FrequencyDomain Data Transmission using Reduced ComputationalComplexity algorithms

Abraham Peled and Antonio Ruiz

Computer SciencesDepartment, IBM T.J. Watson Research Center, P.O. Box 218, Yorktown Heights, N.Y. 10598 N-I Let us also define the complexsequence {D5}5...0 Abstraa

In this paper we describe a frequency domain data transmission method to be used for digital data transmission over analog telephonelines which exploits recently derived reduced computational complexity algorithms,such as the Winograd Fourier Transform, to achieve a significantlylower computational rate than comparable time domain QAM modems implementeddigitally using signal processing techniques. In addition to the lower computational rate, the proposed method also allows for better channel bandwidth utilization by allowingoptimal signal power allocation based on the channel's signal to noise versus frequency characteristics. Experimental results on this method are presented, indicatingthat it may be possible to wnd over 10,000 BPS over an unconditioned
telephone line while maintaininga

n0

A5+jB5
0 AN_fl.-JBN_C

0<n1 2

(1)

f<nNt
(2)

This construction insures that the IDFTof D,J is a real sequence since {D5j {DNfl} ). Therefore in figure (1), N-I N

i0 BER.

x=DejL / 5
n=O

is real, and x(t) is obtained by passing {x1} through a D/A converter, at a rate of Hz., and a low-pass filter,

I. Introduction
We describe here a frequency domain data transmissionscheme in which the modulation process is achieved by an IDFT and the demodulation process by a DFT, using reduced computational (RCC) complexity algorithmssuch as the WinogradFourier transform (WFT). This type of system has been proposed before by Weinstein and Ebert [1]. However, at that time, it could only have beenimplemented in real-time thru the use of' special purpose hardware. With the wide use of digital signal processing techniques,therehas been some emphasis in using recently developed RCC algorithms for implementing signal processing algorithms in real-time in general purpose processors. The processors that we use to implement this system, although not restricted to work only with RCC algorithms, take advantage of them to perform with an attractive cost/performance ratio, A brief description of this processor and its environmentcan be found in Appendix A. We will describe in this paper the basic theory behind the implementation of FDDT over an analog telephone channel. And we will also show how the data transmitted can be allocated according to the channel characteristics so as to maximize the usage of the channel.

N-I

x(t) 0<t<NT,

n=O

Nt

e j2,t (3)

T=*

equivalently,x(t) is a sum of sinusoidal signals whose amplitudes are determined by the sets of data values {A5} and {B5} , as depicted in figure (2). This interpretation makes the transmitter an implementation of a frequency division multiplexing (FDM) system [lj

fn=

If the signal x(t) is transmitted thru a linear channel withimpulse

response h(t) and frequency response H(w) , then the signal at the output of the channel s(t) is determined by the convolutionof x(t) and h(t) , or in the frequency domain by S(w) H(w).X(w).

It appears that at the receiver end, to recover {D5} , we should compute S(nO) ( where 0 = 27TF/N), that is the sampledversion of S(w) , and divide it by H(ntl) , whichcan be obtained

II. FrequencyDomain DataTransmission A) Theory The basic idea is to transmit thru the telephone channel, on a frame by frame basis, multiple low speed channels multiplexedin the frequency domain.
Let and be two data sequences,where A5 and B5 can take on any of a set of 2'' values or constellation points( where r5 is a function of n ).

during an initialization procedurethat will be described later. The idealized formulation above does not hold unless additional steps are taken to insure that certain conditions apply. K-I If we let {h5}5._0be the impulse response coefficientsthat characterize the channel, then {s} is given by K-I (4) S =0h1x,_1 This formulation disregards any channel effects such as: phase modulationjitter, different types of noise, samjitter, amplitude pling jitter, and also differences in clock rates between the receiver and the transmitter. Although many of these effects will be present, we will ignore them for the moment and continue to describe the operation. The box labeled "cyclic extension performs the following map-

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pingon {xj1o (where k denotes the kth frame )

And then taking the K point IDFTof {Or} we obtain {o }

KI

,where

x
{

{hj_p}j=0 (5)
Now, since

K-I

N 1 L Thus, each frame has a length of L = Ni-K-i because of the

the cyclicextension.This causes sequence {x}.0 to appear periodic to a channel with memoryof K samples ( at thesampling rate ), and therefore provides the required conditions for the discrete convolutiontheorem to hold.
L

a physicallyrealizable bandpass channel has a characteristic peak in the middle of its impulse response, we determine
m (the shift) by simply looking for MAX { ABS {o}o
KI

}-

In the receiver, the last K-i terms in each frame are ignored after
synchronizationis achieved. Thus

This has established frame synchronization. It should be noted here that any small error in synchronization( i.e., a few samples will be compensated by the equalizer. The receiver continues to take K point DFT's ( with the new frame and subsequence boundaries ) until we do not get { anymore, at which time we know that the next frame has begun. The receiver takes the N point DFT of the first N points inside

Tj

v =s
NI

(6)

Taking the DFTof {v1} to achieve demodulationwe obtain {Q}

this newly synchronizedframethus obtaining , and proceeds to determine the initial equalizer setting by dividing as
follows,

N-I {Rj

v1 e

.2irlp N

NI

N-i
=

{LT}0

{R},
(7)
I

N-I

(9)

From fig. (1), the equalizer {C) multiplies {Q} to obtain

(D,}

= {Q}. {C,}
N NI

where {LJ} is the DFTof the N point PN sequence above, already storedas reference in the receiver. Having calculated {C} , the initial equalizer setting, we have completed the initialization procedure in only two frames duration.

where, it can be shown that receiver's estimate of the transmitted data B) Initialization

,i.e. {Dlis the

C) Equalizer Update
The equalizer obtained in equation (9) has to be updated in every frame to account for the second ordereffects mentionedin section (ha). However, our initial strategy for updating has originated from the fact that the predominant effect occurring in the system is the clock differencesbetween transmitter and receiver. Overall, for a net error in the sampling clocks, the received unequalized bins for the next frame are given by, where
number.
N (10) {Q} = {Q) eon frame length and clock error, is a constant dependent

The objectives of the initialization procedure are two: synchronization, and initial equalizer setting. The transmitter sends two frames for initialization purposes. Each frame is L samples long, where K of these samples are the cyclic extension as in the mapping of equation (5), and L = K+N-1 asbefore.Also, N is the size of the IDFTand DFT used in the modulation and demodulation processes respectively,

and the angle (2ii/lp)/Nis linearly related to

p the frequency bin

The first frame consists of M identical subsequencesof length K

= N/M , where each subsequence is a pseudo-noise (PN)

Se-

quence whose power spectrum is approximately flat. The second frame consists of a length N PN sequence. The receiver has stored in its memory, as reference, the corresponding complex values of the K and N point DFT's of these two PN sequences.

Using a simple approach to the problem, the formulation for the equalizer update goes as follows. An error signal is formed for every frame, that is { as in figure (1)

r}

(11)

To start the operation, the receiver monitors the channel for energy. Once a preset treshold is exceeded, it assumes that initialization has started and proceeds to take a K point DFT of the received first K point subsequence inside the first received frame. Since the subsequence frame boundary is out of synchronization, the receiver picks-up a time shifted version of the response of the channel to the repeated PN subsequence.

Where

denotes the received constellationpoints after equaliza-

tion and D denotes the decoded constellationpoints. We also define the updated equalizer for the k + 1n frame

Dk k

(12a)

If {t,},.. is the originalPN subsequenceand

thenthe K point DFTtaken

KI

Ki
Ki

its DFT,

where denotes the received constellationpoints before equallzation. Using eqs. (ii) and (7), equation (12a)becomes

by the receiver (V,},...0is also

where

(8) Hr Vr==Tre m is the time shiftand {Hr} is the DFT of the channel's

_j2!m K

c+i

________

k Q Cj,, k Qk p
.

discrete impulseresponse characteristic {h.} ( at intervals r ._.!. ).

By dividing V by Tr we obtain

F K

(1

f)

(12b)

Hr 965

This equation has been empiricallychanged to contain the factor

a ( 0<a1.0)

thusbecoming

tographs of various constellations as the modem was running over a short telephone switched line loop.
a

= C (1.0

(12c)

It can be shown that a provides the best results when set to

higher values for the first few frames and then a lowest steady state value later. This holds true even when the initial equalizer setting is obtained during reasonably noisy channel conditions. Additional work is required on equalizer updating to compensate for other channel impairments.

Ill. Communications aspects of the system In this section we discuss varying the average power and number of different levels per bin to optimally utilize the channel characteristics.

Preliminary results also indicate that we are able to run at a bit at about23 db S/N ratio. For all error rate of better than of the above the processors are doing all the computations in about 60 percent of real-time while using about 65 percent of their data and instruction storage. The total number of operations is approximately60,000 multiplies per second (MPS) and 140,000 additions per second (APS). This compares favorably with approximately 600,000 MPS and 600,000 APS for the digital implementation of the 9,600 BPS quadrature amplitude modulation (QAM) time-domainmodem.

l0

In summary,we have implemented a system that takes advantage of recently developedRCC algorithms in digital signal processing
to transmit more data more efficiently ( i.e. less computations over an unconditioned analog telephone channel, while at the same time taking advantage of the channel's own S/N ratio characteristics to improve the bit error rate performance.
Appendix A

Based on the spectral characteristics of the telephone channel, as obtained in the last survey on analog transmissionperformance of telephone lines [2] , allocation of frequency bins can be made as shown in fig. (3). This allocation is based on = 8khz. , and N = 240 the size of the IDFT and DFT. Accordingly,120 bins are available,out of which, frequencyregion 0 to 234 Hz. will not be used because of the inherent presence of 60 Hz. and its first two harmonics, and also low S/N ratio characteristics. And frequency region 3400 to 4000 Hz. will not be used because of insufficient S/N ratio characteristics for efficient equalization. These constraints translate into having bins 8 thru 102 available

for transmission.

As we indicated in section (ha), each D in eq. (1) can take on any of a set of 2'" complex values. This allocation can be made such that the number of complexlevels and the power on each bin will follow the optimal power allocation strategy as required by the "water filling theorem" of communicationtheory, potentially being able to maximize the utilization of the channel bandwidth [5] . Althoughin our implementationr, is an integer, it need not be that way, and in theory, we can come closer to using the maximum capacityof the channel much betterif r, is allowed to be non-integer also. In summary,we have the capability to allocate a different number of levels to the various spectral bins, thus having more in thoseD 'a in the midband where the S/N ratio is highest, and less in those areas where the S/N ratio is lowest.
The above approach also allows the usage of more of the channel bandwidth while minimizing the noise enhancement problem associated with equalization in time domain data transmission schemes.

The research signal processor (RSP) is a Schottky-TTL imple mented microprocessor specially designed for general signal processing applications. This processor has a 16 bit fixed point arithmetic unit, 64 machine instructions, and a 4 stage pipeline. Storage capabilitiesare 4K. of data store and 5K. of instruction store. An important characteristic is its capability to exploit reduced computational complexity algorithms to achieve high throughput and to efficiently decompose multiplications by constants into shifts and adds using canonical signed digit representation of numbers [4]. Furthermore, synchronousand asynchronousI/O are handled under program control. The programs for the RSP are all written in the processor's own "high level" assembler language, which is facilitated by its single operand instruction format. The RSP runs under host control, mainly program loading and/or data initialization,after which the RSP can carry out a complete signal processing applicationwithout host intervention. Host control is not needed if programs are made to reside in read only instruction storage. References

(1) S. B. Weinstein and P. M. Ebert, "Data Transmission by Frequency-DivisionMultiplexing Using the Discrete Fourier Transform," IEEE Trans. on Comm. Tech., Vol. COM-19, (2) F. P. Duffy and T. W. Thatcher, Jr., "Analog Transmission Performance on the Switched Telecommunications Network," The Bell System Technical Journal, Vol. 50, No. 4, April
1971.

No. 5, October 1971.

IV Experimentalresults,Conclusion

An FDDT systemwas first simulated in fixed-point PL/I on the IBM 370/168 to test all the algorithms,includingthe RCC algorithms. The 40 point and 240 point DFT's and IDFT's used for the simulation and the real-time implementation were WFTA's programmedas in reference [3]. OtherRCC algorithms needed were complex multiplication, complex division, and also scaling

(3) Fl. F. Silverman, "An Introduction to Programmingthe Winograd Fourier Transform Algorithm (WFTA)," IEEE Trans. on Acoustics, Speech, and Signal Processing, Vol. ASSP-25, No. 2, April 1977. (4) A. Peled, "On the Architectural Implications of Recent Reduced Computational Complexity Signal Processing Algorithms,' IBM Israel Scientific Center Technical Report 055, August 1977. (5) R. G. Gallager, "Information Theory and Reliable Communication," Wiley & Sons pub., 1968, pp. 344,389.

and normalization.

The real-time implementationwas programmed on two RSP's, one RSP was programmedas a receiver and the other as a transmitter. Tests and preliminary evaluations of this system were made by

running it on a telephone line impairment simulation instrument, and on shorttelephone line loops ( < 50 miles ) over the switched network. Results obtained for the 11,886 bps. modem as in fig. (3) are illustrated in fig. (4) which shows the corresponding pho-

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TRANSMITTER CYCLIC

H ENCODER HIIDFTF4

RECEIVER

(1.0

Figure

1. FrequencyDomain DataTransmission system

Figure 2. Frequencydivision multiplexing interpretationof system

AVG. NO. OF BITS PER BIN
SAND

0
4

E
4 4

sis/sEc.
8800
9943

0
0

0
3

4
4

0 2

0
0

I: ft

:::

DR

Figure 4. Constellations for the 11,886 bps modem whilerunning on a telephoneline

0.0 0.1

02

0.3

0.4

0.5

TELEPHONE CHANNEL

5hz,

3. Figure Telephonechannel,allocation of frequencybins

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