|
Claims  |
|
|
I claim:
1. A method of linearizing an amplifier to produce an amplified output
sample v.sub.a in response to a predistorted input sample v.sub.d derived
from an input modulation sample v.sub.m, such that v.sub.a
.perspectiveto.Kv.sub.m, where K is the desired constant amplitude gain of
said amplifier, said method comprising the steps of:
(a) deriving the squared magnitude x.sub.m of said input modulation sample
v.sub.m ;
(b) selecting, from a table containing N.sub.t values F.sub.i where i=0, 1,
. . . , N.sub.t -1, each of said entries corresponding to a squared
magnitude values x.sub.mi, and for each of which entries F.sub.i
G(x.sub.mi).vertline.F.sub.i .vertline..sup.2).perspectiveto.K where G(x)
is the complex gain of said amplifier, a table entry F.sub.i for which
x.sub.mi is: (i) the largest table entry less than or equal to x.sub.m ;
or, (ii) the smallest table entry greater than or equal to x.sub.m ;
(c) deriving said predistorted sample v.sub.d as v.sub.d =v.sub.m F.sub.i
in rectangular coordinates, viz:
Re(v.sub.d)=Re(v.sub.m)Re(F.sub.i)-Im(V.sub.m)Im(F.sub.i)
Im(v.sub.d)=Re(v.sub.m)Im(F.sub.i)+Im(V.sub.m)Re(F.sub.i)
where Re(x) is the real component of x and Im(x) is the imaginary
component of x; and,
(d) driving said amplifier with said predistorted input sample v.sub.d.
2. A method as defined in claim 1, further comprising sequentially
repeating each of said steps k times and, after each derivation of said
predistorted sample v.sub.d (k), performing the further steps of:
(a) deriving a sample v.sub.a (k) of said amplifier's output;
(b) deriving an error sample e(k)=(v.sub.a (k)-Kv.sub.m (k)v.sub.m.sup.*
(k)
(c) deriving an adjusted value F.sub.i (k+1) of the table entry F.sub.i (k)
selected during the k.sup.th selecting step, where:
##EQU13##
where .alpha. is an appropriately chosen constant; (d) replacing said
table entry F.sub.i (k) with said adjusted value F.sub.i (k+1); and,
(e) sequentially repeating all of said steps.
3. A method of linearizing an amplifier to produce an amplified output
sample v.sub.a (k) in response to a predistorted input sample v.sub.d (k)
derived from an input modulation sample v.sub.m (k), such that v.sub.a
(k).perspectiveto.Kv.sub.m (k), where K is the desired constant amplitude
gain of said amplifier and k denotes the k.sup.th such samples, said
method comprising the steps of:
(a) deriving the squared magnitude x.sub.m (k) of said input modulation
sample v.sub.m (k);
(b) selecting, from a table of N.sub.t entitles F.sub.i where i=0, 1, . . .
, N.sub.t -1, each of said entries containing a squared magnitude value
x.sub.mi, and for each of which entries F.sub.i
G(x.sub.mi).vertline.F.sub.i .vertline..sup.2).perspectiveto.K where G(x)
is the complex gain of said amplifier, a table entry F.sub.i (x.sub.m (k))
for which x.sub.mi is:
(i) the largest table entry less than or equal to x.sub.m (k); or,
(ii) the smallest table entry greater than or equal to x.sub.m (k);
(c) deriving said predistorted sample v.sub.d (k) as v.sub.d (k) = v.sub.m
(k)F.sub.i (x.sub.m (k)) in rectangular coordinates, viz:
Re(v.sub.d (k))=Re(v.sub.m (k))Re(F.sub.i (k))-Im(v.sub.m (k))Im (F.sub.i
(x.sub.m (k)))
Im(v.sub.d (k))=Re(v.sub.m (k))Im(F.sub.i (x.sub.m (k)))+Im(vhd m(
k))Re(F.sub.i (x.sub.m (k)))
where Re(x) is the real component of x and Im(x) is the imaginary
component of x;
(d) driving said amplifier with said predetermined input sample v.sub.d
(k);
(e) incrementing k by 1; and,
(f) sequentially repeating steps (a) through (e).
4. A method of linearizing an amplifier to produce an amplified output
sample v.sub.a (k) in response to a predistorted input sample v.sub.d (k)
derived from an input modulation sample v.sub.m (k), such that v.sub.a
(k).perspectiveto.Kv.sub.m (k), where K is the desired constant amplitude
gain of said amplifier and k denotes the k.sup.th such sample, said method
comprising the steps of:
(a) deriving the squared magnitude x.sub.m (k) of said input modulation
sample v.sub.m (k);
(b) selecting, from a table of N.sub.t entries F.sub.i (k) where i= 0, 1, .
. . , N.sub.t -1, each of said entries containing a squared magnitude
value x.sub.mi (k), and for each of which entries F.sub.i (k)G(x.sub.mi
(k).vertline.F.sub.i (k).vertline..sup.2).perspectiveto.K where G(x) is
the complex gain of said amplifier, a table entry F.sub.i (k) for which
x.sub.mi is:
(i) the largest table entry less than or equal to x.sub.m (k); or,
(ii) the smallest table entry greater than or equal to x.sub.m (k);
(c) deriving said predistorted sample v.sub.d (k) as v.sub.d (k)= v.sub.m
(k)F.sub.I (k) in rectangular coordinates, viz:
Re(v.sub.d (k))=Re(v.sub.m (k))Re(F.sub.i (k))-Im(V.sub.m (k))Im( F.sub.i
(k))
Im(v.sub.d (k))=Re(v.sub.m (k))Im(F.sub.i (k))+Im(V.sub.m (K)) Re(F.sub.i
(K))
where Re(x) is the real component of x and Im(x) is the imaginary
component of x;
(d) driving said amplifier with said predistorted input sample v.sub.d (k);
(e) deriving said amplified output sample v.sub.a (k);
(f) deriving an error sample e(k)-(v.sub.a (k)-Kv.sub.m (k))v.sub.m.sup.*
(k);
(g) deriving an adjusted value F=hd i(k+1) of said selected table entry
F.sub.i (k), where:
##EQU14##
where .alpha. is an appropriately chosen constant; (h) replacing said
table entry F.sub.i (k) with said adjusted value F.sub.i (k+1);
(i) incrementing k by 1; and,
(j) sequentially repeating steps (a) through (i).
5. A method as defined in claim 1, wherein said selecting step further
comprises selecting that table entry F.sub.i for which the absolute value
.vertline.x.sub.m -x.sub.mi .vertline. is minimized with respect to the
table index i.
6. A method as defined in claim 3, wherein said selecting step further
comprises selecting that table entry F.sub.i (x.sub.m (k)) for which the
absolute value .vertline.x.sub.m -x.sub.mi .vertline. is minimized with
respect to the table index i.
7. A method as defined in claim 4, wherein said selecting step further
comprises selecting that table entry F.sub.i (k) for which the absolute
value .vertline.x.sub.m (k)-x.sub.mi (k).vertline. is minimized with
respect to the table index i.
8. A method of linearizing an amplifier to produce an amplified output
sample v.sub.a in response to a predistorted input sample v.sub.d derived
from an input modulation sample v.sub.m, such that v.sub.a
.perspectiveto.Kv.sub.m, where K is the desired constant amplitude gain of
said amplifier, said method comprising the steps of:
(a) deriving the squared magnitude x.sub.m of said input modulation sample
v.sub.m ;
(b) deriving, by interpolation on a table containing N.sub.t values F.sub.i
where i=0, 1, . . . , N.sub.t -1, each of said entries corresponding to a
squared magnitude value x.sub.mi, and for each of which entries F.sub.i
G(x.sub.mi .vertline.F.sub.i .vertline..sup.2).perspectiveto.K where G(x)
is the complex gain of said amplifier, a value F(x.sub.m);
(c) deriving said predistorted sample v.sub.d as v.sub.d =v.sub.m
F(x.sub.m) in rectangular coordinates, viz:
Re(v.sub.d)=Re(v.sub.m)Re(F(x.sub.m))-IM(V.sub.m)Im(F(x.sub.m))
Im(v.sub.d)=Re(v.sub.m)Im(F(x.sub.m))+Im(V.sub.m)Re(F(x.sub.m))
where Re (x) is the real component of x and Im(x) is the imaginary
component of x; and,
(d) driving said amplifier with said predistorted input sample v.sub.d. |
|
|
|
|
Claims  |
|
|
Description  |
|
|
FIELD OF THE INVENTION
This application pertains to a method of adaptively predistorting a power
amplifier to linearize the amplifier.
BACKGROUND OF THE INVENTION
Mobile communications systems, such as those used for cellular telephone
communication, divide the available frequency spectrum into a multiplicity
of individual signalling channels or frequency bands. Particular channels
are allocated to individual users as they access the system. Each user's
communications are routed by the system through the channel allocated to
that user. Signals broadcast by the system must be carefully regulated so
that they remain within the channels allocated to the various users.
"Out-of-band" signals can spill over from one channel to another, causing
unacceptable interference with communications in the other channels.
To date, mobile communications systems have generally employed frequency
modulation ("FM") techniques which do not require variation of the
amplitude of the transmitted signal. Thus, the frequency of the
transmitted signal carrier changes, but the signal power level remains
constant. Such systems have sufficed for voice communications. However,
there is an increasing desire to expand the capabilities of mobile
communications systems to encompass data as well as voice communications.
Commercially worthwhile data transfer rates require the use of modulation
techniques which are more spectrally efficient than the FM techniques used
for voice communication. This necessitates the use of amplitude modulation
techniques which in turn require linearized modulation.
The electronic amplifiers employed in any communications system inherently
distort signals as they amplify the signals. FM techniques do not suffer
from such distortion because of their constant amplitude. Amplitude
modulation, however, causes the distortion to become dependent on the
input signal, so that the amplifier output signal is no longer simply an
amplified replica of the input signal. Although an amplifier's input
signal may be confined within a particular channel or frequency band, the
distorted, amplified output signal typically includes out-of-band
frequency components which would overlap one or more channels adjacent to
the channel within which the input signal was confined, thereby
interfering with communications in the overlapped channel(s).
Some degree of channel signal overlap due to unregulated amplifier
distortion is acceptable in some cases. However, mobile communications
systems place very stringent restrictions on out-of-band signal emissions
in order to minimize channel-to-channel interference.
To reduce out-of-band signal emissions to an acceptable minimum the
amplifier input signal is conventionally "predistorted" before it is fed
into the amplifier. Before the signal is amplified, an estimate is made of
the manner in which the amplifier will inherently distort the particular
input signal by amplifying that signal. The signal to be amplified is then
"predistorted" by applying to it a transformation which is estimated to be
complementary to the distorting transformation which the amplifier itself
will apply as it amplifies the signal. In theory, the effect of the
predistorting transformation is precisely cancelled out by the amplifier's
distorting transformation, to yield an undistorted, amplified replica of
the input signal. Such amplifiers are said to be "linearized" in the sense
that the output signal is proportional to the input signal, thereby
eliminating the generation of out-of-band components.
Unfortunately, amplifier distortion varies in a complex, non-linear manner
as a function of a wide range of variables, including the amplifier's age,
temperature, power supply fluctuations and the input signal itself.
Accordingly, it is not possible to define a single predistortion
transformation which will cancel out any and all distorting
transformations applied by the amplifier.
One prior art approach to the problem (exemplified by U.S. Pat. No.
4,462,001 issued July 24, 1984 for an invention of Henri Girard entitled
"Baseband Linearizer for Wideband, High Power, Nonlinear Amplifiers") has
been to construct a look up table containing a multiplicity of entries
which define predistortion transformation parameters appropriate for use
with a corresponding multiplicity of different input signals. That is, the
effects of the amplifier's distortion on a range of input signals are
pre-measured, the complementary predistorting transformations
corresponding to each input signal are calculated, and parameters defining
the calculated complementary transformations are stored in the table. In
operation, the fluctuating power level of the signal to be amplified is
continuously measured. The power measurement is then applied to the
electronic embodiment of the table, from which the corresponding
predistortion parameters are derived, so that the input signal sample may
be predistorted before it is fed to the amplifier. However, Girard's
approach accounts only for variation of the input signal, not for
variation of the amplifier's other distorting characteristics. Because the
amplifier's other distorting characteristics in fact vary it is necessary
to continuously "adapt" the lookup table parameters by changing them in
response to changes in the amplifier's other distorting characteristics.
Moreover, Girard's approach is based on separate tables containing
amplitude and phase correction factors. This "polar coordinate"
representation follows naturally from the common practice of representing
amplifier distortion in terms of AM/AM and AM/PM characteristics. So far
as the inventor is aware, all predistortion techniques prior to Girard's
also attempted separate amplitude and phase correction.
Another prior art approach is exemplified by U.S. Pat. No. 4,700,151 issued
Oct. 13, 1987 for an invention of Yoshinori Nagata entitled "Modulation
System Capable of Improving a Transmission System". Nagata uses the real
and imaginary quadrature components of the input signal sample to index
into a lookup table containing predistortion transformation parameters.
The real and imaginary components are each typically defined by at least
10 bits of information. Thus, Nagata employs a 20 bit index, which
requires a lookup table containing 2.sup.20 entries (i.e. over 1 million
entries). The lookup table entries are adaptively changed in response to
variations in the amplifier's distorting characteristics. However, if the
signalling channel is changed (a common occurrence in mobile
communications systems) then every entry in Nagata's lookup table must be
iteratively recalculated. This process can take 10 seconds or longer,
which is unacceptable.
The present invention overcomes the disadvantages of the prior art. By
storing table entries in rectangular coordinate format, it enables the
subsequent predistortion operation to be performed more simply than
Girard's polar coordinate approach. Further, it adapts to amplifier and
oscillator changes, whereas Girard's predistorter does not. In comparison
with Nagata's method, only a comparatively small lookup table is required.
Signal phase rotators (required to stabilize Nagata's circuitry) are not
required. Moreover, the lookup table entries are adaptively changed,
within about 4 milliseconds ("msec.") in response to changes in the
amplifier's distorting characteristics.
SUMMARY OF THE INVENTION
In accordance with the preferred embodiment, the invention provides a
method of linearizing an amplifier to produce an amplified output sample v
in response to a predistorted input sample v.sub.d derived from an input
modulation sample v.sub.m, such that v.sub.a .perspectiveto. Kv.sub.m,
where K is the amplifier's desired constant amplitude gain. The squared
magnitude x.sub.m of the input modulation sample v.sub.m is first derived.
A table entry F.sub.i is then selected from a table containing N.sub.t
values F.sub.i where i=0, 1, . . . , N.sub.t -1. Each table entry
corresponds to a squared magnitude value x.sub.mi ; and, for each table
entry, F.sub.i G(x.sub.mi .vertline.F.sub.i
.vertline..sup.2).perspectiveto.K, where G(x) is the complex gain of the
amplifier. The table entry selected is the one for which the absolute
value .vertline.x.sub.m -x.sub.mi .vertline. is minimized with respect to
the table index i. The predistorted sample v.sub.d is then derived, in
rectangular coordinates, as v.sub.d =v.sub.m F.sub.i, viz:
Re(v.sub.d)=Re(F.sub.i)-Im(V.sub.m)Im(F.sub.i)
Im(v.sub.d)=Re(v.sub.m)Im(F.sub.i)+Im(V.sub.m)Re(F.sub.i)
where Re(x) is the real component of x and Im(x) is the imaginary component
of x.
The aforesaid steps are sequentially repeated k times. After each
derivation of the predistorted sample v.sub.d (k), a sample v.sub.a (k) of
the amplifier's output is derived. An error sample e(k) =v.sub.a
(k)-Km.sub.v (k) is also derived. This facilitates derivation of an
adjusted value F.sub.i (k+1) of the table entry F.sub.i (k) selected
during the k.sup.th selecting step, where:
##EQU1##
where .alpha. is an appropriately chosen constant. Table entry F.sub.i (k)
is then replaced with the adjusted value F.sub.i (k+1), and all of the
steps are again sequentially repeated.
BRIEF DESCRIPTION OF THE DRAWINGS
FIG. 1 is a block diagram of a typical prior art adaptively linearized
amplifier.
FIG. 2 is a block diagram of the mapping predistorter section of a prior
art adaptively linearized amplifier.
FIG. 3 is a graph on which the AM/AM and AM/PM characteristics of a typical
Class AB amplifier are shown with output power (dBm) plotted as the left
ordinate, phase shift (degrees) plotted as the right ordinate, and input
power (dBm) plotted as the abscissa. The solid line plots output power and
the dashed line plots phase shift.
FIG. 4 is a graph on which the complex gain characteristics of a typical
Class AB amplifier are shown with complex gain magnitude plotted as the
left ordinate, complex gain phase (degrees) plotted as the right ordinate,
and input power (mwatts) plotted as the abscissa. The solid line plots
magnitude and the dashed line plots phase shift.
FIG. 5 is a block diagram of a gain based predistorter amplifier
constructed in accordance with the invention.
FIG. 6 is a graph illustrating the manner in which look up table entries
F(x) are optimized according to the relationship F(x.sub.m)G(x.sub.m
.vertline.F(x.sub.m).vertline.)=K.
FIG. 7 is a graph illustrating the performance of the predistorter
amplifier of FIG. 5 in a two tone test, with relative power (dB) plotted
as the ordinate and the ratio f/f.sub.s plotted as the abscissa. The long
dashed line plots the output of the power amplifier ("PA") only with
PBO=0.22 dB; the short dashed line plots the output of the PA only with
PBO=24.6 dB; and, the solid line plots the output of the combined PA and
predistorter ("PD") with PBO=0.22 dB.
FIG. 8 is a graph illustrating the performance of the predistorter
amplifier of FIG. 5 in a noise loading test, with power spectral density
(dB) plotted as the ordinate and the ratio f/f.sub.s plotted as the
abscissa. The line labelled "A" plots the output of the PA with no PD and
OBO =13.2 dB; the line labelled "B" plots the output of the PA with no PD
and OBO=33.7 dB; the line labelled "C" plots the output of the PA with a
32 point PD and OBO=15.0 dB; and, the line labelled "D" plots the output
of the PA with a 64 point PD and OBO=15.0 dB.
FIG. 9 is a graph illustrating the response of the predistorter amplifier
of FIG. 5 to a 16QAM data signal, with power spectral density (dB) plotted
as the ordinate and the ratio f/f.sub.s plotted as the abscissa. The line
labelled "E" plots the output of the PA with no PD and PBO =0.22 dB; the
line labelled "F" plots the output of the PA with a 32 point PD and
PBO=0.22 dB; the line labelled "G" plots the output of the PA with a 64
point PD and PBO=0.22 dB; and, the line labelled "H" plots the output of
the PA with no PD and PBO =30.24 dB.
FIG. 10 illustrates partitioning of the input space (v.sub.m space) of the
mapping predistorter of FIG. 2 into quantization cells.
FIG. 11 is a graph on which the distribution of intermodulation power with
instantaneous output power of the combined amplifier of FIG. 5 and 32
point predistorter is shown, with quantization noise power (mwatts)
plotted as the left ordinate, relative error variance plotted as the right
ordinate, and output power (watts) plotted as the abscissa. The solid line
plots quantization noise power; the short dashed line plots relative error
variance .times.1000; and, the long dashed line plots relative error
variance .times.100,000.
FIG. 12 is a block diagram of a prior art adaptively linearized amplifier
of the type exemplified by U.S. Pat. No. 4,700,151 Nagata.
FIG. 13 is a graph on which the convergence behaviour of the linear and
secant update algorithms is shown, with squared magnitude output error
(watts) plotted as the ordinate, and iteration number plotted as the
abscissa. The long dashed lines illustrate the linear convergence
algorithm's performance at 30 watts. The short dashed lines illustrate the
linear convergence algorithm's performance at 15 watts. The solid lines
illustrate the secant algorithm's performance. The numbers applied to the
dashed lines denote the feedback phase shift whereas those applied to the
solid lines denote the output power in watts.
DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENT
I INTRODUCTION
In the past, both theory and practice of mobile communications have
emphasized constant envelope modulation, such as FM or Gaussian minimum
shift keying ("GMSK"). These techniques allow power amplifiers to be
operated in the nonlinear region near saturation, for power efficiency,
yet they do not generate intermodulation products in nearby channels.
However, continuing pressure on the limited spectrum available is forcing
the development of more spectrally efficient linear modulation methods,
such as 16-ary quadrature amplitude modulation ("16QAM") and quadrature
phase shift keying ("QPSK") with pulse shaping. Since their envelopes
fluctuate, these methods generate intermodulation products in a nonlinear
power amplifier. In the mobile environment, restrictions on out-of-band
emissions are stringent, and the designer is faced with two alternatives:
back off an inefficient Class A amplifier to an even more inefficient, but
linear, operating region; or linearize the amplifier.
The present invention provides a method for linearizing a power amplifier
by predistorting its input. It is particularly well suited to baseband
implementation using digital signal processing techniques, though hybrid
variants are easily concocted. It has a number of advantages compared to
previously published techniques; it is powerful and economical, and it
adapts rapidly to amplifier or oscillator changes.
FIG. 1 shows a generic model for many prior art adaptive amplifier
linearization methods. All signal designations employed herein refer
either to complex baseband signals or the to the complex envelope of
bandpass signals. The linearizer or predistorter 10 creates a predistorted
version v.sub.d (t) of the desired modulation v.sub.m (t), making use of
its measurements v.sub.f (t) of the actual amplifier output v.sub.a (t).
Quadrature modulator 12 creates a real bandpass signal from the components
of v.sub.d (t) for input to power amplifier ("PA") 14. The feedback path
(incorporating bandpass filter 16) directs a portion of the real bandpass
PA output to quadrature demodulator 18 for recovery of the complex
envelope. Its output v.sub.f (t) is a scaled, rotated, and possibly
delayed version of v.sub.a (t). Note that the same oscillator 20 is used
in up and down conversion for coherence, and that some methods require a
phase shifter 22 for stability.
Linearization by Cartesian feedback [see generally Y. Akaiwa and Y. Nagata,
"Highly Efficient Digital Mobile Communications with a Linear Modulation
Method", IEEE J. Sel. Areas in Comms., vol SAC-5, no 5, pp 890-895, June
1987; A. Bateman, D.M. Haines, and R.J. Wilkinson, "Linear Transceiver
Architectures", Proc. IEEE Vehic. Tech. Conf., pp 478-484, Philadelphia,
1988; and, S. Ono, N. Kondoh and Y. Shimazaki, "Digital Cellular System
ith Linear Modulation", Proc. IEEE Vehic. Tech. Conf., pp 44-49, San
Francisco, 1989] has the great virtue of simplicity; the amplifier input
complex envelope v.sub.d (t) is proportional to the difference between the
desired v.sub.m (t) and the measured amplifier output v.sub.f (t), as in a
classical control system. However, its linearity and its bandwidth (i.e.
the gain-bandwidth product) are critically dependent on loop delay.
Linearization by Cartesian feedback is therefore ineffective for
travelling wave tube amplifiers, or if there is additional analog or
digital filtering at any point in the loop. Moreover, the stability of a
system dependent upon linearization by Cartesian feedback depends on
precise adjustment of the phase shifter, an adjustment that depends on the
current channel in use.
A more robust alternative to Cartesian feedback is a predistorter ("PD"),
in which the linearizer produces v.sub.d (t) by applying v.sub.m (t) to a
memoryless nonlinearity complementary to that of the power amplifier.
Feedback is used only for adaptation of the predistorter nonlinearity,
rather than for real time calculation of v.sub.d (t). Because it is
insensitive to loop delay, predistortion has attracted considerable
attention.
Several prior art predistorters [e.g. M. Nannicini, P. Magni, and F.
Oggioni, "Temperature Controlled Predistortion Circuits for 64 QAM
Microwave power Amplifiers", IEEE MTT-S Digest, pp 99-102, 1985; and, J.
Namiki, "An Automatically Controlled Predistorter for Multilevel
Quadrature Amplitude Modulation", IEEE Trans. Commun., vol COM-31 no. 5,
pp 707-712, May 1983] adaptively cancel the dominant 3rd order component
of the PA nonlinearity. The complexity of such structures increases
rapidly if extended to 5.sup.th or 7.sup.th order terms. Moreover, Class
AB amplifiers have significant kinks in the transfer characteristic at low
levels which are not well modelled by a cube law. The predistorter of
Nannicini et al is noteworthy, because the quantity fed back is not
required to be an accurate replica of the PA output v.sub.a (t), but
rather a simple estimate of the power in the 3rd order spectral skirts, as
measured by narrowband filters and envelope detection. This, however,
restricts its use to modulation of a specific bandwidth.
Another prior art predistorter [H. Girard and K. Feher, "A New Baseband
Linearizer for More Efficient Utilization of Earth Station Amplifiers Used
for QPSK Transmission", IEEE J. Selected Areas in Commun., vol. SAC-1, no.
1, pp 46-56, January 1983; see also U.S. Pat. No. 4,462,001 supra] is
based on complex gain, as is the present invention. However, it requires a
dynamic phase shifter, and is not adaptive.
Yet another prior art predistorter [exemplified by U.S. Pat. No. 4,412,337
issued Oct. 25, 1983 for an invention of Robert H. Bickley et al entitled
"Power Amplifier and Envelope Correction Circuitry"] differs from the
present invention in that it makes no attempt to correct distortion due to
AM/PM conversion, and is restricted to digital signalling consisting of a
sequence of pulses.
A number of prior art predistorters [e.g. J. Graboski and R.C. Davis, "An
Experimental MQAM Modem Using Amplifier Linearization and Baseband
Equalization Techniques", Proc. Natl. Commun. Conf., pp E3.2.1-E3.2.6,
1982; U.S. Pat. No. 4,291,277 issued Sept. 22, 1981 for an invention of
Robert C. Davis et al entitled "Adaptive Predistortion Technique For
Linearizing a Power Amplifier For Digital Data Systems"; and, A.A.M. Saleh
and J. Salz, "Adaptive Linearization of Power Amplifiers in Digital Radio
Systems", Bell Syst. Tech. J., vol. 62, no. 4, pp 1019-1033, April 1983]
are restricted to particular modulation formats. Implemented in digital
baseband as RAM lookup tables ("LUTs"), with an entry for each
predistorted point in the signal constellation, they are fast and require
very little memory. However, they are limited to rectangular pulses, or to
pulse shaping implemented by filtering following the PA, an unattractive
arrangement. Saleh et al provide an adaptive version, which requires
conversion between polar and rectangular representations. Davis et al
(supra) also include an adaptation algorithm of the linear type discussed
in Section IV A below. Its convergence is therefore slower than the
present invention, and it requires a phase shifter in the feedback path
for stability (though this is not noted in the Davis et al '277 patent).
The most general and powerful predistorter to date was reported by Y.
Nagata, "Linear Amplification Technique for Digital Mobile
Communications", Proc. IEEE Vehic. Tech. Conf., pp 159-164, San Francisco,
1989 (see also U.S. Pat. No. 4,700,151 supra). Nagata generalizes the
table lookup approach of Grabowski et al and Saleh et al to provide the
predistorted equivalent v.sub.d of any input value v.sub.m (FIG. 2),
thereby mapping the complex plane into itself. Nagata's approach is
therefore unrestricted by order and type of PA nonlinearity (provided it
is memoryless). Since this permits pulse shaping prior to predistortion,
it is not restricted by modulation format, either. Nagata also provided an
update algorithm for adaptation of the table, and a delay compensation
algorithm so v.sub.m (t) and v.sub.a (t) can be compared. Nagata's
technique is hereinafter referred to as the "mapping predistorter"
("mapping PD").
Powerful though it is, the mapping PD has several drawbacks. The lookup
table ("LUT") is 2 Mword long for 10 bit representation of the real and
imaginary parts of v.sub.m, and increases to 8 Mword for 11 bit
representation. It requires a phase shifter in the feedback path for
stability in the adaptation update, and convergence is very slow (10 sec,
at 16 ksym/sec). Moreover, switching to a new channel requires
readjustment of the phase shifter, and reconvergence of the table over
another 10 sec.
The present invention, like the mapping PD, is unrestricted by modulation
format, or by order of PA nonlinearity. In addition, it has some major
advantages compared with the mapping PD. It requires over 4 orders of
magnitude less table memory (typically under 100 complex word pairs). It
reduces convergence time by a similar factor, to a few msec at 16
ksym/sec. It eliminates the reconvergence time following a channel switch,
since the table for each channel is so small that it can be downloaded or
simply retained in memory. Finally, it eliminates the need for a phase
shifter in the feedback loop.
Section II describes the predistorter structure, and demonstrates its
ability to suppress intermod products using only a small table. Section
III analyzes the effect of PD nonidealities (especially limited table
size) on the PA output; to the inventor's knowledge, such an analysis is
missing from all previously published descriptions of linearizers. Section
IV introduces a fast adaptation algorithm. Finally, Section V summarizes
the results and their implications.
II THE GAIN BASED PREDISTORTER
A. Basic Structure
The amplifier can be modelled as a memoryless nonlinearity in several ways.
The most productive for present purposes is as a level dependent complex
gain. That is, the complex envelope of the amplifier's input v.sub.d and
output v.sub.a are related by:
v.sub.a =v.sub.d G(.vertline.v.sub.d .sup.2)=v.sub.d G(x.sub.d)(1)
where x.sub.d denotes the squared magnitude of v.sub.d ; and G(x.sub.d),
the complex gain of the amplifier, summarizes its AM/AM and AM/PM
characteristics. Note that G(x.sub.d) depends only on the instantaneous
power of the input, not on its phase. FIGS. 3 and 4 show the relation
between bench measurements and the complex gain for a typical Class AB
amplifier. The effect of compression is clearly evident at high input
levels, as is the loss of gain at low levels, because of the crossover
point between the push and pull halves of the amplifier.
The gain based PD of the present invention (FIG. 5) is not just modelled,
but actually constructed, according to the complex gain formulation. Its
input and output complex envelopes are related by:
v.sub.d =v.sub.m F(.vertline.v.sub.m .vertline..sup.2)=v.sub.m
F(x.sub.m)(2)
where x.sub.m denotes the squared magnitude of v.sub.m.
For any input power, the optimum value of the PD complex gain F is
determined by equating the composite PD/PA nonlinearity to a nominal
constant amplitude gain K. Normally, K is selected to be a little less
than the amplifier's midrange gain. However, as shown in Section III, the
choice of K has no effect on the signal to quantization noise ratio at the
amplifier output. Combining (1) and (2), we define F implicitly by:
v.sub.m F(.vertline.v.sub.m .vertline..sup.2)G(.vertline.v.sub.m
.vertline..sup.2
.vertline.F(.vertline.vm.vertline..sup.2).vertline..sup.2)=Kv.sub.m(3)
or:
F(x.sub.m)G(x.sub.m .vertline.F(x.sub.m).vertline.)=K (4)
A fast technique for adaptive calculation of F(x.sub.m) is described in
Section IV. Note that the PD complex gain F(x.sub.m) has a real domain, so
that it can be represented by a one-dimensional LUT, rather than the
two-dimensional LUT required by the mapping PD.
In practice, there is little point in trying to linearize an amplifier up
to its saturated output power P.sub.sat, because the distortion increases
drastically in this region, and we are faced with rapidly diminishing
returns on linearization effort. Accordingly, define the span S as the
fraction of saturated power over which linearization is attempted. Thus
the maximum output power is given by S P.sub.sat. Realistic values for the
span are in the range 0.95 to 0.98. An alternative description is the peak
backoff ("PBO") of the PA in dB:
PBO=-10log(S) (5)
The span in turn limits the domain of the linearizer:
0.ltoreq.x.sub.m .ltoreq.p.sub.mm (6)
where the maximum power of v.sub.m is given by:
p.sub.m =S P.sub.sat /K.sup.2 (7)
The implementation of F(x.sub.m) as a LUT with entries equispaced in input
power x.sub.m will now be examined. Although a good case can be made for
nonuniform spacing, as shall be seen in Section III, it would be at the
cost of more computation when implemented digitally. With N table entries,
the step size is given by:
D.sub.g =P.sub.mm /N.sub.t (8)
The range and midpoint of each step, and the corresponding table entries,
are given for i =0, 1, . . . , N -1 as:
X.sub.mi ={x.sub.m :iD.sub.g .ltoreq.x.sub.m <(i+1)D.sub.g }(9)
X.sub.mi =D.sub.g (i+1/2) (10)
F.sub.i =F(X.sub.mi) (11)
That is, the table is optimized according to (4) for the midpoint of each
step, as shown in FIG. 6.
B. Performance of Basic Predistorter
This section demonstrates that even very small PD gain tables can produce
major reductions in intermodulation products, using in all cases the Class
AB amplifier of FIGS. 3 and 4 with a PBO of about 0.22 dB, giving a 95%
span. First, define the output backoff ("OBO") as the dB difference
between P.sub.sat and the average signal output power. Clearly OBO depends
on the peak to average power ratio of the signal, hereinafter called the
"signal backoff"("SBO"):
OBO=PBO+SBO (12)
Large OBO values imply inefficient operation, due either to amplifier peak
backoff or to the signal format. Note that compression lowers the SBO
below the value characteristic of the signal in a linear regime.
The first example is a two-tone test. Although it does not fully exercise
the nonlinearities, it is widely used. Its SBO is 3 dB in the absence of
distortion. FIG. 7 shows the results of two-tone tests of the amplifier of
FIG. 5 with PBO values of 0.22 dB and 24.63 dB (i.e. SBO values of 2.53
and 27.67 dB, respectively). All spectra are normalized to 0 dB for the
desired components at f/f.sub.s =.+-.0.0156, and the frequency is
normalized by the sampling rate. Even with a PBO of 25 dB, the dominant IM
products of the uncorrected amplifier are only 40 dB down. In contrast,
the third curve on FIG. 7 shows that a 64 point predistortion table
achieves the desired PBO of 0.22 dB (OBO of 3.22 dB) with IM products
reduced below 60 dB.
The second example (FIG. 8) is a more demanding noise loading test. Thirty
sine waves with equal amplitudes and randomly selected phases are located
fifteen on each side of a center channel spectral null. The degree to
which the null fills in at the output of the PA indicates the total power
in the nonlinear products. Averaging of a few such output spectra, each
with a new set of phases, is sufficient. The SBO for this signal is 14.8
dB. With a SBO of 13.2 dB (PBO of 0.22 dB), wide IM skirts about 14 dB
down are evident in the uncorrected amplifier, and even an OBO of 33.7 dB
lowers the IM by only about 8 dB. In contrast, both a 32 point and a 64
point PD table introduce a major reduction in intermod power at an OBO of
15 dB (PBO of 0.22 dB).
The final example is more closely related to data transmission. The input
signal is 16QAM, with a square root spectral raised cosine pulse with 25%
rolloff, Hamming windowed to 7 symbols. The SBO is about 6 dB. As shown on
FIG. 9, the 3.sup.rd, 5.sup.th and 7.sup.th order skirts are clearly
visible when the uncorrected amplifier is operated at 0.22 dB PBO (3.0 dB
OBO). Even with an OBO of 33.7 dB (PBO of 30.24 dB), out-of-band emission
is still about 56 dB down, and the power efficiency of the amplifier is
hopelessly small. Again a striking improvement is seen with the gain based
PD operated at a PBO of 0.22 dB: even a 32 point gain table brings the
skirts below 60 dB, and a 64 point table lowers the IM by another 6 dB
(see Section III).
The foregoing examples demonstrate that major improvements in linearity can
be achieved with very small tables: N.sub.t =64 entries, compared with
over 1 million entries in the mapping PD. This dramatic reduction is due
to exploitation of the rotational invariance of the amplifier
nonlinearity. Of the two dimensions required to specify a point in the
plane, only one (radius) need be quantized; the other (phase) remains
continuous.
The cost, compared with the mapping PD, is computation. Both the squared
magnitude of v.sub.m (for the table index) and the complex multiplication
of v.sub.m by the table entry have to be performed at a sampling rate
adequate to represent the highest order nonlinearity of interest. Such a
PD has been implemented on a TMS320C25 at 240 kHz, which is sufficient for
7.sup.th order nonlinearities when the nominal channel bandwidth is 30
kHz.
C. Nonuniform Spacing of Table Entries
The discussion and examples to this point have assumed a table (equations
9-11) optimized according to (4) for the midpoint of each step. In
practice, however, the table entries may not be perfectly adjusted;
indeed, Section III(C) makes an explicit calculation concerning this
point. Although such departures from the condition (4) may degrade
linearization, they nevertheless fall within the scope of the invention.
Accordingly, the condition to be satisfied by the table entries is:
F.sub.i G(X.sub.mi .vertline.F.sub.i .vertline..sup.2).perspectiveto.K(12a)
The foregoing discussion and examples have also assumed table entries
equispaced in power x.sub.mi. The generalization to nonuniform spacing is
straightforward. The table contains N.sub.t distinct values of input power
x.sub.mi, each with an associated F.sub.i value satisfying (12a). In the
case of nonuniform spacing, the table index i is determined for an
arbitrary input power x.sub.m as the one for which the x.sub.mi value is
closest to the x.sub.m value; that is, the table index i minimizes the
absolute value .vertline.x.sub.m -x.sub.mi .vertline.. The range of each
step is therefore given by:
X.sub.mi ={x.sub.m :(x.sub.m,i-1 +x.sub.mi)/2.ltoreq.x.sub.m <(x.sub.mi
+x.sub.m,i+1)/2) (12b)
The ranges and the table entries selected by the predistorter are identical
to those given earlier in (9-11) if the table entries are uniformly spaced
in x.sub.mi.
D. Alternative Selection Criteria
Equations (9-11), which define selection of the best table entry, are
equivalent to:
i=round(x.sub.m /D.sub.g) (12c)
where round(x) is the closest integer to x. Alternative selection criteria
may be desirable in the interests of implementation simplicity:
i=floor(x.sub.m /D.sub.g) (12d)
and:
i=ceil(x.sub.m /D.sub.g) (12e)
where floor (x) and ceil (x) represent, respectively, the greatest integer
less than or equal to x, and the smallest integer greater than or equal to
x. Although these latter two selection criteria will, in general, degrade
linearization accuracy, doubling the table size restores the performance,
as hereinafter shown in Section III(B). All three criteria are considered
to fall within the scope of the present invention.
The general form of the selection criterion, including nonuniform spacing,
follows from (12b-12c). The selected table index is either of the two
values:
1. i, such that x.sub.mi is the largest table entry less than or equal to
x.sub.m ; or,
2. i, such that x.sub.mi is the smallest table entry greater than or equal
to x.sub.m. That is to say, we can select either of the table entries
bracketing x.sub.m.
E. Interpolation of Table Entries
Better use of the table can be made, at the cost of additional computation,
by interpolating a value from the table. By way of example, a linearly
interpolated value results from the calculation:
##EQU2##
where i is such that x.sub.mi is the smallest table entry greater than or
equal to x.sub.m. This, and other standard interpolation formulae [see,
for example, Sgermund Dahlquist and Ake Bjorck, Numerical Methods,
Prentice-Hall, 1974], are also within the scope of the present invention.
III ERROR ANALYSIS OF THE PREDISTORTERS
Section II presented experimental evidence that the gain based PD of FIG. 5
can achieve significant reduction in IM products with a table four orders
of magnitude smaller than that of the mapping PD. This section provides an
analytical explanation for the results, by exploring the relation between
PA characteristics and table size. Among other findings, it will be
demonstrated that table size effects show up as relative error in the gain
based PD, compared with absolute error in the mapping PD; that error power
in the gain based PD decreases inversely as (N.sub.t).sup.2, rather than
inversely as N.sub.t in the mapping PD; and that the jitter in the gain
table entries caused by adaptation can be compensated by a relatively
small increase in table size.
A. The Mapping Predistorter
In the mapping PD, the input v.sub.m, representing the desired output of
the amplifier, is quantized to several bits in its real and imaginary
parts separately, and acts as an index to the table. The table size is
determined by the number of bits; for example, with 10 bit accuracy,
2.sup.20 or about 1 million table entries are needed. Note that the number
of bits in each table entry is a different issue. For simplicity, assume
that the table has infinite precision.
Analysis of the effects of input quantization through two nonlinearities is
surprisingly straightforward. Imagine the input space (v.sub.m space) as
partitioned into quantization cells, as shown in FIG. 10. If there are
N.sub.t entries in the two-dimensional table, then each cell has width:
##EQU3##
since we associate maximum power P.sub.mm with the extreme corner cells.
All v.sub.m values in a cell are mapped onto a single v.sub.d value, which
we assume to be the correct predistorter value for the center of the cell.
Thus the amplifier output v.sub.a has its desired value Kv.sub.m o | | |