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Newsgroups: net.video,net.analog
Subject: How does Beta HiFi audio avoid head switching transients
Message-ID: <116@hydra.UUCP>
Date: Sun, 10-Mar-85 00:00:52 EST
Article-I.D.: hydra.116
Posted: Sun Mar 10 00:00:52 1985
Date-Received: Tue, 12-Mar-85 20:29:32 EST
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	Can anybody explain to me how the Beta HiFi system avoids
corrupting the stereo hi fi sound signals when it switches from
one spinning video head to the other (which should produce some sort
of phase and amplitude transient every 60th of a second in the FM sound
subcarriers). ?

	This transient is invisible in the video because the switch
between heads is in the vertical blanking interval where the machine
generates sync anyway.  Audio, on the other hand, is continuous
and the splice cannot be so easily hidden.

	I suppose the machine might be so designed that the heads contact
the tape for more than 180 degrees of rotation of the drum, ensuring that
during the splice interval when a switch is made from one head to the
other that both heads are solidly in contact with the tape and producing
high quality signals.  Further I suppose that a separate preamp and
sound carrier limiter and detector might be used for each channel
and each head (4 of them) with switching being done between discriminator
outputs rather than FM - RF from the heads as is done with video. 
The problem is, how do you ever get discriminators to track each other
that closely ?  By putting them all on a chip ? (Isn't it a matter of
external components tracking too ?)

	Another scheme is to have separate limiters and RF preamps
as in the first scheme but supply some sort of phase corrector
that would ensure during the splice overlap period that the phase
(and preferably the absolute timing) of the sound subcarriers coming
from both heads was synchronized before a switch was made between sound
carrier source going into the detector.  Such a scheme might be
quite simple if all it had to do was correct phase, correcting absolute
timing would involve some sort of variable delay line, however,
which might be more expensive.

	A third scheme might be to use some sort of blanking and
interpolation that would simply eat the transient by substituting
the average of values before and after it. This might be done
easily with pulse counting detectors.

	Perhaps I have not estimated the magnitude of the switching
transient correctly, but I hear awfully impressive signal to
noise and distortion figures cited for Beta HiFi.

        In any case, short of correcting absolute timing (and aligning
carrier phase) it would seem to me that there would be a very slight
but still measureable phase modulation of audio spectral components
at a 60 hz rate, particularly when playing back tapes recorded
on other machines that might have slightly different head to
head delay.
	
        Can anyone comment ?