ADPCM Equipment for 9.6-Kbps Data
The ADPCM algorithm proposed by OKI Electric of
Japan seems to be a formidable alternative for the
standard.
(an article taken from Telephony magazine, September 1987)
[+] by Yoshihiko Yokoyama
In 1982, the CCITT started work on developing a second
digital encoding standard for speech, after decades of
extensive use of PCM at 64 kbps in the A-law or u-law
formats. The result of that effort was, the encoding
standard of the 32-kbps ADPCM algorithm, known as CCITT
recommendation G.721. It was recognized from the beginning
that the algorithm should maintain adequate performance for
voice-band data signals, although it was acknowledged that
such signals were limited to data rates of up to 4800 bps for
the state-of-the-art ADPCM algorithms. This has resulted in
a virtual hesitation of widespread application of the
standard in the public switched telephone networks (PSTNs),
for which it was intended. Network operators have concluded
that a fast-growing need exists for transmitting data at 9600
bps for their customers, and using G.721 makes that
impossible.
Susequently, the CCITT has embarked on a course of defining
a digital encoding standard for digital circuit
multiplication equipment (DCME), which combines time
assignment speech interpolation (TASI) and a low-rate
encoding technique such as ADPCM to form a very efficient
means of transmitting speech. How to transmit data in such a
system has been the subject of considerable debate and
extensive effort by many experts in the field. It should be
pointed out that, similar to the transcoding standard of
G.721, interfacing with the DCME must be accomplished by
means of an A-law or u-law encoded PCM signal format.
The need for transmitting data up to 9600 bps has been
recognized, and three algorithms have undergone scrutiny by a
group of experts in the field. Two of the algorithms have
the inherent capability of transmitting 9600-bps voice-band
data at the 32-kbps rate, whereas the third algorithm under
consideration is G.721, which does not have that capability.
[+] PRESENT STANDARDIZATION EFFORTS
DCME Aspects
A DCME system is basically an all-digital implementation of
the old concept of TASI. DCME systems operate on the
statistical behavior of a group of talkers in a communication
system. This is characterized by the average time that a
talker on a connection is actually active, nominally assumed
to be 35-40 percent of the total time the circuit is used for
a call. Thus, the remaining time is available for
time-interleaving the speech of other talkers. On the
average, circuit usage can be increased or multiplied by a
factor called digital speech interpolation (DSI) gain. Gain
factors between 2 and 2.5 are commonly used, but these gain
factors are dependent on the actual speech activity exhibited
by the talkers. The larger the group of talkers, the more
statistical stability is attained, and individual
fluctuations in speech activity can be accommodated by the
system. Long talk spurts by one talker are simultaneously
compensated by silence or shourt spurts by another.
Short durations of active speech, more than can be
accommodated by available transmission capacity, do occur.
Without "special means," this would result in what is known
as clipped speech. In DCME, this special means is provided by
instantly reducing the coding rate of one or more channels
(talkers). That is, when the DCME operates nominally with
ADPCM at 32 kbps during overload, this rate is reduced to
24 kbps for one or more channels. As sampling occurs at 8000
times per second, this means that the nominal channel being
encoded at 4 bits/sample is reduced to encoding at 3
bits/sample during overload. This brings about a small
degradation in performance by increased quantizing noise, but
it occurs only sporadically due to the statistical nature of
the phenomenon. Therefore, it is virtually imperceptible as
long as the signal load to the DCME is strictly speech. When
an appreciable part of the DCME load is data (more than 20
percent), special precaution must be taken to prevent
noticeable degradation, because data signals do not exhibit
the same on-off activity as speech. In fact, data are
considered, generally, as being 100 percent active, thus
providing no bearer circuit-sharing capability.
When the DCME load is a mix of speech and data, it is clear
overload will occur more often for the speech signals,
resulting in an associated decrease in performance in the
form of higher quantizing distortion. The choice of ADPCM
algorithm for the DCME has an important bearing on this
problem.
[+] CCITT EFFORTS
The CCITT is considering using the basic G.721 algorithm
for speech at 32 kbps for DCME, but due to that algorithm's
inablity to handle 9600-bps data at 32 kbps, encoding at 40
kbps per channel is needed for data signals at such rates.
This is clearly having a more profound influence on the use
of available bearer transmission capacity than if encoding of
data could be limited to using the 32-kbps bearer rate per
channel. For example, a 60-channel DCME system employing a
proprietary ADPCM developed by OKI Electric of Japan can
accommodate 10 percent data traffic up to 9.6 kbps, whereas
G.721 ADPCM can only accommodate 6.7 percent data and
maintain the same speech performance. Moreover, the DCME
design is considerably simpler with the proprietary ADPCM,
since there is no need to reconfigure the frame structure for
including 5-bit/sample encoding for data.
Another aspect of ADPCM in DCME systems is the need to
tandem such systems for multilink networking purposes. It
can generally be argued that no more than two DCME links
should be allowed to be switched in any end-to-end
connection. If such switching is performed by an analog
switch (asynchronous tandeming), an accumulation of
distortion will be experienced in the second link.
However, if a digital switch would be employed, directly
operating on the PCM output of the first DCME link, passing
it digitally on to the second link (synchronous tandeming),
no additional distortion will be experienced. Both the OKI
ADPCM and the G.721-related technique in DCME application
will have the "synchronous" capability as an inherent part of
the design. A third algorithm, mentioned earlier, does, not
possess that capability, and it will not be discussed.
Digital switching will increasingly be employed in the
public networks. Therefore, the loss of performance due to
asynchronous tandeming, if it occurs at all, may only be
temporarily experienced and should not pose a serious
concern. This aspect of tandeming is not uniquely related to
DMCE systems. Any application of 32 kbps could encounter the
need for tandeming in a network. As digital switching will
be increasingly applied, either by replacing analog switches
or in new installations, the advantage of the ADPCM technique
will be even more evident because of its capability of
transmitting up to 9.6-kbps voice-band data signals.
The CCITT nevertheless has decided to hold on to the G.721
technique, even though a clearly superior technique in now
available.
[+] OKI ADPCM
PERFORMANCE
Data
Extensive performance measurements have been made in a
carefully assembled test bed at COMSAT Laboratories. (This
test bed received approval by the organizations that
submitted ADPCM equipment for evaluation and comparison in a
CCITT context. This made comparisons between algorithms
valid and accurate.) The circuit in which the ADPCM
equipment was tested included a simulated analog access link
which introduced typical distortion effects (analog
impairments) that a voice-band data signal may experience
before being encoded by the ADPCM link.
The typical performance after encoding by OKI ADPCM of a
CCITT V.29 modem (The V.32 modems will perform even better
than V.29 modems because of their inherent design. Thus,
V.29 performance shown (graphs are not shown here in this
text due to the inablility to draw or copy it here with
this word processor) here is more critical to the user.)
in terms of block error rate (BLER), as a function of S/N
ratio of the data signal in the analog impairment circuit
(i.e, just before being encoded), is illustrated in figure 1.
Ther lower curve shown resulted after a single ADPCM
encoding, whereas the higher curve resulted after a second
ADPCM link was added to the first by means of an analog
interconnection between the two links. Thus, this second
curve is the result of asynchronous tandeming of two links.
The curve showing single encoding perfomance applies also for
the case of multiple encodings via digital switches, referred
to as synchronous tandeming. A reference performance
threshold of BLER = 10-2nd power at S/N =30.5 db (this
reference point was selected by an SG XVIII expert group.) is
well met by both curves. This indicates the excellent
capability of the ADPCM equipment for transmitting 9.6-kbps
V.29 signals.
The performance of a V.29 modem operating at the back-off
rate of 4.8-kbps tandem through four asynchronous encodings
of the ADPCM equipment is shown in figure 2. For comparison,
the dashed curve in fig. 2 shows the performance of the same
modem when four asychronous links of G.721 ADPCM equipment
are substituted for the OKI equipment. At S/N values to be
expected in the networks, the OKI advanced ADPCM can perform
two or more orders of magnitude better than G.721. This may
not be required for this modem speed, but it is simply a
consequence of its inherently more powerful predictor than
that employed in G.721, and, as such, it provides an
increased performance margin.
Voice
When considering ADPCM designs, the primary purpose has
always been to provide high performance for voice signals.
This objective has unquestionably been attained by the
G.721 designers. Extensive subjective tests have proven
the algorithm delivers the speech performance required for
the networks.
Similarly, the OKI ADPCM equipment provides the required
performance when speech is transmitted through it. Tests
similar to those used for evaluating the G.721 algorithm have
been performed with the OKI ADPCM equipment, particulary for
the English and Japanese languages.
DCME Gain
As has been pointed out earlier in the article, when
applied in DCME systems, the proprietary ADPCM technique
offers the advantage of encoding all voice-band data by using
only only 4 bits/sample. This offers a bearer-channel
efficiency advantage of up to 20 percent when transmitting 60
channels with 20 percent data. This includes a
bearer-capacity increase to avoid speech degradation. Such
an advantage may be particularly important for countries that
may want to minimize their cost of communication.
It should be emphasized, however, that without DCME, the
main advantage of the propietary ADPCM resides in its
capability of transmitting up to 9.6-kbps voice-band data.
This has an important bearing on networks, since meeting
this requirement is or will become indispensable.
-------------------------------------------------------------
Yoshihiko Yokoyama is the General Representative for OKI
America, Inc., New York office.
--------------------------------------------------------------
TUCoPS is optimized to look best in Firefox® on a widescreen monitor (1440x900 or better).
Site design & layout copyright © 1986-2024 AOH
