George Herold wrote in news:a02bdb85-1047-4ba3-aaec-
:

Your band width is maybe
100kHz? (Do you have any filtering before the A-D)

Sorry, I didn't remember to directly answer that... They just have a
differential input stage that adds the two differential signals (doubles)
then quarters the output in the gain of the second of two amps they do this
with, so halving the total signal. After that it goes through a digitally
controlled analog variable resistor (CS3310-KS) followed by a DC blocking
capacitor and then an op-amp stage to make a differential signal for the ADC
(CS5335) with a 2.2nF cap with two 150R resistors to remove anything above a
few hundred KHz.

Apart from using that variable resistor IC they kept it as simple as they
could. Total dynamic range through the system is 98 dB, so only a tad more
than 16 bits worth, but aiming for better in my DC coupler board means I can
adapt other devices reliably.

On Jul 13, 11:26*am, Lostgallifreyan wrote:
George Herold wrote in news:a02bdb85-1047-4ba3-aaec-
:

Your band width is maybe
100kHz? *(Do you have any filtering before the A-D)

Sorry, I didn't remember to directly answer that... They just have a
differential input stage that adds the two differential signals (doubles)
then quarters the output in the gain of the second of two amps they do this
with, so halving the total signal. After that it goes through a digitally
controlled analog variable resistor (CS3310-KS) followed by a DC blocking
capacitor and then an op-amp stage to make a differential signal for the ADC
(CS5335) with a 2.2nF cap with two 150R resistors to remove anything above a
few hundred KHz.

300 ohms inline with the signal and 2.2nF to ground? Something like
240kHz if I punched the numbers correctly. You could scale the above
noise numbers up a factor of sqrt (2.4). If you want to be real fussy
there is something called the equivalent noise bandwidth. (ENBW)
(Your low pass filter is not a brick wall and higher frequencies add
more noise.) For a simple RC the ENBW is 3.1415/2 times bigger than
the 3dB corner frequency.


Apart from using that variable resistor IC they kept it as simple as they
could. Total dynamic range through the system is 98 dB, so only a tad more
than 16 bits worth, but aiming for better in my DC coupler board means I can
adapt other devices reliably.

OK that's much less of a constraint then. Sounds like you've got more
than 100uV of 'noise' in the system already.

George H.

George Herold wrote in news:a02bdb85-1047-4ba3-aaec-
:

What about the Johnson noise of your 100 kohm resistor?
(forget the 1Meg!) it's got 40 nV/rtHz. or about 14uV of noise...
that's starting to have an impact...

I decided to reduce to 10K. It means I have to stiffen the bias voltage
I'm making for it with a voltage follower but it's easier that way,
especially given that it might have up to ten inputs tapping it.

Lostgallifreyan wrote:

George Herold wrote in news:a02bdb85-1047-4ba3-aaec-
:


What about the Johnson noise of your 100 kohm resistor?
(forget the 1Meg!) it's got 40 nV/rtHz. or about 14uV of noise...
that's starting to have an impact...


I decided to reduce to 10K. It means I have to stiffen the bias voltage
I'm making for it with a voltage follower but it's easier that way,
especially given that it might have up to ten inputs tapping it.
There's that Johnson guy again! Why not Steve, Larry or Moo?

Lostgallifreyan writes:

I'm considering an op-amp for making a DC coupling adapter to a soundcard to
convert it to signal logging purposes while retaining its audio performance.
It uses a passive adder and a gain of 2 to add a bias voltage to the signal
before an ADC input.

The sound card is one with external analog circuitry in a rack unit, it has
20 bit signal conversion, so this op-amp will have to be good to maintain
that and the other specs this unit has.

Audio ADCs usually have bad DC specifications - why wouldn't they. You
may want to verify this before you try too hard to find the perfect
opamp.

[...]


--

John Devereux

John Devereux wrote in
:

Lostgallifreyan writes:

I'm considering an op-amp for making a DC coupling adapter to a
soundcard to convert it to signal logging purposes while retaining its
audio performance. It uses a passive adder and a gain of 2 to add a
bias voltage to the signal before an ADC input.

The sound card is one with external analog circuitry in a rack unit, it
has 20 bit signal conversion, so this op-amp will have to be good to
maintain that and the other specs this unit has.

Audio ADCs usually have bad DC specifications - why wouldn't they. You
may want to verify this before you try too hard to find the perfect
opamp.

[...]



Possibly so, but I have already done somethign similar with a DAC and found
it be be excellent at outputting any abitrary DC or slowly changing voltage
and holding it where it's meant to be, and the ADC's and DAC's in this
equipment were chosen from the same maker. I know I'm making an assumption
that it will work, but I do think it will. Besides, I have to make some
decent effort to get this op-amp decision right, partly to help me learn, and
not least, to make sure than when I do test that assumption, it isn't my
choice of ap-amp or adapter design method that is screwing it up.

"Lostgallifreyan" wrote in message
. ..
I'm considering an op-amp for making a DC coupling adapter
to a soundcard to
convert it to signal logging purposes while retaining its
audio performance.
It uses a passive adder and a gain of 2 to add a bias
voltage to the signal
before an ADC input.

The sound card is one with external analog circuitry in a
rack unit, it has
20 bit signal conversion, so this op-amp will have to be
good to maintain
that and the other specs this unit has.

I looked first at a few audio amps and noticed that their
claims for CMRR and
open-loop gain often fall well short of the claims made
for the equipment
they go into, but never mind, that's another issue for
another day..

Then I looked at a DC instrumentation amp (OPA2277) I'm
using in a laser
power meter design. If I can use it, it saves me buying
varieties of
expensive chips in small quantities. Audio boffs high,
wide and plentiful
will say don't do it, slew rate is slow, etc, but is it??
0.8V/µS. It doesn't
sound a lot when people are saying I need 16V/µS or
whatever, but I
calculated it, and it looks fine to me. The sound unit I'm
adapting to is
considerably better than CD quality, sampling with 20 bits
at up to 48 KHz,
and I calculated that this means a sample at intervals of
a tad over 20 µS.
As 20 µS of 0.8V/µS is 16V, and as the device I'm adapting
to has a ±15V
supply and a differential input design that halves the
input, the largest
possible voltage change will occur, and fully settle, in
the time between
samples at highest sample rate available.

I'm struggling with this maths, perhaps because you've left
quite a bit out. If you're amplifying before any filtering,
why don't you need the same slew rate as the soundcard's
DAC? Take your worst case of a switch between max +ve and
max -ve from one sample to the next. That could happen with
a 24kHz sine input. Assume the DAC outputs a slightly
slew-rate
limited square wave, so that a filter can then be used to
extract a sine wave, at a certain amplitude relative to the
square wave. If your max slew rate is less than the DAC, so
the square tends towards a triangle, then the amplitude of
the sine wave extracted by the same filter may be reduced.

Looked at another way, a 15Vpk 24kHz sine wave has a maximum
slew rate of 2.26V/µS (?). In order for the filter to have
this output, what is the minimum slew rate required at its
input? Is it the same? Could be there's some law that
everyone knows but me. Anyway, in the limiting case of your
argument, where the time to "fully settle" is zero, you
would have a triangle wave, and the fundamental sine would
be considerably attenuated.

I also wonder if it's OK to actively use the slew rate limit
like this? Are there no penalties, like recovery time or
power dissipation? Presumably one stage within the opamp
consumes max current whenever it is in the process if
limiting slew rate.

All in all, this seems close enough so you need to make
sure, by going through each stage of the process from input
to filtered output, to make sure your design gets from one
to the other as well as the original.

Maybe the "10-1" rule of thumb applies somehow...works for
impedance.

Ian

As all the other figures for dynamic range and noise are
so good that they
will allow the original specs for the entire unit to
remain intact, is there
any reason I should not use this op-amp? It's a lot
cheaper than any audio
amp that looks like it will do as well as this. And as I'm
after DC as well
as AC capability, it seems that this is the right
decision, but I'm
interested in other views before I decide anything. (I
could just use
sockets, but for a low profile board I'll be soldering it
in, and don't want
to have to mess with that later.

Although I got your idea exactly backwards, AFAICS now, it's
still true that your slew rate argument is bonkers, as far
as it goes. More or less for the same reason, exactly
backwards.

You seem to argue that, if the opamp is able to slew
full-scale between samples of the following audio ADC, then
the opamp's max slew rate must be adequate for audio input.
Why should this be true?

Perhaps someone could point me in the direction of
enlightenment?

Ian


"Ian Iveson" wrote in
message news:kj6%n.3277$xf1.2298@hurricane...

"Lostgallifreyan" wrote in message
. ..
I'm considering an op-amp for making a DC coupling
adapter
to a soundcard to
convert it to signal logging purposes while retaining its
audio performance.
It uses a passive adder and a gain of 2 to add a bias
voltage to the signal
before an ADC input.

The sound card is one with external analog circuitry in a
rack unit, it has
20 bit signal conversion, so this op-amp will have to be
good to maintain
that and the other specs this unit has.

I looked first at a few audio amps and noticed that their
claims for CMRR and
open-loop gain often fall well short of the claims made
for the equipment
they go into, but never mind, that's another issue for
another day..

Then I looked at a DC instrumentation amp (OPA2277) I'm
using in a laser
power meter design. If I can use it, it saves me buying
varieties of
expensive chips in small quantities. Audio boffs high,
wide and plentiful
will say don't do it, slew rate is slow, etc, but is it??
0.8V/µS. It doesn't
sound a lot when people are saying I need 16V/µS or
whatever, but I
calculated it, and it looks fine to me. The sound unit
I'm
adapting to is
considerably better than CD quality, sampling with 20
bits
at up to 48 KHz,
and I calculated that this means a sample at intervals of
a tad over 20 µS.
As 20 µS of 0.8V/µS is 16V, and as the device I'm
adapting
to has a ±15V
supply and a differential input design that halves the
input, the largest
possible voltage change will occur, and fully settle, in
the time between
samples at highest sample rate available.

I'm struggling with this maths, perhaps because you've
left
quite a bit out. If you're amplifying before any
filtering,
why don't you need the same slew rate as the soundcard's
DAC? Take your worst case of a switch between max +ve and
max -ve from one sample to the next. That could happen
with
a 24kHz sine input. Assume the DAC outputs a slightly
slew-rate
limited square wave, so that a filter can then be used to
extract a sine wave, at a certain amplitude relative to
the
square wave. If your max slew rate is less than the DAC,
so
the square tends towards a triangle, then the amplitude of
the sine wave extracted by the same filter may be reduced.

Looked at another way, a 15Vpk 24kHz sine wave has a
maximum
slew rate of 2.26V/µS (?). In order for the filter to have
this output, what is the minimum slew rate required at its
input? Is it the same? Could be there's some law that
everyone knows but me. Anyway, in the limiting case of
your argument, where the time to "fully settle" is zero,
you would have a triangle wave, and the fundamental sine
would be considerably attenuated.

I also wonder if it's OK to actively use the slew rate
limit like this? Are there no penalties, like recovery
time or power dissipation? Presumably one stage within the
opamp consumes max current whenever it is in the process
if limiting slew rate.

All in all, this seems close enough so you need to make
sure, by going through each stage of the process from
input to filtered output, to make sure your design gets
from one to the other as well as the original.

Maybe the "10-1" rule of thumb applies somehow...works for
impedance.

Ian

As all the other figures for dynamic range and noise are
so good that they
will allow the original specs for the entire unit to
remain intact, is there
any reason I should not use this op-amp? It's a lot
cheaper than any audio
amp that looks like it will do as well as this. And as
I'm
after DC as well
as AC capability, it seems that this is the right
decision, but I'm
interested in other views before I decide anything. (I
could just use
sockets, but for a low profile board I'll be soldering it
in, and don't want
to have to mess with that later.

Here's a followup to this, because it seems to work, so if anyone's still
interested, they might like to know how it went...

First, I noticed noise, about 270µV of it, so I put a LPF filter in the
reference voltage (10K and 10µF ceramic). I also changed the buffer amp from
LF412 to another of the OPA2277A's I'm using. (The other channel buffers a
negative voltage for when my board is to remove a DC offset instead of adding
one). I also had to desolder a pin on the ADC and bend it to a conveniently
grounded pin next to it, and solder it there to disable an onboard digital
HPF.

I now have DC coupling, with noise on an empty channel within 3dB of best
unmodified system performance, which is better than I'd hoped. (-78.3dB as
opposed to -80.8dB originally).

Most of the existing DC offset is in the rest of the original system, I know
this because I can see it change as the device warms up, with all external
signals being absent or constant.

The DC offset remaining is around 700 values on a scale of 32768 so I'm ok
with that, especially as Sound Forge makes a truly neat way to remove it
immediately prior to record. It's so good that when detecting laser power on
the meter all this is aimed at testing, I won't need to tweak its own offset,
I can just record the output and do that in Sound Forge, as well as any extra
filtering I might want.

My conclusion is that modifying a decent studio audio interface for data
logging at arbitrary sample rates from 2000 Hz to 96000 Hz is well worth
doing. One ideal unit is the Echo Layla24, often found on eBay for less than
£100 now. Given the bang per buck, I prefer this to any other method because
I can still use it as a viable multichannel audio I/O when I want to.

(Incidentally, DC coupling on those units is even easier, as they don't have
DC on either side of the DC blocking caps, so just put a wire link where
those are now, and get accurate voltage generation up to around ±13.5V, with
fast and accurate changes, from wave file players or other software... All
kinds of uses for that, no doubt).

One last point: I can get decent audio band through an OPA2277 despite the
modest slew rate, but there are limits. Full scale differential input is
possible for sample rates up to 48 KHz, but for 96 KHz only non-balanced
input will allow this cleanly, so if the input is differential on a system
with a ±V supply, attenuate the signal by 6dB, or choose a faster low noise
amp. The low offset might not seem so important now, but the low noise and
drift still are.

Lostgallifreyan wrote in
:

so if the input is differential on a system
with a ±V supply, attenuate the signal by 6dB, or choose a faster low
noise amp.

Correction: 'with a ±15v supply'...