Classifications

• • H—ELECTRICITY
  • H04—ELECTRIC COMMUNICATION TECHNIQUE
  • H04J—MULTIPLEX COMMUNICATION
  • H04J1/00—Frequency-division multiplex systems
  • H04J1/20—Frequency-division multiplex systems in which at least one carrier is angle-modulated
• • H—ELECTRICITY
  • H04—ELECTRIC COMMUNICATION TECHNIQUE
  • H04M—TELEPHONIC COMMUNICATION
  • H04M11/00—Telephonic communication systems specially adapted for combination with other electrical systems
  • H04M11/06—Simultaneous speech and data transmission, e.g. telegraphic transmission over the same conductors
  • H04M11/062—Simultaneous speech and data transmission, e.g. telegraphic transmission over the same conductors using different frequency bands for speech and other data

Definitions

• the present invention relates to a technique for recovering analog speech and modulated data simultaneously t ransmi tted 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 from the speech by compensating for phase jitter and frequency offset in the recovered data signal.
• 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 July 21, 1981.
• the data and speech signals are separated in the frequency domain and transmitted in respective separate sideband channels, the data sideband channel containing sidebands generated by time coding an otherwise continuous wave signal.
• a spread spectrum arrangement for (de)multiplexing speech signals and nonspeech signals is disclosed in U. S. Patent 4,313,197 issued to N. F. Maxemchuk on January 26, 1982.
• a block of speech signals may be converted from the time domain to a frequency domain by a Fourier transformation.
• a Fourier component may be pseudo-randomly selected from a subset of such components. Responsive to the selected components, a prediction of the component may be substituted therefor, the prediction being thereafter modified, e.g., by its amplitude being incremented or decremented to reflect the multiplexing of a logic 1 or a logic 0 nonspeech signal. The modified prediction may be converted back to the time domain for transmission to the receiver. At the receiver, a parallel demultiplexing occurs for extracting speech signals and nonspeech signals for the multiplexed signals.
• the samples are sent without modification; however, if a logical 1 is present, frequency inversion scrambling of the samples occurs.
• the receiver performs the inverse process to recover both the speech and data.
• 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 by compensating for phase jitter and frequency offset in the recovered data signal.
• the data is detected and is remodulated and then subtracted, via an adaptive filter, from the transmitted and received signal to yield the recovered speech.
• the weights used in the adaptive filter are adjusted by a device implementing the least mean square algorithm to enable maximum removal of the data signal from the received composite speech and data signal.
• FIG. 1 is a block diagram of a preferred transmitter and receiver arrangement for transmitting simultaneous speech and Multilevel Phase Shift Keyed (MPSK) modulated data signals
• FIG. 2 is a plot of the power density (db) vs frequency averaged for exemplary speech spoken by male and female speakers and a predetermined baud rate data signal transmitted in accordance with the present invention
• FIG. 3 illustrates exemplary curves of the Bit Error Rate (BER) vs data-to-speech power ratio (DSPR) for a data bit rate of 500 bits/sec. for BPSK data carrier frequencies ranging from 500 to 2500 Hz and for Gaussian noise; and
• BER Bit Error Rate
• DSPR data-to-speech power ratio
• FIG. 4 are plots of exemplary BER vs DSPR curves for bit rates between 250 and 1000 bits/sec, where the BPSK data carrier frequency is 2500 Hz.
• FIG. 1 A block diagram of a preferred arrangement of a system in accordance with the present invention which transmits analog speech and data signals simultaneously is shown in FIG. 1.
• the system comprises a transmitter 10 which receives a speech signal and a data signal as inputs from external sources not shown.
• the speech signal can be bandpass filtered in optional filter 12 to an exemplary frequency band of, for example, 200 Hz to 3200 Hz if desired.
• the resultant speech signal S (t) is then scaled by a factor ⁇ in multiplier 14 and transmitted to an adder 16.
• the input data signal is modulated in a modulator 18 with a predetermined carrier frequency f c , which hereinafter will take the exemplary form of a Multilevel Phase Shift Keyed (MPSK) carrier within the analog speech signal frequency band of, for example, 2500 Hz to generate a MPSK modulated data signal D (t) which can include raised cosine pulse shaping.
• MPSK Multilevel Phase Shift Keyed
• the resultant exemplary MPSK modulated data signal is added to the weighted speech signal in adder 16 to produce the transmitted signal X (t) over the analog transmission channel 20.
• X (t) passes through an analog transmission channel 20.
• this channel can be described by its impulse response, H ch(t) .
• ⁇ he receiver 30 sees the transmitted signal as the convolution of the channel impulse response and the transmitted signal, i.e.,
• Receiver 30 recovers the data portion of the received signal in a conventional manner using any suitable carrier recovery arrangement 32 and MPSK demodulator 33.
• Demodulator 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 data signal of the received composite signal .
• This phase error information signal includes raw information relating to, for example, long distance microwave or satellite transmission carrier mismatch and local power frequencies and certain harmonics thereof. In the United States, these frequencies would be, for example, 60, 120 and 180 Hz. In Europe, for example, such frequencies might be 50, 100 and 150 Hz.
• the raw phase error signal is processed to generate an appropriate phase error signal in a Phase Error Tracking Circuit 38.
• Phase Error Tracking Circuit 38 can comprise any suitable circuit known in the art 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 U. S. Patent 4,320,526 issued to R. D. Gitlin on March 16, 1982.
• the performance of the data signal recovery portion of receiver 30 depends largely upon the system parameter ⁇ . From equation (1) it can be seen that the data signal D (t) must be detected in the presence of the speech signal S (t) .
• the system parameter ⁇ is adjusted to make the speech power, , small enough for reliable data recovery.
• the speech signal is recovered by subtracting the data signal component from the appropriately synchronized composite signal . This is accomplished by first regenerating the data signal in MPSK remodulator 34, which corresponds in function to MPSK modulator 18 at the transmitter 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 the regenerated data signal in a manner to substantially improve cancellation of the data signal in the resultant recovered speech signal at a second output of the receiver when the regenerated data signal is subtracted from the received composite signal .
• the remodulator can comprise any suitable circuit such as, for example, a first phase encoding section which includes a phase modulator for converting the data into a phase differential encoded signal which is modified by the retrieved phase error information, and a second modulator section which modulates the resultant signal from the first section into the regenerated data signal .
• the data signal is not subtracted directly from the received composite signal to recover the speech signal until the effects of channel 20 have been accounted for. To do this, an estimate of the channel response H ch(t) . must be made after which the speech signal is recovered via
• Carrier Recovery circuit 32 and MPSK demodulator 33 with Gaussian interference is well understood.
• the interference is speech
• the receiver performance requires special attention.
• Gaussian noise has a uniform frequency distribution, 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.
• the MPSK carrier frequency is expected to play an important role in the BER performance since it is only that portion of the interference falling within the same bandwidth as the data signal which contributes to its detriment.
• a typical data signal with a Binary Phase Shift Keyed (BPSK) carrier frequency of 2500 Hz and baud rate of, for example, 250 is also shown in FIG. 2 as curve 41 superimposed on speech signal curve 40.
• BPSK Binary Phase Shift Keyed
• FIG. 4 shows the BER performance for different DSPRs when different data rates are used.
• the BPSK carrier frequency used is the exemplary 2.5 kHz and, as shown, the higher data rates require a higher DSPR for a given BER.
• the parameter a is adjusted to make the speech power small enough for reliable data recovery. The value of ⁇ can be easily determined from the DSPR as
• Adaptive FIR filter 35 configured for adaptive cancellation, is found to be very efficient in solving such problems where the regenerated data signal from remodulator 34 is convolved with an arbitrary impulse response .
• the resultant signal is then subtracted in subtractor 37 from the composite signal X which is synchronized to by any suitable means, such as a delay in the + input leg to subtractor 37 in FIG. 1, leaving the recovered speech .
• a least mean square (LMS) algorithm is used via circuit 36 to update the impulse response , i.e., used by adaptive filter 35.
• the parameter ⁇ controls how fast filter 35 converges. Larger value allows fast adaptation, but if ⁇ is too large, instability occurs. In addition small values of ⁇ yield smaller errors between the final and H ch(t) .
• the theory of the adaptive filter is described in the heretofore mentioned article by Widrow et al in the December 1975 issue of the Proceedings of 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.
• the heretofore described application of the adaptive filter 35 is a special case where the bandwidth of the input data signal does not occupy the entire analog transmission channel bandwidth. In this case, there are many responses which will work with adaptive filter 35.
• the response outside the bandwidth of the data signal is not defined, so a family of solutions exist. After the LMS algorithm from circuit 36 has converged, will continue to change until it arrives at one of the solutions which creates arithmetic errors in the particular hardware implementation.
• a simple solution to this problem is to remove the modulation filter found in the MPSK modulator 34 located at receiver 30. The resulting signal would then be broadband.
• the adaptive filter solution would then be unique and consist of the channel response H ch(t) convolved with the RC filter response.
• the recovered speech is impaired by channel dispersion, additive channel noise, and imperfect cancellation of the data signal.
• the speech signal-to-noise ratio (SNR) is used.
• SNR can be evaluated as
• N ch is the additive channel noise power while N D is the noise power created by the canceled data signal and is the power of the speech signal.
• ⁇ is an important system parameter in deciding the best compromise between recovered data and speech performance.

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• Engineering & Computer Science (AREA)
• Computer Networks & Wireless Communication (AREA)
• Signal Processing (AREA)
• Noise Elimination (AREA)
• Digital Transmission Methods That Use Modulated Carrier Waves (AREA)

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• 1984
  • 1984-04-02 WO PCT/US1984/000483 patent/WO1984004216A1/fr not_active Ceased
  • 1984-04-02 JP JP59501548A patent/JPS60501086A/ja active Pending
  • 1984-04-02 EP EP19840901557 patent/EP0138975A4/fr not_active Ceased
  • 1984-04-09 CA CA000451522A patent/CA1203030A/fr not_active Expired
  • 1984-04-10 IT IT20477/84A patent/IT1176002B/it active

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