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TUCoPS :: Phreaking Technical System Info :: adpcm.txt

Information on ADPCM







                     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.

         --------------------------------------------------------------

  


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