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