November 29th 05, 10:09 AM
posted to uk.rec.audio
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Got to laugh
On Tue, 29 Nov 2005 10:08:49 -0000, "Keith G"
wrote:
"Don Pearce" wrote
(The world runs on bull**** and two things that not many people want are
the
absolute truth and absolute freedom...!! ;-)
The best of all are the magic cleaning products on the shopping
channels. I just don't know how they are *still* getting away with it.
Having got a letter from my dentist this morning telling me to take a hike
because I won't sign up to their ****ing 'Denplan' scam I'm in the mood to
say the worst aspect of the Thatcherite legacy is the 'proliferation' of
scams and rip-offs that seem almost to get 'official approval' by dint of
the *absolute lack* of any sort of crackdown on them simply becausse they
are 'businesses'....!!! (ie 'entrepreneurial'.....)
Changing the subject a tad - what happens if I swap a 100K log volume pot on
a valve amp for a 50K one - wheels fall off or summat?
How much have you paid for the 50k? If it is an expensive one, you
will get better rhythm and deeper sound staging - sorry, couldn't
resist.
Just go for it. The tolerance on the average pot is pretty huge, and
although there will be a slightly increased loading on the previous
stage, it won't be as much as the change you get as you wind the
slider along the track. I wouldn't go lower than 50k, though.
If you find that you never wind the pot up past about ten o,clock, you
could always add a 50k resistor in series with the top (live) end of
it to preserve the full impedance load. You just turn the wick up a
bit to compensate.
d
Pearce Consulting
http://www.pearce.uk.com
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November 29th 05, 10:19 AM
posted to uk.rec.audio
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Got to laugh
"Don Pearce" wrote
Changing the subject a tad - what happens if I swap a 100K log volume pot
on
a valve amp for a 50K one - wheels fall off or summat?
How much have you paid for the 50k? If it is an expensive one, you
will get better rhythm and deeper sound staging - sorry, couldn't
resist.
It's an Alps, so I expect a shimmering treble, deep, tight bass with
sparkling transients and a visceral sense of 'being there', naturally....
Just go for it.
OK. Thought so - I've used it externally, so I think I know what to expect
anyway.
The tolerance on the average pot is pretty huge, and
although there will be a slightly increased loading on the previous
stage, it won't be as much as the change you get as you wind the
slider along the track. I wouldn't go lower than 50k, though.
OK.
If you find that you never wind the pot up past about ten o,clock, you
could always add a 50k resistor in series with the top (live) end of
it to preserve the full impedance load. You just turn the wick up a
bit to compensate.
OK, another dumb question - is two 100K resistors in parallel the same as
50K? (Fuctiff I know - I only got 100Ks...)
Also, if I said sod it and wanted to take the pot out of the loop entirely
(making the amp 'full power' only and continue with external attenuation),
do I simply connect the wires together (as it were) or would you connect
them via a 100K resistor?
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November 29th 05, 10:22 AM
posted to uk.rec.audio
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Got to laugh
On Tue, 29 Nov 2005 11:19:36 -0000, "Keith G"
wrote:
OK, another dumb question - is two 100K resistors in parallel the same as
50K? (Fuctiff I know - I only got 100Ks...)
Yup - two hundreds in parallel makes fifty.
Also, if I said sod it and wanted to take the pot out of the loop entirely
(making the amp 'full power' only and continue with external attenuation),
do I simply connect the wires together (as it were) or would you connect
them via a 100K resistor?
Just join the wires - no resistors. That is what happens at full
volume setting.
d
Pearce Consulting
http://www.pearce.uk.com
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November 29th 05, 11:40 AM
posted to uk.rec.audio
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Got to laugh
In article ,
Keith G wrote:
Yup - two hundreds in parallel makes fifty.
OK, thought so but wasn't *sure*...??
(I'm at that stage where yesterday's *knowledge* is today's hazy
recollection...!! :-)
Resistors in series. R = R1 + R2
R1 x R2
Resistors in parallel. R = -------
R1 + R2
--
Dave Plowman London SW
To e-mail, change noise into sound.
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November 29th 05, 11:50 AM
posted to uk.rec.audio
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Got to laugh
"Dave Plowman (News)" wrote in message
...
In article ,
Keith G wrote:
Yup - two hundreds in parallel makes fifty.
OK, thought so but wasn't *sure*...??
(I'm at that stage where yesterday's *knowledge* is today's hazy
recollection...!! :-)
Resistors in series. R = R1 + R2
R1 x R2
Resistors in parallel. R = -------
R1 + R2
--
Nice one Plowie - beats clonking through this little lot:
http://www.play-hookey.com/dc_theory...resistors.html
:-)
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November 29th 05, 12:13 PM
posted to uk.rec.audio
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Got to laugh
On Tue, 29 Nov 2005 09:27:36 +0000 (GMT), Jim Lesurf
wrote:
In article , Don Pearce
wrote:
On Mon, 28 Nov 2005 09:13:40 +0000 (GMT), Jim Lesurf
wrote:
If someone had employed me to design a DAC with the requirement to use
the clock embedded in the SPDIF stream, and omit a LO then I'd probably
have used the approach I outlined.[1] However I have no idea which DACs
use what methods in general as my experience is only with the Meridian
ones, and a lot of what appears in reviews sheds no reliable light on
such matters.
And unless instructed to make a DAC in the above way, I'd simply use a
LO, PLL, and buffer... :-)
That would certainly seem to be the obvious - not to mention easy - way
to do it. Using the SPDIF stream as a clock would really be quite hard.
Not really very 'hard' in itself, but it would be difficult to get the
results to have a smooth clock. The SPDIF essentially has the clock XORd in
it already, so this can be extracted using something like a rectifier and
filter. That gives a bit-rate clock which you then use just as you would an
LO - which you'd be likely to have locked to the SPDIF embedded clock rate
anyway. The bit rearrangements and word assembly, etc, can then be handled
in the same ways regardless of how the DAC clock signal was produced.
I mean, you need some memory in which to perform error correction, and
the processor for that needs a clock.
Not sure what you mean by error correction here. I think that would have
been done prior to SPDIF transport.
Sorry - crossed connections here. The raw data has already been
through a processor to carry out error correction and interpolation.
By the time it gets as far as S/PDIF it has already been re-clocked -
which is where we came in.
d
Pearce Consulting
http://www.pearce.uk.com
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November 29th 05, 01:06 PM
posted to uk.rec.audio
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Got to laugh
"Keith G" wrote in message
"Dave Plowman (News)" wrote in
message ...
In article ,
Keith G wrote:
I wouldn't expect anything less - would you?
How many people keep packaging for small low value
items?
Anybody with half a brain buying them with a view to
giving them a *trial* and hoping/expecting to get their
money back, I would have thought - no...??
People who buy fancy cables don't with a view to giving
them a trial. They've already decided they will be an
improvment. Hence, of course, they are.
If I could only be as certain of things as you seem to
be....
Actually, this is all about dueling perceptions of
certainty. One is based on ignorance, and one is based on a
fairly detailed and correct understanding of the relevant
technology. I'll leave it to the reader to figure out which
is which! ;-)
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