Nokia Corporation v. Apple Inc.
Filing
379
NOTICE of Issuance of Amended Subpoena upon The Institute of Electrical and Electronics Engineers, Inc. ("IEEE") by Apple Inc. (Attachments: # 1 Exhibit 1 part 1, # 2 Exhibit 1 part 2, # 3 Exhibit 1 part 3, # 4 Exhibit 1 part 4, # 5 Exhibit 1 part 5, # 6 Exhibit 1 part 6, # 7 Exhibit 1 part 7, # 8 Exhibit 1 part 8)(Moore, David)
1064
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IEEE TRANSACTIONS ON COMMUNICATIONS, YOLo COM-25, NO. 10, OCTOBER 1977
IEEE TRANSACTIONS ON COMMUNICATIONS,VOL. COM-25, NO. 10,
1977
Concatenated Coding Systems Employing a
Concatenated Coding Systems Employing
Unit-Memory Convolutional Code and a
Unit-Memory Convolutional Code and
Byte-Oriented Decoding Algorithm
Byte-Oriented Decoding Algorithm
LIN-NAN LEE,
Abstract-Concatenated coding systeq1s utilizing aconvolutional
convolutional
Abstruct-Concatenated
coding
systems
code as the inner code and aa Reed-Solomon code as the outer code are
and Reed-Solomon code as the outer code
code as the
considered. In order to obtain very reliable communications over a very
considered. In order to obtainvery reliable communications
a
noisy channel with relatively modest coding complexity, it is proposed
noisy channel with relatively modest
is
to concatenate a byte-oriented unit-memory convolutional code withwith
to concatenate a byte-oriented unit-memory convolutionai code
an RS outer code who~ symbol size is one byte. It is further proposed
an RS outer code whose symbol size is one byte. It is further proposed
to utilize aareal-time minimal-byte-error probability decoding algorithm,
to utilize real-time minimal-byte-error probability decoding algorithm,
together }Vith feedback from the outer decoder, in the decoder for the
together pith feedback from the outer decoder, in the decoder for the
inner convolutional code. The performance
inner convolutional code. The performance of the proposed concateof the proposed concatenated coding system is studied, and the improvement over conventional
nated
is studied, and the improvement over conventional
concatenated systems due to to
concatenated systems due each additional feature is isolated.
each additional feature is isolated.
I. INTRODUCTION
I. INTRODUCTION
coding systems
HE COMPLEXITY of
systems
T exponentially withthetheconventional codingcodes(or grows
THE COMPLEXITY ofblocklengthforblock block codes (or
block length for
exponentially with
with the constraint length forfor convolutib.nal codes). To overwith the constraint length convolutibpal
codes). To overcome the complexity of very long codes; the idea of cascading
come the complexity
very long codes; the idea cascading
two ormorecodesof
two or more codes of lesser complexity to achieve highly
lesser complexity to achieve
reliable communications was considered first by Elias [1] ,,and
reliable communications was considered first by Elias [ 11 and
later by Forney [2]. Forney's technique of using two or more
later by Forney [2]. Forney's technique of
two or more
block codes over different alphabets to obtain a very low error
block codes over different alphabets to obtain a very low error
rate is known as concatenated coding.
rate is known as concatenated coding.
Guided by the premise that a convolutional code generally
the premise
Guided
a convolutional
performs better than a block code of the same complexity,
code
performs better than a
the
Falconer [3],andlaterJelinekandCocke[4],
Falconer [3], and later Jelinek and Cocke [4], considered
cascading an outerblockcodewithaninnerconyolutional
cascading an outer block code with an inner convolutional
code. Figure 1 shows a general representation of ~uch a blocka general representation such blockcode. Figure 1
convolutional concatenated coding ,system. In both the
convolutional
concatenated
codihg system.both
In the
Falconer and Jelinek-Cocke schemes, ~equential decoding was
Falconer and
schemes, sequential
was
used for the inner decoder; and the outer block coding system
used for the inner decoder; and the outer block coding
system
was used only to intervene when the sequential decoder
was used only t o intervene when the sequential decoder
experienced computational overflow. Therefore, these systems
experienced computational overflow. Therefore, these
can be regarded as primarily sequentially decoded convolucan be regarded as
sequentially decoded
tional coding systems.
tional coding systems.
Maximumlikelihood (i.e., Viterbi [5]) decoding of conMaximum likelihood (i.e.,
[5])
convolutional codes with a moderate constraint length can provide
volutional codes with a moderate constraint length can
anerror rate of less thanat10- 2 at a rate slightly higher than
an error
rate
of less than
a rate slightly
Paper approved by thethe Editor for Communication Theory of the
Paper approved by Editor for Communication Theory of the
IEEE CommunicationsSocietyforpublicationafterpresentation
IEEE Communications Society for publication after presentation in
in
partatat IEEEInternationalSymposiumonInformationTheory,
part thethe IEEE International Symposium on Information Theory,
21-24, 1976. Manuscript received October
Ronneby, Sweden, June 21-24, 1976. Manuscript received October 19,
Ronneby, Sweden,
1976; revised May 6, 1977. This work was supported by the National
This work was supported by the National
1976; revised May 6,
Aeronautics and Space Administration under
Aeronautics and Space Administration under NASA Grant NSG 5025
NASA Grant
5025
the Goddard Space
at the University of Notre Dame in liaison with the Goddard Space
at the University of Notre Dame in liaison
submitted to the
Flight Center. This paper forms part of a dissertation submitted to the
This paper forms part of a
Flight
Graduate School of the University of Notre Dame in partial fulfillment
Graduate School of
University of Notre Dame in
fulfillment
of the requirements for the Ph.D. degree.
of the requirements for the
Ph.D. degree.
The author was with the Department of
Engineering,
The authorwaswiththeDepartmentofElectrical Electrical Engineering,
University of NotreDame,NotreDame,IN46556.He
University of Notre Dame, Notre Dame, IN 46556. He is nowwith
is now with
the LINKABIT Corporation, San Diego, CA 92121.
the LINKABIT Corporation, San Diego, CA 92121.
MEMBER, IEEE
Data
SOYrCe
Block
Encoder
L
Fig. 1. A concatenated coding system employing a a convolutional
coding system employing convolutional
Fig. 1.
code as the inner code and a block code as the outer code.
code as the inner code and a block code
as
code.
R comp ofthenoisy
the noisy channel.Forney's
Forney's work [2]] suggested
Rcomp
[2
that a concatenated coding system with a powerful outer code
code
can perform reasonably well when its inner decoder is operated
well
its inner decoder
with a probability of error in the range between 10- 2 and
in
the
and
10- 3 . It was natural thenforfor Odenwalder [6] to choose a
was natural then
[6]
Viterbidecoder
decoder fortheinnercoding
for the inner coding system in his blockin
blockconvolutional concatenated coding system.
convolutional concatenated coding
Because the output error patterns of Viierbi-type decoders
output error patterns
Because
Viterbi-type decoders
for convolutional codes are bursty, block codes over a large
convolutional codes
block codes
alphabet, suchthatmanybits bits of theinnercodeformone one
inner code form
alphabet, such that many
symbol of the outer code, appear very attractive fortM outer
very
for.thc?
symbol
codingsystem. Reed-Solomon
coding system. The Reed-Solomon (RS) block codes are
The
codes
particula~ly appealing because they can be decoded by relabe decoded by
particularly
tively siJrlple procedures (such as the Berlekamp.,.Massey [7],,
siryple procedures
as
Berlekamp-Massey [ 7 ]
[8] algorithm) and have optimum distance properties [17] .
[17].
[8]
Because, the lengths of the bursts of output errors made byby
bursts
errors made
Because,the
Viterbi decoders are widely distributed, it is generally necessary
Viterbi
are
distributed,
to interleave the inner convolutional code so that errors in the
to
so
errors in the
individual RS-symbols ofoneblock are independent; otherone block
otherindividual RS-symbols
wise, a veiy long block code would be required to operate the
long
required
wise,
operate the
system efficiently. Because the most likely length of the outmost
outefficiently.
put er~or patterns made by the inner decoder are on the order
ertor
inner decoder
the order
of the constraint length, K,, of the convolutional code, Odenconstraint length, K
convolutional code, Odenwalder chose the RS symbol alphabet to be GF(2 K ).
GF(2K).
RS symbol alphabet
In block-convolutional
a block-convolutional concatenanted coding system
concatenanted
coding
such as Odenwalder's employing a Viterbi decoder with
such as
employing
a
decoder
with
unlikely that the
conventional convolutional codes, it is very unlikely ~hat the
conventional convolutional
is
beginning of a decoding error burst is always aligned with the
is
boundary between two RS symbols; in fact, such a burst only
RS symbols; in
two bits long may affect two RS symbols. This fact led us to
RS symbols. This
consider using good convolutionalcodes whichare symbolconvolutional codes
are symboloriented rather than bit-oriented. In [9], we reported a
[9],
rather bit-oriented.
than
In
class ofunit-memoryconvolutional
unit-memory convolutional codes for which ko-bit
class
k,-bit
information segments are encoded no-bit no-bit encoded
are encoded into
into encoded
segments. It was shown there that an (no, k o ) convolutional
(no, k,)
segments.
was shown there that
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YSTEMS LEE: CONCATENATED CODING SYSTEMS
LEE: CONCATENATED CODING
code with unit memory always achieves the largest free distance
code with unit memory
always achieves
free distance
possible for codes of the same rate ko/no and the same number
for codes the same
ko/n,
the
2Mk o of encoderstates, where M is 'theencodermemory.
states,
2Mko
the encoder memory.
The unit-memory codes are naturally byte-oriented with byte
with byte
The unit-memory
are
size equal to k o information bits. It will beshown that the
shown
the
size equal to ko
improved free distance andthe sYl11bol-oriented nature of
the
free
distance
symbol-oriented
nature
these codes provides an improvement of approximately 0.3 dB
these codes
an improvement approximately 0.3
in the overall performance of the concatenated coding system
in the overall performance
when these codes replace bit-oriented convolutional codes.
bit-oriented convolutional
the
Anotherimprovement is to modifythe decoder for the
improvement is
for the
convolutional code so that the decoder emits notonlythe the
the
code so
emits not only
most-likely estimated symbol, but also reliability information
but also
estimated
about the estimated symbol. The outer decoder may then use
estimated symbol.
outer
about
either “erasures-andthis reliability information to performeither "erasures-andthis
errors" decoding or "generalized-minimum-distance" (GMD)
errors”
“generalized-minimum-distance’’
decoding as suggested by Forney [2]. Zeoli [10] and Jelinek
Forney [2].
Jelinek
decoding as
[IO]
[11] proposed to extract reliability information by annexing
by
[I 11
a long tail to the original convolutional code and using this
long tail to
original
added tail to provide anerrordetection
the
added
to
an error detection capability forthe
estimate made by the Viterbi decoder for the original shorter
decoder for
estimate
convolutional code. This approach requires thefeedbackof of
feedback
convolutional code. This approach
symbols previously decoded by the Viterbi decoder and, more
decoder
symbols
decoded
more
importantly, uses the output of the outer
restart
importantly, uses the output of the outer decoder to restart
by
the inner Viterbi decoder whenever an error is corrected by
decoder
error is
the inner
the outer decoder. It will be shown that the error detecting
the outer
It
shown that the error detecting
capability used with an "erasures-and-errors" outer decoder
capability
“erasures-and-errors”
provides an improvement of 0.2 dB and thatthefeedback
that the feedback
an improvement
0.2 dB
from the outer decoder further improves the performance by
performance
from the outer
further
0.3 dB.
0.3 dB.
An alternative approach to extracting reliability information
approach
from the inner decoder is
compute the posteriori
from the inner decoder is to compute the a posterion probability of
for
from the
ability of correctness for each decoded symbol fromthe
decoder for the short constraint length convolutional code and
for the short constraint length
and
then use this probability as
then use this probability as the reliability information provided
to the outer coding system. It will be shown that, when used
system.
outer
with an errors-and-erasures
scheme
decoder,
with an errors-and-erasures outerdecoder, this scheme imdB
0.1 dB compared
proves performance by only 0.05 dB to 0.1 dB compared to
only
is less
Zeoli’s
hard-decision decoding and hence is less powerful than Zeoli's
tail annexation
tail annexation scheme; yet itsperformance is undoubtedly
its performance
optimal among all schemes employing
the short
optimal among all schemesemploying onlytheshortcon- constraint
length
convolutional (with
code
straint length convolutional code (with no annexed tail).
However,
shown
However, it will be shown that, in conjection with the use of
in conjection with the
feedback
feedback from the outer decoder, the a posteriori probability
outer decoder, the
inner
0.2 dB
improvement than
inner decoder provides about 0.2 dB more improvement than
does the
decoder
does the Viterbi decoder aided byfeedback.Infact,the the a
feedback. In fact,
posteriori inner decoder, used with feedback from the outer
decoder,
feedback from the outer
offers
a
Zeoli’s scheme;
decoder, offers a slight improvement over Zeoli's scheme;
the inner encoder and the decoder
moreover the inner encoder and the innerinner decoder have the
same
so
inner decoder
same constraint length so that the inner decoder generally and
automatically returns to
few branches
automatically returns to normal operation only a few branches
after making an error.
after making an error.
The plan
is as follows.
The plan of this paper is as follows. In Section II, a "real
Section 11, “real
time” decoding
algorithm
for
unit-memory
time" decoding algorithm for unit-memory convolutional
codes is
calculates
codes is developed which calculates the a posteriori probability
for
value
decoded.
byte
Sections 111,
for each value of the byte being decoded. In Sections III, IV,
and V, the
several
and V, the performances of several block-convolutional concatenated coding
catenated coding systems having unit-memoryconvolutional
convolutional
codes are
having
inner codes are compared with similar systems haVing conven-
convolutional
tional bit-oriented convolutional inner codes. In each case, we
chose the (1 8 , 6 ) unit-memory convolutional code as the inner
(18,6)
convolutional code
practically
complexity in terms
code because it has practically minimum complexity in terms
of decoder implementation, and because of its reasonably large
of its reasonably
and
Reed-Solomon codes
free distance (dfree= 16). We chose the Reed-Solomon codes
distance (dfree =
over GF(2 6 ), with block length 63 symbols, as the outer codes
GF(26),
block length
RS
would
matched
so that the symbol size of the RS codes would be matched to
the byte-size (six bits) of the unit-memory code. In Section VI,
of
unit-memory code. In Section
the degradation of performance, when the rate 1/3 inner
degradation
rate
convolutional
convolutional code is replaced byrate rate 1/2 convolutional
a a
code, is considered in order to demonstrate tradeoff
in
thethe tradeoff
between bandwidth expansion and signal-energy-to-noise ratio.
In Section VII, the 95%confidence intervals for the simulation
Section VII, the95%
simulation
results are obtained and interpreted.
and interpreted.
II. REAL-TIME MINIMAL-BYTE-ERROR PROBABILITY
11.
DECODING OF UNIT-MEMORY CODES
We now develop an alogrithm for real-timeminimal-byteminimal-byteof
unit-memory convolutional
error probability decoding of the unit-memory convolutional
probability
codes described in [9] .
in [9].
the byte
of k o
Let at (tt = 1,2, -.) denotes the byte (or subblock) of ko
( = 1 , 2, ...)
informationbits to be encodedatat time t , andlet bt ( t =
bits
t,
let b t (t =
1,, 2, ...) be the corresponding encoded subblock ofof no bits.
corresponding
subblock
1 2, ..*)
For a unit-memory code,
code,
b , =a,Go
+at-lG1
(1)
(1 1
G1
ko X
and where,
of
where Go and Gl are ko X no matrices and where, by way of
convention, a, = 0 . We assume that the sequence b , ,, b , ,, ...
ao = O.
the
bl
2 ...
has been transmitted over a discrete memoryless channel and
discrete memoryless channel
that fr t, (t = 1,2, -) is the received subblock corresponding to
corresponding
(t = 1, 2, ...)
transmitted
subblock
denote
the transmitted subblock b,. We shall write a[ t, t' 1 to denote
bt .
[at,a,+l,...,a,~];similarlyforb[t,t~lf[t,t'l'
[at, at+1 , "',ad; similarly fOf b[t,t'l andr[,,,’l.
and
By real-timedecoding withdelay A, we mean thatthe
feal-time decoding
delay fl,
the
decoding decision for at is madefromthe the observation of
from
observation of
f[ 1, t + A l .
The real-time minimal-byte-error probability
r [ l , t+ Al'
minimal-byte-error
probability
(RTMBEP) decoding
chooses
(RTMBEP)decoding rule then is that which chooses its estimate C as that value of a, which maximizes P(at I rC1,+ A ] )
i
,
,
at
of at whichmaximizesP(atlf[I,t+Al)
for t = 1,, 2,, .... To find a recursive algorithm for this decoding
= 1 2 -.. find
decoding
rule, we begin by noting that
that
peat
If[l,t+Al)=p(at'f[1,t+Al)/~P(a,r[1,t+Al)
(2)
where we have used Q! to denote a running variable for at. It
a denote running
It
suffices then to find a recursive method for calculating P(at,
to find
for
peat,
r[l,t+Al) ).
‘[l,t+AI .
We next observe that
peat, f[l, + A 1 = peat, fO, ] ( r [ t + l , t + A ]
P(‘t> r [ l , tt+A1 )) =P(at>r [ l , tt]p)P(f[t+l,t+A 1
I at, r [ l , t ] )
r[l, t])
=P(at,f[l,tl)p(r[t+l,t+Al 1lat)
=P(atp
r[l,t])P(r[t+l,t+A]
%)
(3)
where we have used the facts that the channel is memoryless
facts that the
and that the code has unit memory. It remains to find recurthat the code
memory. It
sive
obtaining the two
the
sive rules for obtaining the two probabilities on the right in (3).
probabilities
Obtaining the recursion for P ( a t , r c l ,,tl) is quite standard
peat> r[1, I )
standard
[12] - ~ 4 1 ;
P I -[14];
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IEEE TRANSACTIONS ON COMMUNICATIONS, VOL. COM-25, NO. 10, OCTOBER 1977
VOL. COM-25, NO. 10, OCTOBER 1977
IEEE TRANSACTIONS ON
COMMUNICATIONS,
peat, 'p,t]) = ~ P(a[t-l,t], '[l,t])
=
~
r[ l,t] )
(10)
= P(r[t+i,t+. '[l,t-1])
h(a)
°t~l
(4)
But also
also
peat, 't I at-I, ,[ l,t-l])
The R TMBEP Decoding Algorithm for Unit-Memory Codes
RTMBEP Decoding Algorithm Unit-Memory Codes
tor
=P(at I at-I> 't-l)P('t I a[t-l,t]' '[l,t-l])
= 2- kO P('t I b(a[t-l,t]»
i
decremented from A to 1 as the algorithm
~
where i will bedecrementedfrom
progresses. (Of course, the received segment r[t+l,t+A] must
(Of
the
r[ t+l, t+.t.] must
also be stored so that P(rt+i I ~ ( Q [ , + ~ - ~ , ~ can I also be
P(rt+ilb(a[t+i-l,t+i]» + ~ > )
found for i = A, A - 1, -., 1.) We may now state:
i = ~, ~ ... ,
for
state:
Step 0: Set[(0)=2 ko ~ndset[(a)=Ofora*O.Sett=
StepO:Setf(0)=2koandsetf(ol)=Ofora#0.Sett=1. 1.
Step 1: Make the replacement, for all states &,
a,
1:
.
replacement, for
(5)
where we have written b(a[t-l,t]) for the value ofb t deterb(act-l,tl) for the
of bt determined by (1) from a[t-l',t],, and where we assume here and
and
by
from
hereafter that all informaHon sequences are equally likely (as
hereafter that all information
correspondstomaximum-likelihood
corresponds to maxirrium"likelihood decoding.) Substituting
(5) into (4),,we have our desired recursion
(5) into ( )
4
p(at"[l,t]~=2-kO ~
[(01.) "'-,2- kO
,
~ [(OI.')P('t I b(OI.',
Q'
a».
'
Step 2: Set i = A and, for all states a,set
i = ~
a,
for
h(a) = r k o
44 = 2- k o
P(at-l,r[l,t-l])
E
x
P(rt+~ b(OI., a')).
P(rt+a I b(e, a’)).
Q'
a‘
0(--:1
(6)
3:
make the replacement, for
Step 3 Decrease i by 1 and make the replacement, for all
:
states a,
a,
We now turn to the quantity P(r[t+l,t+.
byte ao
Step 4: Emit, as theestimate. of ut, that byte a0 which
reliability
the
maxirriizes f(a)h(ol), and emit, as the reliability indicator, the
maximizes t(OI.)h(a),
probability
Proceeding in the same manner that (6) was obtained from (4),
from (4),
the
that (6)
we
desired
recursion
we find the desireq recursion
peat =
01.0
I r[l,t+. memory
11S.61
1
:l
code
code
Svstem VI w r th
System VI vii t h
(18.6) Unit(lS.6) Unltmemory System
memory code
code
i
System IV Hith (18.6)
(18,6)
Unit-memory code
Uni t-memory
System V wlth
V with
(18.6)
(18.6)
uni
l l n l t-memory code
code
1.41
1.1
15
. L
Fig. 8.
8.
The relative dB gains among the concatenated coding
concatenated coding
studied.
systems studied.
the softfeedbackfrom outer outer decoder. Among them,thesoftfeedback from the
the
Among
decision RTMBEP decoder with feedback performs the best.
qecoder
feedback
best.
modification
Intermsof of hardware implementation, Zeoli's modification
terms
unit-memory code and
with the unit-memory code and the the hard-decision RTMBEP
decoder are of approximately the same complexity. However,
the
decoder
since the operation of the Viterbi decoder for Zeoli's system
operation of
feedback from the outer decoder, there
depends on the correct feedback from the outer decoder, there
outer
is always a slim chance that the outer decoder fails to provide
the
decoder. Since theencoder
encoder
correct decisions to the Viterbidecoder.Since
constraint length is much larger than the decoder constraint
the
length
conlength, this can cause endless errors as if a catastrophic conthis
cause
volutional code were used. Thus, it is necessary to send
Thus,
it
synchronization signals inthe Zeoli schemeperiodically to
the
periodically
signals
reset the Viterbi decoder to guarantee restoration of normal
of
Viterbi
the
operation. the RTMBEP decoder has''the same constraint
the
has
length as that of the encoder; therefore, the decoder is able to
encoder; therefore,
recover from errors in a few branches by itself without feederrors
a few branches
itself
The feedback from the outer decoderonly speeds this
only speeds this
back. The feedback from the outer
back, most
therefore,
process uptherefore, when an error is fedback,thethe most
damage it can cause is for the RTMBEP decoder to make a
the
a
few more errors before it recovers by itself. This is certainly a
errors
very desirable advantage for a concatenated codingsystem.
system.
can restore
operaMoreover, because the decodercan restore its normal operaquickly,
the
required this
for
tion quickly, the degree of interleaving required for this
the
Reed-Solomon block
scheme is considerably less than the full Reed-Solomon block
the Zeoli's
length interleaving required for the Zeoli's scheme.
Finally, as a remark to information theorists, we note that
remark
theorists,
for System I11 (the RTMBEP inner decoder for the rate 1/3
System III
the rate 1/3
code concatenated with
(18, 6) unit-memorycodeconcatenatedwiththe the (63, 51),
6-error-correcting
with feedback from the RS
6-error-correcting RS code with feedback from the RS errorsdecoder)
byte-error-probability
only decoder) we can achieve a byte-error-probability of
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IEEE TRANSACTIONS O N COMMUNICATIONS, VOL. COM-25, NO. IO, OCTOBER 1 9 7 7
COM-2S,
10,
1977
TRANSACTIONS ON COMMUNICATIONS,
10- 66 at Eb/No of 2.17 dB, or,equivalently,at at Es/No of
lop
Eb/No
dB, or, equivalently,
EJN,
3.52 dB. The ,cut-off rate, R compp , of this 8-level quantized
rate, R c o m ,
3.52 dB. The
8-level
AWGN channel is 0.275 whereas its channel capacity is 0.44.
capacity
AWGN
is 0.275
its
The overall rate of the concatenated coding system is 0.27. It
concatenated
system
The overall
seems that the cut-off rate, rather th& thethe channel capacity,
the cut-off rate, rather th~
seems
is still the practical limit of rate for reliable communications,
limit
for
is still the
communications,
even for a very sophisticated concatenated coding system, just
even for a
sophisticated concatenated
as it is ina a conventional convolutional coding system emas
is
conventional
codingsystememploying sequential decoding
sequential decoding [16]. The advantage of the concon[16]. The
catenatedcoding system resides only in the elimination of
coding
"deleted data" such as is always presenta in a sequential
“deleted such
data”
as
is
in
sequential
decoding system because of the latter's highly variable
system
latter’s
computation.
computation.
ACKNOWLEDGMENT
ACKNOWLEDGMENT
The author would like to express his gratitude to his DisThe author
gratitude to
DisAdvisor, Professor James
sertation Advisor, Professor James L. Massey for his patient
and
and generous guidance of this research and for his suggestions
guidance
research
to match byte-oriented convolutional codes with Reedmatch
byte-oriented
convolutional
codes
Solomon codesand to develop the "real-time minimal-byteSolomon codes and
‘‘real-time minimal-byteerror probability (RTMBEP) decoding algorithm."
probability
algorithm.”
1. N. Lee, "Short, Unit-Memory, Byte-Oriented, Binary ConL.
“Short,Unit-Memory,Byte-Oriented,
ConTransvolutionalCodes Having MaximalFree Distance,”
Codes
Free Distance," IEEE Transactions on Information Theory, Vol. IT-22, No. 3, pp. 349-352,
No.3,
349-352,
May 1976.
1976.
"Coupled Decoding of Block-Convolutional
G. W. Zeoli,“CoupledDecodingofBlock-ConvolutionalCon- ConG.
Transactions
Codes,"
catenated Codes,” IEEE Transactions on Communications, Vol.
COM-21, pp. 219-226, March 1973.
COM-21,
"Bootstrap
Decoding,"
F. Jelinek, “Bootstrap Trellis Decoding,” IEEE Transactions on
Information Theory, Vol. IT-21, pp. 318-325, May 1975.
Theory,
318-325,
1. R. Bahl, J . Cocke, F. Jelinek and J . Raviv, “Optimal Decoding
J.
and 1.
"Optimal
L.
of Linear Codes for Minimizing Symbol Error Rate," IEEE
IEEE
for
Error
Rate,”
Transactions on Information Theory, Vol.IT-20,pp.284-288,
Information Theory,
IT-20, pp. 284-288,
Transactions
March 1974.
C. L.
"M.A.P. Bit
P. 1. McAdam, 1. R. Welch and C. 1. Weber,“M.A.P.BitDe-DeL.
L.
coding of Convolutional Codes," presented at 1972 International
ofConv91utional Codes,” presented at 1972 International
Symposium on Information Theory, Asilomar, California,
Information
Theory,
California,
January, 1972.
1972.
1.N. Lee, "Real-Time Minimal-Bit-Error ProbabilityDecoding
L. N.
“Real-TimeMinimal-Bit-Error Probability Decoding
of Convolutional Codes," IEEE Transactions on CommunicaofConvolutionalCodes,”
COM-22,
146-151, Feb. 1974.
tions, Vol. COM-22, pp. 146-151, Feb. 1974.
J.. 1. Massey and M. K.Sain,“InversesofLinear‘Sequential
Sain, "Inverses of Linear Sequential
J L.
Circuits," IEEE Transactions on Computers, Vol. C-17, pp. 330330Transactions
Circuits,”
337, April 1968.
"Sequential Decoding for Efficient CommunicaL
I. M. Jacobs,“SequentialDecodingforEfficientCommunicafrom Deep Space,"
Communication
tions from Deep Space,” IEEE Transactions on Communication
COM-IS, pp.
Technology,
Technology, Vol. COM-15, pp. 492-501, Aug. 1967.
W. W. Peterson and E. J . Weldon, Error Correcting Codes,
1.
and
E.
Second Edition MIT Press, Cambridge, Mass., 1972.
MIt
Cambridge,
1. N. Lee, "Concatenated Coding Employing
Systems Employing Un!tL.
“Concatenated
Coding
Systems
UnitConvolutional Codes and Byte-Oriented Decoding AlMemoryConvolutionalCodesandByte-OrientedDecoding
gorithms", Ph.D. Dissertation, Department En- Electrical Engorithms”,
Ph.D.
Dissertation,
Department Electrical of
of
of
gjneering, University of Notre Dame, June 1976.
gbeering,
Dame, June 1976.
[9]
[10)
[11)
[12]
[l3]
[l4)
[15]
[16)
[17]
[18)
REFERENCES
REFERENCES
[1)
[2)
[3)
[4)
[5)
[6)
[7)
f8)
P. Elias, “Error-Free Coding,” IRE Transactions on Information
P. Elias, "Error-Free Coding,"
Transactions
19S4.
Theory, Vol. IT-4, pp. 29-37, Sept. 1954.
Theory, Vol.
Mass.: M.I.T.
G. D. Forney, Jr., Concatenated Codes. Cambridge, Mass.: M.LT.
G.
Jr.,
Codes.
Press, 1966.
'
Press, 1966.
D. Falconer, HybridSequentialand
D. D. Falconer,“A"A Hybrid Sequential and AlgebraicDecoding
Decoding
Scheme,” F’h.D.
Scheme," Ph.D. Dissertation,Dept.ofElectricalEngineering,
Dept. of Electrical Engineering,
M.I.T.,
1966.
M.LT., Cambridge, Mass., 1966.
F. Jelinek and J.. Cocke, “Bootstrap Hybrid Decoding forfor Symand J
"Bootstrap Hybrid Decoding SymF.
Control,
metrical
metrical Binary Input Channels,” Information and Control, Vol.
Channels," Information
18, pp.
1971.
18, pp. 261-298, March 1971.
A. J Viterbi, "Error Bounds Convolutional Codes and
A. J.. Viterbi, “Error Bounds for for Convolutional Codes and An
ZEEE TransacOptimal Decoding Algorithm," IEEE TransacAsymptoticallyOptimalDecodingAlgorithm,”
tions on Information Theory,
IT-13, pp. 260-269, April
tions on InformationTheory, Vol. IT-l3,pp.260-269,
1967.
1967.
J. P.
J. P. Odenwalder, “Optimal Decoding of Convolutional Codes,”
"Optimal Decoding of Convolutional Codes,"
Ph.
Ph. D. Dissertation, School of Engineering and Applied Sciences,
Dissertation, School of Engineering and Applied Sciences,
University of California, Los Angeles, 1970.
of California,
1970.
E. R.
Coding Theorx.
E. R. Berlekamp, Algebraic Coding' Theory. New York: McGrawHill, 1968.
Hill,1968.
J L. Massey, "Shift-Register
Synthesis
and
Decoding,”
J.. 1. Massey,“Shift-Register Synthesis and BCH Decoding,"
Transactions on Information
Theory,
IT-15’,
IEEE Transactions on Information Theory, Vol. IT-IS', pp.
122-125, Jan. 1969.
172-125, Jan. 1969.
*
*
(S'73-M'76)
born in Kaohsiung,
Lin-nan Lee (S’73-M’76) was born in Kaohsiung,
Taiwan,China, on February 24, 1949.
February 24, 1949. HerereChina,
ceived the B.S.degreefromNationalTaiwan
degree from National Taiwan
University, Taipei, Taiwan, China, in in 1970, and
Taipei, Taiwan, China, 1970, and
the MS. and the Ph. D. degrees from the UniM.S.
Ph.
versity of NotreDame,NotreDame,
Dame, Notre Dame, IN, in
,.:
“I ”,,”
1973 and 1976, respectively, allElectrical
1973 and 1976, respectively, all in Electr'ical
in
1
Engineering.
From 1970 to 1971, he he was aaCommunica1970 to 1971,
Communications Officer in the Chinese Air Force. Between
Officer in the Chinese
Force: Between
a research assistant
1971 and 1975, he he wasaresearchassistantat at
and 1975,
the University of Notre Dame
thll yniversity of Notre Dame engagedin graduate studies on digital
in
studies on digital
communications with emphasis on information and coding theories.
communications with emphasis on information and coding theories. In In
1975, hejoinedthe the LINKABIT Corporationwherehe he is presently
he joined
where
of coding, modulation
research
engaged in study, research and development of coding, modulation and
in
multiple access techniques for satellite,communication systems.
for satellite/communication systems.
mult)p1e
I
,
Authorized licensed use limited to: UNIVERSITY NOTRE DAME. Downloaded on January 6, 2010 at 14:14 from IEEE Xplore. Restrictions apply.
AppDel0008218
Exhibit 7
GSM ENHANCED FULL RATE SPEECH CODEC
K. Jarvinen, J. Vainio, P. Kapanen, T. Honkanen, P. Haavisto
Nokia Research Center, Tampere, Finland
kari.jarvinen@research.nokia.com
R. Salami, C. Lajlamme, J-P. Adoul
University of Sherbrooke, Sherbrooke, Canada
ABSTRACT
This paper describes the GSM enhanced full rate (EFR) speech
codec that has been standardised for the GSM mobile
communication system. The GSM EFR coclec has been jointly
developed by Nokia and University of Sherbrooke. It provides
speech quality at least equivalent to that of a wireline telephony
reference (32 kbit/s ADPCM). The EFR coldec uses 12.2 kbit/s
for speech coding and 10.6 kbit/s for error protection. Speech
coding is based on the ACELP algorithm (Algebraic Code
Excited Linear Prediction). The codec provides substantial
quality improvement compared to the existing GSM full rate and
half rate codecs. The old GSM codecs lack behind wireline
quality even in error-free channel conditions, while the EFR
codec provides wireline quality not only for terror-free conditions
but also for the most typical error conditions. With the EFR
codec, wireline quality is also sustained in the presence of
background noise and in tandem connections (mobile to mobile
calls).
1. INTRODUCTION
The background for introducing wireline speech quality to GSM
is the increasing use of the GSM system in communications
environments where it competes with fixed or cordless systems.
To be competitive also with respect to speech quality, GSM must
provide wireline speech quality which is robust to typical usage
conditions such as background noise and transmission errors.
The standardisation of an enhanced full irate (EFR) codec for
GSM started in European Telecommunications Standards
Institute (ETSI) in 1994 with a pre-study phase. The pre-study
phase was undertaken to set essential requirements for the EFR
codec and to assess the technical feasibility of meeting them.
During the pre-study phase, wireline speech quality was set as a
development target for the EFR codec [I]. Wireline quality (with
ITU-T (3.728 16 kbit/s LD-CELP as a reference codec) was
required not only for error-free transmissilon, but also in low
error-rate conditions (C/I=13 dB) as well as in background noise
(error-free conditions). Wireline performance was required also
for speaker independence and for speaker recognisability. For
more severe error conditions (C/I=lO clB and C/I=7 dB)
significant improvement to the existing GSM full rate (FR)
codec was required. In extreme error condiiiions (C/1<7 dB) the
requirement was to provide graceful degradation without
annoying effects. In error-free self-tandem (mobile to mobile
calls) the EFR codec should perform equal to G.728 in tandem.
In erroneous tandem at C/I=lO dB, the EFR. codec was required
to perform significantly better than the FR codec.
0-8186-7919-0/97
$10.00 8 1997 IEEE:
Parameter
2 LSP sets
ACB index (lag)
FCB pulses
FCB gain
Total
771
2nd and 4th
subframe
1st and 3rd
subframe
I
9
35
5
I
Total per
frame
38
6
30
35
5
140
20
244
frame
subhame
\ ; /
I
\;
Figure 1: Block diagram of the GSM EFR encoder.
2.1 Linear Prediction
A 10th order linear prediction (LP)analysis is carried out
twice for each 20 ms frame using two different asymmetric
windows of length 30 ms. Both LP analyses are performed for
the same set of speech samples without using any samples from
future frames (no lookahead). The two sets of LP parameters are
converted into line spectrum pairs (LSP).First order moving
average prediction is used for both LSP sets. The LSP residual
vectors are jointly quantised using split matrix quantisation
(SMQ) with 5 submatrices of dimension 2x2 (two elements from
both sets). The submatrices are quantised with 7, 8, 9, 8, and 6
bits, respectively. A total of 38 bits are used for LSP
quantisation. For the 1st and 3rd subframes, LP parameters
interpolated from the adjacent subframes are used in the codec.
2.2 Pitch Analysis
The adaptive codebook is searched for the lag range [ 17 3/6,
1431 with a combined open-loop/closed-loop search [2]. A
fractional lag with 116th resolution is used for lag values below
95 in the 1st and 3rd subframes and for all lag values in the 2nd
and 4th subframes. The codebook search consists of the
following steps:
-~
An open-loop search for integer lag values is carried out
once every 10 ms from the weighted original speech. Small
lag values are preferred to avoid pitch multiples.
A closed-loop search for integer lag values is performed on
subframe basis. For the 1st and 3rd subframe the search is
carried out in the vicinity of the found open-loop lag [To]- 3,
To,+ 31 and for the 2nd and 4th subframe in the vicinity of
+
the lag found for the previous subframe [T,, - 5, Tps 41.
Fractions are searched around the closed-loop lag if it is less
than 95 (and always in the 2nd and 4th subframes).
The lag is quantised with 9 bits for the 1st and 3rd subframes
and with 6 bits for the other two subframes where the lags are
coded differentially. The codebook gain is quantised to 4 bits.
2.3 Fixed Codebook
An algebraic codcbook with 35 bits is used as the fixed
codebook. Each excitation vector contains 10 non-zero pulses,
with amplitudes +1 or -1. The 40 positions in each subframe are
divided into 5 tracks where each track contains two pulses. In
the design, the two pulses for each track may overlap resulting in
a single pulse with amplitude +2 or -2. The allowed positions for
pulses are shown in Table 11.
Table 11: Allowed pulse positions for each track.
The optimal pulse positions are determined using a nonexhaustive analysis-by-synthesis search:
For each of the five tracks, the pulse positions with
maximum absolute values of the sum of normalised
backward filtered target and normalised long-term prediction
residual are searched. From these the global maximum value
for all the pulse positions is selected. The first pulse p 0 is
always set in the position corresponding to the global
maximum value.
Five iterations are carried out in which the position of pulse
p l is set to one of the five track maxima. The rest of the
pulses are searched in pairs by sequentially searching each of
the pulse pairs (p2, p31, (p4, p 5 ) , (p6, ~ 7 1 ,
(p8, p 9 ) in
nested loops. For each iteration, all 9 initial pulse positions
are cyclically shifted so that the pulse pairs are changed and
the pulse p l is placed at a local maximum of a different
track. The rest of the pulses are searched also for the other
positions in the tracks. In the search, at least one pulse
position is located corresponding to the global maximum and
one pulse is located in a position corresponding to one of the
5 local maxima.
For subframes with a lag less than the subframe length,
adaptive pitch sharpening filter is used. The two pulse positions
in each track are encoded with 6 bits and the sign of the first
pulse in each track is encoded with one bit. The fixed codebook
gain is coded using moving average prediction and quantised to
5 bits.
772
2.4 Error Concealment
5. COMPLEXITY AND DELAY
Error concealment in the EFR codec is based on partially
replacing the parameters of the received bad frame with values
extrapolated from the previous good frames:
The LSPs of the previous good frame are used but shifted
towards their mean values.
The codebook gains are replaced by attenuated values
derived from the previous subframes using median filtering.
The amount of attenuation is different for the two codebooks
and depends on which state the error concealment is in. The
lag values are replaced by the lag value from the 4th
subframe of the previous good frame.
The received excitation pulses of the fixed codebook are
used as such.
In case a good frame is preceded by a bad frame, the
codebook gains for the good frame are limited below values used
for the last good subframe.
3. CHANNEL CODEC
The EFR channel codec is almost the same as the FR channel
codec because the design aim was to keep it as unchanged as
possible. During the GSM EFR codec standardisation, the use of
the existing FR channel codec (or any existing GSM generator
polynomials) was encouraged since this minimises hardware
changes in the GSM base stations and speeds up the introduction
of the EFR codec. In the PCS 1900 EFR coldec standardisation,
the use of the existing FR channel codec was an essential
requirement. Therefore, the FR channel codlec was included in
the EFR channel codec as a module together with additional
error protection. The additional error protection consists of an 8bit CRC parity check and a repetition code. The FR channel
codec module protects the 182 most important bits with 1R-rate
convolution code and it uses a 3-bit CRC that covers the 50 most
important bits. The bits in the EFR codec are divided into
protected and unprotected bits according to their subjective
importance to speech quality. Only 66 bits :are left unprotected.
These consist of the least significant bits of pulse positions in the
algebraic code. The 8-bit CRC covers the 65 most important
bits. It was included in the EFR codec to achieve reliable bad
frame detection and, consequently, to reduce the number of
undetected bad frames. These are a major source of audible
degradations in current digital cellular systems.
4. VAD/DTX
The EFR codec contains also the functioins of discontinuous
transmission (DTX) and voice activity. detection (VAD). DTX
allows the radio transmitter to be switched off during speech
pauses in order to save power in the mobile station and also to
reduce the overall interference level over thle air interface. VAD
is used on the transmit side to detect speech pauses, during
which characteristic parameters of the background acoustic noise
are transmitted to the receive side, where similar noise, referred
to as comfort noise, is then generated. The comfort noise
parameters in the EFR codec consist of averaged LSP parameters
and an averaged fixed codebook gain. Locally generated random
numbers are used on the receive side as excitation pulses. During
comfort noise generation, the adaptive codebook is switched off.
The estimated average speech channel activity for the EFR codec
is 64% [3].
The complexity of the EFR codec has been estimated from a
C-code implemented with a fixed point function library in which
each operation has been assigned a weight representative for
performing the operation on a typical DSP [3]. The theoretical
worst case complexity of the EFR codec has been estimated
during ETSI EFR verification phase to be 18.1 WMOPS
(weighted million operations per second) [3]. This is below that
of the GSM half rate codec (21.2 WMOPS) [3], [4]. Memory
consumption estimated for data RAM (4.7k 16-bit words), data
ROM (5.9k 16-bit words) and program ROM are each below
those of the GSM half rate codec.
The delay of the EER codec is approximately the same as
that of the FR codec. Both codecs have a buffering delay of 20
ms without any lookahead. The round-trip delay (uplink delay +
downlink delay) for EFR taking into account all system and
processing delays of the GSM network is 191.0 ms while for the
FR codec it is estimated to be 188.5 ms [3]. The difference is
unnoticeable.
6. SUBJECTIVETEST RESULTS
The most extensive subjective tests of the performance of the
EFR codec are from the PCS 1900 EFR codec validation tests
[ 5 ] . These were carried out to characterise the performance of
the EFR codec after selection for PCS 1900. The standardisation
process in ETSI also included pre-selection tests which were
carried out in six laboratories, but no common analysis
averaging the scores over all laboratories exists [3]. The results
from both tests are well in line with each other. Both show that
the EFR codec has basic speech quality at least equal to that of a
wireline reference (G.721 32 kbit/s ADPCM in PCS 1900 tests
and G.728 16 kbit/s LD-CELP in ETSI tests).
The PCS I900 EFR codec validation test results are
discussed first. The EFR codec was tested in three channel error
conditions with C/I-ratios 13, 10 and 7 dB. The channel bit
error-rates for these are approximately 2%, 5% and 8%,
respectively. The two lowest error-rate conditions correspond to
operating well inside a cell while the 7 dB C/I condition
corresponds to operating at a cell boundary. Figure 2 shows the
results from channel error test. These show that the EFR codec
performs much better than the FR codec in the error-free case
and in all the tested error conditions. In error-free and low errorrate conditions (C/I=13 dB), the performance of the EFR codec
is statistically better (based upon 95% confidence intervals) than
the performance of the error-free reference ADPCM codec. At
medium error-rate conditions (C/I=lO dB) the EFR codec
performs equally well to (error-free) ADPCM. Only at medium
to high error-rates (C/I1.5
2
3
I
14
Error-free
Cll=13 dB
1-GSM
EFR -GSM
C/I=lO dB
FR
--
ADPCM (error-free)
C l k 7 dB
I
Figure 3: Results from background noise test.
Figure 2: Results from channel error test.
Tandem Test (ACR, MOS)
Talker Dependency Test (DCR, DMOS)
4.5
4
U)
3.5
0
=
3
OGSM FR
2.5
2
GSM EFR
GSM FR
ADPCM
Figure 4: Results from tandem test.
Figure 5: Results from talker dependency test.
EFR codec was found statistically better than ADPCM.
The ETSI pre-selection tests consisted of five experiments:
transmission errors (C/I=IO and 7 dB), tandeming (C/I=IO dB),
background noise (music 20 dB and vehicle 10 dB), talker
dependency and high error conditions (C/I=4 dB). The
performance of the EFR codec was found equal to G.728 for
error-free transmission, speech in background noise (for both
noise types) and talker dependency. No testing was carried out
for the low error-rate condition C/I=13 dB. In erroneous
transmission at CLI=lO dB and 7 dB, the EFR codec was found
clearly better than the FR codec. At C/I=lO dB, the EFR codec
performed equal to or better than MNRU 24 dB in half of the
tests. For the outside-a-cell error condition of C/I=4 dB, the
results show that the EFR codec has approximately the same
performance as the FR codec. The EFR codec was tested in
error-free self-tandem in one laboratory and was found
equivalent to (3.728. In self-tandem at C/I=10 dB, the EFR
codec performed clearly better than the FR codec and equal
performance to the FR codec in single encoding at C/I=lO dB
was demonstrated in all laboratories except one.
Verification tests complementing the pre-selection tests were
carried out in ETSI for such items as DTMF and network
information tones, frequency response, complexity, delay,
different languages, music and special input signals (e.g.,
different input levels, sine waves, noise signals etc.). In all the
verification tests, the EFR codec performed well [3]. For DTMFtones, the EFR codec was found 100% transparent.
associated previously only with fixed networks to the end users
of mobile communication systems. The GSM EFR specifications
have been completed in 1996 and the codec is expected to be
deployed in GSM, DCS 1800 and PCS 1900 networks in 19971998.
ACKNOWLEDGEMENTS
The authors acknowledge the efforts of M. Fatini and J. Hagqvist
in PCS 1900 EFR codec standardisation, the algorithmic
contribution of B. Bessette and the efforts of members of ETSI
SMG2 SEC, especially those who have carried out pre-selection
and verification tests for the EFR codec.
REFERENCES
[ 1J “Enhanced Full-Rate Speech Codec - Study Phase Report,”
[2]
[3J
[4]
7. CONCLUSIONS
The GSM EFR codec has met and even exceeded the essential
requirements set for the development of the EFR codec. It
provides substantial improvement in speech quality compared to
the existing GSM full rate codec and brings high speech quality
[SI
774
Report of the ETSI SMGl sub-group on the EFR codec,
Version 0.5.3, December 19, 1994.
R. Salami, C. Laflamme, J-P. Adoul, and D. Massaloux, “A
toll quality 8 kb/s speech codec for the personal
communications system (PCS),” IEEE Trans. Veh. Technol.,
vol43, no. 3, pp. 808-8 16, Aug. 1994.
“Digital cellular telecommunications system; Performance
characterization of the GSM enhanced full rate speech codec
(GSM 06.55),” ETSI Technical Report (In preparation),
Draft ETR 305, version 1.O.O.
Complexity Evaluation Subgroup of ETSI TCH-HS (Alcatel
Radiotelephone, Analog Devices, Italtel-SIT, Nokia),
”Complexity Evaluation of the Full-Rate, ANT and
Motorola Codecs for the Selection of the GSM Half-Rate
Codec,” Proceedings of the 1994 DSPx Exposition &
Symposium, June 13-15, San Fransisco (USA), 1994.
“COMSAT Test Report on the PCS1900 EFR Codec,”
Source: Nokia, Temporary Document 47/95 of ETSI SMG2SEC, June 1995.
Exhibit 8
A GOlOD JOB WELL DONE:
THE LATEST WIRELESS CODECS DELIVER WIRELINE QUALITY
Leigh A. Thorpe & Paul V. Coverdale
N ortel, Ottawa, Canada
Abstract
The results of a listening test conducted at Nortel on behalf
of the CDMA Development Group (CDG) suggest that the
new crop of wireless codecs are all able to deliver voice
quality comparable to wireline with single encoding over a
clean wireless transmission channel.
The codecs were tested with clean speech, varying input
level, background noise, and tandem encoding. Given these
findings, we hope to provoke discussion among the: audience
about the next challenges facing speech coding researchers
and codec designers. Potential issues: codec robustness to
errors, noise reduction, and very-low bit rate codecs for a
variety of applications.
Introduction
How good is the general quality of the latest codecs intended
for the three major wireless technologies? How does that
quality compare to the quality of wireline equipment?
Listening tests were conducted to evaluate the performance of
the current standard codecs for the three major wireless
technologies. The codecs tested are the latest standards for
CDMA, North American TDMA, and GSM (PCS1900).
The tests examined performance with clean Speech and
robustness to input impairments and tandem encoding.
Method
Four codecs intended for wireless and two codecs used in
wireline networks were evaluated. Clean source speech and
speech with input impairments was processed through each
codec in a non-real-time software simulation. Ratings were
gathered from listeners in an absolute category rating (ACR)
procedure[ 11.
Test and reference codecs
The following codecs listed below in the tests. Mu-law
PCM and ADPCM were included to represent the range of
wireline quality. ADPCM at 32 kb/s is generally taken to
represent the lower boundary of wireline quality.
Test codecs:
TDMA EFRC: 8 kb/s enhanced full-rate codec (EFRC) for
TDMA (IS-641).
CMDA EVRC: the new 8 kb/s variable-rate coding standard
for CDMA systems. (This codec includes a noise
reduction algorithm applied to the input speech signal.)
CDMA Q13: The 13 kb/s variable-rate codec proposed by
the CDG for CDMA systems.
GSM EFWC:
13 kb/s enhanced full-rate (EFR) for GSM
and PCS 1900.
Reference codecs:
p-law PCM:
G.711 at 64 kb/s. Used in North American
digital wireline switching and transmission.
ADPCM:
G.726 at 32 kb/s. Used on many international
wireline calls and private networks; also in CT2 and
DECT wireless standards.
Test cases
The test cases examined the contribution of varying input
level, background noise level and degradation from
asynchronous tandem operation. (Because of the difficulty of
defining comparable channel degradation for the various
wireless systems, we did not try to compare the effects of
transmission impairments across wireless platforms.) The
test cases included the following:
Clean speech: SNR > 45 dB; -20 dBmO input level
Input levels:
SNR > 45 dB; -10 dBmO and -30 dBmO
Background noise: car interior noise at 10 and 20 dB SNR;
street noise at 15 dB SNR; babble at 20 dB SNR; and
Hoth noise at 20 dB SNR (white noise filtered to model
long-term average room noise).
Asynchronous tandem:
clean speech input; nominal level.
The source file is encoded, decoded, converted to analog,
re-digitized, encoded and decoded a second time.
Additional reference cases processed through the modulated
noise reference unit [ 2 ] were also included in the session.
Samples with dB Q values of 5, 10, 15, 20, 25, 30 and 35
were used.
Preparation of source files and test samples
Source speech from four talkers was prepared from highquality 16-bit linear PCM recordings from each of four
talkers (two men and two women). To maintain equivalent
speech levels for each sample, the filtered source files were
equalized for speech power as determined by the P.56
algorithm (SV6) [3]. The samples were then filtered with
the modified IRS transmit filter [ 11 and processed through plaw PCM before processing through the test codecs to
simulate the effects of the wireline portion of a connection.
For speech-plus-noise samples, the filtered, equalized speech
was mixed with filtered noise scaled to the appropriate level
to achieve the intended signal-to-noise ratio.
85
Figure 1.
4
3
E
Mean ratings given to the clean
speech case for each of the test and
reference codecs. The results for all
of the wireless codecs fall between
those for p-law PCM and ADPCM,
often taken as defining the range of
wireline quality.
3
2
Bad 1
plaw AD EVRC Q13 TDMA GSM
EFRC EFRC
PCM PCM
The chart shows the results for the clean speecwclean
transmission test case. All of these codecs were found to
provide voice quality better than ADPCM. In addition, Q13,
EVRC, and GSM EFRC showed voice quality equivalent to
or nearly equivalent to y-law PCM with clean input speech.
Source files were processed through each codec in a non-realtime software simulation. Asynchronous tandem processing
was done in digital simulation using up-sampling, applying
an all-pass filter, and downsampling again [4].
Listening test procedure
Results for cases with background noise showed that all the
test codecs performed as well or better than ADPCM with all
the noise types tested. Because of its on-board noise reduction algorithm, the EVRC performed better than any other
codec in the noise cases. Finally, voice quality remains
acceptable even with asynchronous tandem processing.
Speech samples for each test case were rated by 60 typical
telephone users in a carefully controlled listening test.
Listeners rated the overall audio quality of the sample on a 5point scale. The rating scale was defined as 1-bad, 2-poor,
3-fair, &good, and S-excellent. All listeners rated every test
sample once in a classic repeated-measures design. Such a
design allows variation due to differences in subject rating
criteria to be partialled out in an analysis of variance.
These data demonstrate clearly that the latest codecs for each
of the major wireless standards (Q13 and IS-127 for CDMA,
IS-641 for TDMA, and EFRC for GSM) can deliver voice
quality equivalent to wireline under good transmission conditions. Given these findings, we hope to provoke discussion among the audience about the next challenges facing
speech coding researchers and codec designers. These perhaps
include: robustness to transmission impairments, noise
reduction, and the development of half-rate codecs with wireline equivalent quality for wireless and other applications.
The test samples were played back to listeners over one
channel of high-fidelity headphones. The playback signal
was filtered to simulate an ideal handset receiver response,
and was presented at 79 dB SPL. Listeners were told to wear
the headphones with the live speaker at their telephone ear.
Listeners were run in groups of three. Each group received a
different randomization of the test samples. Randomization
was done using a randomized block design, with each test
case presented once in each of four blocks. This controls
both for effects of order of presentation and for criterion
shifts due to effects of familiarization and fatigue. The
randomization was also constrained so that samples from the
same talker were never presented consecutively.
References
[ l ] Recommendation P.830 (1 996). Subjective performance
assessment of telephone-band and wideband digital codecs.
Geneva: International Telecommunications Union.
[ 2 ] Recommendation P.8 1 (1993). Modulated Noise Reference
Unit (MNRU). Geneva: International Telecommunications
Union.
Listeners were given written instructions to read at the
beginning of the listening session, and completed a short
practice session before the test session began. The whole
session took about one hour to complete.
[3] Recommendation G.191 (1993). Software tools for speech
and audio coding standardization. Geneva: International
Telecommunications Union.
Results
[4] Recommendation P.56 (1993). Objective measures of active
Listener ratings for each test case were averaged to obtain the
Mean Opinion Score (MOS). Data were collapsed over
talkers in computing these means. In an experiment with 60
listeners and 240 judgments per test case, differences of
greater than about 0.1 MOS are statistically reliable.
speech level. Geneva: International Telecommunications
Union.
86
Exhibit 9
Exhibit 10
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