System for simultaneous analog and digital communications over an analog channel

At a transmitting end, frequencies used to a construct a digital signal are substantially removed from an analog signal by a notch filter circuit to produce an interim signal which is then combined with the digital signal as by combining to produce a composite analog signal that is transmitted to a receiving end. At the receiving end the frequencies used to construct the digital signal are substantially removed from the composite analog signal by a notch filter circuit. In this way the digital signal can be transmitted simultaneously with the analog signal without errors that could be introduced by the analog signal, and with only a slight change to the frequency spectrum of the analog signal. In a second embodiment, the system provides clear and error free continuous transmission of digital signals over an analog channel simultaneously with an analog signal by periodically analyzing the analog signal and, during any given period, transmitting the digital signals in open bands, i.e. bands that are not used, or are slightly used, by the analog signal in the given period.

BACKGROUND OF THE INVENTION 
This invention relates in general to devices for providing simultaneous 
transmission of analog communication and digital data over an analog 
channel, and in particular to such devices that remove from the channel's 
bandwidth available only those frequencies necessary to accomplish the 
data transmissions and only during the data transmissions, or in the 
alternative such devices that continuously adapt to send data in one or 
more bands not used, or slightly used, by the analog signal. 
A need exists for a way to transfer, from time to time, bursts of digital 
data over an analog channel simultaneously with analog communications 
without degrading the fidelity of the analog communications except only 
slightly during the bursts of digital data. Such a need exists in the art 
of telephone communications, particularly with the introduction of a 
feature commonly called "caller identity delivery" also known as "caller 
i.d." (hereinafter sometimes referred to as "CID"). This feature provides 
a user with certain identifying information of a caller, such as telephone 
number and/or the name of the caller. This identifying information, i.e. 
data, is transmitted digitally from a telephone company to its users' 
telephone equipment via the same voice frequency channels used for voice 
communication for display on their respective equipment. In the case where 
the user is not using the telephone when a caller rings, the CID 
information is transferred between the first and second ring signals. The 
user can then view the information to decide whether to answer the 
telephone. 
The techniques for transmitting digital data over an analog channel such as 
a voice frequency channel are well known in the art. For example 
transmission of the digital data can be, and is most commonly, 
accomplished using a technique called "FSK" or frequency shift keying. For 
another example, digital data can also be transmitted over such channels 
by a technique called "PSK" or phase shift keying. Both of these 
techniques utilize a portion of the frequency spectrum available on the 
channel. 
A conflict can arise when a user has the CID feature and also has a 
telephone feature commonly called "call waiting." (The combination of the 
two features is commonly called "caller i.d. on call waiting" hereinafter 
sometimes referred to as "CID/CW"). If a user's equipment is connected to 
another party, the call waiting feature alerts the user to the presence of 
a waiting call by a distinctive audible indication. The user can then 
elect to receive the waiting call by a known keying operation. The 
conflict arises whenever the user is telephoned by a caller while the user 
is engaged in a telephone conversation with another party and the caller's 
CID information is transmitted to the user's receiver. The CID 
transmission takes place almost automatically after the user is alerted to 
the new call so that this information can be used to help decide whether 
to take the new call. Since both the analog signals of the conversation 
between the user and the other party, and the digital signals 
corresponding to the caller's CID information are transmitted over the 
same analog channel, there can be audible interference between the two 
while the CID information is being transmitted. The user and the other 
party involved in the conversation can be subjected to annoying and 
uncomfortable bursts of sound generated by the CID data transmission. 
A heretofore system that has addressed this CID/CW problem uses a technique 
of muting all sound to the conversing parties while the caller i.d. 
information is being transmitted. This system has the advantage of 
providing an error free channel for the CID data communication, but it has 
a significant disadvantage in that it causes the conversation to be muted 
for the three to four seconds it takes to transmit the CID information. 
Another heretofore proposed system for CID/CW involves using a spread 
spectrum signal for CID data transfer mixed with the voice signal. This 
system is too expensive and too complicated to implement. Moreover, the 
spread spectrum signal will be heard as noise to the user reducing the 
clarity of the voice communication during CID transfer. 
A first embodiment of this invention avoids the muting without creating 
noise and is able to transmit CID information over a channel with no 
interruption in voice communication and without subjecting the conversing 
parties to the aforesaid annoying and uncomfortable burst of sound. It has 
the further advantage of not reducing the band of frequencies available on 
the channel for voice communications except only slightly during the brief 
CID data transmissions. It also provides an error free path for the data 
communications. 
A second embodiment invention provides means for continually transmitting 
data and voice simultaneously over an analog channel at the highest 
possible data rates with the good audio quality. Conventional systems 
accomplish this task by means of digitizing and digitally compressing both 
the voice and data, sending it in one data stream and decompressing the 
data at the receiving end. The drawbacks to this is that no matter how 
well the compression algorithm works, analog representation is more 
compressed and provides more information than any digitally compressed 
data. Also, this method is more cost-effective since it does not require 
high-speed compression and re-expansion circuits. 
The second embodiment operates by means of detecting the voice signal and 
determining how much of the voiceband width can be selectively or 
adaptively notched, i.e. eliminated, to provide a varying degree of data 
transmission rates. This differs from the first embodiment in that an 
array of notch filters are continually adjusted to provide maximum 
bandwidth for data and not degrade the voice quality significantly. 
Other advantages and attributes are either discussed in, or can be gleaned 
from a reading of, the text hereinafter. 
Previous methods of achieving simultaneous transmission of speech and data 
have been disclosed in U.S. Pat. Nos. 4,523,311, 4,512,013 and 4,280,020. 
These inventions multiplex the data and the analog signal thereby 
continuously degrading the quality of the analog signal. 
SUMMARY OF THE INVENTION 
An object of this invention is to maintain the maximum possible fidelity of 
an analog signal traversing an analog channel while providing error free 
data communications during without muting the analog signal. 
A further object of this invention is to provide a system for transmitting 
a digital signal simultaneously with an analog signal without errors that 
could be introduced by the analog signal, and with only a slight change to 
the frequency spectrum of the analog signal. 
A further object of this invention is to provide a system for clear and 
error free intermittent data communications by switching into the system 
notch filters, i.e. band elimination filters, that attenuate the data 
transfer frequencies, and after the data communications are complete, the 
notch filters are switched out of the system thereby providing maximum 
analog fidelity of the voice signal. 
A further object of this invention is to provide a system for clear and 
error free continuous transmission of digital signals over an analog 
channel simultaneously with an analog signal by periodically analyzing the 
analog signal and, during any given period, transmitting the digital 
signals in open bands, i.e. bands that are not used, or are slightly used, 
by the analog signal in the given period. 
These and other objects are achieved by a system for simultaneously 
transmitting analog and digital signals via an analog channel comprising a 
transmitting end comprising means for removing from the analog signal 
those frequency components used to construct the digital signal resulting 
in a first interim signal, the removal of said frequency components being 
for the purpose of preventing errors which could be caused by the presence 
of said frequency components in the analog signal, means for producing a 
second interim signal comprising at least a combining of the first interim 
signal and the digital signal, and means for transmitting the second 
interim signal; and a receiving end comprising means for receiving the 
second interim signal, and means for removing from the second interim 
signal those frequency components used to construct the digital signal. In 
a second embodiment a voice signal being transmitted is periodically 
analyzed, as by a fast fourier transform algorithm, for "holes" in the 
voiceband, and during each period the holes found are used to transmit 
data for that period. In any given period, data is transmitted via data 
subchannels disposed within the holes existing in that period. There is a 
fixed and known number of data subchannels spaced in frequency across the 
voiceband. The number and width of the holes in the voiceband in any given 
period is dependent on the amount of voiceband occupied by the voice 
signal during the period. A data subchannel is always open at the high end 
of the voiceband for periodically sending bursts of information (band 
select data) for synchronizing a receiver with a transmitter as to open 
holes and their widths. By defining holes the information also defines 
active data subchannels because they are fixed as to frequency and 
encoding techniques.

DESCRIPTION OF THE PREFERRED EMBODIMENT 
Referring to FIG. 1, the portion of a CID/CW system according to this 
invention that resides in a telephone company's central switching facility 
1 is illustrated to have a switching means comprising a switch 2 and a 
switch control circuit 4. The switch is illustrated diagrammatically and 
is preferably a conventional semiconductor analog signal switch integrated 
with the switch control circuit. The switch provides two alternative paths 
for an incoming analog voice signal ("VOICE SIGNAL") originating from a 
telephone instrument (not shown) remote from the switching facility: a 
first path through a notch filter circuit 6, preferably a pair of serially 
connected notch filters, and a second path 8 that bypasses the notch 
filter circuit. Depending on the setting of the switch the incoming voice 
signal, either filtered or unprocessed, is communicated to one input of a 
signal adder 10, the output of which is amplified if necessary and 
transmitted via telephone lines to a second telephone instrument 
(partially illustrated in FIG. 2) also remote from the switching facility. 
Referring again to FIG. 1, the switch control 4 is caused to select the 
notch filter path for the incoming voice signal at times when CID data is 
being transmitted to the second telephone instrument by the switching 
facility 1. The CID data is transmitted by combining a corresponding CID 
signal ("CID SIGNAL") with the processed incoming voice signal by means of 
adder 10. The notch filter circuit 6 substantially removes the notch 
frequency or frequencies from the frequency spectrum available on the 
channel for voice communication. The frequencies removed are those that 
are used by the switching facility to construct the CID data signals. For 
example, if the switching facility uses an FSK method of sending the CID 
data, then the notch filter circuit effectively removes from the spectrum 
available for voice communications those frequencies, plus and minus a 
tolerance, that are used for the FSK "mark" and "space" signals. In a PSK 
method, a single notch filter substantially makes the carrier frequency 
unavailable for voice communications. This is done in order to avoid 
errors in the transmission of the CID signal components of the composite 
analog signal ("COMPOSITE ANALOG SIGNAL") sent to the second telephone 
instrument. Errors could occur if the voice signal contained relatively 
strong components of the CID transmission frequencies. 
Referring again to FIG. 1, for illustrative purposes only the switch 
control 4 selects the notch filter path in response to a signal ("SELECT 
CID FILTERING") indicating the start of the CID transmission. It should be 
recognized that the signal SELECT CID FILTERING is not necessarily one 
signal but may be a composite of a plurality of signals which combine to 
accomplish the aforesaid filtering of the incoming voice signal. The means 
for generating SELECT CID FILTERING or its equivalent is conventionally 
available at switching facilities 1 implementing the CID/CW feature or can 
be created thereat from conventional circuits by one of ordinary skill in 
the pertinent art. 
Referring to FIG. 2, the composite analog signal sent by the switching 
facility is received by the user's telephone equipment and communicated to 
a switch 12 which provides two alternative paths for the incoming signal: 
a path through a notch filter circuit 14 and a path 15 that bypasses the 
notch filter circuit. Depending on the setting of the switch the incoming 
signal is either processed, preferably by a pair of serially connected 
notch filters 14, or alternately unprocessed before it is communicated to 
a conventional speech network 16. If CID data is not present, then the 
switch is configured to bypass the notch filter circuit. The switch is 
preferably a conventional semiconductor analog signal switch integrated 
with a switch control circuit 18. 
Conventionally the user's telephone instrument is informed by the telephone 
company's switching facility that CID data is going to be sent by means of 
a special alerting tone comprising a 2130 Hz tone and a 2750 Hz tone sent 
simultaneously for approximately 50 milliseconds. This tone is used to 
alert the user's instrument that the CID data is forthcoming. When the 
user's instrument recognizes the alerting tone, it responds by sending 
back to the switching facility an acknowledgment tone conventionally 
called a "DTMF D" tone for 50 to 55 milliseconds. The alerting and 
acknowledgment tones are both well known in the art of telephone 
communication. 
Referring to FIGS. 1 and 2, according to this invention, the two tones 
comprising the alerting tone and their simultaneous duration are 
recognized by means of an analog interface 20 and a digital signal 
processor ("DSP") 22. The composite analog signal received by the user's 
instrument is communicated to the analog interface which preferably is an 
integrated circuit such as the TLC32046 which is a wide-band analog 
interface circuit manufactured by Texas Instruments and conventional 
support circuits. By well known techniques, the analog interface quantizes 
the composite analog signal and presents the quantized data to a serial 
port 24. Communicating with this port is the digital signal processor 
which is preferably a TMS320C40 processor, and conventional support 
circuits, manufactured by Texas Instruments, and which is programmable by 
well known techniques to perform fourier analysis on quantized signals 
such as the alerting tone and the CID signals which follow the alerting 
tone. Using the TLC32046 and the TMS320C40 in combination to analyze 
analog signals in real time is well known. In response to the alerting 
tone, the DSP communicates a signal 26 to the switch control 18 which 
causes the notch filter 14 path to be selected for the incoming composite 
analog signal. After the CID data has been sent the DSP removes the signal 
26 which causes the bypass path 15 to be selected by the switch 12. Also 
in response to the alerting tone, the DSP causes the analog interface, by 
well known techniques, to send the DTMF D acknowledgement signal back to 
the switching facility. 
In operation, the notch filter circuit 14 effectively removes from the 
incoming composite signal those frequencies used by the switching facility 
to construct the CID data signals. For example, if the switching facility 
combined FSK encoded CID data in the composite analog signal, those 
frequencies, plus and minus a tolerance, that were used in the FSK 
technique are effectively removed from the composite signal before it 
reaches the speech network. In a PSK method, a single notch filter removes 
the carrier frequency. In this way, the user is not subjected to the 
audible frequencies used to transmit the CID data, but the overall 
fidelity of the voice signal is only insignificantly reduced. 
Referring again to FIGS. 1 and 2, the composite analog signal communicated 
between the telephone company's switching facility and a user's telephone 
instrument is conditioned by a DAA (Data Access Arrangement) circuit 28 
required by the Federal Communications Commission (FCC) and permits 
bilateral communication between the switching facility and the user's 
telephone equipment over the same channel. These DAA circuits are 
conventional and can be made from a DS2249PH manufactured by Dallas 
Semiconductor. These DAA circuits provide a bidirectional signal interface 
and necessary circuit isolation. At the switching facility end 
conventional network interface circuits 30 are used to accomplish the same 
end, bilateral communications over the same channel. Referring to FIG. 3, 
illustrated is the general form of frequency response curves for the notch 
filters adapted to be centered at 1200 Hz and 2200 Hz respectively. These 
are the frequencies arbitrarily selected for "mark" and "space" 
frequencies for an FSK method of transferring the CID data. They are 
certainly not the only pair of frequencies that could have been selected 
without departing from the spirit and scope of this invention. The 
requirement for CID/CW as set by the telephone regulating authorities is 
to maintain the overall present telephone speech fidelity from 200 to 3200 
Hz. The present invention only degrades the speech path during the 3 to 4 
second interval of the actual CID data transmission. 
Referring to FIG. 4, a computer 32 is in data communication with an 
input/output controller 34. The computer can be any source of data to be 
transmitted across the analog channel and need not necessarily be a 
computer. The computer sends data to the input/output controller in 
predetermined units of data, for example in bytes of data. For each unit, 
the input/output controller responds by sending a data acknowledgement 
signal back to the computer. In this way the flow of data is controlled by 
the input/output controller's data acknowledgement signal. The 
input/output controller is preferably a microcomputer with program memory 
and data memory or a digital signal processor also with program memory and 
data memory. The data received by the controller is buffered in its data 
memory. 
Referring again to FIG. 4, an acoustic transducer, illustrated as a 
microphone 36, receives sound pressure waves and converts them to a signal 
indicated as a "Voice" signal. It should be understood that the source of 
the sound need not necessarily be a voice. The signal from the microphone 
is communicated to a high end notch filter 38, i.e. a band elimination 
filter, that eliminates a small band, relative to the size of the 
voiceband, of frequencies at the uppermost end of the voiceband spectrum 
and produces a filtered signal 40 which is communicated to a first summing 
node 42. A second input to the summing node is a signal 44 from a high end 
modulator 46. The high end modulator receives band select data 48 from the 
input/output controller and modulates it with a frequency from a frequency 
source (not shown) according to a selected modulation scheme such as FSK, 
PSK or any other conventional modulation scheme for sending digital data 
across an analog channel. The significance of the band select data will be 
discussed below. 
Referring again to FIG. 4, the voice signal from the microphone is also 
communicated to a voice power spectrum detection unit 50 which 
periodically analyzes the voice power across the spectrum of the voiceband 
to determine bands of frequencies within the spectrum in which the voice 
power is below a preselected threshold, and which encompass one or more 
"data subchannels." A data subchannel is a unique frequency or set of 
frequencies used to encode data for transmission via an analog channel. 
For example, the two frequencies of FIG. 3 used for FSK encoding define a 
unique data subchannel. As another example, a frequency and surrounding 
band used for PSK encoding would also define a unique data subchannel. The 
detected frequency bands containing voice power below the threshold and 
containing one or more data subchannels can be called "holes" in the 
voiceband spectrum. Preferably there are a plurality of predetermined 
subchannels across the voiceband with known encoding schemes so that for 
any given subchannel, both the transmitter and the receiver will be aware 
of how the data being sent via the subchannel is encoded. 
Referring again to FIG. 4, the detection unit 50 can be, for example, a 
fast fourier transform circuit, or an algorithm performed by a digital 
signal processor (DSP). Preferably a complete voice analysis takes place 
once each 0.5 milliseconds and is accomplished in 0.5 microseconds. The 
output of the detection unit is a map 52 of holes across the voiceband. 
The operation of the detection unit is controlled by the input/output 
controller via control lines 54. As will be explained, these maps of holes 
in the analog signal spectrum determine the band select data 48 
communicated to the high end modulator. 
Referring again to FIG. 4, the input/output controller 34 communicates a 
set of array select signals 56 to a notch filter array 58 and a modulator 
array 60. The notch filter array is preferably implemented as a DSP 
algorithm and comprises an adaptable plurality of notch filters, a set 
(zero, one or more) of which is selected, via the array select signals. 
The set selected at any given time matches those holes in the voice 
spectrum that have been detected by the voice power spectrum detection 
unit 50. In other words, each time the detection unit makes an analysis of 
the then incoming voice signal and finds holes in the voiceband spectrum, 
the input/output controller configures its array select signals to select 
notch filters centered in the detected holes. This results in elimination 
of the voice signal in these holes at the output 62 of the notch filter 
array. This output is communicated to a second summing node 64. 
Referring again to FIG. 4, the input/output controller also communicates 
data 66 to the modulator array 60. The data 66 is the digital data to be 
communicated via the analog channel. The modulator array comprises a 
plurality of modulators selectable by the array select signals 56. Each 
modulator corresponds to a unique data subchannel and modulates data from 
the input/output controller according to the subchannels known modulation 
scheme, e.g. FSK, PSK, etc. The modulators are selected by the array 
select signals and the set of modulators selected matches the set of data 
subchannels found in the then current holes detected in the voiceband. The 
data from the input/output controller is orderly, for example in sequence, 
modulated by the selected modulators and communicated to the summing node 
64 where it is summed with the output of the notch filter array 58. The 
outputs of the summing nodes 42 and 64 are also summed at node 70 and 
communicated to telephone line 72 as a voice/data stream. 
Referring FIGS. 4 and 5, a receiving portion of this invention ("receiver") 
is preferably an integrated member of a transmitting portion 
("transmitter") of this invention so that when two devices according to 
the preferred embodiment of this invention are communicating across an 
analog channel, two-way analog and digital data communications can occur 
simultaneously. However, the transmitter and the receiver can be at 
opposite ends of the analog channel for one way communication if so 
desired. In the preferred embodiment, however, the computer 32 receives 
data from the analog channel 72 via a bandpass filter array 74 which 
receives the voice/data stream as an input argument. The filter array also 
receives a control input comprising a plurality of array select signals 76 
from the input/output controller 34. The array select signals are 
configured according to band select data 78 received by the controller 
from a high end demodulator 80. The high end demodulator receives as an 
input the output of a high end bandpass filter 82 which is in 
communication with the analog channel 72. The band select data is that 
which was sent across the channel by a transmitter according to this 
invention. As explained above, the band select data comprises information 
concerning holes in the voiceband left by the voice signal being sent to 
the receiver, information derived from the periodic analysis of the voice 
signal by the detection unit 50. By specifying the holes, the band select 
data also specifies the currently active data subchannels. From this 
information, the input/output controller can select a set of bandpass 
filters from the bandpass filter array which correspond to the active data 
subchannels, and thereby allow only those frequencies of the active 
subchannels to pass to a demodulator array 84 which also receives the 
array select signals. It is the band select data sent via the high end 
data subchannel that synchronizes the set of modulators used by the 
transmitter and the set of bandpass filters and demodulators used by the 
receiver at opposite ends of the channel. Preferably, the band select data 
is a sequence of information defining at least the then current holes in 
the voiceband, and an input/output controller receiving the band select 
data considers each hole defined by the band select data to be opened 
until that hole is no longer designated in a subsequent burst of band 
select data received by it. Preferably a burst of band select data is sent 
each time an analysis of the voice signal is completed, for example every 
0.5 milliseconds. 
Referring again to FIG. 5, the voice/data stream of the analog channel 72 
is communicated to the input of a high end notch filter 86 which basically 
eliminates the band select data from the stream. The output of the high 
end filter is communicated to a summing node 88. The voice/data stream is 
also communicated to a notch filter array 90 and processed by a set of 
selected notch filters, selected according to the array select signals 76. 
The selected set of notch filters basically eliminates the modulated data 
from the stream, and the outputs 92 of the set of selected notch filters 
are communicated to the summing node 88. The output of this summing node 
is communicated to a speaker 94 which transduces the summed signals to 
sound pressure waves. 
Referring again to FIG. 5, a second voice power detection unit 96 is shown 
in phantom as monitoring the voice/data stream and providing an output to 
the input/output controller 34. This provides the controller, in its 
receiving function, with an optional capability of analyzing which 
subchannels of incoming data are currently active without relying entirely 
on the band select data sent by the transmitter. This can be used to 
verify or supplant the band select data, especially in situations where 
the analog channel is very noisy. The second detection unit can be a 
separate unit, or the two units can be one unit time shared by the 
controller. 
Preferably many or all the functions indicated in FIGS. 4 and 5 are 
implemented by a preprogrammed DSP or other microcomputer, except the 
source and destination of the digital data illustrated as computer 32, and 
during pauses in the voice signal, as when a party momentarily stops 
talking, the entire voiceband is detected as a hole and is used to 
transmit data. Preferably the band select data is periodically sent in 
packets at a high end frequency such as 3500 Hz. In this way, data is 
continually being sent from transmitter to receiver indicating current 
holes openings and the width of the open holes for sending data. 
While the foregoing discussion of the first embodiment centered around the 
application of this invention to sending CID data simultaneously with a 
voice signal, this invention is equally useful for other telecommunication 
applications. As shown in the second embodiment, it is useful for sending 
digital data via analog channels in general, including modem applications. 
Using this invention, two computers can be engaged in continuous two-way 
communication of digital data, such as by modems, while their operators 
can be engaged in voice communication, both over the same telephone line. 
The foregoing description and drawings were given for illustrative purposes 
only, it being understood that the invention is not limited to the 
embodiments disclosed, but is intended to embrace any and all 
alternatives, equivalents, modifications and rearrangements of elements 
falling within the scope of the invention as defined by the following 
claims.