Re: Multiplex decoder nearly rewired.

In article <437C5E56.59006049@xxxxxxxxxxxxxxxxxx>, Patrick Turner
<info@xxxxxxxxxxxxxxxxxx> wrote:

> I saw the filter in the Quad circuit and i don't know what F it is for, 55khz,
> or 19kHz, since there is no 19kHz filter indicated.

It is a low pass filter, presumably cutting off starting somewhere around
55 kHz, with a null tuned to 67 kHz according to the alignment
instructions in the QUAD manual. The 67 kHz null was necessary in the US
to eliminate the 67 kHz SCA sub carrier components, plus it gives the
filter a sharper cutoff, although at the expense of ultimate attenuation
at higher frequencies. The QUAD stereo mpx design does not include a 19
kHz filter, 19 kHz filters were uncommon in FM stereo tuners prior to the
introduction of Dolby noise reduction into cassette recorders.

> > > Good that you have mentioned the 55kHz filter.
> > > When I looked at the Quad decoder I didn't spot this filter.
> > > I will try such a filter to see if it reduces the much larger hiss
level in
> > > stereo
> > > that I am getting compared to mono, even when no modulation is applied
> > > at my transmitter.
> >
> > You might want to look into combining your 19 kHz pilot filter with the
> > low pass filter as was done in the Marantz 10B. A former Marantz employee
> > said that with proper design the phase shifts introduced by the 19 kHz
> > pilot filter can be used to compensate for the phase shifts introduced by
> > the low pass filter into the upper sidebands of the 38 kHz DSBSC sub
> > carrier.
>
> But filtering to separate the L+R from the DSB is not required.
> Scott also don't separate the composite.

Yes, both the Scott circuits and the QUAD circuit keep the composite
signal together and do not separate the L+R from the DSB, however it is
still necessary in the US to eliminate the 67 kHz SCA sub carrier, as well
as high frequency random noise resulting from the FM demodulation
process. Today audiophiles feel that 19 kHz filters are a necessity least
the residual 19 kHz bother their golden ears. The 19 kHz filter is
commonly inserted in the two audio paths coming out of the decoder, but
some times it is placed before the decoder as you have done. In the
design of the famed 10B Marantz combined the 19 kHz null with the low pass
function and the 67 kHz SCA null into a single integrated filter design,
which has been claimed to offer improved phase compensation.

> The whole idea of the Quad circuit and some othr similar ones, and including
> the Scott circuits is to keep all the components of the composite signal
together,
> so that good separation is possible across a wide band.
> The basic idea is to simply add the missing 38khz carrier to the
existing composite
> from the
> ratio detector, or whatever detector is used.

Yes that is how it is usually, but not always, done. The low pass and 67
kHz null filters used between the ratio detector and the mpx decoder cause
phase shifts or time delays that are not constant across the band up to 53
kHz occupied by the stereo composite signal. This nonlinear time delay
causes a loss of stereo separation, keep in mind that modern audiophiles
are not content with the 20 to 30 dB of separation that you are willing to
settle for, and demand separation figures in the 60 dB range least the
"sound stage" be compromised.

Marantz simply observed that by combining the 19 kHz null with the 67 kHz
null, and the 54 kHz low pass in a single coordinated package, they were
able to make the phase shifts due to the 19 kHz null filter compensate for
the phase shifts introduced by the 67 kHz null and low pass filter, or so
a former Marantz employee has claimed. I haven't gone through the math to
verify this, but I assume that the idea is that the phase shift of the 19
kHz null at the upper end of the L+R audio band is designed to complement
the phase effects on the demodulated L-R audio resulting from the
combination of the phase shift of the 67 kHz null and LPF on the upper
sideband of the 38 kHz sub carrier and the 19 kHz null on the lower
sideband of the 38 kHz sub carrier. The idea being that the L+R and L-R
audio would line up correctly in the decoder ring and subsequent auxiliary
matrix. This would not however imply that the time delay would be
constant vs. frequency for the demodulated left and right audio signals.
This is just my speculation as to what Marantz was trying to do with their
filter, I have not gone through the analysis, so my speculation could
easily be all or partly wrong. Comments about the design of the Marantz
composite filter from anyone that knows more would be welcomed.

> The phase of the carrier relative to the subcarrier waves is easily
> swayed to line up exactly.

Yes, you can sway the phase of the carrier relative to the sub carrier
waves so that it effectively matches the phase of the original suppressed
carrier, but I don't believe that necessarily implies that the phase of
the demodulated L-R audio and the L+R audio will correctly line up in
phase to give maximum separation, especially across the entire audio band,
that depends on the constancy of the time delays in the composite filter
comprised of the 19 kHz null filter, 53 kHz low pass filter, and the 67
kHz null filter.

> Any attempt to filter out the DSB and thus gain the L-R signal so that
it may be
> applied
> to LPF filtered phases of L+R, and -L-R results in poor separation at HF
due to
> inevitable phase shifts caused by what must be more than first orer filters.
> I tried all this in many efforts but always the phase shifts prevented
more than
> 12dB sep at 10kHz, and
> almost no sep at 13kHz.
>
> The phase effects of the 19kHz notch filter between 23kHz and 53 kHz and
between
> 20Hz and 15kHz is utterly negligible.

I'm not sure what you mean by "utterly negligible", clearly something is
causing your poor high frequency separation even if it isn't your 19 kHz
filter. In any case the phase effects of the 67 kHz null and 53 kHz low
pass filters are not negligible, I take it what Marantz was trying to do
in the 10B is to also make the phase effects of the 19 kHz null filter non
negligible and then take advantage of the phase effects of the 19 kHz null
to compensate for the effects in the decoded audio resulting from the 67
kHz null and LPF.

> It can be shunted out and all that does is in my similar to Quad circuit
is dump
> a whole lot of 19khz in each L&R output, plus probable IMD products.
>
> Much of what we listen to, especially the HF portions, is at a level below the
> pilot tone levels.
> IN my case I am able to banaish the damn pilot tone after a cathode
follower, and
> thus
> drive drive the diodes with a low impedance.
> There *is* a small effect of the 19kHz filter elements after the follower
> but that is easily compensated for with a cap on the cathode R of the
gain before
> the CF.
> We are not looking for much compensation, about 3dB max.

I wouldn't call 3 dB an "utterly negligible" effect! Are you sure it is
the "small effect of the 19kHz filter" that your "compensation" capacitor
is correcting for? That 3 dB figure is suspiciously close to the
theoretical 3.9 dB difference in level between the L+R and L-R signals in
the FM stereo composite signal.

> > > > 4.) It isn't clear how you are compensating for the fact that the fact
> > > > that the L-R sub carrier signal is effectively transmitted at a lower
> > > > level than is the main channel L+R signal? QUAD used an auxiliary
matrix
> > > > to perform this function, feeding inverted L+R audio into the outputs of
> > > > the decoder to cancel some of the L+R signal.
> > >
> > > Indeed you are correct about the Quad additional matrix.
> > >
> > > But I saw that as a compensation network because separation levels
of L and R
> > > modulation
> > > recovered tend to be poor because the amplifying of the ratio
detector signals
> > >
> > > tends to lessen separation; the composite signal envelope is mauled by the
> > > RC couplings and gain variations between the audio frequencies and the
> > > subcarrier F.
> > >
> > >
> > > > There are at least two
> > > > other was to perform this function, are you following the QUAD
lead, or is
> > > > the "compensation" capacitor in the cathode circuit of the input
amplifier
> > > > a crude solution to this problem.?
> > >
> > > The compensation cap isn't as crude as you think; it tweaks the
phase of the
> > > supressed carrier double sideband signal and slightly increase the gain at
> > > around 38kHz.
> >
> > It is a crude approach to equalizing the differing levels of the L+R and
> > 38 kHz L-R signals.
>
> Well my system works very well to give over 30db of separation, and is simple.

I thought you said the separation was only 20 dB at higher frequencies?

> > > Then you should find the separtaion from the diode ring are either L or R,
> > > without any need for Quad's method of compensation.
>
> The diode ring seems to cause some loss of separation and there would only be
> perfect
> sep if there was perfect detection, as theory suggests it should happen,
but it just
> doesn't occur.
> The CF and the oscillator have to provide the power to the matrix, and
> the some losses seem to occur.

There is a theoretical "problem" with the FM stereo signal that you
haven't considered yet that is responsible for most of this loss of
separation, even if the diode ring were perfect you would still see the
effect.

> > > This is understood if you just modulate the FM signal on R audio
channel only
> > > with nothing on the L channel.
> > > The magnitude of the L+R signal = that of the modulation carried by
the DSB
> > > signal,
> > > which is L-R.
> > > The signal applied to one side of the diode ring will then appear to be
> > > an AM 38khz carrier but which has the top of the envelope modulated
with the R
> > > channel
> > > signal, and the bottom should be the L channel modulation, so it
should be a
> > > flat line of mod
> > > because there is no L channel mod in tis test case.
> > >
> > > The other side of the diode ring has an inverted version of the
other side,
> > > flat line on top, R mod on the bottom.
> > >
> > > The levels of L-R transmitted at the station must be fixed to be the
same for
> > > all stations for all tuners to give the same separation.
> >
> > The point you are missing is that the peak amplitudes of the L+R audio and
> > the 38 kHz sub carrier are the same, but the demodulated L-R audio is
> > lower in level than the L+R audio after it is multiplied by the 38 kHz
> > switching function. The math to show this is relatively simple.
>
> Let me see if I have it straight.
> If you modulate just one channel with a 1 kHz sine wave, 1Vrms, then the
the L+R
> audio wave from the ratio detector
> has an amplitude equal to the amplitude of the L-R modulation
> contained by the double sideband signal, ie, the envelope modulation
shape has an
> amplitude of 1Vrms.
>
> The composite signal when viewed will then have an amplitude of 2Vrms,
consisting
> of the +ve 1/2 1kHz waves filled with 38kHz waves, between its arch and
0V, and
> then followed by
> the -ve 1/2 waves with 38kHz filling between the 0V line and the arch of
the sine
> wave.

Yes, if I followed your explanation correctly the signal is as you have
described it. To correctly demodulate this signal without an auxiliary
matrix or "compensation" capacitor to boost the sub carrier gain, it is
necessary to use an impulse sampling function, where as with your high
amplitude sine wave you are effectively using a square wave sampling
function, which results in the L+R component of the signal having a higher
amplitude than the demodulated L-R component, hence poor separation.

> There is a strong tendency for this signal to try to loose the amplitude
> relationships between DSB quantity and L+R audio quantity.

It is more than just some vague "strong tendency", the lower amplitude of
the recovered DSB quantity is a direct result of the way the FM stereo
signal is mathematically defined.

> So if the 38kHz wave is accurately applied to the composite,
> the resulting mainly 38khz wave form will have the L signal on the top
> of it, and R signal on the bottom of it.

Yes, one channel will appear on the upper peaks of the 38 kHz carrier,
while the other will appear on the lower peaks, but that doesn't imply
that you can completely separate the two channels with a square wave
switching function of the type you are using, for that you need an impulse
type sampler. When you multiply the composite signal by a 38 kHz
switching function as you are, the L+R signal is attenuated by a factor of
1/2, while the L-R signal demodulated from the 38 kHz side bands is
attenuated by a factor of 1/PI. This is a difference in level of 3.9 dB
and is suspiciously close to the approximately 3 dB compensation you say
your "compensation" capacitor introduces.

> Simple diode detection could be used to retrieve the signal but the use of the
> matrix
> and balancing results in a staircase stepped wave at the caps off the
diodes, not a
> saw tooth
> "ripple voltage" like one sees in a power supply, or diode & RC detector
> in an AM set.
> Filtering the steps out is easier than filtering out 38kHz of ripple.

It sounds to me like "ripple" is just another word for "steps"?

> > 2.) The BA1404 signal generator may be generating a nonstandard stereo
> > signal that helps to compensate for separation problems in your stereo
> > decoder.
>
> Nope, after years of testing other tuners, the BA1404 provides a signal
to them all
> which
> could be separated very well.

I am sure you are correct here, and that the BA1404 includes a
compensation circuit although the data *** doesn't mention it. I think
I was confused about the conclusion I had drawn about the BA1404 many
years ago, thinking about it again I think the problem was that the BA1404
puts out a square wave 38 kHz DSB signal, which if feed through a very
wide band tuner, or directly into a mpx decoder without a 53 kHz low pass
filter, will impact the separation because there will be L-R sidebands
around the third harmonic of 38 kHz which if not filtered out at some
point will be demodulated by a square wave decoder increasing the level of
the demodulated L-R signal with a consequent impact on the separation.
This may partly explain why you only required 3 dB of compensation rather
than theoretical 3.9 dB, or even more when potential high frequency losses
in a real circuit are considered.

> > What you are describing is the design used in Scott's early decoders,
> > check this link for a later Scott tube decoder design:
> >
> > http://hhscott.com/pdf/345.pdf
>
> My god, 4.6Mb.
> Can't you reduce the file size of stuff like this?

It wasn't me, I just provided the link.

> Its also a little hard to follow, but it seems little different in principles
> to what have been used commonly except the 38Hz seems to be
> created by amplifiying the rectified 19kHz, rather than have a 38kHz which is
> then syncronised. Either will work fine, especially in good signal areas.
>
> The locked oscillator is better imho, because
> any amplitude changes in the 19khz, or audio spuriae
> at 19kHz will not alter the amplitude of the 38khz so badly.
> Its very important for fidelity that the 38khz has a steady amplitude.

I think if you look again you will see that there is positive feedback
from the plate circuit to the grid, making it an oscillator, unless what
you are saying is that there isn't enough positive feedback to make it an
oscillator, and the positive feedback is only there to increase the gain?

Notice also that in the later Scott designs like this one the out of phase
"compensation" signal for the auxiliary matrix is taken from the "bottom"
side of a floating ratio detector.

> The front end of my set is ex-trio with 2 x 6AQ8, 2 x 6BA6(IF), 6AU6 limiter,
> and the rest are in the MPX.
>
> It could have another limiter; some FM sets did have two limiters,

Doesn't your set use a ratio detector? The 6AU6 plus the ratio detector
make a total of two limiters.


Regards,

John Byrns


Surf my web pages at, http://users.rcn.com/jbyrns/
.