wherein a first one of said matrixed channels
is said left channel; wherein a second one of said matrixed channels is
said right channel; and wherein a third one of said matrixed channels is a
function of said center channel and a prediction of said center channel,
that prediction being a function of said left and right channels.
20. The invention of claim 19 wherein said prediction is a function of
perceptually encoded versions of said left and right channels.
21. The invention of claim 19 wherein said prediction is a function of the
product of a) a perceptually encoded version of said left channel with b)
a respective prediction coefficient, and is further a function of the
product of a) a perceptually encoded version of said right channel, with
b) a respective prediction coefficient.
22. The invention of claim 15 wherein a first one of said matrixed channels
is a function of the sum of said left and right channels; wherein a second
one of said matrixed channels is a function of the difference between said
left and right channels; and wherein a third one of said matrixed channels
is said center channel.
23. The invention of claim 15 wherein a first one of said matrixed channels
is a function of a sum channel and a prediction of said sum channel;
wherein a second one of said matrixed channels is a function of a
difference channel and a prediction of said difference channel; and
wherein a third one of said matrixed channels is said center channel; said
sum channel being a function of the sum of said left and right channels,
said difference channel being a function of the difference between said
left and right channels, and each said prediction being a function of said
center channel.
24. The invention of claim 23 wherein each said prediction is a function of
a perceptually encoded version of said center channel.
25. The invention of claim 23 wherein each said prediction is a function of
the product of a) a perceptually encoded version of said center channel,
with b) a respective prediction coefficient.
26. The invention of claim 15 wherein a first one of said matrixed channels
is a sum channel; wherein a second one of said matrixed channels is a
difference channel; and wherein a third one of said matrixed channels is a
function of said center channel and a prediction of said center channel;
said sum channel being a function of the sum of said left and right
channels, said difference channel being a function of the difference
between said left and right channels, and said prediction being a function
of said left and right channels.
27. The invention of claim 26 wherein said prediction is a function of
perceptually encoded versions of said left and right channels.
28. The invention of claim 26 wherein said prediction is a function of the
product of a) a perceptually encoded version of said left channel with b)
a respective prediction coefficient, and is further a function of the
product of a) a perceptually encoded version of said right channel, with
b) a respective prediction coefficient.
29. The invention of claim 15 wherein said set of input channels further
includes left surround and right surround channels.
30. The invention of claim 29 wherein an individual one of said matrixed
channels is a function of said left surround channel and a prediction of
said left surround channel, and wherein a further one of said matrixed
channels is a function of said right surround channel and a prediction of
said right surround channel.
31. The invention of claim 30 wherein said left surround channel prediction
and said right surround channel prediction are respective functions of one
or more of said left, right and center channels.
32. The invention of claim 30 wherein said left surround channel prediction
and said right surround channel prediction are respective functions of
perceptually encoded versions of one or more of said left, right and
center channels.
33. The invention of claim 29 wherein an individual one of said matrixed
channels is a function of the sum of said left surround and right surround
channels; and wherein a further one of said matrixed channels is a
function of the difference between said left surround and right surround
channels.
34. The invention of claim 29 wherein an individual one of said matrixed
channels is a function of a sum surround channel and a prediction of said
sum surround channel; and wherein a further one of said matrixed channels
is a function of a difference surround channel and a prediction of said
difference surround channel; said sum surround channel being a function of
the sum of said left surround and right surround channels, and said
difference surround channel being a function of the difference between
said left surround and right surround channels.
35. The invention of claim 34 wherein said sum surround channel prediction
and said difference surround channel prediction are respective functions
of one or more of said left, right and center channels.
36. The invention of claim 34 wherein said sum surround channel prediction
and said difference surround channel prediction are respective functions
of perceptually encoded versions of one or more of said left, right and
center channels. |
|
 |
|
 |
Claims |
|
|
|
Description |
|
|
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.
More recently, the art has turned its attention to the perceptual coding of
more-than-two-channel audio, such as five-channel audio. (As will apparent
to those skilled in the art as this description continues, the invention
can, however, be implemented in a system having other than five channels.)
The input channels of a five-channel audio system typically comprise three
"front" channels and two "back" channels. The front channels include the
conventional left and right stereo channels plus a center channel whose
frequency domain representation over time is denoted herein as C. These
channels are intended to be reproduced at speakers positioned in front of
the listener-at the left, at the right and directly in front,
respectively. The back channels are referred to as the "left surround" and
"right surround" channels whose frequency domain representations over time
are denoted herein as LS and RS, These channels are intended to be
reproduced at speakers positioned behind the listener--at the left and at
the right, respectively.
SUMMARY OF THE INVENTION
The above-mentioned teachings of the Johnston patent relating to switching
between coding modes for the coding of stereo, i.e., two-channel audio,
can be applied to a five-channel system, as well, in order, again, to
provide further compression over that provided by the perceptual coding
itself. For example, one can switch the front channels between two modes
and the back channels between two modes. The two coding modes for the
front channels would be a) a mode whose set of matrixed channels comprises
L, R, and C, and b) a mode whose set of matrixed channels comprises S, D
and C. Similarly, the two coding modes for the back channels would be a) a
mode whose set of matrixed channels comprises LS and RS, and b) a mode
whose set of matrixed channels comprises back sum/difference channels SS
and SD, given by SS=(LS+RS)/2 and SD=(LS-RS)/2.
We have invented, however, a more sophisticated mode-switching approach for
more-than-two-channel, e.g., five-channel, coding. In accordance with the
invention, yet additional compression is achieved by switching among a
plurality of modes at least one of which includes in its matrixed channel
set at least one matrixed channel given by an input channel or
sum/difference channel from which has been subtracted a prediction of
itself. If a prediction is a "good" prediction, i.e., closely matches to
the channel being predicted, the number of bits needed to represent their
difference will be substantially less than that required to represent the
predicted channel directly, thereby providing the aforementioned
additional compression.
An example of such a mode, for the front channels, comprises the set of
three matrixed channels
L-.alpha.C, R-.beta.C C
where .alpha.C and .beta.C are predicted values of input channels L and R,
respectively, as described in further detail hereinbelow. Another example
of such a mode, for the back channels, comprises the set of two matrixed
channels
SS-.eta..sub.1 L-.kappa..sub.1 R, SD-.eta..sub.2 L+.kappa..sub.2 R
where .eta..sub.1 L-.kappa..sub.1 R and .eta..sub.2 L+.kappa..sub.2 R are
predicted values for sum/difference channels SS and SD, respectively.
In preferred embodiments, as can be seen from the examples above, the
prediction for a particular input channel or sum/difference channel is
derived from at least one other, "predicting" channel. For the front
channels, there are illustratively a total of six coding modes (explicitly
laid out in the Detailed Description below), in which 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, there are
illustratively a total of eight coding modes (also explicitly laid out
below), in which all three front channels are used as predicting channels
for LS, RS, SS and DS, either by themselves or in various combinations.
Predictions are preferably based on the coded values of the predicting
channels rather than their actual values, e.g., C rather than C. This
allows the input channels to be decoded without introducing a quantization
artifact.
In preferred embodiments, the selection of which coding mode is to be used
for each coder band is made by determining which of the modes will require
the fewest bits to encode it.
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 implementing the principles of the invention;
FIGS. 4-6 illustrate a novel aspect of the perceptual audio coder relating
to the use of a global masking threshold;
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
convened 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), Is(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 generated in the
course of implementing the present invention 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.
In accordance with the present invention, however, 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 carded 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 matrixed channel which includes a prediction of that input or
sum/difference 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 carded
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 carded 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.
A further novel feature of coder 104 of FIG. 2 relates to the generation of
the thresholds for the five input channels and the four sum/difference
channels. That feature of the coder, which uses a so-called global masking
threshold, 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.
In order to understand the use of this feature, we can 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-beating 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 aforementioned novel feature of the coder, an
approach to the bit reservoir depletion which takes psycho-aco | | |
