Plural telephony channel baseband signal demodulator for a broadband communications system

A communication system for coupling telephony or other digital networks to a broadband network such as a CATV network. The system transmits a multiplex of telephony signals in the forward band of the broadband network, with individual signals directed to an addressed subscriber. Telephony signals returning from subscribers are modulated onto the reverse band of the broadband network in a frequency division multiplex (FDM). The modulated telephony signals are received at a telephony network interface coupled to the broadband network. A selected plurality of modulated telephony signals are frequency shifted to baseband. The baseband-shifted telephony signals are sampled to generate a plurality of time division multiple access telephony signal samples. A plurality of time division multiple access telephony signal samples are channelized into a serial data stream with a channelizer. The serial data stream is processed with a single digital signal processor (DSP) to derive a plurality of demodulated telephony signals. The demodulated telephony signals are coupled to the telephony network.

FIELD OF THE INVENTION 
The present invention pertains generally to communicating telephony 
signals, and other or similar signals, over CATV and equivalent networks, 
and is more particularly related to processing of telephony signals 
transmitted in the reverse path of a CATV network. 
CROSS REFERENCE TO RELATED APPLICATIONS 
This application is related to application Ser. No. 08/219,848, filed Mar. 
30, 1994, entitled "Frequency Agile Broadband Communications System", now 
U.S. Pat. No. 5,594,726, which is a continuation-in-part of application 
Ser. No. 08/123,363, filed Sep. 17, 1993, entitled "Broadband 
Communications System", now U.S. Pat. No. 5,499,241, the disclosures of 
which are incorporated herein by reference and made a part hereof. 
BACKGROUND OF THE INVENTION 
In order to introduce the present invention and the problems that it 
solves, it is useful to overview a conventional CATV broadband 
communication system, and then examine certain prior approaches to 
problems encountered when attempting to introduce telephony signals into 
the broadband environment. 
CONVENTIONAL CABLE TELEVISION SYSTEMS (CATV) 
Cable television systems, sometimes referred to as community-antenna 
television (CATV) systems, are broadband communications networks of 
coaxial cable and optical fiber that distribute television, audio, and 
data signals to subscriber homes or businesses. In a typical CATV system, 
a single advantageously located antenna array feeding a cable network 
supplies each individual subscriber with a usable television signal. It is 
estimated that CATV networks currently pass approximately 90% of the 
population in the United States, with approximately 60-65% of all 
households actually being connected. 
A typical CATV system comprises four main elements: a headend, a trunk 
system, a distribution system, and subscriber drops. 
The "headend" is a signal reception and processing center that collects, 
organizes and distributes signals. The headend receives 
satellite-delivered video and audio programming, over-the-air broadcast TV 
station signals, and network feeds delivered by terrestrial microwave and 
other communication systems. In addition, headends may inject local 
broadcasting into the package of signals sent to subscribers such as 
commercials and live programming created in a studio. 
The headend contains signal-processing equipment that controls the output 
level of the signals, regulates the signal-to-noise ratio, and suppresses 
undesired out-of-band signals. Typical signal-processing equipment 
includes a heterodyne processor or a demodulator-modulator pair. The 
headend then modulates received signals onto separate radio frequency (RF) 
carriers and combines them for transmission over the cable system. 
The "trunk system" is the main artery of the CATV network that carries the 
signals from the headend to a number of distribution points in the 
community. A modern trunk system typically comprises of a combination of 
coaxial cable and optical fibers with trunk amplifiers periodically spaced 
to compensate for attenuation of the signals along the line. Such modern 
trunk systems utilizing fiber optics and coaxial cable are often referred 
to as "fiber/coax" systems. 
The "distribution systems" utilize a combination of optical fibers and 
coaxial cable to deliver signals from the trunk system into individual 
neighborhoods for distribution to subscribers. In order to compensate for 
various losses and distortions inherent in the transmission of signals 
along the cable network, line-extender amplifiers are placed at certain 
intervals along the length of the cable. Each amplifier is given just 
enough gain to overcome the attenuation loss of the section of the cable 
that precedes it. A distribution network is also called the "feeder". 
There is a strong desire in the CATV and telecommunications industry to 
push optical fiber as deeply as possible into communities, since optical 
fiber communications can carry more signals than conventional networks. 
Due to technological and economic limitations, it has not yet proved 
feasible to provide fiber to the subscriber's home. Present day "fiber 
deep" CATV distribution systems including optical fibers and coaxial cable 
are often called "Fiber-To-the-Serving-Area" or "FTSA" systems. 
"Subscriber drops" are taps in the distribution system that feed individual 
75.OMEGA.coaxial cable lines into subscribers' television sets or 
subscriber terminals, often referred to as "subscriber premises equipment" 
or "customer premises equipment" ("CPE"). Since the tap is the final 
service point immediately prior to the subscriber premises, channel 
authorization circuitry is often placed in the tap to control access to 
scrambled or premium programming. 
Cable distribution systems were originally designed to distribute 
television and radio signals in the "downstream" direction only (i.e., 
from a central headend location to multiple subscriber locations, also 
referred to as the "forward" path). Therefore, the component equipment of 
many older cable systems, which includes amplifiers and compensation 
networks, is typically adapted to deliver signals in the forward direction 
only. For downstream transmissions, typical CATV systems provide a series 
of video channels, each 6 MHz in bandwidth, which are frequency division 
multiplexed across the forward band, in the 50 MHz to 550 MHz region of 
the frequency spectrum. As fiber is moved more deeply into the serving 
areas in fiber/coax and FTSA configurations, the bandwidth of the coax 
portion is expected to increase to over 1 GHz. 
The advent of pay-per-view services and other interactive television 
applications has fueled the development of bidirectional or "two-way" 
cable systems that also provide for the transmission of signals from the 
subscriber locations back to the headend. This is often referred to as the 
"upstream" direction or the "reverse" path. This technology has allowed 
cable operators to provide many new interactive subscriber services on the 
network, such as impulse-pay-per-view (IPPV). In many CATV systems, the 
band of signals from 5 MHz to 30 MHz is used for reverse path signals. 
However, the topology of a typical CATV system, which looks like a "tree 
and branch" with the headend at the base and branching outwardly to the 
subscriber's, creates technical difficulties in transmitting signals in 
the upstream direction back to the headend. In the traditional tree and 
branch cable network, a common set of downstream signals are distributed 
to every subscriber home in the network. Upstream signals flowing from a 
single subscriber toward the headend pass by all the other upstream 
subscriber homes on the segment of distribution cable that serves the 
neighborhood. 
The standard tree and branch topology has not proven to be well suited for 
sending signals from each subscriber location back to the headend, as is 
required for bidirectional communication services. Tree and branch cable 
distribution systems are the most efficient in terms of cable and 
distribution usage when signals have to be distributed in only the 
downstream direction. A cable distribution system is generally a very 
noisy environment, especially in the reverse path. Interfering signals may 
originate from a number of common sources, such as airplanes passing 
overhead or from Citizens Band (CB) radios that operate at a common 
frequency of 27 MHz, which is within the typical reverse channel bandwidth 
of CATV networks. Since the reverse direction of a tree and branch 
configuration appears as an inverted tree, noise is propagated from 
multiple distribution points to a single point, the headend. Therefore, 
all of the individual noise contributions collectively add together to 
produce a very noisy environment and a communications problem at the 
headend. 
Present day FTSA systems facilitate the communication of signals in the 
reverse direction by dividing the subscriber base of a cable network into 
manageable serving areas of approximately 400-2500 subscribers. This 
allows for the reuse of limited reverse band frequency ranges for smaller 
groups of subscribers. The headend serves as the central hub of a star 
configuration to which each serving area is coupled by an optical 
communications path ending in a fiber node. The fiber node is connected to 
the serving area subscribers over a coaxial cable distribution sub-network 
of feeders and drops in each serving area. In the FTSA configuration, some 
of the signals in the forward direction (e.g., television program signals) 
are identical for each serving area so that the same subscriber service is 
provided to all subscribers. In the reverse direction, the configuration 
provides an independent spectrum of frequencies confined to the particular 
serving area. The FTSA architecture thus provides the advantage of 
multiplying the bandwidth of the reverse portions of the frequency 
spectrum times the number of serving areas. 
The Desire for Telephony Service 
The ever-expanding deployment of fiber optic technology in CATV systems 
across the country has cable operators looking to provide a whole new 
range of interactive services on the cable network. One area that is of 
particular interest is telephony service. Because of recent advances in 
technology as well as the loosening of regulations, the once distinct 
lines between the cable television network and the telephone network have 
blurred considerably. Currently there is a great demand for a broadband 
communication system that can efficiently provide telephone service over 
the existing cable distribution network. 
Moreover, there is substantial interest expressed by telephone system 
operating companies in the idea of increased bandwidth for provision of 
new services to telephone subscribers, such as television; interactive 
computing, shopping, and entertainment; videoconferencing, etc. Present 
day "copper" based telephony service (so called because of the use of 
copper wires for telephone lines) is very bandwidth limited--about 3 
kHz--and cannot provide for such enhanced services by the telephone 
companies without massive changes to the telephone networks 
infrastructure. 
Existing communications systems, however, have not proven to be well suited 
for the transmission of telephony signals on the cable network. A system 
for transmitting telephony signals must be configured to allow single 
point to single point distribution (i.e., from a single subscriber to a 
single subscriber). However, unlike the telephone companies with their 
well-established national two-way networks, the cable industry is 
fragmented into thousands of individual systems that are generally 
incapable of communicating with one another. The cable network is instead 
ideally configured for single point to multiple point signal transmission 
(i.e., from a single headend downstream to multiple subscriber locations). 
Moreover, CATV systems do not have the switching capabilities necessary to 
provide point to point communications. A communications system for the 
transmission of telephone signals must therefore be compatible with the 
public switched telephone networks ("PSTN") operated by the telephone 
operating companies. To be useful in the carriage of telephony signals, a 
CATV network must be able to seamlessly interface to a telephony network 
at a point where it is commercially viable to carry telephony signals. It 
must also provide signals that can pass to other parts of the 
interconnected telephone systems without extensive modulation or protocol 
changes to thereby become part of the international telephone system. 
Telephony on Broadband Network 
One approach taken to provide a bidirectional broadband communications 
system is shown in the above-referenced related U.S. patents which are 
owned by the assignee of the present invention. These patents describe 
broadband communication systems that utilize two different modulation 
schemes for communicating information between a central headend and a 
plurality of subscriber nodes. For downstream communications from the 
headend, telephony signals are transmitted in a plurality of 3 MHz 
bandwidth channels utilizing QPR modulation, with each 3 MHz band carrying 
the equivalent of 96 DS0 telephony channels. For upstream communications, 
one system uses a frequency-agile quadrature phase shift keyed (QPSK) 
modulation scheme that transmits each subscriber's outgoing DS0 telephony 
channel in one of 480 separate 49.5 kHz bands in the 5-30 MHz reverse 
band. 
Telephony signals that are carried on the broadband network are coupled to 
the telephony network at various points, such as the headend unit (HIU) or 
at separate network interface connection points. In a distributed 
architecture where a limited number of telephony signals are carried in 
the broadband domain, it is important that the telephony network interface 
be inexpensive, compact, and structurally simple. Nonetheless, each 
network interface should be structured so as to provide connection to the 
telephony network in a customarily accepted format such as in DS1 or T1, 
DS2, SONET, or similar multiple telephony channel formats. 
The need for efficient signal processing of the upstream telephony channels 
is most acute in the upstream channel, since a given voice channel 
originating with a subscriber can appear anywhere in the 5-30 MHz 
spectrum, at different frequencies at different times, as a result of the 
operation of the frequency reassignment scheme used when the communication 
environment becomes noisy. Although processing the signals in the digital 
domain would be preferable from an implementation and hardware standpoint, 
with present day technologies it is not cost-effective to handle the QPSK 
signal digitally at the carrier frequency. On the other hand, if the 
telephony channel could be converted to the relatively low frequency 
baseband (such as approximately 50 kHz for QPSK), then it is possible for 
plural DS0s to be handled by a single digital signal processor. 
Therefore, there is a need for a broadband communications system that is 
compatible with the existing public switched telephone networks and that 
is not sensitive to noise or other interference issues, particularly in 
the reverse path. 
There is also a need for a broadband communications system that is 
bandwidth efficient and provides a higher spectral efficiency than present 
systems, thereby increasing the number of subscribers that may be served 
by each broadband network with telephony and enhanced services offered by 
CATV system operators, telephone company operating companies, and others. 
There is also a need for a telephony network/broadband communication 
network interface that allows handling of plural telephony channels in an 
upstream communication path in an efficient and cost-effective manner. 
There is also a need for a digital signal processing scheme that allows the 
processing power of a digital signal processor to operate upon plural 
telephony channels, which are typically bandwidth limited to about 8 kHz, 
to obtain a savings in density, cost and space. 
There is also a need for a broadband communication system telephony system 
network interface that allows efficient retrieval of plural telephony 
channels assigned to predetermined and variable frequency assignments 
because of frequency agile operation, that efficiently collects and 
multiplexes plural telephony channels into a predetermined telephony 
signal format for communication on the telephony network. 
That the present invention achieves these objects and fulfills the needs 
described hereinabove will be appreciated from the detailed description to 
follow and the appended drawings. 
SUMMARY OF THE INVENTION 
The invention includes methods and apparatus for providing broadband 
communications, including bidirectional telephony communications, over a 
cable distribution network. In particular, the present invention provides 
an integrated CATV/telephony system that is compatible with today's public 
switched telephone networks and can also deliver video, data, security 
monitoring, and other services without affecting current in-home wiring or 
equipment. 
Briefly described, the present invention provides a system for coupling 
telephony signals communicated from a subscriber via a broadband 
communication network to a telephony network interface. An analog front 
end comprising a plurality of frequency converter circuits is provided for 
downconverting the center frequency of a given bandwidth comprised of a 
predetermined number n of DS0 channels to DC. A channelizer is provided 
for taking a composite of the predetermined number n DS0 channels in a 
given band and producing n baseband DS0 channels. The channelizer provides 
serial data corresponding to time division multiple access (TDMA) sampled 
signals corresponding to the modulated telephony signals at baseband. A 
baseband demodulator constructed from a single digital signal processor 
(DSP) is provided for demodulating the TDMA sampled signals and deriving 
demodulated telephony signals. The demodulated telephony signals are 
coupled to the telephony network an output interface such as a headend or 
separate telephony network interface. 
More particularly described, the present invention provides an improved, 
multi-DS0 channel baseband processing demodulator that can process a 
plurality (six in the disclosed embodiment) baseband DS0 channels with a 
single DSP. The system including the DSP includes an input port for 
receiving a serial data stream comprising a plurality of digital signal 
samples representing a plurality of QPSK-modulated DS0 telephony channels 
at baseband. The processing stages effected in the DSP include an 
automatic gain control (AGC) stage for adjusting the gain of the digital 
signal samples for each of the independent DS0 telephony channels. The DSP 
also carries out a symbol timing recovery (STR) stage for extracting 
timing information for each of the QPSK-modulated DS0 telephony channels 
from the digital signal samples and delaying the sampling of the digital 
signal samples at a decoding stage to a time such that the sampling will 
be at optimized symbol instants. Further, the DSP provides a carrier phase 
recovery stage for locking the demodulator to the frequency of the carrier 
for each of the QPSK-modulated DS0 telephony signals. Finally, the DSP 
carries out a symbol decoding and formatting stage for sampling the 
digital signal samples and providing a digital signal output corresponding 
to the each of the demodulated DS0 telephony channels. 
In the disclosed embodiment, telephony signals are communicated from a 
telephony network to CATV subscribers in the forward band of the cable 
network and telephony signals are communicated from the CATV subscribers 
to the telephony network in the reverse band of the cable network. 
Further, the subscriber telephony signals to the telephony network are 
digitized and individually modulated on a carrier in the reverse band of 
the CATV system. As an illustrated example, a subscriber DS0 telephony 
line is QPSK modulated into an approximately 50 kHz bandwidth signal (e.g. 
49.5 kHz) and frequency division multiplexed on the reverse band of the 
CATV network. The individual telephony signals are multiplexed into a 
standard time-division multiplexed (TDM) telephony signal which can be 
adapted to couple directly into a SONET port or other standard telephony 
connection, such as a DS1, DS2, or DS3 format signal, of the telephony 
network. 
It is an object of the present invention to provide a flexible, digital 
design for handling baseband processing of plural DS0 telephony channels 
in a single programmable digital signal processor, to obtain improvements 
in density, cost, and space. 
Advantageously, by translating a plurality of reverse channel telephony 
signals to baseband, a single digital signal processor can handle a 
significant number, six in the preferred embodiment, of DS0 telephony 
channels. This provides a low cost, compact demodulator that can be used 
in larger systems where a large number of telephony signals are coupled 
between the telephony network and the broadband network. 
Furthermore, all the mathematical constants (sin and cosine values) 
required for QPSK demodulation can be stored within the memory of the DSP, 
providing a very memory-usage efficient demodulator that can handle a 
number of DS0 channels simultaneously. 
Also, handling the AGC, STR, CPR, and symbol decoding functions digitally 
at the symbol rate rather than in the analog domain or with an 
oversampling approach results in a compact, efficient, multi-channel 
demodulator. 
By using the reverse band of the CATV network in small increments of about 
50 kHz, the flexibility of the reverse signaling band is not compromised. 
The system operator can still provide interactive TV services, IPPV 
services, and other reverse path signals while providing telephony service 
.

DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS 
With respect now to FIG. 1, there is shown a broadband communications 
system in which the present invention is utilized The system will be 
described in connection with the communications of telephony signals, but 
it will be evident that other signals of similar or equivalent types can 
also be used. Further, while digital telephony signals are described, the 
system is also capable of communicating analog telephony signals or other 
types of digital signals. Telephony signals from the telephony network are 
coupled to the CATV network 12 and are communicated over the CATV network 
to an addressed subscriber premises 30. The addressed subscriber 30 
communicates telephony signals back over the CATV network 12 which are 
then coupled to the telephony network 10. The system serves as an 
extension of the telephony network 10 where subscribers can call out to 
the telephony network 10 or receive calls from the telephony network. This 
service is in addition to the conventional video, audio, data and other 
services provided to each subscriber by the CATV network 12. 
By "headend", we do not mean to be limited to a conventional coaxial CATV 
headend such as 14, but also consider that an optical fiber node such as 
16 or other communication node that can serve the functions of receiving 
multiplexed communication signals from a source of signals, such as a 
telephony central office, and communicating such signals to subscribers in 
the broadband network. As will be seen in the following discussion, a CATV 
headend 16 is the preferred embodiment for effecting these functions. 
A preferred implementation of the broadband communications system is 
illustrated in FIG. 1. The system includes the telephony network 10 which 
interfaces through an input interface 32 to the CATV network 12. The CATV 
network 12 further interfaces with the telephony network 10 through an 
output interface 34. Telephony signals are communicated to subscribers of 
the CATV network 12 through the input interface 32 to a subscriber 
premises 30. Telephony signals from the subscriber premises 30 of the CATV 
network 12 are communicated over the CATV network 12 and through the 
output interface 34 to the telephony network 10. The broadband 
communications system does no switching and thus takes advantage of the 
strength of the CATV network 12 for its broadband communications path and 
the strength of the telephony network 10 for its connection and switching 
capability. 
The CATV network 12 is illustrated as having a fiber to the serving area 
(FTSA) architecture. A headend 14 provides CATV programming which is 
distributed via a distribution network to a plurality of subscribers at 
their subscriber premises 30. The distribution network serves a plurality 
of "serving areas", such as the one referenced at 20, which are groups of 
subscribers that are located proximate to one another. Each serving area 
is comprised of groups ranging in size from about 50 homes to about 2500 
homes. The headend 14 is coupled to each serving area in a star 
configuration through an optical fiber 18 which ends in a fiber node 16. 
The CATV programming and telephony signals are converted from an RF 
broadband signal to light modulation at the headend 14, transmitted over 
the optical fiber 18, and then converted back to an RF broadband signal at 
the fiber node 16. Radiating from each of the fiber nodes 16 throughout 
its serving area 20 is a coaxial sub-network of feeders 22 having 
bidirectional amplifiers 24 and bidirectional line extenders 25 for 
boosting the signal. 
The RF broadband signal is distributed to each of the subscriber premises 
30 by tapping a portion of the signal from the nearest feeder 22 with a 
tap 26, which is then connected to the subscriber premises through a 
standard coaxial cable drop 28. The CATV network thus provides a broadband 
communications path from the headend 14 to each of the subscriber premises 
30, which can number in the several hundreds of thousands. 
While one preferred embodiment of the invention shows the input interface 
32 coupled to the fiber node 16 and the output interface 34 coupled to the 
headend 14, it is evident that the insertion and extraction of the RF 
telephony signals need not be limited to this single architecture. Both 
the input interface 32 and an output interface 38 (shown in phantom) can 
be connected at the fiber node 16. Alternatively, both an input interface 
36 (shown in phantom) and the output interface 34 can be coupled to the 
headend 14. Moreover, the input interface 36 can be coupled to the headend 
14, while the output interface 38 can be coupled to the fiber node 16. For 
cable architectures which do not conform to a star configuration, it is 
generally most advantageous to insert the RF telephony signals at the 
headend and to extract them from the system at the headend. 
The input and output interfaces 32 and 34 produce a facile method for 
inserting the telephony signals in one direction and extracting the 
telephony signals in the other. The telephony signals are transformed into 
compatible RF signals which can be inserted or extracted from the CATV 
network 12 in much the same manner as other programming at various points 
in the network. The compatibility of RF telephony signals with the 
previous RF signals on the CATV network 12 allows their transmission in a 
transparent manner over the network without interference to the other 
signals or special provision for their carriage. 
Theoretically, the broadband communications path provided by the CATV 
network 12 is bidirectional so that information can be passed in each 
direction. However, because of convention and the single point to 
multipoint nature of most networks, the reverse path, i.e., communications 
originating from the subscriber premises 30 and communicated to the 
headend 14, is much more limited. Normally, the reverse amplifiers 25 are 
bandwidth limited and include diplexers which separate the CATV spectrum 
into forward and reverse paths based on frequency. 
In summary, the described system provides for broadband communications 
including digital communications, telephony, and telephony-related 
services by utilizing a CATV system in an efficient manner, while not 
requiring extensive switching equipment and a redesign of such systems. 
The broadband communications system requires no switching in the normal 
context when connecting telephony based calls from a subscriber or to a 
subscriber. A multiplicity of calls can be placed through the system 
efficiently using the broad bandwidth of the CATV network to utilize its 
best features and having the switching for the connection of the calls 
performed by the telephony network to utilize its best features. 
There are two types of telephony calls in the broadband communications 
system, where one is an incoming call and the other is a outgoing call. 
With combinations of these types of calls, all the necessary connections 
to or from another telephony set and to or from a CATV network subscriber 
can be made. The subscriber may call (or be called by) another subscriber 
within the CATV network system, may call (or be called by) a local 
telephone set within the local area of the telephone network, or may call 
(or be called by) the telephone network to interface to the long distance 
and international telephony systems. 
An incoming call is directed to a particular subscriber of the CATV network 
by the telephony network recognizing that the call is directed to one of 
the group of subscribers belonging to the CATV network. The call is then 
switched by the telephony network to the OC-1 or other standard telephony 
signal coupled to the CATV network in the time slot assigned to that 
subscriber. The addressing and control system of the CATV network then 
decodes the multiplexed information and translates it into a frequency and 
time position in the forward multiplex that has been assigned to the 
particular subscriber. The addressing and control system further provides 
the necessary control for causing the subscriber equipment to ring or 
alert the subscriber of an incoming call. 
The telephony network and CATV network maintain the connection until there 
is an indication of an "on hook" signal by one of the parties or another 
signal that indicates that the communication is complete, such as an end 
of message data pattern or the like. What is meant by maintaining the 
connection is that the telephony network continues to place the called 
party's data packets into the assigned DS0 position in the standard 
telephony signal and the broadband communications system continues to 
convert them to the location and frequency in the forward multiplex that 
is directed to the particular subscriber. 
For outgoing calls, the telephony network recognizes from the DS0 position 
in the standard telephony signal which data packet belongs to a particular 
originating subscriber of the CATV network. This is an assigned position 
and the CATV system converts data on whatever carrier frequency is input 
to the demodulators to that assigned position in the reverse multiplex. 
Therefore, for outgoing calls the telephony network will consider the 
standard telephony signal as a group of individual DS0 signals, whose 
location in the reverse multiplex identifies the originating subscriber. 
FIG. 2 illustrates a preferred implementation of the broadband 
communication system configured as an extension to a telephony network. 
For connection to the telephony network 10, a class 5 switch 41 is used. 
The switch 41 has suitable circuitry for handling conventional local, 
trunk and interconnect signals which integrate the switch into the local 
area, national and international calling grids. The switch 41 has a 
switching network of crosspoints which may switch any of a plurality of 
inputs to any plurality of outputs. Particularly, the switch 41 has 
equipment to provide DS1 format interfaces. 
As known to those skilled in the art, a "DS0" signal is a standard 
telephony format corresponding to a 64 kb/s digital channel which can be 
used for voice, data, audio, etc. Thus a single DS0 telephony signal can 
be viewed as a single telephone conversation. Likewise, a "DS1" signal 
corresponds to a 1.544 Mb/s digital channel that contains 24 DS0 channels. 
For a summary of the bit rates of the standard digital telephony formats 
and their relationships to one another, see TABLE 1 below: 
TABLE 1 
______________________________________ 
Digital 
Signal Bit Rate DS0 DS1 DS3 
______________________________________ 
DS0 64 kb/s 1 1/24 1/672 
DS1 1.544 Mb/s 24 1 1/28 
(also T-1) 
DS1C 3.152 Mb/s 48 2 1/14 
DS2 6.312 Mb/s 96 4 1/7 
DS3 44.736 Mb/s 672 28 1 
OC-1 51.84 Mb/s 672 28 1 
______________________________________ 
Additionally, the switch 41 has means for demultiplexing DS1 signals into a 
plurality of DS0 signals which then can be routed to outgoing points. The 
system uses a forward path which receives a plurality of the DS1 channels 
at the input interface 32 and connects them over the CATV network 12 to 
the subscriber premises 30. The subscriber premises 30 transmits telephony 
signals over the CATV network 12 to the output interface 34 which converts 
them back into the same number of DS1 signal channels for transmission to 
the switch 41. If the switch 41 is located proximately to the input 
interface 32 and the output interface 34, then they can be coupled 
directly. Alternatively, as will be the most prevalent case, where a 
headend or fiber node is not located proximately to the class 5 switch, an 
optical fiber link can be used to connect the switch 41 and interfaces 32 
and 34. 
In the forward direction, a fiber optic transmitter (FOT) 43 converts the 
plurality of DS1 telephony signals into an optical signal which is 
transmitted to a fiber optic receiver (FOR) 45. The fiber optic receiver 
45 converts the optical signal back into the DS1 format telephony signals. 
Likewise, the fiber optic transmitter 49 in the reverse path converts the 
outgoing DS1 telephony signals into an optical signal which is received by 
the fiber optic receiver 47 for conversion back into the DS1 telephony 
format signals. 
The DS1 telephony signal format was chosen because it is a standard 
telephony format, and conventional optical links to do the conversion and 
transmission are readily available for the transmitters 43, 49 and for the 
optical receivers 45, 47. 
The system uses this bidirectional mode of communication where each DS1 
signal contains 24 DS0 channels, which can be considered groups of 64 kb/s 
digital data channels. The 64 kb/s channels can either be used for voice, 
data, audio (music, stored information), etc. In general, for telephony 
type signals, each DS0 channel derived from a connected DS1 link is 
addressed to and associated with a particular subscriber. The preferred 
embodiment provides transport from each DS0 signal in the connected DS1 
link to the particular subscriber, by transmitting incoming telephony 
signals downstream in a selected DS0 downstream channel in the broadband 
system forward path, and has a corresponding DS0 upstream channel assigned 
to that subscriber in the broadband system reverse path for outgoing 
telephony signals. Received DS0 signals from subscribers are then routed 
to the corresponding DS0 time slot in the DS1 link for outgoing signals. 
This permits the switch 41 to connect any of the local, trunk or 
interconnect calling points to any of the DS0 channels in the forward path 
and its associated DS0 channel in the reverse path to the same local, 
trunk or interconnect points for completing the communications path. Each 
of the subscribers 30 appears as another DS0 subscriber connected directly 
to the class 5 switch 41. The distribution system of the CATV network 12 
is transparent to the switch 41 and does not need any further 
communication, information or connection to the broadband communication 
system. 
FIG. 3A illustrates a typical frequency allocation for many of the 
installed split band CATV networks. The frequencies used for programming 
which generate the revenues for the system operator are carried in the 
forward band from 50 MHz to about 550 MHz. Although, the frequencies above 
550 MHz are not presently used, there has been increased interest in 
providing additional services in this unused forward bandwidth, currently 
considered to extend to about 1 GHz. Conventionally, the forward band 
comprises a series of video channels, each 6 MHz in bandwidth, which are 
frequency division multiplexed across the forward band. Several areas are 
not used and each video channel has a 1.5 MHz guard band between other 
adjacent channels. 
In combination with the forward band, the typical CATV spectrum includes a 
reverse band from about 5-30 MHz. These frequencies have been allocated 
for signals returning from the subscriber to the headend. This band has 
traditionally been relatively narrow because of the high noise from the 
funneling effects of the multiplicity of the multipoint signals adding to 
a single point. Further, in the past bandwidth taken from the forward band 
has meant less revenues from other services. The present invention 
provides a solution to these problems by providing a system where the 
telephony signals to a subscriber premises are communicated in the forward 
band of the spectrum and the telephony signals from a subscriber premises 
are communicated in the reverse band of the CATV system. 
As seen in FIG. 3B, the broadband communications system utilizes a 
plurality of frequency division multiplexed carriers in the forward band 
to communicate the telephony signals to the subscribers. In the 
illustrated embodiment, seven (7) channels of approximately 3 MHz are used 
to carry incoming telephony signals from the telephony network 10. Each 
forward channel is a QPR modulated carrier, where the modulation occurs as 
a 6.312 Mb/s digital data stream in three DS1 telephony signals including 
72 DS0 telephony signals. The carriage capacity of such a system is then 
at least 20 DS1 channels, or enough for at least 480 DS0 voice channels. 
Each of the reverse band signals are about 50 kHz in bandwidth (49.5 kHz in 
the presently preferred embodiment), which is narrow enough to be easily 
placed at different frequency division multiplexed positions in the 
frequency spectrum. The modulators are frequency agile and can reallocate 
frequencies based upon traffic over the system, noise, channel condition, 
and time of use. The 49.5 kHz wide carriers can be placed anywhere in the 
reverse band that there is space for them. Depending upon the CATV system, 
i.e., whether there is a reverse amplification path in the distribution 
network, they could also be allocated to frequencies normally reserved for 
forward band transmissions. Further, such system is expandable by 
bandwidth for other uses besides the individual telephony signals. For 
example, if a particular subscriber required a return path of a greater 
bandwidth than 49.5 kHz, then the bandwidth could be easily allocated to 
this use without a complete reconfiguration of the system. Such uses may 
include high speed data transmissions, trunk connections for small central 
offices, video services originating from the telephony network, and other 
uses requiring a nonstandard bandwidth. 
There are a number of advantages with the broadband communications system 
as described. It uses the reverse band efficiently and uses only that 
portion of the forward band which is necessary. Digital QPR and QPSK 
modulation is used to permit digital and telephony services to the 
subscriber and provide a robust signaling method allowing the forward or 
reverse signals to be placed anywhere in the CATV band, either at high or 
low frequencies without signal to noise ratio concerns. Moreover, in the 
forward direction, the carrier signals are minimized so that carrier 
overloading does not occur and that the 3 MHz channels can be placed where 
space is found. 
FIG. 3C illustrates an alternative frequency allocation for a split band 
CATV network. As in the other systems, the frequencies used for television 
programming that generate the revenues for the system operator are 
generated in the forward band from about 50 MHz and above. The spectrum in 
FIG. 3C includes the reverse band from about 5 MHz to about 30 MHz. The 
5-30 MHz band is used for upstream telephony signals in the form of 388 
DS0's, combined to form DS0 pairs and QPSK modulated in 128 kHz upstream 
channels or subbands designated UP1, UP2, . . . UP194, where each upstream 
channel UPn carries 2 DS0's. Thus, in order to accommodate 388 DS0's, 194 
QPSK carriers or channels are required. Each of the upstream channels UPn 
consumes 128 kHz bandwidth, comprising 108 kHz of modulated signal space 
and 20 kHz of guard band. 
The downstream telephony is provided in downstream channels DN1, DN2, . . . 
DN480, each DN corresponding to a DS0. In one preferred alternative 
embodiment, a total of 21 MHz of bandwidth is provided in 3.168 MHz 
subbands, each 3.168 MHz subband carrying the equivalent of three DS1 
telephony signals (72 DS0's), in QPR modulation. 
FIG. 3D, which will be discussed in greater detail below, illustrates the 
frequency downconversion of the reverse band QPSK modulated DS0 telephony 
signals into baseband before processing with the system of the present 
invention. As will be discussed later, the QPSK modulated DS0 telephony 
signals in the reverse band, e.g. DS0-1 at 14.949 MHz, DS0-2 at 20.205 
MHz, etc. are converted to baseband signals, e.g. DS0-1 at CH3, DS0-2 at 
CH6, etc., before processing. 
A detailed block diagram of the input interface 32 is illustrated in FIG. 
4. The function of the input interface 32 is to convert the 20 DS1 
telephony signals into the seven QPR modulated RF signals which are sent 
to the subscribers in the forward band of the CATV system 12. The input 
interface 32 is connected to an optical interface 40, comprising a fiber 
optic receiver 45 and a demultiplexer 44. The fiber optic receiver 45 
operates to convert the optical signal into an RF digital signal of a 
standard telephony format. The demultiplexer 44 receives the digital DS3 
telephony signal and separates it into its 28 component DS1 signals, where 
each DS1 signal comprises 24 DS0 signals. The optical interface 40 also 
allows an addressing and control unit 42 to decode and strip overhead and 
framing bits from the signal. 
The input interface 32 comprises a series of five multiplexers 46, which 
each take four of the DS1 signals from the demultiplexer 44 and combine 
them with signaling and addressing bits from the addressing and control 
unit 42 to form a 6.312 Mb/sec serial digital signal. Each of the five 
digital signals is modulated on a selected carrier frequency by an 
associated QPR modulator 48. The five telephony channels from the outputs 
of the modulators 48 are frequency division multiplexed together in an RF 
combiner 50 before being inserted conventionally on the CATV network 12. 
The output interface 34 will now be more fully described with reference to 
FIG. 5. The output interface 34 functions to convert the 480 DS0 digital 
signals which are QPSK modulated on the reverse band carriers into the 
optical format for coupling to the telephony network 10. The output 
interface 34 extracts the reverse band signals in a conventional manner 
and fans them out with a signal divider 60 to a plurality of 
tuner/demodulators 62. Each of the tuner/demodulators 62 is adapted to 
tune one of the carrier frequencies of the reverse band signals and 
demodulate it into a DS0 format digital signal. The tuners of the 
tuner/demodulators 62 can be variable or fixed, or can be adapted to tune 
only certain bands of the reverse spectrum. The output of the 
tuner/demodulators 62 is 480 DS0 signals which are concentrated into 
groups of DS1 signals by a group of multiplexers 64 under the control of 
addressing and control unit 66. 
In accordance with the preferred embodiment of the present invention, the 
tuner/demodulators 62 are constructed to provide up to 24 DS0 signals from 
four DSPs, utilizing the baseband processing systems and methods as 
described herein. The construction of such an arrangement is described in 
greater detail below. 
Each of the multiplexers 64 inputs 24 DS0 formatted signals and outputs one 
DS1 formatted signal to a fiber optic transmitter 49. At the fiber optic 
transmitter 49, the 20 DS1 signals are concentrated by a multiplexer 68 
into a single DS3 digital signal which is input to the optical transmitter 
70. The addressing and control unit 66 adds the necessary control 
information in the optical transmitter 70 before communicating the digital 
DS1 signals in an optical format. The optical transmitter 70 also converts 
the RF signal into light so the optical fiber of the telephony network can 
transmit it. 
A detailed block diagram of the system equipment at the subscriber premises 
30 is shown in FIG. 6. Generally, the subscriber will want to maintain 
CATV video or other services and has a CATV terminal 84 for this purpose 
connected between the CATV drop line 28 and a television receiver 88. The 
CATV terminal is connected to a splitter/combiner/diplexer 80 coupled to 
the drop 28 from one of the CATV coaxial subnetwork feeders. 
Because the presently described broadband communications system does not 
interfere with or displace the conventional CATV programming and frequency 
allocations, the CATV terminal 84 can generally be used with no 
modification or change in operation of the installed terminal base. The 
system operator does not need to change or reconfigure its distribution 
network operation and the new telephone service is compatible with its 
installed CATV subscriber terminal base. 
The broadband communications service is provided by coupling a telephony 
terminal, also called a "customer interface unit" 82, between the 
splitter/combiner/diplexer 80 and the telephone equipment 86. The customer 
interface unit 82 converts the incoming telephony signals to a subscriber 
into analog signals which can be used by a standard telephone handset 86 
over a pair of twisted wires 85. Further, the customer interface unit 82 
converts the analog signals, representing outgoing telephony signals from 
the handset 86, into a QPSK modulation which is coupled to the CATV 
network. A standard telephone handset 86 is shown for the purpose of 
illustration but could in fact be any equipment normally connected to a 
telephone line for digital communications purposes. 
The telephony terminal 82 has two communication paths. The first path for 
incoming signals comprises a tuner/demodulator 92, demultiplexer 96, and a 
portion of a line card 98, and a second path for outgoing signals 
comprises a portion of the line card 98 and a modulator 94. The 
tuner/demodulator 92, modulator 94, demultiplexer 96, and line card 98 are 
under the control of an addressing and control unit (CPU) 90. 
For incoming telephony signals which are received in the 3 MHz channels 
modulated on an FDM carrier, the control unit 90 causes the 
tuner/demodulator 92 to tune the carrier on which the particular call 
information directed to the subscriber is carried. The carder defines one 
of the seven 3 MHz channels having 3 DS1 or 3 E-1 telephony signals QPR 
modulated thereon. 
The telephony signals are demodulated by the tuner/demodulator 92 into a 
serial digital stream containing the 3 DS1 or 3 E-1 telephony signals 
before being input to the demultiplexer 96. The demultiplexer 96 selects 
the particular DS0 digital telephony channel assigned to the subscriber at 
the input rate of 64 kb/s and inputs the data to an input terminal of the 
line card 98. The control unit 90 determines which forward telephony 
channel to tune and which DS0 signal to select from that channel from the 
signal and addressing information it receives by its connection to the 
splitter/combiner/diplexer 80 via line 89. 
The DS0 digital format provides a voice channel with sufficient bandwidth 
for voice quality communications. The DS0 format is a 64 kb/s data stream 
of bytes forming timed samples of an analog voice signal. This produces a 
voice signal quantized to 8-bits per sample (256 values) at a sampling 
rate of 8 kHz and with a bandwidth of 4 kHz. 
The line card 98 receives the digital telephony signal in the DS0 format 
and converts it to the proper analog voltages and signals to drive the 
telephone handset 86. In addition, the line card 98 provides ringing 
current, terminal identification, and other standard functions under the 
direction of control unit 90. The line card 98 receives the analog 
telephony signals from the telephone handset 86 and converts them into a 
digital DS0 format. Dialing signals and other addressing and control 
signals from the handset 86 are also digitized by the line card 98. The 
digitized outgoing telephone signals are then combined and formatted by 
the line card 98 into a DS0 format at 64 kb/s and input to the modulator 
94. 
The modulator 94 under the regulation of the control unit 90 selects a 
carrier frequency in the reverse band and QPSK modulates the DS0 telephone 
signal thereon. The QPSK modulated carrier having a bandwidth of 49.5 kHz 
is coupled on the CATV network through the splitter/combiner/diplexer 80. 
The QPSK modulated outgoing telephony signal is then received at an output 
interface 34 (either on a stand-alone basis or associated with a headend 
unit 14) and demodulated into DS0 format for coupling to the telephony 
network. 
Before leaving FIG. 6, it will be understood that the nature of the 
telephony service that is provided at any given CIU 82 must be 
preidentified and prestored in memory in the HIU 14 or circuitry that is 
utilized as the telephony network interface, so as to enable provision of 
the selected service upon demand. In response to a request for service 
either originating with a subscriber at a selected CIU, or a request for 
incoming service to a subscriber originating externally to the network, 
status signals such as the subscriber going off hook, or a ringing 
condition on an incoming line, the system causes the selection and 
allocation of appropriate bandwidth, DS0 channels, reverse channels, 
carriers, etc., required to provide the selectably variable bandwidth 
commensurate with the selected service. 
Incoming data from the broadband network is derived from the forward 
channel demodulator 92, which is operative to monitor a preassigned 
channel in the QPR-modulated forward channel utilized for incoming 
telephony signals. The preferred forward demodulator 92 demodulates a QPR 
modulated forward channel signal in the designated telephony downstream 
subband of 15.840 MHz, and monitors the directory channel and signaling 
channels provided as a part of the overhead data. 
It will be understood that a plurality of reverse channel modulators 94 may 
be required to provide the appropriate bandwidth required for a given 
level of service. For example, if a selected service at a given CIU 82 
entails the equivalent of four DS0's, then there is the need for four 
reverse channel modulators 94. Furthermore, it will be recalled that each 
modulator 94 is frequency agile and is not necessarily operating at a 
given fixed upstream carrier frequency, since upstream channels can be 
reassigned dynamically and in response to changing conditions such as 
noise level and reallocation of bandwidth in response to the subscriber's 
needs. 
It will also be understood that the CIU 82 can be physically configured 
either as separate customer premises equipment located in or near a 
subscriber's telephony punch blocks, or as a CATV set top terminal 
including one or more RJ-11 or similar telephone connectors. Moreover, the 
CIU, since it includes a computer (as a part of the addressing and control 
unit 90) and associated circuitry can be used for conventional CATV signal 
management such as pay-per-view control, descrambling, etc. Therefore, the 
preferred CIU, whether settop or separate circuitry enclosure, preferably 
includes a control connection provided from the addressing and control 
unit 90 to a switch (not shown in FIG. 6) associated with the signal line 
between the splitter 80 and the subscriber's television 88. This allows 
the programming signals to be disconnected from a subscriber in the event 
of non-payment or election not to receive a certain programming. 
Finally, each CIU 82 is associated with a unique predetermined serial 
number for identification purposes in the network. This serial number is 
preferably maintained internally in a read-only memory. Also, within a 
particular network configuration, each CIU is assigned a unique 16-bit 
address by the HIU. The address of the CIU is provided in the upstream 
channel to the HIU whenever the CIU requests service. The address 
information is utilized by the HIU to examine a service level table or 
data array, described below, to identify the subscriber associated with 
the address information and determine the appropriate and authorized level 
of service to be provided. For example, when a telephone connected to the 
CIU goes off hook, the address of the CIU is transmitted in association 
with the off hook status information in the upstream channel to the HIU 
(or network interface), where it is received and examined to determine the 
appropriate service level, DS0 assignments, frequency assignment, etc. 
The service level table or data array stored by the comprises an array of 
data fields, suitable for storage in a database maintained by the HIU's 
CPU 308. Preferably, this table is maintained in RAM for rapid access. 
Furthermore, it is preferred that the table be indexed utilizing 
conventional database indexing methods so that the table may be rapidly 
search by subscriber name, subscriber address, telco DS0 number, upstream 
carrier frequency, etc. Use of indexed methodologies ensures rapid lookup 
of service level and minimized response time when a subscriber requests 
service. 
From the foregoing, it will be understood and appreciated that the 
frequency agile CIU is operative for modulating telephony and other 
signals from a subscriber in a plurality of frequency subbands in the 
upstream band of a broadband subscription network so as to provide 
selectably variable bandwidth in the upstream band commensurate with a 
selected subscriber communication feature such as single voice line, 
multiple voice lines, ISDN, security monitoring services, and the like. In 
the preferred embodiment, the bandwidth is selectably allocated in 
discrete unit of DS0's, which will be understood can be combined to 
provide for higher capacity digital data channels in response to varying 
needs of subscribers. 
Furthermore, it will be understood that the frequency agile CIU is 
operative to reassign signals in a selected subband, such as UP1 . . . 
UPn, to another subband at another frequency in response to a 
determination that the noise level in a particular selected subband 
exceeds a predetermined level. 
Finally, there is provided one upstream data link for each carrier that is 
utilized by the CIU 82 to provide a general purpose data transport for 
alarm conditions, configuration information, etc. Each CIU 82 is normally 
assigned at least one upstream frequency (either the DS0-1 or the DS0-2 of 
the 128 kHz channel), which comprises a portion of the 1.333 kbps data 
channel that is combined with two 64 kbps data channels to form 72 kbps 
for each upstream frequency subband. The 1.333 kbps data link carries the 
subscriber's address as well as status information associated with a 
subscriber's address. 
Turning next to FIG. 7, the preferred embodiment of a headend interface 
unit (HIU) 301 constructed in accordance with the present invention will 
be described. A HIU constructed as shown in FIG. 7 may be utilized to 
carry out the invention as an alternative to providing a separate input 
interface 32 and output interface 34 as shown in FIG. 1. Stated in other 
words, a HIU 301 constructed as in FIG. 7 may be utilized to implement the 
combination of a headend 14, input interface 36, and output interface 34 
shown in FIG. 1. 
The HIU 301 is suitable for use either as equipment comprising the headend 
14 or equipment comprising the fiber node 16 shown in FIG. 1, both of 
which are operative for receiving multiplexed digital telephony signals in 
a standard telephony format such as DS3, DS2, DS1, and coupling such 
signals to an input interface 32, 36 or an output interface 34, 38. 
Although the preferred embodiment is described in connection with a 
coaxial line HIU, it will be understood that the principles are applicable 
for an optical-fiber based HIU that employs methods for communicating 
broadband signals via amplitude modulation (AM) methods, such as described 
in U.S. Pat. No. 5,262,883, which is owned by the assignee of the present 
invention. Briefly described, the HIU 301 is operative for connecting to a 
telephone company (telco) standard multiplexed telephony signal, directing 
incoming telephony signals to subscribers downstream on the broadband 
network using QPR modulation in the forward path, and receiving outgoing 
telephony signals from subscribers upstream on the broadband network in 
one or more selected subbands within the reverse path spectrum, 
commensurate with service levels or features elected by subscribers. 
The alternative HIU 301 shown in FIG. 7 is a presently preferred embodiment 
involving the use of digital line cards 303a, 303b, . . . 303n that 
provide digital signals to a digital bus or backplane 305, operating 
together with a central processing unit (CPU) 308 corresponding to the 
address and control unit 42 as shown in FIGS. 4 and 5. 
The HIU 301 comprises a plurality of DS1 line cards 303a, 303b, . . . 303n, 
where n is 17 in the disclosed embodiment, for connection to the telephony 
network 10 or to a higher level multiplexer/demultiplexer capable of 
handling higher level multiplexing such as DS2 or DS3. It will be recalled 
that each DS1 corresponds to a T1 line, each T1 line comprises 24 DS0 
standard telephony channels. For provision of 388 DS0's, therefore, 
slightly more than 16 DS1's must be accommodated. With 17 DS1 line cards 
303, a number of lines are provided as spares. 
Each DS1 line card 303 provides interfaces compatible with ANSI Doc. T1.403 
(1989 version), which is incorporated herein by reference and made a part 
hereof. Each line card 303 provides a digital output signal that is 
coupled to the digital backplane 305. The backplane operates to connect 
all of the line cards 303 and route signals between the line cards and the 
forward and reverse path modulators, to be described. The backplane 305 
preferably comprises up to five 8-bit serial digital busses each clocked 
at 8.192 MHz. Each bus thus provides an 8.192 Megabit per second (Mb/s) 
digital pathway that is operative to receive digital signals from each of 
the line cards in a time division multiple access (TDMA) format. It will 
be appreciated that five 8.192 Mb/s digital busses in parallel are 
sufficient to handle the 388 separate 64 kbps signals. 
The backplane 305 further includes a CPU bus coupled between a CPU 308 
utilized as a database controller and each of the line cards 303 The CPU 
308 is operative to control the assigned relationships between particular 
telephony lines, ingoing and outgoing, with predetermined carrier 
assignments in the reverse path and in the forward path, monitor the noise 
level in the reverse path, and assign DS0 channels in the reverse path 
commensurate with subscriber features and the like. Further, the CPU 308 
is operative to carry out steps of monitoring noise in the reverse pathway 
channels, dynamically allocating bandwidth, and to maintaining in memory a 
service level table that indicates the correspondence between reverse 
channel carrier frequencies, subscriber identification, service level, 
telco DS0 identification, signaling status, error count for noise 
monitoring, and the like. 
The preferred CPU 308 is a Motorola 68360 32-bit microprocessor with 
built-in memory (DRAM) controller and is operatively connected to 2 MB of 
random access memory (RAM). Details of the preferred CPU are available in 
the literature supplied by the manufacturer. 
Still referring to FIG. 7, the backplane 305 further includes a signaling 
channel bus connected between the CPU 308 and each of a plurality of 
forward channel modulators 320 and reverse channel demodulators 330. The 
signaling channel bus communicates status information associated with a 
telephony line such as off hook, on hook, busy, ring, security status, and 
the like. Bits associated with particular status states of the 
subscriber's telephone and of the associated telco line are included and 
combined with digitized telephony signals and transmitted to the CIU's 82. 
In the disclosed embodiment, the HIU 301 comprises a plurality of forward 
channel modulators 320a . . . 320n and a plurality of reverse channel 
demodulators 330a . . . 330m The forward modulators 320 couple outgoing 
telephony signals to the broadband network in the forward spectrum, while 
the reverse channel demodulators receive telephony signals from CIU's in 
the reverse spectrum via the broadband network. Each of the forward 
channel modulators 320 is connected to a combiner 322 that is operative to 
combine the RF signals from the forward channel modulator and provide an 
output to a diplex filter 325. The diplex filter 325 is preferably a 
bandpass filter that passes signals outward within the 15.840 MHz 
frequency forward spectrum provided in the alternative embodiment whose 
spectral allocation is shown in FIG. 3C. The output of the bandpass 
filter, whose frequency is centered at an appropriate location along the 
spectrum allocated for forward or downstream telephony signals, is then 
coupled to a multiway splitter 340 that is coupled to the broadband 
communication network. 
It will be appreciated that the broadband communication network (not shown) 
connected to the multi-way splitter can either be a coaxial cable network, 
or alternatively can be an additional fiber optic link that is amplitude 
modulated to carry the broadband signal in a manner known to those skilled 
in the art. 
Still referring to FIG. 7, the HIU 301 further comprises a plurality of 
reverse channel demodulators 330a . . . 330m that are connected to receive 
signals from the multiway splitter 340. The reverse channel demodulators 
330 are similarly constructed, as described in connection with FIG. 8. A 
separate reverse channel demodulator constructed as described herein is 
provided for each group of 24 DS0 telephony signals. 
The multiway splitter 340 preferably includes at least one lowpass filter 
segment that isolates the signals in the 5-30 MHz range designated in the 
alternative embodiment for reverse path telephony signals. 
It will be recalled from the discussion above that the CPU 308 in the HIU 
stores a service level table or data array for associating frequency 
assignments with particular subscribers and other information required for 
system maintenance. The service level table or data array comprises an 
array of data fields, suitable for storage in a database maintained by the 
HIU's CPU 308. Preferably, this table is maintained in RAM for rapid 
access. Furthermore, it is preferred that the table be indexed utilizing 
conventional database indexing methods so that the table may be rapidly 
search by subscriber name, subscriber address, telco DS0 number, upstream 
carrier frequency, etc. Use of indexed methodologies ensures rapid lookup 
of service level and minimized response time when a subscriber requests 
service. 
From the foregoing, it will be understood and appreciated that the 
frequency agile CIU is operative for modulating telephony and other 
signals from a subscriber in a plurality of frequency subbands in the 
upstream band of a broadband subscription network so as to provide 
selectably variable bandwidth in the upstream band commensurate with a 
selected subscriber communication feature such as single voice line, 
multiple voice lines, ISDN, security monitoring services, and the like. In 
the preferred embodiment, the bandwidth is selectably allocated in 
discrete unit of DS0's, which will be understood can be combined to 
provide for higher capacity digital data channels in response to varying 
needs of subscribers. 
Turning next to FIG. 8, a reverse path demodulator 330 (REV DEMOD) is 
operative to receive the filtered broadband signal from the multiway 
splitter 340 (FIG. 7), which can occur any where in the 5-30 MHz band, 
downconvert the center of a 1.584 MHz band comprised of 24 useful DS0 
channels to DC, channelize the composite signal into 24 useful DS0 
channels each having a 49.5 kHz bandwidth by weighting and FFT methods, 
time multiplex a group of six channels into serial form for transmission 
to a digital signal processor's serial port for QPSK demodulation, derive 
demodulated DS0 telephony signals, provide time division multiplexed DS0 
telephony signals to a framer for proper telephony signal formatting, and 
output the selected DS0 signals to the line cards 303 (FIG. 7). 
The broadband signal from the multiway splitter 340 is first provided to a 
low pass filter 351, which removes signal components in excess of the 30 
MHz band. The output of the low pass filter 351 is then provided to an 
analog front end 352 which includes bandpass filters for forming sixteen 
1.584 MHz subbands and mixers for downconverting the filtered signals 
within each subband to DC. The output of the analog front end 352 is 
provided to a 12-bit digital to analog (D/A) converter which samples a 
selected 1.584 MHz subband at 6.335 MHz and downsamples the digital 
signal. The output of the A/D converter is then provided to a channelizer 
360 for conversion into the baseband DS0 channels. 
The channelizer 360 includes a fast Fourier transform (FFT) circuit 354 
that separates each of the 24 digitized DS0 signals within the 1.584 MHz 
subband into a separate baseband channel, CH1, CH2, . . . CH24. The FFT 
operates by applying predetermined weightings in the known manner to 
frequency shift any signals within the 1.584 MHz subband to DC. The 24 
separate outputs or channels of the FFT 354 CH1, CH2, . . . CH24 are then 
provided to a serializer 355 that takes a group of six channels and 
generates a time division multiplexed serial signal on one of lines 362 
that is provided as the output of the channelizer to the digital signal 
processors constructed in accordance with the present invention. 
FIG. 3D illustrates the operation of the analog front end and channelizer 
in forming the baseband DS0 signals. It will be understood that any given 
DS0 signal can appear anywhere within the 5-30 MHz reverse band. For 
example, a signal identified as DS0-1 may be modulated on a carrier at 
14.949 MHz, while DS0-2 may be modulated on a carrier at 29.205 MHz. The 
analog front end 352 and channelizer 360 are operative to downconvert any 
given DS0 signal to baseband, and group six DS0 channels together for 
transmission to a DSP for further processing. As shown in the middle part 
of FIG. 3D, DS0-1 may be associated with any give channel, e.g. CH3, DS0-2 
with CH6, etc., and the samples of the signals in the channels serialized 
and time division multiplexed for further processing. 
Accordingly, it will be understood that the channelizer 360 is operative to 
multiplex six of the QPSK-modulated 49.5 kHz baseband signals to a single 
DSP. The output of the channelizer 360 is alternate samples of each of the 
six 49.5 kHz baseband signals successively in a TDMA fashion on each of 
lines 362a-362d. Each serial data line 362 therefore provides time 
division multiple access sample digital data for demodulation of the QPSK 
signals at baseband. 
Each of the serial data lines 362, four total, is provided to a separate 
digital signal processing (DSP) circuit 370; therefore, four separate DSPs 
370a-370d are required to handle 24 channels. The operation of the DSPs 
370 is described in greater detail below. The output of the DSPs 370 is a 
TDM serial stream of data provided on line 372a-372d to framer circuits 
375a-375d, respectively. 
Each framer circuit 375 is operative to separate the TDMA demodulated DS0 
telephony data, combine it with appropriate formatting bits, check bits, 
etc. required for the standard DS0 telephony format, and provide serial 
digital signal to a connected line card 303 (FIG. 7), where the signals 
are coupled to the telephony network, or to such other telephony equipment 
as is required for further digital multiplexing and/or combination into 
other digital telephony formats such as DS1, DS2, etc. Thus, each framer 
375 provides collects six separate DS0 signals, for example, the framer 
375a collects the signals DS0-1, DS0-2, . . . DS0-6 and frames the data up 
for communication via the HIU bus 305 (FIG. 7) to a line card. Each framer 
handles six DS0s, so four framer circuits 375a-375d are provided in the 
demodulator 330 to handle 24 total DS0s. 
The preferred DSPs 370 are ADSP2171, manufactured by Analog Devices, Inc., 
Norwood, Mass. Details of the preferred DSP chips are found in the 
literature supplied by the manufacturer. Each DSP 370 also provides a 
16-bit host interface port shown on line 378 which is used to communicate 
signals to the database controller 308 associated with the HIU. The two 
asynchronous serial ports (SPORT0 and SPORT1) provided in the preferred 
DSPs are used to receive input data from the channelizer 360 and output 
data to the framer 375, respectively. In addition, an external interrupt 
(not shown) from the channelizer is provided to the IRQ2 input of the 
ADSP2171 to provide synchronization to the channelizer input. 
The channelizer 360 is preferably constructed with a field programmable 
gate array (FPGA), and provides 16-bit input data to the DSP 370 in a 
19.008 MHz serial bitstream to serial port SPORTO. The serial data clock 
and transmit frame synchronization signals are configured for external 
sourcing from the channelizer. The frame synchronization signals for 
SPORT0 are preferably configured for active high signals and alternate 
framing mode. In addition, the serial port SPORT0 is preferably configured 
to automatically buffer sixteen values from the serial port in a data 
memory circular buffer. 
The DSPs 370 output data to the framers 375 in a 24 word time-division 
multiplexed serial bitstream. The output serial port SPORT1 is configured 
for multichannel operation on all DSPs. A 2.592 MHz clock for the 
multichannel serial bitstream from the DSP is preferably generated 
externally by the channelizer FPGA 360. A Receive Frame Synchronization 
(RFS) signal is generated by the first DSP 370a, and the remaining DSPs 
370b, 370c, and 370d are initialized for external sourcing of the RFS 
signal from the first DSP 370a. The RFS signal is configured to be active 
high and for a multichannel frame delay of zero. Each DSP is configured to 
source data for six of the 24 channels, and provide 3-bit output data 
consisting of 2 data bits and one status bit. 
Still referring to FIG. 8, the host interface port 378 of each DSP 370 is 
used to boot the DSP and to send channel status information to the HIU 
database controller 308 (FIG. 7). To receive channel status information, 
the database controller 308 writes a request into a data register HDR0 
within the DSP and waits for a response from the DSP. The DSP firmware 
responds to a request at most once per 36 kHz cycle. The response will 
consist of channel status information written to DSP data registers HDR0 
through HDR5. 
At least three sources of interrupts are used on the DSPs 370:SPORT1 
Receive Buffer Full, SPORT0 Transmit Buffer Empty, and IRQ2. As mentioned, 
the IRQ2 interrupt is used to synchronize the input data with the 
channelizer. 
FIG. 9 illustrates the basic functions carried out within each of the DSPs 
370 to provide for baseband processing of plural channels. Briefly stated, 
each DSP carries out the functions of automatic gain control (AGC), symbol 
timing recovery (STR), and carrier phase recovery (CPR), for each of six 
baseband DS0 channels. These functions are carried in a TDM fashion on 
quadrature I and Q data values provided from the channelizer 360. All 
incoming I and Q data values are stored in the internal data memory of the 
preferred DSP 370. In addition, the sine and cosine values employed in the 
baseband phase rotator portion of carrier phase recovery are stored in the 
internal program memory of the DSP. 
In FIG. 9, the automatic gain control functional block comprises an 
exponent detection and shifting component 404, a multiplier 406, and an 
amplitude detector and filter 408. The amplitude detector and filter 
generates a multiplication factor M that is provided to the multipliers 
406. 
After the signals from the channelizer 360 are adjusted for gain by the AGC 
circuit 402, they are provided to a symbol timing recovery (STR) function 
420. The STR function 420 comprises a variable group delay filter 425 and 
a correlator 428 that generates a filter coefficient b that is provided to 
the variable group delay (VGD) filter 425. The output of the variable 
group delay filter in the STR block is downsampled by two and then 
provided to a carrier phase recovery (CPR) function 450, which is 
operative to lock to the carrier frequency in a conventional phase lock 
loop fashion. The CPR function 450 includes a baseband phase rotator (BPR 
460) that adjusts the phase difference between the phase of the input I 
and Q values and the phase of the QPSK carder. 
Also, the CPR function 450 comprises a carrier phase recovery (CPR) 
function 470, which is operative to detect and lock to the QPSK carrier 
and adjust for frequency variations that might occur between the 
receiver's local oscillator and the carrier oscillator. 
The outputs of the CPR function 450 are separate I and Q values at one 
sample per symbol rate from the baseband DS0 signal that are then provided 
to a symbol decoding and differential phase decoding function 480 that is 
operative to derive the DS0 data, frame it, and format it appropriately as 
an output on line 372 at 72 kbits/sec to the framer circuit 375 (FIG. 8). 
The arithmetic functions provided for the preferred DSP 370 in 1.15 fixed 
format, i.e. the numbers have one sign bit and 15 fractional bits; 
therefore, during processing numbers can range from -1 to 1-2.sup.-15. In 
order to perform the calculations for the functions inside DSP correctly 
without any overflow, at some points during processing the inputs or 
outputs of some of these functions may need to be shifted. The nominal 
amplitude level at different points of processing are summarized in Table 
2 below: 
TABLE 2 
______________________________________ 
Nominal Amplitude level throughout processing 
Processing section Nominal level 
______________________________________ 
Shifter input 2.sup.-7 
Shifter output 2.sup.-1 
Multiplier output 2.sup.-3 
AGC filter output, M 2.sup.-2 
STR filter output, b 2.sup.0 
Phase accumulator output in BPR 
2.sup.0 
______________________________________ 
The nominal input data level for all the functions succeeding the 
multiplier is 2.sup.-3 ; therefore, the AGC filter output M must be 
shifted left by 1 so that the nominal value of M sent to multiplier is 
2.sup.-2. The STR correlator filter output, b, must be shifted left by 3 
so that the maximum value of b sent to variable group delay filter is 1. 
The phase accumulator output in BPR is also shifted left by 3 to limit the 
maximum phase to 1 cycle/sec, which corresponds to a hexadecimal value of 
EFFF for addressing the sine and cosine values in the lookup table. 
The DSP 370 receives samples at 72 Ksamples/sec from the channelizer on 
line 362 through its serial port and performs five tasks. These tasks are 
as follows: carrier phase recovery, symbol timing recovery, automatic gain 
control, decoding and formatting the symbol decisions, and providing 
various information about each channel through host interface port 378 to 
the HIU processor. The symbol decisions are formatted and sent to the 
framer on line 372 by a second serial port. Each DSP is capable of 
performing the tasks for 6 channels; therefore, for all 24 channels, four 
DSP's are required. 
FIG. 10 is a flow chart illustrating the sequence of operations taken by 
each DSP 370 to carry out the baseband processing functions described 
above. Those skilled in the art will understand that the steps shown in 
FIG. 10 are implemented as the software for the DSP, and is preferably 
stored in the internal program memory for the DSP. It will be understood 
that the steps described are taken for all six (6) DS0 channels being 
processed, in time division multiplexed fashion. Each described processing 
function is performed for all channels before processing of the next 
function, to save overhead in coding of the DSP. 
Starting at step 501, the first step taken is to initialize the DSP upon 
boot-up and to clear all internal registers and memory locations. At step 
510, a digital sample of the I and Q values is read from the channelizer 
for the six channels. 
At step 515, an inquiry is made whether a 6 millisecond (ms) interval has 
passed, indicating that it is time to conduct an automatic gain control 
function, the AGC function preferably being carried out every 6 ms. If a 6 
ms period has not expired, the "no" branch is taken to step 535, discussed 
below. If the 6 ms period has expired, the "yes" branch is taken to step 
518. 
At step 518, steps required to implement the AGC functions (402 in FIG. 9) 
are initiated. At step 518, the exponent of the I and Q sample values is 
detected, and an inquiry is made at step 522 whether the amplitude A of 
the detected signal is less than 2.sup.-9, which would require that the 
values must be fight-shifted more than six places. If so, the "yes" branch 
is taken to step 525, where a message is sent to the HIU processor that 
the amplitude of the incoming signals is too low. This can be indicative 
of noise in the channel or other signal degradation. The HIU processor may 
respond by changing the frequency for a given DS0 or by ordering a 
selected CIU transmitter to increase its transmitter power. 
If the detected signal level is not less than 2.sup.-9, or after sending a 
message the amplitude is too low, the inquiry is made at step 530 whether 
the detected signal level A is greater than 2.sup.-5, indicating a 
shifting factor greater than three places. If so, the "yes" branch is 
taken to step 532, where a message is sent to the HIU processor that the 
amplitude of the incoming samples is too high. The HIU processor may 
respond by changing the frequency for a given DS0 or by ordering a 
selected CIU transmitter to decrease its transmitter power. 
If the "yes" branch was taken from step 515, or if the detected signal 
level at step 530 is less than 2.sup.-5 and greater than 2.sup.-9, the 
"no" branch is taken to step 535, where the sample values are shifted in 
an appropriate amount left or right to provide for gain shifting. 
At step 540, the I and Q samples are multiplied by a multiplication factor 
M provided by an amplitude detecting and filtering method described in 
greater detail below. The I and Q values after the AGC functions are then 
provided to steps involved in symbol timing recovery (420 in FIG. 9). 
The first step taken for symbol timing recovery is shown at 550. At this 
step, a routine to execute a group delay filter is executed. This involves 
steps shown in greater detail below. After executing the group delay 
filter, the I and Q values are utilized in carrier phase recovery function 
(450 in FIG. 9). The first step taken for carrier phase recovery is to 
execute a baseband phase rotator (BPR) routine 560, utilizing a phase 
increment value PI provided from a carrier phase recovery (CPR) routine 
(470 in FIG. 9). 
After the baseband phase rotator step at 560, the I and Q values from the 
BPR step are provided to the symbol decoding and formatting steps shown at 
565 (which corresponds to the function 480 shown in FIG. 9), wherein a 
decoded symbol is transmitted to the framer circuit. After the symbol is 
output to the framer, the carrier phase recovery (CPR) routine 570 is 
executed to derive a new phase increment PI. 
After executing the CPR routine at step 570, an inquiry is made at step 580 
whether a predetermined time period has elapsed for carrying out automatic 
gain control functions. The predetermined time period in the preferred 
embodiment is 10 symbol periods. If the predetermined time period has not 
elapsed, the "no" branch is taken from 580 to step 587, where the symbol 
timing recovery routine (STR) is carried out to determine a new value of 
the filter coefficient b. If the predetermined time period has elapsed, 
the "yes" branch is taken to step 585, where the AGC control routine is 
executed to obtain a new multiplier M. Control then passes to step 587. 
After step 587, the program loops back to step 510 for receipt and handling 
of the next I and Q values. 
The foregoing general flow diagram may be broken down into a number of 
separate subroutines, which will be described next. 
Automatic Gain Control (AGC) Functions 
Turning next to FIG. 11, the automatic gain control (AGC) function 402 
comprises the basic steps of detecting the exponent of the input samples, 
shifting, and then multiplying the samples by a gain multiplying factor M 
that is derived periodically (every 10 symbol periods in the disclosed 
embodiment). 
The automatic gain control functions adjust the system gain by first 
finding the exponent for a block of incoming data, determining if the 
input amplitudes are too high or too low, and adjusting the gain using a 
shifter and a multiplier. The shifter output amplitude level is preferably 
somewhere between 0.5 to 2 times the nominal input signal level. 
Preferably, further steps are taken for peak detection and filtering to 
eliminate any residual amplitude error by multiplying the output of 
shifting operation by the estimated multiplication factor M. 
The incoming I, Q values from the channelizer are 16 bit words with the 
nominal value of 2.sup.-7 -2.sup.-8 for positive signals and -2.sup.-7 for 
negative signals, but these signals may have higher or lower amplitudes 
depending on the amount of the transmitted power and the channel gain loss 
or noise. As long as the absolute value of the signal amplitude is between 
2.sup.-9 and 2.sup.-5, the incoming signal is reliable. If signal 
amplitude is below 2.sup.-9 or above 2.sup.-5, the AGC reports the signal 
to the HIU processor. First, the exponent of the signal amplitude A is 
determined. Assuming that the incoming signal level from channelizer is 
between 2.sup.-9 and 2.sup.-5, an appropriate shifting operation is done 
such that the nominal value of the signal out of the shifter is 2.sup.-1. 
In the preferred embodiment, the exponent in a block of data containing 
eight in.sub.-- phase (I) and eight quadrature (Q) values is found by 
averaging I.sup.2 +Q.sup.2, and then using a lookup table to pick the 
appropriate shifting factor such that the nominal value of the signal 
level out of the shifter is 2.sup.-1 In other words: 
##EQU1## 
The signal amplitude level in the DSP after the shifting operation is 
limited to upper and lower limits: 
EQU 2.sup.-2 .ltoreq.amplitude after shifting.ltoreq.1 
which corresponds to signal levels of: 
EQU (2.sup.-1 .times.nominal value).ltoreq.amplitude after 
shifting.ltoreq.(2.times.nominal value) 
It will be understood that finding the shift factor is done only at 
initialization, but shifting is done on all incoming samples throughout 
the processing. The output of the shifter function 535 is the input to the 
multiplier 540 (FIG. 10). The multiplication factor in the DSP is: 
EQU 2.sup.-3 .ltoreq.M (Multiplication factor).ltoreq.2.sup.-1 
The initial value of the multiplication factor is 2.sup.-2 ; therefore, the 
multiplier output range is: 
EQU 2.sup.-4 .ltoreq.amplitude after multiplication.ltoreq.2.sup.-2 
The nominal amplitude out of the multiplier is 2.sup.-3 ; therefore, the 
multiplication operation corresponds to: 
EQU (0.5.times.nominal value).ltoreq.amplitude after 
multiplication.ltoreq.(2.times.nominal value) 
The multiplication factor M is updated by the AGC amplitude detection and 
filtering function (408 in FIG. 9). The residual amplitude error will be 
eliminated after AGC reaches the steady state. 
FIG. 12 illustrates the steps taken to implement the program for the 
amplitude detection and filtering process 408 described in connection with 
FIG. 9 (585 in FIG. 10). When considering this figure, it should be 
understood that in every 10 symbol periods, one STR update and one AGC 
update take place. The STR update algorithm occupies the first 4 symbol 
periods and the AGC update algorithm occupies the other 6 symbol periods. 
In determining the amplitude, at 601 the absolute values of the in.sub.-- 
phase (I) and quadrature signals (Q) from the multiplication 406 (FIG. 11) 
are normalized to 1 by summing their values at 605. Then, an amplitude 
error is determined at 609 by first subtracting the absolute values of the 
in.sub.-- phase (I) and quadrature signals (Q) from 2. The error term is 
then averaged at step 611 over 3 past values using a 3 tap rectangular 
window to output an averaged error. The signal is downsampled by 10 at 
step 613. 
The averaged error is input to a stabilizing filter at 615. The gain of the 
stabilizing filter is C1=0.11 in the first 4.2 milliseconds (150 symbol 
periods) for fast convergence of the amplitude level during the 
acquisition time. After the first 4.2 milliseconds, the gain is changed to 
C1=0.05 to minimize the low amplitude jitter at steady state. The 
stabilizing filter output is accumulated at 618 to generate the 
multiplication factor M that is limited to between 0.25 and 4, i.e. the 
values are clipped to these limits at 620. The multiplication factor M is 
provided to the multiplication stage 406. 
A GAIN LOCK signal for the channel being processed is produced by steps 
starting at 625. At step 625, the absolute value of the average amplitude 
error is determined. At step 627, the average of the absolute value of the 
past four amplitude error values is determined. At step 630, this average 
is compared to a threshold value u, which corresponds to an amplitude 
error of about 2.5 % of the ideal amplitude level. Note that the nominal 
preferred threshold value of 0.05 corresponds to a amplitude of 2 (I.sub.n 
+Q.sub.n =2). The initial value of the accumulator is 2.sup.-2, which 
corresponds to a normalized nominal value of 1. This value is preferably 
updated once in the AGC operating time interval (or every 10 symbol 
periods). If the average exceeds the threshold value, the GAIN LOCK signal 
is true for the channel, and this signal is provided to the HIU processor 
as status information as to the channel. 
Symbol Timing Recovery (STR) Methods 
Returning for a moment to FIG. 9, after the automatic gain control (AGC) 
processing 402 comes the symbol timing recovery (STR) processing 420. 
Turning next to FIG. 13, the first processing to occur within the STR 
stages is the variable group delay filter (VGDF) 425. 
First, it will be recalled that the STR methods involve the VGDF function 
425 and a correlation and filtering function 428 that determines a filter 
coefficient b that is used to delay the incoming signal from the 
channelizer 360 such that the sampling would be at the symbol instants. 
Second, the STR methods are carried during a predetermined operating 
interval measured in symbol periods. 
The STR operating time interval is defined as the time that STR algorithm 
is estimating the b value. The first 6 symbol periods inside every 10 
symbol period interval is the STR operation time. The other 4 symbol 
period interval is the time for operation of the AGC (amplitude detection 
and filtering) algorithm, described above. During the STR time, the output 
of the AGC algorithm is held constant and equal to the last AGC filter 
output before the start of the STR operating time interval. During the AGC 
time, the output of the STR algorithm is held constant and equal to the 
last STR algorithm output before the previous STR operating time interval 
ended. 
The symbol timing recovery (STR) function (420 in FIG. 9)(587 in FIG. 10) 
is operative to extract timing information in the form of a group delay, 
and the variable group delay filter to delay the input data (at 2 samples 
per symbol rate) such that the sampling will be at symbol instants. 
The variable group delay filter 425 is a first order allpass filter with a 
variable coefficient b. The transfer function for this filter is: 
EQU H(z)=(b+z.sup.-1)/(1+bz.sup.-1) 
By changing the value of b, the filter introduces different group delays 
without changing the gain. In FIG. 13, the filter 425 is implemented by 
summing the input I or Q value at 640 with the fed-back value from a 
z.sup.-1 unit delay 642, subjecting the sum to a second z.sup.-1 unit 
delay 645, multiplying the sum with the filter coefficient b at 650, and 
adding the operations of the multiplication results and the delayed sum at 
653. 
Referring now to FIG. 14, the VGD filtering is done on both in.sub.-- phase 
(I) and quadrature input (Q) signals at 2 samples per symbol rate. A 
SAMPLE SELECT signal, generated by the symbol timing recovery function 
(FIG. 15), is employed at a switch 660 to select whether the input signal 
to the variable group delay filter 425 is delayed by half a symbol period 
or not. The switch is preferably implemented with selectable storage 
registers within the DSP. The output of the filter 425 is then downsampled 
by 2 at step 665 to reach the symbol rate. 
In the preferred embodiment, a group delay for the coefficient b of b=0 
corresponds to a delay of half a symbol period, while b=1 corresponds to a 
zero delay. Since b is the value of the pole, it must be less than 1, and 
for b&lt;-0.045, the allpass filter shows phase nonlinearity. Thus, b is 
preferably limited by upper and lower bounds: -0.045.ltoreq.b.ltoreq.0.96. 
For this range of b, the group delay of the filter is almost constant in 
the band of interest. This range of b corresponds to a delay range of: 
EQU 0.01.times.Tsymb.ltoreq..tau..ltoreq.0.55.times.Tsymb 
where Tsymb is the symbol timing offset. If the symbol timing offset is 
outside this range, the SAMPLE SELECT signal will toggle its previous 
value to cause a delay (or advance, depending on the previous value of 
sample select) of Tsymb/2 to accommodate for all symbol timing offsets 
from zero to Tsymb. At the same time, the value of b will be changed such 
that the new value of b and the inserted Tsymb/2 delay (or advance) 
correspond to the same symbol timing offset: 
EQU SAMPLE SELECT toggles=&gt;0.51.times.Tsymb.ltoreq..tau..ltoreq.1.05.times.Tsym 
b 
The preferred initial value of b used for processing is b=0.6. This value 
corresponds to .tau.=0.133.times.Tsymb or .tau.=0.633.times.Tsymb 
(depending on whether the selected sample is x.sub.n or x.sub.n-1). The 
initial value of b=0.6 is chosen to be somewhere close to the midpoint 
between b=-0.045 and b=0.96 to prevent the possibility of initial 
oscillation of the SAMPLE SELECT signal which may be due to initial noise 
from other control loops and also to decrease the acquisition time for all 
possible timing offset values. 
The correlator function 428 of the symbol timing recovery (STR) function 
420 is shown in FIG. 15. The method employed is a minimum variance error 
algorithm based on the work of Mueller and Muller. The known Mueller and 
Muller method works on one sample per symbol rate. The timing information 
is derived from the symmetry error of the sampled impulse response by 
extracting the correlation between every two consecutive sample values and 
their estimates. However, the present invention differs from the Mueller 
and Muller approach as follows: instead of IIR filtering the correlator 
output at the symbol rate, in the present invention the correlator output 
is averaged over a number of symbol periods (five in the preferred 
embodiment), and the IIR filtering is conducted at 1/6 of the symbol rate. 
This decreases the required processing power for STR functions such as IIR 
filtering, updating the value of the filter coefficient b, and generating 
the SAMPLE SELECT signal since they now may be effected six times more 
slowly. 
The correlator 428 uses the new in.sub.-- phase symbol I.sub.n, and the 
previous in.sub.-- phase symbol I.sub.n-1 to generate an error signal at 
line 705 that is related to the timing offset between the symbol clocks in 
the transmitter and receiver. The correlator performs the following 
calculations shown at the block 702 to generate the error signal: 
EQU E(n)=I.sub.n .times.sign (I.sub.n-1)+I.sub.n-1 .times.sign (I.sub.n-1) 
This error will be averaged at step 708 with a 5 tap rectangular window to 
reduce the effect of noise from other control loops on the symbol timing 
recovery loop during acquisition time. This filtering is done at the 
symbol rate by averaging a block containing 5 past input error values 
E(n-5) . . . E(n-1). At the next STR operation interval, the window slides 
over the next block of past inputs. Therefore, one averaged output is 
produced once in each STR operation time interval (i.e., every 10 symbol 
periods), as shown at step 710 by downsampling by 10. 
The averaged error then is sent to a two-pole IIR filter at step 713 for 
loop stabilization and timing jitter control. The gain C2 of this filter 
is set to C2=0.115 during the first 4 milliseconds for fast convergence of 
the loop during acquisition time. After the first 4.2 milliseconds, the 
gain value is set to C2=0.03 to reduce the timing jitter at steady state. 
The output of the IIR filter 713 is then sent to a sample select logic 716 
to determine the coefficient value b for the variable group delay filter 
and the SAMPLE SELECT signal. A summary of the operations of IIR filter 
and sample select logic is as follows: 
______________________________________ 
u = input to the sample select logic 
b = output of the sample select logic to be sent to the VGD 
filter 
e.sub.-- str = averaged STR error input to the IIR filter 
SAMPLE SELECT = output of the sample select logic 716 
u(n) = C2 .times. e.sub.-- str(n) + 1.3 .times. b(n-1) - .3 .times. 
b(n-2) 
If (u.sub.n .ltoreq. -.045) OR (u.sub.n .gtoreq. .96) ==&gt; toggle SAMPLE 
SELECT 
signal 
Else ==&gt; SAMPLE SELECT(n) = SAMPLE SELECT(n-1) 
If u.sub.n-1 .ltoreq. -.045 ==&gt; b.sub.n = u.sub.n + 0.8 
Elseif u.sub.n .ltoreq. 0.96 ==&gt; b.sub.n = u.sub.n - 0.95 
Else ==&gt; b.sub.n = u.sub.n 
______________________________________ 
The bottom blocks 720, 722, 725 in FIG. 15 are used for generation of a 
symbol timing lock signal SYMBOL CLOCK LOCK. This signal is generated by 
determining the absolute value of the error values out of the downsampler 
710 at step 720, averaging the absolute value of the past 4 error values 
at step 722, and comparing the average at step 725 to a threshold value. 
If at step 725 the absolute value of this averaged error is less than 0.2, 
the timing offset is approximately within 5 percent of the symbol period, 
which indicates that the symbol timing is locked. The initial output value 
of the sample select logic 716 to the variable group delay filter is 
b=0.6, and SAMPLE SELECT is initial set to zero. 
Carrier Phase Recovery (CPR) Methods) 
Turn next to FIG. 16 for a discussion of the carrier phase recovery (CPR) 
processes (450 in FIG. 9), which comprises a baseband phase rotator 
process (BPR) 460 and a phase lock loop 470. The carrier phase recovery 
450 takes the residual carrier phase from the complex baseband signal by 
estimating the residual phase using a phase lock loop (also known as a 
Costas loop) and rotating the phase of the complex waveform with the 
baseband phase rotator (BPR). 
The baseband phase rotator 460 receives a phase increment (PI) value at 730 
from the phase lock loop and accumulates the phase increments at step 734 
to generate the residual carder phase values at the symbol rate. This 
value is then quantized to 9 bits to form an address for a sine and cosine 
table 740 that is stored in the program memory of the preferred DSP. The 
phase resolution is defined by: 
EQU Phase resolution=2.pi..times.LSB=2.pi..times.2.sup.-9 =0.012272 radians 
This resolution results in a maximum symbol error of -44.2 dB and an rms 
symbol error of about -63 dB. The number of sine and cosine points stored 
in the program memory is 640, which corresponds to 1.25 cycles of a sine 
waveform, 1.25.times.2.sup.9 =640. Since cos .theta.=sin(.pi./2+.theta.), 
the address for the cosine values are obtained by adding 512/4=128 to the 
address of the sine values. The sine and cosine values are used to rotate 
the complex waveform by estimated residual carrier phase by the following 
calculations carried out in steps 743: 
EQU I.sub.out =I.sub.in .times.cos (PHASE)+Q.sub.in .times.sin (PHASE) 
EQU Q.sub.out =Q.sub.in .times.cos (PHASE)-I.sub.in .times.sin (PHASE) 
As shown in FIG. 9, the phase-rotated I and Q are provided to the phase 
lock loop function 470. 
Turning now to FIG. 17, the phase lock loop function 470 is operative to 
determine when the system is locked to the carrier, indicated by the 
signal CARRIER PHASE LOCK, and to generate the phase increment signal PI. 
The in.sub.-- phase I and quadrature Q components from the BPR 460 are 
input to the phase lock loop at the symbol rate to produce an error term 
by step 751 which is passed on to a loop filter 752. The output of the 
loop filter is the phase increment value PI to be used in baseband phase 
rotator. 
A limiter block 755 is used to prevent a false lock. If the carrier phase 
error is such that the phase increment value goes beyond 90 degrees for 
QPSK, a symbol that represents a point on the constellation may rotate to 
another point on constellation and be detected falsely as a correct 
decision. To prevent this false lock for QPSK modulation techniques, the 
phase increment value must be limited to some threshold value which is 
less than 90 degrees. The threshold value used in the preferred embodiment 
is .+-.30 degrees, which corresponds to a 3 kHz offset of the local 
oscillator with the reference frequency: 
EQU .pi./6 radians=2.pi..times.3 kHz/Symbol Rate. 
The estimated error values from steps 751 are sent to the loop filter 752 
to generate a phase increment value. The bandwidth of the closed loop 
transfer function of the loop filter is about 1 kHz, which means that loop 
can track a phase jitter with a frequency below 1 kHz. Further, the 
absolute value of the error values from step 751 is determined at step 758 
and averaged at step 760 over the past 8 values. This average is compared 
at step 762 with a threshold value of 0.11, which corresponds to a phase 
error of 12.7 degrees. If the averaged error is below this threshold, the 
carrier phase is deemed locked. A lock indication signal CARRIER PHASE 
LOCK is then generated and passed to the HIU host system through the host 
interface port. The phase increment output PI of the phase lock loop is 
input to the baseband phase rotator section at the symbol rate. 
Decoding and Formatting 
The final stage of the baseband processing is symbol decoding, differential 
phase decoding, and formatting, shown at 480 in FIG. 9 (565 in FIG. 10). 
Symbol decoding of course entails demodulating the QPSK modulated DS0 
signal and determining the instantaneous values of the signal. As will be 
recalled from prior discussion, the I and Q input at the modulator are 
encoded by differential phase encoding to eliminate the phase ambiguity of 
the received symbols. Therefore, the received I and Q must be 
differentially phase decoded. First, the output of the BPR process 460 are 
mapped to binary decisions, A and B. For QPSK, the mapping is a simple 
operation: 
EQU If I.sub.BPR &gt;0.fwdarw.A=1 
EQU If I.sub.BPR .ltoreq.0.fwdarw.A=0 
EQU If Q.sub.BPR &gt;0.fwdarw.B=1 
EQU If Q.sub.BPR .ltoreq.0.fwdarw.B=0 
The differential encoder takes the present and previous values of A and B 
and maps them into binary symbol decisions I.sub.out, and Q.sub.out. The 
truth table for the mapping function of the differential decoder is shown 
in Table 3 below: 
TABLE 3 
______________________________________ 
mapping function of the differential decoder 
A.sub.n-1 
B.sub.n-1 A.sub.n 
B.sub.n I.sub.out 
Q.sub.out 
______________________________________ 
0 0 0 0 1 1 
0 0 0 1 0 0 
0 0 1 0 1 0 
0 0 1 1 0 1 
0 1 0 0 1 0 
0 1 0 1 1 1 
0 1 1 0 0 1 
0 1 1 1 0 0 
1 0 0 0 0 0 
1 0 0 1 0 1 
1 0 1 0 1 1 
1 0 1 1 1 0 
1 1 0 0 0 1 
1 1 0 1 1 0 
1 1 1 0 0 0 
1 1 1 1 1 1 
______________________________________ 
The values of I.sub.out and Q.sub.out are preferably obtained by a 16-word 
look up table located in program memory of the DSP. Each word is formatted 
as a 3 bit words. The MSB corresponds to I.sub.out, the next bit 
corresponds to Q.sub.out, and the LSB is zero or can be used as a status 
bit. The presence of the third bit is inevitable since the minimum word 
length for the preferred DSP circuit's serial ports is 3. 
FIG. 18 illustrates the timing of the various routines of the shifter, 
multiplier, VGDF, BPR, phase lock loop, STR, and symbol decoding. Taking 
in conjunction with the flow chart of FIG. 10, those skilled in the art 
will be enabled to program the preferred DSP utilized in the disclosed 
embodiment to carry out the invention. 
While there has been shown and described the preferred embodiments of the 
invention, it will be evident to those skilled in the art that various 
modifications and changes may be made thereto without departing from the 
spirit and scope of the invention as set forth in the appended claims and 
equivalents thereof.