31 5
FAST ALGORITHMS FOR TMUX DESIGNS
G. J. Uriarte, R. A. Carrasco
Staffordshire Polytechnic, U.K.
ABSTRACT
Different approaches of the decomposition of
the digital SSB modulation process into a
transform processor and a polyphase network
are researched. The application of the latest
DCT algorithm to established transmultiplexer
(TMUX) algorithms and a new approach based on
the Hartley transform are presented. It is
shown that the resulting structures reduce
the total computational requirement in
comparison with the most efficient TMUX
algorithm published.
INTRODUCTION
In November 1981, Scheuermann and Gockler (1)
published a survey of transmultiplexing
methods, where they classified and compared
them into groups having different criteria.
They showed that those algorithms
characterized by the simultaneous processing
of all channels in a transform of the Fourier
type are, by far, the most efficient in terms
of number of operations per second and per
channel. Providing that commercially
available IC's are employed for the hardware
implementation, the number of multiplications
and additions to be carried out per second is
a suitable measure for the computational
complexity of the algorithm.
The decomposition of the multiplexing
algorithm into two processing units, the fast
transform processor and the polyphase
network, was first suggested by Bellanger and
Daguet ( 3 ) in 1974. Since then, several
different approaches for the implementation
of each of these units has been reported. In
the latest publications, the focus has been
directed to the polyphase network: this paper
describes the different approaches of the
fast transform processor unit and suggests
the application of different transform
algorithms in order to improve the
efficiency. Following, the fundamentals
behind the modulation process are introduced
and the different techniques are compared.
Since the demodulation process may be derived
from the modulation process by transposition,
only the latter will be treated.
DIGITAL SSB MODULATION CONCEPTS
Sampling and sampling rate conversion can be
viewed as modulating processes which produce
in the spectrum of the digital signal,
periodic repetitions of the baseband signal
(images) spaced at harmonics of the sampling
frequency. These processes can be used
advantageously when dealing with bandpass
signals by associating the bandpass signal
with one of those images instead of with the
baseband (Crochiere and Rabiner (2)) . This
permits the realization of the digital SSB
modulation-demodulation bank using a bandpass
filter bank and therefore, the application of
digital filter bank and spectrum analysis and
synthesis theory to the TMUX design.
Fig. 1.a illustrates the basic framework of
the digital SSB modulation process for K
channels. The pk (v) signals represent the
digital voice signals sampled at 8 kHz
obtained from a demultiplexing of the PCM
signal. Before their frequency multiplexing,
they may be pre-processed with the intention
of modifying their original frequency spectra
shown in fig. 1.b into any of the others
shown in the same figure.
The output of the modulating filter bank can
be either real or complex: in the latter case
the subsequent post-processing converts it
into real. Depending on the spectrum of the
input signals, two different types of channel
stacking arrangements can be expected: the
even type (where the centres of the channel
bands are at frequencies Wr=2nk/2K for
k=O..K-1) or the odd type (where the centres
of the channel bands are at frequencies
wr=(2nk/2K)+(n/2K) for k=O. .K-1).
In order to SSB modulate K real channels
sampled at their minimum sampling rate, they
must be interpolated by a factor MZK. If M=K,
the system is said to be critically sampled
(2). and if M>K it is said to be over-
sampled. In TMUX applications the system is
chosen to be critically sampled, where the
total number of channels is composed of voice
channels and empty or dummy channels. The
latter are used as guard bands in the
analogue modulation process which translates
the FDM signal produced into the standard
frequency position, and occasionally, as
extra channels for the forcing of K to a
value suitable for later algorithmic
simplifications.
Since the SSB modulation can be expressed as
the addition of two complex conjugate
modulations, the following discussion-
considers the complex modulation and treats
the SSB modulation as a special case. In the
next section, the equivalence between the
digital modulation and bandpass filtering is
formally presented.
DIGITAL COMPLEX MODULATING PROCESS
INTERPRETATIONS
There are two different interpretations of
the complex digital modulation process: the
complex modulating and the complex bandpass
filtering interpretations.
In the complex modulating bank, each channel
is first interpolated by either a bandpass
or a lowpass filter, and then a complex
modulations step translates them all to their
respective frequencies. Fig. 2.a shows the K
channel complex modulator, where the general
modulating function, t(k,n), takes the form:
t(k,n) = EXP [2nj((n+no)(k+ko)+a)/K].
ko and no represent the frequency and time
origins and a , a time shift on the carrier

signal. The introduction of the ko and no
constants provides for those cases where it
is desirable to use time and/or frequency
origins not equal to zero; for example,
modification of the time origin is sometimes
useful in the design of IIR filter banks for
improving the frequency response
characteristics of the filter bank.
Alternatively, modification of the frequency
origin leads to designs with different
channel stacking arrangements. Finally,
different values of a lead to different types
of discrete transformations, allowing the one
with the fastest computation algorithm to be
chosen.
Denoting yk (n) as the output of the kth
channel complex modulator, the output of the
bank can be expressed as:
K-1 K-1
y(n) = E yk(n) = I t(k,n) Z Xk (m) h(n-mK)
k=O k=O m (1)
where Xk (m) represents the pre-processed
input channels, h(n) the impulse response
of the interpolating filter, and m takes the
implied discrete range from --
to +-.
The complex bandpass filtering interpretation
can be developed using the identities:
t (k,n) = t [k, (n-mK)I t [k,mKJ
for non critically sampled systems, and
t(k,n) = t[k, (n-mK)]
for critically sampled systems (M=K).
Since the system under consideration is
critically sampled, eq. 1 can be rewritten:
K-1
y(n) = E E Xk(m) h(n-mK) t[k,n-mK1
k=O m
K-1
= E Xk (m) hk (n-mK).
k=O m
This expression implies the structure shown
in fig. 2.b, where the complex bandpass
filter eliminates the appropriate harmonic
images of the signal after the sampling rate
increase has taken place. When processing
real input signals, for the even channels the
upper sideband is selected, whereas for the
odd channels the lower sideband is selected;
if the same sidebands are desired for both
even and odd channels, one of them must be
modulated by (-l)=, which corresponds to
frequency shifting by 4 kHz.
In the case of an SSB modulation, a third
interpretation can be resolved. In order for
the output of the modulating filter bank to
be real, it must be provided with two complex
conjugate frequency bands in each of the
input channels and therefore the filter bank
must be provided with two modulating branches
per input channel. In pursuance of the
requirement of the outputs of these branches
being complex conjugates, both the modulating
function t(k,n) and the input signals xr(m)
must be complex conjugate. Following the
complex modulator bank interpretation, the
addition of the outputs of those two branches
corresponding to the kth channel will be:
yak (n) = yk (n) -k y*k (n)
31 6
= W k (n) t(k,n) + W'k (n) t* (k,n)
K-1
= 2 I RE[Wk (n)] RE[t(k,n)l
k=O
- IM[wr (n)I IM[t (k,n)1 1 (3)
where W k (n) represents the output signal from
the interpolating filter corresponding to the
k'h channel. The operation described in eq. 3
bears resemblance to the last two steps of
the digital Weaver modulator: the second
modulation process and the addition of the
two paths. This relation becomes clearer if,
in a block diagram corresponding to eq. 3,
the conversion of the real input signal pr (VI
into the complex signal X k (m), is indicated:
fig. 2.c shows the k'h channel of the filter
bank following the resulting Weaver modulator
interpretation.
EFFICIENT REALIZATION O p CRITICALLY SAMPLED
MODULATING FILTER BANKS
The fast transform TMUX algorithms are
achieved through the reformulation of the
basic equations of the modulating filter bank
in such a way that they include a discrete
transform computable through a fast
algorithm. Examining eqs. 1 and 2 , it can be
seen that the factor t(k,n), which is a
periodic discrete complex exponential
function, appears in both: the variable k
takes the values lO..(K-l)l, while n takes
the values IO..-I. Defining T. as the period
of the t(k,n) function, the variable change
n=rT.+p leads to the expression:
y(rT.+p) = I h(rTo+p-mK) XP (m) (4)
m
K- 1
where xp (m) = I t(k,p) Xk (m) (5)
(2) k=O
and p = O..Tn-1, r = O..-.
Since the range of the variables p and k
variables is finite, eq (5) takes on the
form of a discrete transform calculable
through a fast algorithm. Assuming that no=O,
then ko for the even stacking arrangement is
0 and for the odd stacking arrangement is
1/4.
For ko=O, Tn=K and therefore the variables k
and p have the same range of values in the
discrete transform. Examining eq. 4 , it can
be seen that for the generation of each
output sample y(n)=y(rL+p) (for p=O..L-l), a
different subset of the filter coefficients,
hp(r)=h(rK+p), is used. After L outputs are
generated, the coefficient pattern repeats:
thus y(L) is generated using the same set of
coefficients ho (r) as ~ ( 0 ) :y(L+l) uses the
same set of coefficients ht (r) as y(l), and
so on. Therefore, for ko=O the interpolating
filter h(n) is a periodic time-varying
function with period L.
For ko= l/4, Tn=4K and the range of p becomes
four times that of the k variable, which is
not ideally suited to the use of a fast
algorithm for the discrete transform. Using
symmetries of the t(k,n) periodic function,
the range of p can be further reduced to
(O..2KI or IO..KI leading to the expression:
K- 1
y(rL+p) = Z h(rL+p-mK)f(r) I X k ( m ) t(k,p)
m k=O (6)

where t(k,n) = t(k,p) f(r)
p = O..(L-l), r = O..-,
and f(r) = EXP[jnrJ for L=2K
f(r) = EXP[jnr/2] for L=K
The f(r) function is periodic in "r", and has
a period with respect to this variable of 2
for the case L=2K, and 4 for the case L=K;
hence, in these cases, the period of h(n) f (r)
with repect to "n" is 4K.
To illustrate this graphically, it is
possible to represent the output signal as
the result of an interleaving of the p*h
filter outputs calculated in K parallel
branches, where each is derived from a
convolution of the outputs of the transform
block XP (m) with:
hp (r) for ko = O
h'p (r)=hp(r)f (r) for k0=1/4
This structure is called a polyphase network
and fig 3.b shows its commutator model. As an
example, fig. 3.a shows the case for L=2K,
where it can be seen that the polyphase
network inputs change from the first set of K
outputs of the fast transform processor to
the second, and back, for every K outputs of
the y(n) signal. The polyphase filter
coefficients are updated every time the
"commutator" takes the top position. The
resulting structures for different values of
ko and no are similar, and they will not be
addressed here.
Generation of the real FDM sianal
Thus far, the K channel complex modulation
process has been described. In order to
produce the SSB modulated assembly, the
complex conjugate of the complex FDM signal
must be generated. To achieve this a complex
conjugate modulation process of the K complex
conjugate input signals can be used. The
mathematical expression of the real output
signal is:
yn (rL+p) = y(rL+p) + y* (rL+p)
K- 1
= Z h(rL+p-mK)f(r) Z Xk(m)t(k,p)
m k=O
K-1
+ Z h* (rL+p-mK)f' (r) Z X * k (m)t*(k,p).
m k=O
Whenever the interpolating filter is real,
the filtering process for both terms in eq. 7
becomes the same and therefore it is possible
to add the output of the two transforms and
filter the resulting signal. Eq. 7 can be
rewritten as:
yn (rL+p) = Z h(rL+p-mK) X R " (m)
m algorithms using wide transition band
where XRP (m) = xp (m) + x A P(m)
R-1 K-l
= Z X k (m) t(k,p) + Z X * k (m) t' (k,p).
k=O k=O
Then, depending on the characteristics of the
modulating function t(k,p), the two K point
complex transforms can be reformulated in
order to reduce the computational
requirements. For example, two K point
standard DFTs can be combined into one 2K
point standard DFT (31.
317
It is interesting to note that in the
particular case where X k (m) is real, since
the input to the two branches corresponding
to sach channel is the same, both branches
can be combined into one where the modulating
function becomes real:
K-1
X R P(ml = 2 I Z Xk (n) RE[t(k,p)] I (10)
k=O
If the filter h(n) has real coefficients, it
transpires that all the operations are real.
In the case of t(k,n)=EXPlj(2n(2k+l)n/2K)I,
then RE[t(k,n)] = COS(2n(2k+l)n/2K), and so
eq. 10 represents a K point discrete cosine
transform. Narasimha and Peterson (4),
Marshall ( 5 ) , Yam and Redman ( 6 ) and knsari
and Liu ( 7 ) have followed this approach.
Using the Weaver modulator interpretatian, a
further alternative for computing X K (m)
~ can
be obtained. Expressing eq. 9 as a function
of the real and imaginary parts of Xh (m) and
t(k,n), results in:
K-1
XRP (m) = 2 Z f RElxr (m)I RE[t(k,p)l
k=O
- IM[xp (m)I IM[t(k.p)l I . (11)
Therefore, the two K point complex transforms
can be substituted by two different K point
real transforms which operate upon the real
and imaginary parts of thc. complex input
signals x k (m).
----
CHOICE OF THE FASTEST TRANSFORM APPROACH
As explained in the previous sections. the
SSB modulating process has been reduced to
the bandpass filtering of the interpolated
X k (n) signals and the resulting filter bank
has been decomposed into a polyphase network
and a fast transform processor.
The operations involved in the polyphase
network are defined by the interpolating
filtar Characteristics which depend mainly
upon the transition band allowed in the
filter design. For example, the algorithms
using the input signal spectra shown in fig.
1 . c and l.f, can use low order wide
transition band filters, while the rest cf
the spectra require high order, narrow
transition band filters. The former case
(7) represents a shift of the computational load
from the polyphase network into the input
signal pre-processing; that is, while narrow
transition band filters perform the multiple
function of selecting one harmonic image, and
one sideband within that image, followed by
sDectral limiting of the sideband to the
frequency range of 300-3400 Hz (300-3700 Hz
for real prototype filters), these functions
(8) are distributed over two stages in the
filters. In the pre-processing, a filter
(operating at an 8 kHz rate) rejects the
unwanted sideband and shapes that desired. In
the second stage, the interpolating filter
(9) selects the desired image (or complex
conjugate images in S S B r,oduiation).
A restriction on the use of wide transition
band filters is that they must operate on S
kHz rate signals, making it impossible to
take advantage of the saving resulting from
the 4 kHz rate input signals. On the other
hand, they make possible the use of FIR
interpolating filters with a realizable
 318
number of taps, which generate small round-
off noise and require only small word length
representation (an important issue in
hardware implementations). In 1981,
Wakabayashi et a1 (8) built a 24 channel
TMUX using a wide transition band 56 taps
real FIR filter which enabled the
implementation of the translation algorithm
with 16 bit digital signal processing.
Another alternative is the use of either real
or complex coefficient filters. In the latter
case, the frequency response need not be
constrained by any symmetry requirements;
consequently the passband is defined to span
the exact range of 300-3400 Hz and the
stopband is optimally specified to reject
specific frequencies. Following this approach
Rossiter et a1 ( 9 ) designed a 24 channel TMUX
using a sixth order wide transition band IIR
filter.
In general, while the efficiency of the
alternative algorithms based on narrow and
wide transition band filters is approximately
the same, those using complex coefficients
require far more operations than those with
real coefficients. Furthermore, if in the
generation of the real FDM signal a real
filter is used, the two required complex
discrete transforms can be combined into one.
Then, if the input signals are real, i.e. the
filter is of the narrow transition band type,
the two K point complex discrete transforms
can be substituted by a K point real
transform. Therefore, the real narrow
transition band filters give more flexibility
in the transform processor design when
generating a real FDM signal.
With reference to the fast transform
processor, as indicated above, the possible
alternatives for the generation of the real
or complex FDM signal are:
- one K point complex transform
- two K point complex transforms
- one 2K point complex transform
- one K point real transform
- two K point real transforms,
where the complex transforms operate over
either real or complex data and the real
transforms operate over real data.
The efficiency of the fast transform
processor and, therefore of the TMUX, depends
on the efficiency of the fast algorithm
available for computing the transform. For
example, in 1974 Bellanger and Daguet ( 3 )
used a 2K point standard FFT because at that
time it was the fastest algorithm available
for computing a discrete transform of the
Fourier type. Since such a transform produces
an even stacking arrangement, the input
channel had to be pre-processed (i.e. shifted
by 2 kHz), and therefore translatsd into
complex form. The use of a 2K point transform
over a K point one allowed the complex
processing in the polyphase network to be
avoided and so to lead to a real FDM signal.
The interpolating filter was real with a
narrow transition band.
Subsequently, Bonnerot et a1 (101, using the
same interpolating filter, produced an odd
stacking arrangement, avoiding the pre-
processing of the input signals. To achieve
this, a 2K point 02DFT with a real symmetric
input was used, where the kernel of the
transform was the function:
t(k,p) = EXP[jn(2k+l) (2n+l)i4KI.
Taking advantage of the fact that the input
signals were real, the two 02DFT's required
for the modulation and demodulation were
reformulated into a K/2 point standard DFT
with extra pre- and post-processing.
Equivalent algorithms were obtained by
exchanging the complex 02DFT with the DCT (4-
7 ) . producing algorithms of equivalence
efficiency.
The application of a fast DCT algorithm
(Malvar (111,Vetterli and Nussbaumer (12)
which makes the resulting TMUX design faster
than those previously published is suggested
by the authors. The 64 point DCT transform
over a real input sequence requires 512
additions and 193 multiplications, while the
previously faster DCT required 547 additions
and 227 multiplications (12). The relation of
these savings to the total modulation process
will be commented on in the next section.
It should be noted that in the above
mentioned three previously fastest
algorithms, none have generated a complex FDM
signal since methods following this approach
need complex arithmetic in a specific stage
of the modulation process and so, a far
greater number of operations is required than
when real arithmetic is employed.
THE HARTLEY TMUX APPROACH
An original algorithm based on a different
transform is now presented.
Considering the modulating function:
t(k,n) = EXP[2njkp/K +n/41
eq. 11 can be rewritten as:
K- 1
Xup (m) = 2 1 Z RE[Xk (m)1 COS (2nkp/K+n/4)
k=O
-IM[xk (m)] sin(2nkp/K+n/4) I
= xupi (m) - X R P Z (m) 121
where :
K- 1
Xupl (m)= J2 I X RE[Xk (m)] cms(2nkp/K) 13 1
k=O
K-1
X U ~ (m)=
Z J 2 f X IM[xr (m)] cas(2nkp/K), 14)
k=O
and,
cms wt = (cos wt - sin wt)
cas wt = (cos wt + sin wt)
are sinusoidal functions in phase quadrature.
Therefore, X U P (m) is obtained through the
addition of two real transforms: X R P I (m) and
X R P Z (m). Neglecting scaling factors, the
second describes a Hartley transform and
several fast algorithms have already been
suggested for it's computation ((ll),
Sorensen (13)). Adopting one of them, a fast
algorithm for the first, which will be called
the CMS transform, has been developed by the
authors, requiring the same amount of
computation as a Hartley transform (which in
the case of a 64 point transform needs 124
multiplications and 416 additions) - Therefore
the total amount of operations for computing
X R ~(m) is 248 multiplications and 832
additions, which is greater than those
required in the DCT.
 31 9
However, the advantage of a Hartley TMUX
approach arises from the fact that the
Hartley and CMS transforms can be computed at
a decimated sampling rate (i.e. 4 kHz)
without requiring filtering or time consuming
pre-processing of the real signals. Since the
modulating function used leads to an even
stacking arrangement, the channels must be
shifted by 2kHz. If a complex modulation of
the form EXP [jnt/2] is used in order to
produce the spectrum shown in fig. l.d, the
real part of the complex sequence xh (m) is
obtained by multiplying the pk (VI input
signal by the sequence { l , O , - l , O , i ,. . . I , and
the imaginary part is obtained by multiplying
it by iC,1.0,-1,0,... I . Therefore whenever
RE[xk (m)] is nonzero, I M [ X k (m)] is zero. and
vice versa. This means that X R P (m) is the
result of an interleaving of the outputs of
two different transforms functioning at a
decimated rate of 4 kHz.
The Weaver modulator interpretation
represents an interesting perspective of this
new approach. Considering the input sequences
pk (VI as a combination of two 4kHz rate
interleaved Sequences, X I k (m)=pk ( 2 v ) and
x2 k (m)=pk (2v+1), then the real part of x k ( V I
is equal to x ~ (m)h (-l)m interpolated by a
factor two, and the imaginary part of X k (VI
is equal to x ~ (m)k (-lIm interpolated by the
same factor, where m is the index of the
decimated sequences. Substituting this
interpretation of the pk (v) signal pre-
processing in the block diagram of the Weaver
modulator shown in fig. 2.c, the structure
illustated in fig. 4.a results, where the
cascade of the 2 and M factor interpolators
has been substituted by a 2M factor
interpolator. Fig. 4.b shows the general
block diagram of ths resulting fast algorithm
dascribed by eq. 8 and 12.
Using the same modulating function, but with
a = -ni4. a similar structure is obtained.
Table 1 shows the number of operations
required in the fast transform unit of the
modulation and demodulation processes
corresponding to a 60 channel TMUX using the
the following transforms: the OZDFT (which
corresponds to the fastest published TML'X
( 1 0 ) ) , the latest DCT reported in the
previous section, and the Hartley and CMS
transforms here presented. Since the
interpolating filter specifications in the
three TMUX approaches are the same, the
number of computation in table 1 are
proportional to the total number of
operations in the TMUX algorithms.
ADDITIONS MULTIPLICATIONS
O2 DFT 7,168,000 6,208,000
(100%) (100%)
DCT 8,208,000 3,072,000
(114%) (50%;
HARTLEY 6,656,000 1,364,000
(93%) (32%;
TkBLE 1- Number of operations & the fast
transform Drocessor of both
modulatinq/demodulatina Drocesses.
As can be seen above, the introduction of the
DCT halves the number of multiplications in
the transform processor of the TMUX, but
increments by 14% the number of additions.
Conversely, the fast transfor-m processor of
the new Hartley TMUX approach achieves a
reduction in the number of additions to 93%
and multiplications to 32% over the OZDFT.
Since the fast transform block in TMUX
approaches using an IIR interpolating filter
represents approximately 30% of the total
number of multiplications, then the
introduction of the DCT algorithm will reduce
them by 15% in the overall modulation
process, while the new Hartley method will
reduce them by 20%.
CONCLUSIONS
The voice channels SSB modulation process was
presented as a particular case of complex
modulation, which was reformulated a5 a
critically sampled bandpass filtering
process. An efficient structure of the
bandpass filter bank, i.e. its decomposition
into a polyphase network and a discrete
transform processor was described. After
considering the alternatives for the
polyphase network and the discrete transform
processor, a review of the most efficient
contemporary TMUX algorithms was presented. A
saving of 50% of the multiplications required
in the fast transform processor through the
application of the latest DCT algorithm to
established TMUX approaches is suggested.
Finally, a new TMUX approach based on the
Hartley and CMS transforms was presented,
which improves upon the efficiency of all
previously published TMUX algorithms.
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1. Scheuermann H., and Gockler, H., 1981,
Proc. IEEE, 6 9 , 1419-1450.
2. Crochiere, R.E., and Rabiner, L.R., 1983,
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Jersey.
3. Bellanger, M.G., and Daguet, J.L., 1974,
IEEE Trans. on COM, 22, 1199-1205.
4. Narasimha, M.J., and Peterson, M.D., 1979,
IEEE Trans. on ASSP. 2 7 , 752-762.
5. Marshall, .T.G., 1981, Int. Conf. Comm.
Denver, Colorado, 7.7.1-7.7.4.
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7. Ansari, R., and Liu, B.. 1982, IEEE Trans.
on COM, 30, 1569-1574.
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M.I., and Humphrey, L.D., 1982, IEEE
Trans. on COM, 30, 1540-1551.
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and Nussbaumer, H.I., 1984,
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_ ___
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 320

I a)

PK-1

IM [Xk (m)1 I
C) IM [t(k,n)l

Figure 1.a Basic block diagram of the K Figure 2.a Complex modulator interpretation
channel SSB modulator. Figure 1.b - 1.f Figure 2.b Bandpass filter interpretation
Alternative spectra for the Xk(m) signal. Figure 2.c Weaver modulator interpretation

cms (2nkn/K)

xo Yk (n)
x1

XK-1 f2K h(n)

(-1). cas (2nkn/K)

Figure 4.a Weaver modulator structure working

Figure 3.a Modulating filter bank for no=O, at the minimun rate (m=v/2). Figure 4.b Basic
ko=1/4 and L=2K. Figure 3.b Commutator model block diagram of the Hartley TMUX approach.
of the polyphase structure.