CA1203030A - Simultaneous transmission of speech and data over an analog channel - Google Patents

Abstract

SIMULTANEOUS TRANSMISSION OF SPEECH AND DATA OVER
AN ANALOG CHANNEL

Abstract The present invention relates to a technique for recovering each of an entire analog speech signal and a modulated data signal simultaneously received over a transmission channel such as a common analog telephone speech channel. In the received composite signal, the entire modulated data signal is multiplexed within the normal analog speech signal frequency band where the speech is present and its signal power density characteristic is at a low level. Separation of the speech and data signals at the receiver is effected by recovering the modulation carrier frequency and demodulating the received signal to recover the data signal. The data signal is then (a) remodulated with the recovered carrier, (b) modified to cancel phase jitter and frequency offset errors detected during the data demodulating process and (c) convolved with an arbitrary channel impulse response in an adaptive filter whose output signal is subtracted from the received composite data and speech signal to generate the recovered speech signal. To improve the recovered speech signal, a least mean square algorithm is used to update the arbitrary channel impulse response output signal of the adaptive filter.

Description

3~

SIMULTANEOUS TRANSMISSION OF SPEE:CH AND DATA OVER
AN ANALOG CIIANNEL

The present invention relates to a technique for 5 recovering analog speech and modulated data simultaneously transmitted over an analog channel with the capability at the receiver of separating the two simu~taneously received signals and substantial~y improving the cancellation of the data from the speech by compensating for phase jitter and frequency offset in the recovered data signal.
~ xisting analog transmission facilities wo~ld be more efficient if speech and data could be simultaneously transmitted over the same channel. Preferably, such a proposal shou~d not compromise the recovered speech and data qualities, neither should there be an expansion in the bandwidth requirements. At the same time, ;t is desira~le to have a system which is simple and cost-effective~
A method of transmitting data and speech signals in a telephone system in which communication is effected via a radio link is disclosed in U. S. Patent 4,280,020 issued to L. E. Schnurr on Jul~ 21, 1981. There the data and speech signals are separated in the frequency domain 3nd transmitted in respective separate sideband channels, the data sideband channel containing sidebands generated by time coding an otherwise continuous wave signa~.
A spread spectrum arrangement for (de~mul~ipl~xing speech signals and nonspeech signals is disclosed .n U. S. Patent 4,313,1~7 issued to N. F.
Maxemchuk on January 26, 1982. There, at the transmitter, a block of speech signals may be converted from the time domain to a frequency domain by a Fourier transformationO
A Fourier component may be pseudo-randomly selected from a subset of such components. Responsive to the selected components7 a prediction of the component may be substituted therefor~ the prediction being thereafter modified~ e.g.9 by its amplitude being incremented or A~
, .

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- 2 -decremented to reflect the multiplexing of a logic 1 or a logic 0 nonspeech signal~ The modified prediction may ~e converted back to the ~ime ~omain for transmission to the receiverO At the receiver~ a parallel demu~tiplexing - 5 occurs for exteacting speech signals and nonspeech signals for the multiplexed signals.
Recentiy several systems have been proposed to send speech and data simultaneously which exploit the properties of the Short Time Fast Fourier Transform (FFT) and the statistical properties of speech. For example~ in the article "Si~ultaneous Trans~ission of Speech and Data using Code-Breaking Techniques" by R. Steele et al in BSTJ
VolO 60, No~ 9, Novembe~ 1981 at pages 2081-2105, a system whereby speech is used as a da~a carrier is proposed. More particularly, the speech, sampled at 8 kHz, is divided into blocks of N samples, and provided the corre~ation coeficient and mean square value of the samples exceed system thresholds, data is allowed to be transrnitted. If the data is a logical 0, the samples are sent without modification; however~ if a logical 1 is present, frequency inversion scramblins of the samples occurs. The receiver performs the inverse process to recover both ~he speech and data. These techniques can be quite co~plex and require careful timing and non-dispersive channels~
The problem remaining is to provide a technique for the simultaneous trans~ission of speech and data over an analog channel while compensating at the receiver for various effects produced by the analog channel which technique is simple and cost effective and does not require an expansion in bandwidth requirements.
The foregoing problem has been solved in accordance with the present invention which relates to a technique for the simultaneous transmission of analog speech and modulated data over an analog channel with the capability at the receiver of separating the two simultaneously received signals and substantially improving the cancellation of the data signal from the speech signal ~2~3~3~3 by co~pensating for phase jitter and freq~ency offset in the recovered data signal.
It is an aspect of the present invention to provide a receiver for recovering from an analog ~ 5 transmission channel, which includes a predetermined channel bandwidth, each of a simultaneously received analog speech signal and a modulated data signal, where the analog speech signal includes a predetermined power density characteristic over the bandwidth of the analog transmission channel and the data signal is received in the portion of the analog transmission channel frequency band where the power density characteristic of the analog speech signal is at a minimal level~ At the receiver ~he data is detected and is remodulated and then subtracted, via an adaptive fil~er, from the transmitted and received signal to yield the recovered speech. The weights used in the adaptive filter are adjusted by a device i~ple~enting the least mean square algorithm to enable maximum removal of the data signa~ from the received composite speech and data signal. In the process of data detection, information relating to phase jitter and frequency offset is generated and used in remodulating the data in order to substantially improve cancellation of the data signal from the recovered speech signal.
Other and further aspects of the present invention will become apparent during the course of the following description and by refer~nce to the accompanying drawings.
Referring now to the drawings, in which like numerals represent like parts in the several views:
TIG. 1 is a block diagram of a preferred transmitter and receiver arrangement for transmitting simultaneous speech and Multilevel Phase Shift Keyed (MPSK3 modulated data signa~s;
FIG. 2 is a plot of ~he power density (db) vs frequency averaged for exempiary speech spoken by male and female speakers and a predeteemined baud rate ~ata si~nal ~L2~3~t3~

transmitted in accordance with the present inventlon;
FIGo 3 illustrate~ exemp.lary curves of the Bit Error Rate (BER) vs data-to-speech power ratio (DSP~) for a data bi~ rate of 500 bits/sec. for BPSK data carrier frequencies ranging from 500 to 2500 Hz and for Gaussian noise; and FIG. 4 are plots of exemplary BER vs D5PR curves for bit rates between 250 and 1000 bits/sec~, where the BPSK data carrier frequency is 2500 Hz, A block diagram of a preferred arrangement of a system in accordance With the present inventlon which transmits analog speech and data signals simultaneously is shown in FIG. l. The system comprises a transmitter lO
which receives a speech si~nal and a data signal as inpu~s from external sources not shown. The speech signal can be bandpass fil~ered in optional filter 12 to an exemplary frequency band or, for example, 200 H~ to 3200 Hz if desired. The resu~tant speech signal S(t) is then scaled by a factor ~ in mul~iplier 14 and transmitted to an adder 16. The input data signa~ is ~odulated in a modu~ator 18 with a predetermined carrier frequency fc~
which hereinafter will take the exe~plary form of a Multiievel Phase Shift Keyed (MPSK) carrier within the ana70g speech signal frequency band of 9 for exampler 2500 Hz to generate a MPSK modulated data signal ~(~) which can include raised cosine pulse shaping. The resultant exemplary MPSK modulated data signa~ is added to the weighted speech signa~ in adder 16 to produce the transmi~ted signal X(t) over the ana~og trans.mission channel 200 The transmitted sign~l can be defined as X(t)=D(t)~aS(t)~
In the present system, the transmitted signal X~t) passes through an anàlog transmission channel 20. To a first approximation, this channel can be described by its impulse response; HCh(t). The receiver 30 sees the transmitted signal ~(t) as the convolution of the channel i~pulse response and the transmitted signal, iOe., ~L2~3~3~

X(t~ = (D(t)-~S(t)) ~Ich(t) ~ (D(t~*HCh(t))~ s(t) Hch(t)) (1) Receiver 30 recovers the data portion of the received signal Xtt~ in a conventional manner using any suitable carrier recovery arrangement 32 and MPSK
demodulator 33~ Demoduiator 33 comprises a decoder and decision section which has the capability of (a) decoding the received data signal for transmission to both a first output of the receiver and a remodulator 34 and (b) generating phase error information relating to phase jitter and frequency offset in the da~a signal of the received composite signal X(t). This phase error information signal includes raw information relating to, for example, long distance microwave or satellite transmission carrier mismatch and local powee frequencies and certain harmonics thereof. In the United States, these frequencies would be, for examp~e, 60, 120 and 180 Hz. In Europe, for example, such frequencies might be 50, 100 and 150 H2~ The raw phase error signal is processed to generate an appropriate phase error si~nal in a Phase Error Tracking Circuit 38.
~ is ~o be understood that Phase Error Tracking Circuit 38 can comprise any suitable circuit known in the ar~ as, for example, separate bandpass filters for each of the frequencies of interest; a low pass filter to, for example, pass up to 500 Hz; or an Adaptive Phase~Jitter Tracker disclosed, for example, in UO S~ Patent 4~320J526 issued to R~ D. Gitlin on March 16, 1982. The performance of the data signal recoveey portion of receiver 30 depends largely upon the system parameter u~ From equation (1) it can be seen that the data signal D(t) must be detected in the presence of the speech signal Stt~. The system parameter is adjusted to make the speech power, aS, sma~l enough for :~p~

reliable data recovery.
The speech si~nal is recovered by subtracting the data signal D(~) component from the appropriately synchronized composite signal X(t). This is accomplished by first regenerating the data signal D(t) in MPSK
remodulator 34, which corresponds in function to MPSK
modulator 18 at the transmi~ter 10. Timing for the MPSK
remodulator 34 is obtained from the carrier recovery circuit 32. In addition, the phase error information from Phase Error Tracking Circuit 38 is introduced into ~he regenerated data signal D(t) in a manner to substantially improve cancellation of the data signal in the resu:Ltant recovered speech signal at a second output of the receiver when the regenerated data signal is subtracted from the received composite signa~ ~(t) The remodulator can comprise any suitable circuit such as, for exa~ple, a first phase encoding section which inc1udes a phase modulator for converting the data into a phase differentia~ encoded signal which is modified by the retrieved phase error information, and a second modulator section which modulates the resultant signal ~rom the first section into the regenerated data signal D(t). The data signal ~(t) is not subtracted directly from the received composite signal X
to recover the speech signal S(t) until the effects of channal 20 have been accounted fo~. To do this, an estimate o~ the channel response HCh~t) must be made after which the speech signal ~(~) is recovered via t~ - [(a(t) Hch~t)~+(~S~t~ Hch~t))~ (~(t) ~Ich~t))~(2) The problem of estimating the channel response HCh(t) knowing the data signal D~t) and not knowing the random variable speech signal S(t) is solved in a~cordance with the present invention by the use of an adaptive filter 35. Presently~ an adaptive Finite Impulse Response (FIR) filter whose weights are adjusted by the least mean square ~LMS) algorithm via device 36 is used for adaptive ~203n3q:1 filter 35. A typical arrangement is shown in FIG. 29 of the article "~daptive Noise Cancelling: Principles and Applications" by B~ Widrow et al in Proceedings of the IEEE, Vol. 63, No. 12, December 1975 at paye 1709.
-The per~ormance of a MPSI~ receiver, comprising Carrier Recovery circuit 32 and MPSK dernodulator 33, with Gaussian in~erference is well understood. However, when the interference is speech, the receiver performance requires special attention. White Gaussian noise has a uniform frequency distrib~tion, so when the data bit-error-rate (BER) is looked at, the MPSK carrier frequency is not important. The power density of speech is not uniform with frequency, but rather decreases rapidly as the frequency increases as shown in FIG. 2 for curve 40. In this case the MPSK carrier frequency ;s expected to play an important role in ~he BER performance since it is only that portion o the interference falling within the salne bandwidth as the data signal which contributes to its detriment, A typica~ data signal with a Binary Phase Shift Keyed-(BPSK) carrier frequency of 2500 Hz and baud rate of, for example, 2~0 is also shown in FIG. 2 as curve 41 superimposed on speech signal curve 40.
Ik has been found that for a given data-to-speech power ratio (DSPR), better BER performance is obtained when a higher carrier frequency is selected as shown in FIG. 3 using a matched filter receiver. FIGo 4 shows the BER
performance for different DSPRs when different data rates are usedO In FIG. 4, the BPSK carrier frequency used is the exemplary 2O5 kHz and, as shown, ~he hi~her data rates require a higher DSPR for a given BER. As mentioned hereinbefore, ~he parameter ~ is adjusted to make the speech power small enough for reliable data recovery. The value of ~ can be easily determined ~rom the DSPR as DSPRdb ~ (3) a = 10 ~2~!3~

The heretofore described application of the adaptive filter 35 is a special case where the b~ndwidth of the input data signal D(t) does not occupy the entire analog transmission channel bandwidth. In this case, there are many responses H~t) which wi~l work with adaptive filter 35. The response outside the bandwidth of the data signal D(t) is not defined, so a family of solutions exist.
After the L~S algorithm from circuit 36 has converged, ~(t will continue to change until it arrives at one of the solutions which creates arithmetic errors in the particular hardware implementation. A simp~e solution to this problem is to remove the modu~ation filter found in the MPSK
modulator 34 located at receiver 30. The resulting signa~
D(t) would then be broadband~ The adapti~e filter solution would then be unique and consist of the channel response HCh(t) convolved with the RC filter response.
It is to be understovd that the recovered speech is impaired by channel dispersion, additive channel noise~
and imperfect cancellation of the data si~nal. To quantify the recovered speech quality, the speech signal-to-noise ratio (SNR) is used. The SNR can be evaluated as 2S SNR = 10 log ~ S___ (5) NCh is the additive channel noise power while ND ;s the noise power created by the canceled data signa:L ~(t) and a~ is the power of the speech signal. Hereinbefore~ it was stated that a sma~ler value of ~ yields a better B~R.
However, from Equation (5) it can be seen that ~he recovered speech SNR decreases with ~ and that, if ~ must be very small, poor recovered speech quality is expectedO
Therefore, a is an important system parameter in deciding the best compromise between recovered data and speech performance.

~3~3~

The degree to which the speech signal can be recovered fro~ the composite data and speech signal received in receiver 30 is limited primarily by how well the channel 20 response HCh(t) can be estimated using eq~ation (2). Adaptive FIR filter 35, configured for adaptive cancellation, is found to be very efficient in solving such problems where the regenerated data signal S(t) fro~ remodulator 34 is convolved with an arbitrary impulse response H(t). The resultant signal is then subtracted in subtractor 37 from the composite sigrial X
which is synchronized to ~(t) by any suitable ~eans, such as a delay in the ~ inpu~ leg ~o subtractor 37 in FIG. 1, leaving the recovered speech g(t)- To improve the estimate of the recovered speech, a least mean square (LMS) algorithm is used via circuik 36 to update the impulse response H(t)~ i.e., ^ ^ ~
H(t~,l)=H(t) tlls(t)D(t) ' (~) used by adaptive filter 35. After many iterations, H(t) converges from its arbitrary response ~t) to HCh(t)~ and the recovered speech at the output of subtractor 37 contains little or no noise attributed to the data signal D(t). The re-introduction of phase error information into the remodulated data signal D(t) assists the adaptive fil~er 35 which is not capable of compensating for the phase error.
The parameter ~ controls how fast filter 35 converges. Larger v~lue allows fast adaptation, but i ~
is too large, instability occurs~ In addition small values of ~ yield smaller errors between the final H(t) and HCh~t). The theory of the adap~ive ilter is described in the heretofore mentioned article by Widrow et al in the December 1975 issue of the Proceedings o~ the IEEE. As a typical example, a FIR filter length of 64 and a ~ of 10 9 was used to achieve a data cancellation in the neighborhood of 33 db.

~%(~3~3~

It is to be understood that ~he above-described embodiments are simply illustrative of the principles of the inventionO Various other modifications and changes may be made by those skilIed in the art which will embody the principles of the invention and fa~l within the spirit and scope thereof. It is ~.o be understood that analog transmission channel 20 can comprise many forms such as, for example, a common telephone channel which operates within the 0-4000 H~ range with unknown amplitude and frequenc~y distortionsO

Claims (7)

Claims

1. A receiver characterized by:
an input terminal capable of simultaneously receiving an analog speech signal which includes a predetermined power density characteristic over a predetermined bandwidth, and a modulated data signal which is received in a portion of the received analog speech signal bandwidth where the analog speech signal is present and the power density characteristic of the analog speech signal is at a low value, a first and a second output terminal means capable of demodulating the data signal from a received composite analog speech and modulated data signal for transmission to the first output terminal and for generating phase error signals detected in the received data signal;
means capable of remodulating the recovered data signal at the output of the demodulating and recovering means while introducing said phase jitter and frequency offset information signals for generating an output signal corresponding substantially to the data signal received at the input terminal of the receiver including said phase error signal;
adaptive filtering means capable of generating a first signal representative of an estimate of an impulse response of a channel connected to the input terminal of the receiver, and convolving said first signal with the remodulated data output signal from the remodulating means to generate a resultant output signal; and means capable of subtracting the resultant output signal generated by the adaptive filtering means from the composite analog speech and modulated data signal received at the input terminal of the receiver for substantially canceling the data signal forming part of the composite received signal and generating a resultant output signal at the second output terminal of the receiver which comprises the recovered analog speech signal.

2. A receiver in accordance with claim 1 CHARACTERIZED BY
a phase error tracking means responsive to the phase error signals from the demodulating means capable of tracking the phase error at either one of selected frequencies or a selected frequency band associated with interference signals that introduce phase error into the composite signal propagating in the channel connected to the input terminal of the receiver.

3. A receiver in accordance with claim 2, CHARACTERIZED IN THAT
the remodulating means comprises phase encoding means capable of converting the demodulated data signals at the output of the demodulating means to a phase differential encoded signal and reintroducing said phase error signal into the encoded signal; and modulating means responsive to a phase differential encoded signal at the output of the phase encoding means to generate said output signal of the remodulating means.

4. A receiver in accordance with claim 1, CHARACTERIZED IN THAT
the adaptive filtering means comprises means capable of generating said first signal and convolving said first signal with the remodulated data output signal from the remodulating means; and means responsive to the resultant output signal from the subtracting means and the remodulated data output signal from the remodulating means for causing a modification of the first signal generated by the generating and convolving means for producing a resultant output data signal of the adaptive filtering means which best cancels the data signal at the second output terminal of the receiver.

5. A receiver in accordance with claim 4, CHARACTERIZED IN THAT
the modification means of the adaptive filtering means comprises an arrangement for implementing a least mean square algorithm on sequential synchronized samples of the output signals of both the remodulating and the subtracting means for producing control signals to the generating and convolving means which converge the estimate of an impulse response of the channel connected to the input terminal of the receiver to an actual channel impulse response.

6. A receiver in accordance with claim 1, CHARACTERIZED IN THAT
in the adaptive filtering means is adapted to generate a first signal which is an estimate of an impulse response of an analog transmission channel.

7. A receiver in accordance with claim 1, CHARACTERIZED IN THAT
the received modulated data signal at the input terminal is a multilevel phase-shift-keyed data signal.