L+R separation system

An L+R separation system separates the L+R component of a composite stereo signal, such as a BTSC signal or an FM broadcast stereo signal, from the L-R suppressed subcarrier component by sampling the composite stereo signal at a rate which is twice the frequency of the L-R component subcarrier frequency. The sampling signal, which is derived from the composite stereo signal, is synchronized with the subcarrier frequency such that the sampling occurs at the zero crossings of the subcarrier frequency, eliminating the L-R component from the sampled output. A simple lowpass filter recovers the L+R component.

BACKGROUND OF THE INVENTION 
The present invention relates to signal separation circuits, and more 
particularly to an L+R measurement system for separating the L+R component 
from a composite stereo signal. 
When decoding stereo signals in either the broadcast FM or BTSC stereo 
systems, it is often desired to obtain the monaural L+R component from the 
composite signal. This may be done to measure the amplitude or other 
characteristics of this component, or as part of a decoding process. The 
BTSC system for television defines a main audio L+R channel up to 15 kHz, 
a pilot signal at the horizontal sweep frequency of 15.734 kHz, and a 
double sideband, suppressed carrier stereo subchannel symmetrically about 
a subcarrier frequency which is twice the pilot signal frequency from 
about 16.470 kHz to 46.47 kHz. The BTSC stereo signal can be written as: 
EQU Vc(t)=(Vl(t)+Vr(t))+2K(Vl(t)-Vr(t)) Sin (2pi*Fs*t)+Ve 
where Vc(t)=BTSC composite stereo signal; Vl(t)=left channel signal=L; 
Vr(t)=right channel signal=R; K=gain coefficient of the dbx noise 
reduction system; 2 pi=6.283183 . . . ; Fs=stereo subcarrier frequency; 
and Ve=everything else such as the pilot, the second language channel and 
the operations channel (SCA in broadcast FM). The user of a modulation 
monitor desires to read the amplitude of the stereo sum term L+R. To 
design lowpass filters for this system requires filter performance several 
orders of magnitude beyond performance adequate for an FM broadcast stereo 
system. The guardbands between the main channel, the pilot signal and the 
subchannel are much narrower, and a dbx noise reduction compressor is 
placed in the subchannel, increasing the potential for subchannel to main 
channel crosstalk. The guardbands between the main channel and the pilot 
signal and between the pilot signal and the subchannel are only 734 Hz 
each, with the separation between the main channel and the subchannel 
being 1.468 kHz. Thus, a significant factor in channel separation is the 
subchannel to main channel crosstalk. 
Subchannel to main channel crosstalk occurs when the lower sideband of the 
subchannel leaks into the main channel due to inadequate low pass 
filtering of the audio that modulates the subchannel. This crosstalk is 
nonlinear, i.e., it is highly offensive to the ear because it is not 
harmonically related to the main channel signals. Additionally when the 
signal levels are low but the program material contains substantial L-R 
content, the dbx noise reduction compressor can cause subchannel levels to 
be 20-30 dB higher than main channel levels. In this situation the main 
channel has negligible ability to psychoacoustically mask the crosstalk. 
Further the greatest gain by the dbx noise reduction system is likely to 
be produced at high frequencies--the very frequencies that appear at the 
edge of the lower sideband and which are most likely to cause audible 
crosstalk. 
Filters that will pass the sum term and reject the difference term must 
have such sharp cutoff characteristics that they will ring and overshoot 
when pulse tested, resulting in erroneous peak readings. High performance 
filters which are adequate for the BTSC system also are quite complex, one 
such having as many as 29 poles of filtering overall, are expensive and 
require great stability. 
What is desired is an L+R separation system which separates the L+R 
component from the composite stereo signal without having stringent 
filtering requirements. 
SUMMARY OF THE INVENTION 
Accordingly the present invention provides an L+R separation system which 
obtains the L+R monaural component from a composite stereo signal, such as 
BTSC or FM broadcast stereo, by time domain sampling the composite signal 
at appropriate points. A sampling signal is generated from the composite 
stereo which has a frequency twice the subcarrier frequency of the L-R 
component of the composite stereo. The sampling signal is used to sample 
the composite stereo at the zero-crossing times of the subcarrier, i.e., 
at times when the L-R component is zero. The resulting signal is then 
filtered using a simple low pass filter to obtain the L+R component.

DESCRIPTION OF THE PREFERRED EMBODIMENT 
Referring now to FIG. 1 a composite stereo signal is input to an input 
filter circuit 10 and to a sampling signal generator 12. The input filter 
circuit 10 suppresses the pilot frequency, if any, which for the BTSC 
system is the television horizontal sync rate of 15.734 kHz, and passes 
the L+R and L-R components to a sampling circuit 14. The sampling signal 
generator 12 outputs a sample pulse stream at a frequency which is twice 
the subcarrier frequency for the suppressed subcarrier (L-R) component of 
the composite stereo signal and which is synchronized with the subcarrier 
frequency such that the sample pulses occur at points where the L-R 
component is zero, i.e., the zero-crossing points of the subcarrier 
frequency. The resulting output, devoid of the L-R component, is then 
filtered by a lowpass filter 16 to recover the L+R component. 
The circuit of FIG. 1 treats the composite stereo signal after removal of 
the extraneous signals by the input filter circuit 10, leaving only the 
stereo sum and difference terms. The unique feature of the stereo 
difference signal is that it is amplitude modulated onto a suppressed 
carrier. At the zero crossings of the suppressed carrier the instantaneous 
value of the stereo difference term is zero, regardless of the amplitude 
of the left or right signals impressed upon it. Therefore if the composite 
signal is sampled at a 2Fs rate in time with the zero crossings of Fs, the 
stereo sum term can be recovered and easily separated from the rest of the 
signals. 
To show the effect of the sampling system the time representation of the 
sampling waveform is transformed into the frequency domain, modified and 
transformed back into the time domain in a new form. The sampling waveform 
Vs is represented as a summation of a series of impulses which are offset 
from each other by the period of the sampling frequency: 
##EQU1## 
where .delta.(x)=a unit impulse at x=0, .delta.(t-n/Fa) being a series of 
impulses separated by t=1/Fa. The Fourier transform of this is another 
series of impulses separated in frequency by Fa (ignoring any scaling 
constants): 
EQU Vs(s)=.SIGMA..delta.(s-n*Fa) for all n. 
Observing that the inverse Fourier transform of .delta.(s) is 1 and that 
the inverse transform of .delta.(s-x)+.delta.(s+x) is 2 Cos (2pi*t), 
transforming the sampling series back into the time domain (again ignoring 
any scaling constants) produces: 
##EQU2## 
The sampling of the composite signal may be expresse as: 
EQU Vo(t)=Vc(t)*Vs(t) 
EQU Vo(t)={(L+R)+2K(L-R) Sin (2pi*Fs*t)*{1+.SIGMA. Cos (2pi*n*Fa*t)} 
Setting K=1, Fa=2Fs and doing a term by term expansion of the first few 
terms of the infinite series: 
##EQU3## 
All the Sin terms cancel, leaving: 
##EQU4## 
The result of the sampling is the stereo sum term L+R plus this same term 
modulating the even harmonics of the subcarrier frequency. The bandwidth 
of the stereo sum term is about Fs/2 and the modulation sidebands extend 
down from 2Fs by Fs/2. Therefore the lowpass filter 16 has to pass Fs/2 
and reject 3Fs/2, a ratio of three to one, requiring a greatly simplified 
filter. 
FIGS. 2 and 3 illustrate graphically the above mathematical derivation. 
FIG. 2 illustrates a composite stereo waveform which is sample at the zero 
crossings of the subcarrier frequency to obtain the L+R component while 
FIG. 3 illustrates that at the zero crossings of the subcarrier frequency 
the values of the L-R component are zero. 
FIG. 4 illustrates an embodiment of the present invention which derives the 
sampling signal from the difference component. The composite stereo signal 
is input to a pass filter 20 which is centered around the subcarrier 
frequency. The output of the pass filter 20 is input to a multiplier 22 
configured as a squaring circuit. The output of the multiplier 22 is then 
input to another pass filter 24 centered around a frequency which is twice 
the subcarrier frequency. The resulting frequency from the second pass 
filter 24 is used to injection lock an oscillator 26 having a frequency 
which is twice that of the subcarrier. The output of the oscillator 26 is 
the sampling signal which is input to a phase adjustment circuit 28 to 
compensate for any phase shifts introduced by the previous circuitry so 
that the sampling signal is in synchronism with the suppressed subcarrier 
of the difference component of the composite stereo signal. 
The composite stereo signal is also input to a sampling circuit 30 where it 
is sampled by the sampling signal from the phase adjustment circuit 28. 
Since the sampling signal is in synchronism with the suppressed carrier, 
the composite stereo signal is sampled at the zero crossing points of the 
subcarrier, i.e., where the L-R component is zero. The remaining component 
from the sampling circuit 30 is input to a low pass filter 32 to recover 
the L+R component as an output. 
As shown in FIG. 5 a composite stereo signal having a pilot signal, such as 
the BTSC signal, is input to a pilot canceller circuit 40 to remove the 
pilot signal and then to a lowpass filter 42 to remove extraneous channels 
at higher frequencies, as is conventional. The composite stereo signal is 
also input to a phase lock loop 44 having a phase detector 46, a filter 48 
and a voltage controlled oscillator (VCO) 50. The output of the VCO 50 is 
input to clock a control circuit 52, which may typically be a programmable 
array logic () integrated circuit. The control circuit 52 generates 
control signals in the form of pulse series of various frequencies. One 
output 54 of the control circuit 52 closes the phase lock loop 44 by 
inputting to the phase detector 46 for comparison with the pilot signal. A 
second output 56 from the control circuit 52 is input to the pilot 
canceller circuit 40 to eliminate the pilot signal from the composite 
stereo signal to be processed. A third output 58 is a sampling waveform 
with a sample rate of twice the subcarrier frequency. A fourth output 59 
is the sampling waveform offset in time by a small amount dt. 
The sampling output 58 is input to a track/hold circuit 60 to which is also 
input the composite signal containing only the L+R and L-R components of 
the composite stereo signal. Since the sampling waveform 58 is 
synchronized with the pilot signal via the phase locked loop 44, the 
pulses of the sampling waveform gate the track/hold circuit 60 at the zero 
crossing times of the subcarrier frequency. As shown in FIG. 6 the 
resulting output of the track/hold circuit 60 is a series of steps having 
some noise at the sampling frequency at the step transitions. The step 
output is then input to a deglitch circuit 62 which converts the series of 
steps, in response to the fourth output 59, into a series of pulses whose 
amplitude is equivalent to the amplitude of the respective steps. The time 
increment dt is chosen so the pulses are output after the settling time of 
the noise prior to any appreciable decay of the step. The L+R steps are 
input to a conventional lowpass filter 64 which converts the pulses into 
the analog L+R waveform upon which desired measurements can now be taken. 
Thus the present invention provides an L+R separation system which 
synchronously samples a composite L+R and L-R signal at the zero crossings 
of the L-R subcarrier frequency to eliminate the L-R component, the 
remaining L+R component being readily recovered with a simple lowpass 
filter.