Perceptual noise shaping in the time domain via LPC prediction in the frequency domain

A method and apparatus for the perceptual coding of audio signals in which perceptual noise shaping is achieved in the time domain by performing a (linear) prediction (i.e., filtering) in the frequency domain. As a result, the temporal spread of quantization noise is reduced. Specifically, according to one illustrative embodiment of the present invention, a method comprises decomposition of the audio signal into a plurality of spectral component signals; generating a prediction signal representative of a prediction of one of said spectral component signals, said prediction based on one or more other ones of said spectral component signals; comparing the prediction signal with said one of said spectral component signals to generate a prediction error signal; coding said one of said spectral component signals based on the prediction error signal to generate a coded spectral component signal; and generating the encoded signal based on the coded spectral component signal.

FIELD OF THE INVENTION 
The present invention relates to the field of audio signal coding and more 
specifically to an improved method and apparatus for coding audio signals 
based on a perceptual model. 
BACKGROUND OF THE INVENTION 
During the last several years so-called "perceptual audio coders" have been 
developed enabling the transmission and storage of high quality audio 
signals at bit rates of about 1/12 or less of the bit rate commonly used 
on a conventional Compact Disc medium (CD). Such coders exploit the 
irrelevancy contained in an audio signal due to the limitations of the 
human auditory system by coding the signal with only so much accuracy as 
is necessary to result in a perceptually indistinguishable reconstructed 
(i.e., decoded) signal. Standards have been established under various 
standards organizations such as the International Standardization 
Organization's Moving Picture Experts Group (ISO/MPEG) MPEG1 and MPEG2 
audio standards. Perceptual audio coders are described in detail, for 
example, in U.S. Pat. No. 5,285,498 issued to James D. Johnston on Feb. 8, 
1994 and in U.S. Pat. No. 5,341,457 issued to Joseph L. Hall II and James 
D. Johnston on Aug. 23, 1994, each of which is assigned to the assignee of 
the present invention. Each of U.S. Pat. Nos. 5,285,498 and 5,341,457 is 
hereby incorporated by reference as if fully set forth herein. 
Generally, the structure of a perceptual audio coder for monophonic audio 
signals can be described as follows: 
The input samples are converted into a subsampled spectral representation 
using various types of filterbanks and transforms such as, for example, 
the well-known modified discrete cosine transform (MDCT), polyphase 
filterbanks or hybrid structures. 
Using a perceptual model one or more time-dependent masking thresholds for 
the signal are estimated. These thresholds give the maximum coding error 
that can be introduced into the audio signal while still maintaining 
perceptually unimpaired signal quality. 
The spectral values are quantized and coded according to the precision 
corresponding to the masking threshold estimates. In this way, the 
quantization noise may be hidden (i.e., masked) by the respective 
transmitted signal and is thereby not perceptible after decoding. 
Finally, all relevant information (e.g., coded spectral values and 
additional side information) is packed into a bitstream and transmitted to 
the decoder. 
Accordingly, the processing used in a corresponding decoder is reversed: 
The bitstream is decoded and parsed into coded spectral data and side 
information. 
The inverse quantization of the quantized spectral values is performed. 
The spectral values are mapped back into a time domain representation using 
a synthesis filterbank. 
Using such a generic coder structure it is possible to efficiently exploit 
the irrelevancy contained in each signal due to the limitations of the 
human auditory system. Specifically, the spectrum of the quantization 
noise can be shaped according to the shape of the signal's noise masking 
threshold. In this way, the noise which results from the coding process 
can be "hidden" under the coded signal and, thus, perceptually transparent 
quality can be achieved at high compression rates. 
Without further precautions, however, a perceptual coder may not deliver 
transparent signal quality when coding transient signals such as, for 
example, castanet or glockenspiel sounds. This problem results from what 
is commonly known as the "pre-echo" problem, familiar to those skilled in 
the art. In particular, while the signal to be coded may contain strong 
signal components in only portions of the time window processed by the 
coder's analysis filterbank and a given instant, the resultant coding 
error typically becomes spread out across the entire window length. Thus, 
the quantization noise may be distributed over a period of, for example, 
20 milliseconds or more, and it may thereby exceed the magnitude of 
original signal components in certain signal regions. Given, for example, 
a castanet signal with an "attack" in the middle portion of an analysis 
window, the noise components of the coded signal may be stronger than the 
original signal components in the portion of the window immediately before 
the "attack." 
It is known that, due to the properties of the human auditory system, such 
"pre-echoes" are masked only if no significant amount of the coding noise 
is present longer than approximately 2 ms before the onset of the signal. 
Otherwise the coding noise is likely to be perceived as a "pre-echo" 
artifact--i.e., a short noise-like event preceding the signal onset. 
A number of techniques have been proposed in order to avoid pre-echo 
artifacts in an encoded/decoded signal produced by a perceptual audio 
coding system: 
1) One technique which has been used is to increase the coding precision of 
the spectral coefficients of the filterbank window that first covers the 
transient signal portion. This is known as "pre-echo control," and is 
incorporated, for example, in the MPEG1 audio standard. Since this 
approach requires considerably more bits for the coding of these frames, 
such a method cannot be easily applied in a constant bit rate coder. To a 
certain degree, local variations in bit rate demand can be accounted for 
by using the conventional technique known as a "bit reservoir," also 
incorporated, for example, in the MPEG1 audio standard. This technique 
permits the handling of peak demands in bit rate by using bits that have 
been set aside during the coding of earlier frames--thus, the average bit 
rate still remains constant. In practice, however, the size of the bit 
reservoir needs to be unrealistically large in order to avoid artifacts 
when coding input signals of a very transient nature. 
2) A different strategy used in many conventional perceptual audio coders 
is known as adaptive window switching. This technique, also incorporated 
in the MPEG1 audio standard, adapts the size of the filterbank windows to 
the characteristics of the input signal. While portions of the signal 
which are relatively stationary will use a long window length (as is 
usual), short windows are used to code the transient portions of the 
signal. In this way, the peak bit demand can be reduced considerably 
because the regions for which a high coding precision is required are 
constrained in time. 
One major disadvantage of the adaptive window switching technique is that 
it introduces significant additional complexity into the coder and 
complicates its structure. Since the different window sizes require 
different parameters and encoding strategies, a coder using window 
switching in fact consists of essentially two coders, one for the longer 
window size and one for the shorter window size. Moreover, this technique 
cannot be used efficiently in the case of a "pitched" signal consisting of 
a pseudo-stationary series of impulse-like signals, such as, for example, 
human speech, without incurring a substantial penalty in coding 
efficiency. Due to the mechanism of speech production, the temporal spread 
of quantization noise would only be adequately avoided with use of this 
technique by permanently selecting the shorter window size. This would, in 
turn, lead to a significant decrease in coder efficiency due to the 
decreased coding gain and increased side information overhead. 
3) A third technique which has been used to avoid the temporal spread of 
quantization noise is to apply a gain change/modification to the signal 
prior to performing the spectral decomposition. The underlying principle 
of this approach is to reduce the dynamics of the input signal by applying 
a gain modification prior to its encoding. The parameters of the gain 
modification are then transmitted in the bitstream--using this information 
the process may be reversed on the decoder side. 
In order to perform well for most signals, however, the processing has to 
be applied to different parts of the frequency spectrum independently, 
since transient events are often present only in certain portions of the 
spectrum. This can be done using more complex hybrid filterbanks that 
allow for separate gain processing of different spectral components. In 
general, however, the interdependencies between the gain modification and 
the coder's perceptual model are often difficult to resolve. 
SUMMARY OF THE INVENTION 
In accordance with an illustrative embodiment of the present invention, a 
method and apparatus which overcomes the drawbacks of prior art techniques 
is provided. In particular, perceptual noise shaping is achieved in the 
time domain by performing a (linear) prediction (i.e., filtering) in the 
frequency domain. As a result, the temporal spread of quantization noise 
is reduced. Specifically, according to one illustrative embodiment of the 
present invention, the following processing steps are applied in an 
encoder for use with monophonic signals: 
The audio signal to be coded is decomposed into spectral coefficients by a 
high-resolution filterbank/transform (such as that used for the "longer 
block" in conventional perceptual coders which employ adaptive window 
switching). 
Using a perceptual model, one or more time-dependent masking thresholds for 
the signal are estimated. These thresholds give the maximum coding error 
that can be introduced into the audio signal while still maintaining 
perceptually unimpaired signal quality. 
The encoding of the spectral values is then performed using a 
quantization/coding scheme based on Differential Pulse Code Modulation 
(DPCM) that operates on the filterbank outputs in frequency. As in 
conventional perceptual coders, the target for the required coding 
precision may be given by the perceptual model. 
Finally, all relevant information (e.g., the coded spectral values and the 
generated side information) is packed into a bitstream and transmitted to 
the decoder. In particular, the generated side information includes a flag 
indicating the use of DPCM coding and, if used, information about the 
target frequency range and the filter employed for encoding. 
Similarly, a corresponding illustrative decoder in accordance with an 
illustrative embodiment of the present invention performs the following 
processing steps: 
The bitstream is decoded and parsed into coded spectral data and side 
information. 
The inverse quantization of the quantized spectral values is performed. In 
particular, this may include the DPCM decoding of spectral values if the 
use of DPCM has been flagged in the side information. 
The spectral values are mapped back into a time domain representation using 
a synthesis filterbank. 
The selection of the type of DPCM quantization/coding scheme 
(predictor/quantizer combination) may yield different advantages for the 
overall system behavior. Specifically, and in accordance with a first 
illustrative embodiment of the present invention, a closed-loop DPCM 
system is employed. Although this first embodiment results in a coding 
gain for transient signals, in a preferred approach in accordance with a 
second embodiment of the present invention, an open-loop DPCM system is 
employed. This second embodiment will advantageously result in a 
time-shaped quantization error at the output of the decoder. Specific 
processing is applied to spectral coefficients, the quantization noise in 
the decoded signal (after the inverse filterbank is applied in the 
decoder) will be shaped in time, thereby keeping the quantization noise 
under the actual signal. In this manner, temporal problems with unmasking, 
either in transient or pitchy signals, are advantageously avoided without 
the need for substantial overcoding and its commensurate expenditure of 
bits.

DETAILED DESCRIPTION 
The instant inventive method and apparatus overcomes the drawbacks of prior 
art techniques by effectively replacing the use of a conventional Pulse 
Code Modulation (PCM) quantization/coding scheme as is typically used in 
conventional perceptual audio coders with a quantization/coding scheme 
based on Differential Pulse Code Modulation (DPCM), wherein the DPCM 
scheme operates on the filterbank outputs in the frequency domain. (Both 
PCM coding and DPCM coding techniques in general are well known to those 
skilled in the art.) 
FIG. 1 shows a conventional perceptual encoder for use in coding monophonic 
audio signals. The encoder of FIG. 1 performs the following steps: 
The input signal x(k) is decomposed into spectral coefficients by analysis 
filterbank/transform 12, resulting in "n" spectral components y(b,0) . . . 
y(b,n-1) for each analysis block "b," where "n" is the number of spectral 
coefficients per analysis block (i.e., the block size). Each spectral 
component y(b,j) is associated with an analysis frequency or frequency 
range according to the employed filterbank. 
Perceptual model estimates the required coding precision for a perceptually 
transparent quality of the encoded/decoded signal and generates one or 
more masking thresholds. This information may, for example, comprise the 
minimum signal-to-noise ratio (SNR) required in each frequency band, and 
is provided to PCM encoder 16. 
Each spectral component y(b,j) is quantized and mapped to transmission 
indices i(b,0) . . . i(b,n-1) by quantizers 16-0 . . . 16-(n-1), 
respectively (performing quantizations Q.sub.0 . . . Q.sub.n-1, 
respectively). These quantizers perform a PCM quantization/coding of the 
spectral coefficients in accordance with the perceptual masking thresholds 
generated by perceptual model 14. 
The index values i(b,0) . . . i(b,n-1) are passed to bitstream encoder 18 
together with (optional) side information, and are subsequently 
transmitted (e.g., to a decoder) in the encoded bitstream. Alternatively, 
the encoded bitstream may be stored on an audio signal storage medium such 
as a Compact Disc (CD) or a Digital Audio Tape (DAT) for later retrieval. 
In accordance with certain illustrative embodiments of the present 
invention, the encoding apparatus of FIG. 1 may be advantageously modified 
by replacing PCM encoder 16 with a DPCM-type encoder wherein the DPCM 
encoding is performed in the frequency domain. FIGS. 3 and 4 show two such 
illustrative embodiments of the present invention. In particular, an 
illustrative embodiment of the present invention may be realized by 
replacing PCM encoder 16 of the conventional encoding apparatus of FIG. 1 
with module 32 as shown in FIG. 3, thereby resulting in an encoding 
apparatus in accordance with a first illustrative embodiment of the 
present invention. Similarly, another illustrative embodiment of the 
present invention may be realized by replacing PCM encoder 16 of the 
conventional encoding apparatus of FIG. 1 with module 42 as shown in FIG. 
4, thereby resulting in an encoding apparatus in accordance with a second 
illustrative embodiment of the present invention. In each case the input 
to the quantizer/coding kernel is given by the series of the spectral 
coefficients y(b,0) . . . y(b,n-1). That is, the DPCM encoding is 
performed across the frequency domain, as opposed to, for example, 
predictive coding across the time domain as is performed by conventional 
subband-ADPCM coders, well known to those skilled in the art. 
Specifically, rotating switch 33 of the illustrative encoder of FIG. 3 and 
rotating switch 43 of the illustrative encoder of FIG. 4, each are used to 
bring the spectral values y(b,0) . . . y(b,n-1) into a serial order prior 
to quantization/encoding by DPCM encoders 34 and 44, respectively, and 
rotating switch 35 of the illustrative encoder of FIG. 3 and rotating 
switch 46 of the illustrative encoder of FIG. 4 each are used to bring the 
respective resulting index values i(b,0) . . . i(b,n-1) into a parallel 
order thereafter. Although in each of the illustrative encoders shown, the 
processing of the spectral values y(b,0) . . . y(b,n-1) is advantageously 
performed in order of increasing frequency, other illustrative embodiments 
may perform the processing either in order of decreasing frequency or in 
other alternative (e.g., non-monotonic) orderings. Moreover, only a subset 
of the spectral values (rather than all "n" of them, as shown herein) may 
be provided to DPCM encoders 34 and 44 for differential coding. 
More specifically, FIG. 3 shows a first illustrative embodiment of an 
encoder according to the present invention in which a closed-loop 
prediction scheme is used. Closed-loop prediction is a conventional 
technique well known to those of ordinary skill in the art. In the 
illustrative perceptual audio encoder of FIG. 3, however, a closed-loop 
prediction is applied to the spectral values (i.e., in the frequency 
domain). In particular, a prediction filter (shown in the figure as 
comprising predictor 36 and adder 39) is driven by the quantized output 
values generated by quantizer 37, and the predicted value is subtracted 
from the input signal by subtractor 38 so that only the prediction error 
signal is advantageously quantized/encoded. Note that quantizer 37 
performs quantizations Q.sub.0 . . . Q.sub.n-1, respectively, for each of 
the spectral component values y(b,0) . . . y(b,n-1) which are provided 
thereto by rotating switch 33 (via subtractor 38). The use of the 
illustrative encoder of FIG. 3 will advantageously result in a coding gain 
if the encoder input signal x(k) has a transient characteristic. 
FIG. 4 shows a second illustrative embodiment of an encoder according to 
the present invention in which an open-loop prediction scheme is used. 
Open-loop prediction is a conventional technique well known to those of 
ordinary skill in the art. In the illustrative perceptual audio encoder of 
FIG. 4, however, an open-loop prediction is applied to the spectral values 
(i.e., in the frequency domain). In particular, predictor 47 is driven by 
the unquantized input values and the predicted value is then subtracted 
from the input signal by subtractor 48 so that only the prediction error 
signal is advantageously quantized/encoded (by quantizer 45). Note that 
quantizer 45 performs quantizations Q.sub.0 . . . Q.sub.n-1, respectively, 
for each of the spectral component values y(b,0) . . . y(b,n-1) for which 
corresponding prediction error signals are provided thereto by rotating 
switch 43 (via subtractor 48). 
Like the illustrative encoder of FIG. 3, the use of the illustrative 
encoder of FIG. 4 will also advantageously result in a coding gain if the 
encoder input signal x(k) has transient characteristics. In addition, 
however, the use of a perceptual audio encoder employing the open-loop 
approach of FIG. 4 will advantageously produce a timeshaped quantization 
error in the final reconstructed output signal x'(k) of a corresponding 
decoder. This follows from the fact that open-loop prediction has been 
applied to spectral coefficients so that the quantization noise appears as 
shaped in time, thereby putting the noise level under the signal level. In 
this way, temporal problems with unmasking, either in transient or in 
pitchy signals, are advantageously avoided without the need for 
substantial overcoding and its commensurate expenditure of bits. 
Since in the above-described illustrative embodiments of the present 
invention predictive coding is applied to spectral domain data, certain 
relations known for classic prediction are valid with time and frequency 
domain swapped. For example, prediction gain is achieved depending on the 
"envelope flatness measure" of the signal (as opposed to the "spectral 
flatness measure"). Moreover, in the open-loop case shown in FIG. 4, the 
prediction error is shaped in time (as opposed to frequency). In effect, 
therefore, the above-described open-loop technique may, for example, be 
considered equivalent to applying an adaptive time domain window by 
prediction in the frequency domain, effectively using convolution by a few 
elements in the frequency domain to instantiate time-domain noise shaping. 
Although in the above-described embodiments the prediction process is 
performed over the entire frequency spectrum (i.e., for all spectral 
coefficients), in other illustrative embodiments the prediction may be 
performed for only a portion of the spectrum (i.e., for a subset of the 
spectral coefficients). In addition, different predictor filters can be 
advantageously employed in different portions of the signal spectrum. In 
this manner, the instant inventive method for time-domain noise control 
can be applied in any desired frequency-dependent fashion. 
In order to provide for the proper decoding of the encoded signal, the 
bitstream generated by the illustrative encoders of FIGS. 3 and 4 
advantageously includes certain additional side information, shown, for 
example, as an additional input to bitstream encoder 18 of FIG. 1. In 
various illustrative embodiments of the present invention, for example, 
one field of side information may indicate the use of DPCM encoding and 
the number of different prediction filters used. Then, additional fields 
in the bitstream may be transmitted for each prediction filter signalling 
the target frequency range of the respective filter and its filter 
coefficients. 
FIG. 6 shows a flow chart of a method of encoding monophonic audio signals 
in accordance with an illustrative embodiment of the present invention. 
The illustrative example shown in this flow chart implements certain 
relevant portions of a perceptual audio encoder with open-loop prediction 
and a single prediction filter. Specifically, step 61 performs a 
conventional calculation of the spectral values by an analysis filterbank 
(as performed, for example, by analysis filterbank/transform 12 of the 
conventional encoder of FIG. 1). Then, the order of the prediction filter 
is set and the target frequency range is defined in step 62. These 
parameters may, for example, be illustratively set to a filter order of 15 
and a target frequency range of from 4 kHz to 20 kHz. With these 
illustrative parameter values, pre-echoes and post-echoes will be 
advantageously removed when coding pitchy signals. 
In step 63, the prediction filter is determined by using the range of 
spectral coefficients matching the target frequency range and applying a 
conventional method for predictive coding as is well known for DPCM 
coders. For example, the autocorrelation function of the coefficients may 
be calculated and used in a conventional Levinson-Durbin recursion 
algorithm, well known to those skilled in the art. As a result, the 
predictor filter coefficients, the corresponding reflection coefficients 
("COR" coefficients) and the expected prediction gain are known. 
If the expected prediction gain exceeds a certain threshold (e.g., 2 dB), 
as determined by decision 64, the DPCM coding procedure of steps 65 
through 67 is used. In this case, the prediction filter coefficients are 
quantized (in step 65) as required for transmission to the decoder as part 
of the side information. Then, (in step 66) the prediction filter is 
applied to the range of spectral coefficients matching the target 
frequency range where the quantized filter coefficients are used. For all 
further processing the given range of spectral coefficients is replaced by 
the output of the filtering process. Finally (in step 67), a field of the 
bitstream is transmitted signalling the use of DPCM coding ("prediction 
flag" on), and the target frequency range, the order of the prediction 
filter and information describing its filter coefficients are also 
included in the bitstream. If, on the other hand, the expected prediction 
gain does not exceed the decision threshold, step 68 transmits a field in 
the bitstream signalling that no DPCM coding has been used ("prediction 
flag" off). Finally, in either case, the quantization process is applied 
to the spectral coefficients (step 69), where the quantization is based on 
the perceptual masking thresholds generated by the perceptual model of the 
encoder. 
Using an open-loop encoder embodiment of the present invention (e.g., as 
shown in the illustrative apparatus of FIG. 3 and in the illustrative 
method of FIG. 6), a straightforward temporal noise shaping effect can be 
achieved for certain conventional block transforms including the Discrete 
Fourier Transform (DFT) or the Discrete Cosine Transform (DCT), both 
well-known to those of ordinary skill in the art. If, for example, a 
perceptual coder in accordance with the present invention uses a 
critically subsampled filterbank with overlapping windows--e.g., a 
conventional Modified Discrete Cosine Transform (MDCT) or another 
conventional filterbank based on Time Domain Aliasing Cancellation 
(TDAC)--the resultant temporal noise shaping is subject to the time domain 
aliasing effects inherent in the filterbank. For example, in the case of a 
MDCT, one mirroring (i.e., aliasing) operation per window half takes place 
and the quantization noise appears mirrored (i.e., aliased) within the 
left and the right half of the window after decoding, respectively. Since 
the final filterbank output is obtained by applying a synthesis window to 
the output of each inverse transform and performing an overlap-add of 
these data segments, the undesired aliased components are attenuated 
depending on the used synthesis window. Thus it is advantageous to choose 
a filterbank window that exhibits only a small overlap between subsequent 
blocks so that the temporal aliasing effect is minimized. An appropriate 
strategy in the encoder can, for example, adaptively select a window with 
a low degree of overlap for critical signals of very transient character 
while using a wider window type for stationary signals providing a better 
frequency selectivity. The implementation details of such a strategy will 
be obvious to those skilled in the art. 
FIG. 2 shows a conventional perceptual decoder for use in decoding 
monophonic audio signals corresponding to the conventional perceptual 
encoder of FIG. 1. The decoder of FIG. 2 performs the following steps: 
The incoming bitstream is parsed and the index values i(b,0) . . . i(b,n-1) 
are extracted by decoder/demultiplexer 22. 
Using inverse quantizers 24-0 through 24-(n-1) (performing inverse 
quantizations IQ.sub.0 . . . IQ.sub.n-1, respectively), the quantized 
spectral values yq(b,1) . . . yq(b,n-1) are reconstructed by PCM decoder 
24. 
The quantized spectral values yq(b,1) . . . yq(b,n-1) are mapped back to a 
time domain representation by synthesis filterbank 26, resulting in 
reconstructed output signal x'(k). 
In accordance with an illustrative embodiment of the present invention, the 
conventional decoding apparatus of FIG. 2 may be advantageously modified 
by replacing PCM decoder 24 with a DPCM-type decoder wherein the DPCM 
decoding is performed in the frequency domain. FIG. 5 shows one such 
illustrative embodiment of the present invention. In particular, an 
illustrative embodiment of the present invention may be realized by 
replacing PCM decoder 24 of the conventional decoding apparatus of FIG. 2 
with module 52 as shown in FIG. 5, thereby resulting in an decoding 
apparatus in accordance with an illustrative embodiment of the present 
invention. Specifically, the input to DPCM decoder 55 is given by the 
series of index values i(b,0) . . . i(b,n-1), which are brought into a 
serial order prior to decoding by rotating switch 53. The resulting 
spectral values yq(b,0) . . . yq(b,n-1) are brought into a parallel order 
after the DPCM decoding by rotating switch 56. 
DPCM decoder 55 comprises inverse quantizer 54, predictor 57 and adder 58. 
Inverse quantizer 54 performs inverse quantizations IQ.sub.0 . . . 
IQ.sub.n-1, respectively, for each of the index values i(b,0) . . . 
i(b,n-1) which are provided thereto by rotating switch 53. Note that, if 
the illustrative open-loop encoder of FIG. 4 has been used to encode the 
audio signal, the combination of predictor 57 and adder 58 of the 
illustrative decoder of FIG. 5 effectuate a noise shaping filter which 
advantageously controls the temporal shape of the quantization noise. 
Again, although the illustrative decoder of FIG. 5 advantageously performs 
the processing of the index values i(b,0) . . . i(b,n-1) in order of 
increasing frequency, other illustrative embodiments may perform the 
processing either in order of decreasing frequency or in other alternative 
(e.g., non-monotonic) orderings, preferably in a consistent manner to the 
ordering employed by a corresponding encoder. Moreover, only a subset of 
the index values (rather than all "n" of them, as shown herein) may be 
provided to DPCM decoder 55, and/or several different predictor filters 
may be used for different portions of the signal spectrum, again 
preferably in a consistent manner with the specific technique employed by 
a corresponding encoder. Note also that, in the latter case, for example, 
in order to execute a proper decoding of the incoming bitstream, a decoder 
in accordance with the present invention may advantageously evaluate 
additional side information which has been transmitted by a corresponding 
encoder. In this manner, the decoder may apply DPCM decoding in each 
specified target frequency range with a desired corresponding decoder 
prediction filter. 
FIG. 7 shows a flow chart of a method of decoding monophonic audio signals 
in accordance with an illustrative embodiment of the present invention. 
The illustrative example shown in this flow chart implements certain 
relevant portions of a perceptual audio decoder with a single prediction 
filter. Specifically, step 71 performs a conventional reconstruction of 
the spectral coefficient values by inverse quantization. Then, decision 72 
checks the bitstream information to determine if the use of DPCM coding is 
indicated ("prediction flag" is on). If it is, then the extended decoding 
process shown in steps 73 and 74 is applied. Specifically, the transmitted 
side information in the bitstream is decoded to determine the target 
frequency range of the DPCM coding, the order of the prediction filter, 
and information describing its filter coefficients (step 73). Then, the 
inverse prediction filter is applied to the range of spectral coefficients 
matching the specified target frequency range (step 74). For all further 
processing, the given range of spectral coefficients is replaced by the 
output of the filtering process. Finally (and regardless of the 
determination made by decision 72 described above), a conventional 
synthesis filterbank is run from the spectral coefficients in step 75. 
Although a number of specific embodiments of this invention have been shown 
and described herein, it is to be understood that these embodiments are 
merely illustrative of the many possible specific arrangements which can 
be devised in application of the principles of the invention. For example, 
although the illustrative embodiments which have been shown and described 
herein have been limited to the encoding and decoding of monophonic audio 
signals, alternative embodiments which may be used for the encoding and 
decoding of multichannel (e.g., stereophonic) audio signals will be 
obvious to those of ordinary skill in the art based on the disclosure 
provided herein. In addition, numerous and varied other arrangements can 
be devised in accordance with these principles by those of ordinary skill 
in the art without departing from the spirit and scope of the invention.