Sign in with
Sign up | Sign in
Your question

Looking for schematic to build a -10db to +4dbu converter

Last response: in Home Audio
Share
Anonymous
August 3, 2004 1:06:24 PM

Archived from groups: rec.audio.pro (More info?)

I need to build 8 channel of this. Wondering if someone could email me
a schematic. I found one on the net describing the circuit from an old
Electronic Musician, but no schematic. I have some nice Linear
Technologies opamps or Burr Browns on hand for the circuit. Just need
unbalanced in to balanced out. I want to take the tape returns of my
Tascam 388 into my AD converter. Have tons of old files to translate.
So I want to build a really high quality little box instead of paying
a couple of hundred.

Thank you,
Chris
Anonymous
August 3, 2004 4:12:41 PM

Archived from groups: rec.audio.pro (More info?)

The Analog Devices SSM2142 is a super clean, super quiet, super SIMPLE balanced line driver
that is great for doing unbal to bal conversion. If you have trouble finding them, I think I
have at least 8 on hand that I got cheap. All you need is a clean +/-12 to 15v split power
supply & a tiny handful of passive components.

--
Stephen Sank, Owner & Ribbon Mic Restorer
Talking Dog Transducer Company
http://stephensank.com
5517 Carmelita Drive N.E.
Albuquerque, New Mexico [87111]
505-332-0336
Auth. Nakamichi & McIntosh servicer
Payments preferred through Paypal.com
"Christian Serig" <cserig@artic.edu> wrote in message
news:2f4d4600.0408030806.43b78fd5@posting.google.com...
> I need to build 8 channel of this. Wondering if someone could email me
> a schematic. I found one on the net describing the circuit from an old
> Electronic Musician, but no schematic. I have some nice Linear
> Technologies opamps or Burr Browns on hand for the circuit. Just need
> unbalanced in to balanced out. I want to take the tape returns of my
> Tascam 388 into my AD converter. Have tons of old files to translate.
> So I want to build a really high quality little box instead of paying
> a couple of hundred.
>
> Thank you,
> Chris
Anonymous
August 3, 2004 8:04:00 PM

Archived from groups: rec.audio.pro (More info?)

It would appear, then, that the time it takes to build such
a box is not a consideration.

Remember the old, tired, worn-out saw:

"Cheap, quick, good. Pick two."

Or you could look for a Fostex 5030 ($75 - $150 on eBay) and
spend the time actually transferring the files and getting
something accomplished. But, of course, there is the
satisfaction of building the box...



TM

Christian Serig wrote:
>
> I need to build 8 channel of this. <snip> I want to take the tape returns of my
> Tascam 388 into my AD converter. Have tons of old files to translate.
> So I want to build a really high quality little box instead of paying
> a couple of hundred.
Related resources
Anonymous
August 3, 2004 9:55:52 PM

Archived from groups: rec.audio.pro (More info?)

"Christian Serig" <cserig@artic.edu> wrote in message
news:2f4d4600.0408030806.43b78fd5@posting.google.com...
> I need to build 8 channel of this. Wondering if someone could email me
> a schematic. I found one on the net describing the circuit from an old
> Electronic Musician, but no schematic. I have some nice Linear
> Technologies opamps or Burr Browns on hand for the circuit. Just need
> unbalanced in to balanced out. I want to take the tape returns of my
> Tascam 388 into my AD converter. Have tons of old files to translate.
> So I want to build a really high quality little box instead of paying
> a couple of hundred.

If you're planning on building this yourself with opamps, the circuit is
trivial but you'll discover that the PCB layout is critical to achieving low
distortion. In particular, make sure that the grounds of your feedback
networks (e.g., the noninverting pin of the opamp if you use an inverting
topology) are connected directly to the point where the power supply ground
comes in, rather than sharing those traces with any other current.

Assuming the devices are physically close, there is not much point in trying
to create a voltage-balanced output. You're going to add as much noise and
inaccuracy (by adding a second opamp) as you gain in extra headroom; and
you're not concerned, in this application, with crosstalk to other channels
or with power loss. So, what you really care about is getting balanced
*impedance*, so that the A/D's input stage can do its noise rejection thing.
Look at the Rane white papers; your simplest and cleanest solution will be
to connect the cold leg of the output to ground, through an R-C network that
is identical to that on the hot leg.

In any event, be aware that -10dBV and +4dBu are different standards;
they're not 14dB apart, so you don't need 14dB of gain. dBV is relative to
one volt; dBu is relative to "the voltage that produces 1mW into 600 ohms",
which is 0.7746V. So, -10dBV = 0.316V; +4dBu = 1.228V; the ratio is 3.882,
or 11.8dB. In other words, your amplifier needs a gain of 3.882.

Another possibility, producing excellent quality for somewhere around $150,
would be to pick up a pair of the appropriate Jensen transformers. See
their web site for examples and schematics.

Are you sure you need this? Does your A/D not have sufficient input
sensitivity to work with the -10dBV, or does the 388 not have sufficient
output drive? Are you having problems with hum or noise if you simply
connect the devices together with appropriate cable?
Anonymous
August 3, 2004 10:41:35 PM

Archived from groups: rec.audio.pro (More info?)

In article <ceokhv$2bc$1@reader2.nmix.net> bk11@thuntek.net writes:

> The Analog Devices SSM2142 is a super clean, super quiet, super SIMPLE balanced
> line driver
> that is great for doing unbal to bal conversion.

He'll also need some gain. Can the SSM2142 do that too, or will he
need another stage ahead of it? And since he wants nominal +4 dBu out
with enough headroom to drive his converter to distraction, he'll
definitely need a +/- 15V supply, or 18V if the chips will take it.

Chips, power supply, handful of components, connectors, and chassis
and it's going to be a pretty expensive and extensive project for a
beginner who's still at the level of asking for a schematic (rather
than suggestions for a design/build approach).

To Christian:

What's your A/D converter, and does it have enough gain to work with a
-10 dBV input? If you're worried about just balancing an unbalanced
source, forget it. If you have a problem with noise pickup, you'll
just move the problem from the input of the A/D converter to the input
of your converter box. If your project is really worth the expense and
effort, then go for it. Otherwise, try to work with what you have.






If you have trouble finding
> them, I think I
> have at least 8 on hand that I got cheap. All you need is a clean +/-12 to 15v
> split power
> supply & a tiny handful of passive components.
>
> --
> Stephen Sank, Owner & Ribbon Mic Restorer
> Talking Dog Transducer Company
> http://stephensank.com
> 5517 Carmelita Drive N.E.
> Albuquerque, New Mexico [87111]
> 505-332-0336
> Auth. Nakamichi & McIntosh servicer
> Payments preferred through Paypal.com
> "Christian Serig" <cserig@artic.edu> wrote in message
> news:2f4d4600.0408030806.43b78fd5@posting.google.com...
> > I need to build 8 channel of this. Wondering if someone could email me
> > a schematic. I found one on the net describing the circuit from an old
> > Electronic Musician, but no schematic. I have some nice Linear
> > Technologies opamps or Burr Browns on hand for the circuit. Just need
> > unbalanced in to balanced out. I want to take the tape returns of my
> > Tascam 388 into my AD converter. Have tons of old files to translate.
> > So I want to build a really high quality little box instead of paying
> > a couple of hundred.
> >
> > Thank you,
> > Chris
>
>

--
I'm really Mike Rivers (mrivers@d-and-d.com)
However, until the spam goes away or Hell freezes over,
lots of IP addresses are blocked from this system. If
you e-mail me and it bounces, use your secret decoder ring
and reach me here: double-m-eleven-double-zero at yahoo
Anonymous
August 3, 2004 10:52:14 PM

Archived from groups: rec.audio.pro (More info?)

"Walter Harley" <walterh@cafewalterNOSPAM.com> wrote in message
news:ceojj8$p0$0

> Another possibility, producing excellent quality for somewhere around
$150,
> would be to pick up a pair of the appropriate Jensen transformers. See
> their web site for examples and schematics.

That could be problematical in two ways. First, the OP needs 8 channels, so
that's more like $600. Second, since the impedance ratio is proportional to
the square of the voltage/turns ratio, a typical 10k input impedance on the
A/D will look like 10,000 / 3.882^2 at the input of the transformer, or
about 664 ohms. This is kinda low for gear like the 388, which is not likely
to have a hefty output driver.

Stephen's suggestion of the SSM2142 is probably more practical.

Peace,
Paul
Anonymous
August 4, 2004 4:07:22 AM

Archived from groups: rec.audio.pro (More info?)

"Paul Stamler" <pstamlerhell@pobox.com> wrote in message
news:y9RPc.169767$OB3.5584@bgtnsc05-news.ops.worldnet.att.net...
> That could be problematical in two ways. First, the OP needs 8 channels,
so
> that's more like $600. Second, since the impedance ratio is proportional
to
> the square of the voltage/turns ratio, a typical 10k input impedance on
the
> A/D will look like 10,000 / 3.882^2 at the input of the transformer, or
> about 664 ohms. This is kinda low for gear like the 388, which is not
likely
> to have a hefty output driver.

Whoops, sorry, wasn't paying close enough attention to the OP's gear. My
apologies.
Anonymous
August 4, 2004 4:17:05 PM

Archived from groups: rec.audio.pro (More info?)

The levels coming off to tape are good to bad. These are recording's
going back to when I started recording so some levels are low, before
I started hitting tape hard. The AD converter in question is a
Panasonic WZ-AD96. I tried cranking the gains on the converter and it
did not do much to anything pleasable. I could just use the line in's
on my board (Ward Beck) but that still is not completely up to par.
Still recapping and I want to preserve these recordings to the best
possible within reasonable means.

Would you all recommend me just taking these outputs unbalanced into 4
channels of Symetrix 202 and 4 channels of Sytek to boost the gains? I
have those on hand. I don't mind the trouble of a box like this
because it always could be used at a later date.

Also I probably will not build it. I will hook up all the wires to
the jacks, but a buddy of mine who I do alot of work with just spends
his days now learning electronics working up new modules for his
modular synth. He would do the actual work. I just thought I would
help him with a schematic.
Anonymous
August 4, 2004 10:41:30 PM

Archived from groups: rec.audio.pro (More info?)

In article <2f4d4600.0408041117.6d3e7957@posting.google.com> cserig@artic.edu writes:

> The levels coming off to tape are good to bad. These are recording's
> going back to when I started recording so some levels are low, before
> I started hitting tape hard.

OK, another unknown (unless I forgot). What tape deck? What's it's
nominal output level? Is it properly aligned so that all the channels
are putting out the level they're supposed to? Heads cleaned?

> The AD converter in question is a
> Panasonic WZ-AD96. I tried cranking the gains on the converter and it
> did not do much to anything pleasable.

Can you be more specific?

> Would you all recommend me just taking these outputs unbalanced into 4
> channels of Symetrix 202 and 4 channels of Sytek to boost the gains?

No, not really. Those are mic preamps. The input may not be a good
match for the recorder's output and you might end up just about where
you started only with more noise and more distortion.

What peak and "eyeball average" levels are you seeing on the
converter's meters with the tape playing? You may be trying to hit
them too hard.

--
I'm really Mike Rivers (mrivers@d-and-d.com)
However, until the spam goes away or Hell freezes over,
lots of IP addresses are blocked from this system. If
you e-mail me and it bounces, use your secret decoder ring
and reach me here: double-m-eleven-double-zero at yahoo
Anonymous
August 4, 2004 10:56:21 PM

Archived from groups: rec.audio.pro (More info?)

"Mike Rivers" <mrivers@d-and-d.com> wrote in message
news:znr1091565458k@trad...
> He'll also need some gain. Can the SSM2142 do that too, or will he
> need another stage ahead of it? And since he wants nominal +4 dBu out
> with enough headroom to drive his converter to distraction, he'll
> definitely need a +/- 15V supply, or 18V if the chips will take it.

This has me puzzled, why does he need to drive the converter into clipping?
One nice thing about digital is that you know exactly how much level you
need, and extra headroom is unnecessary. You certainly don't need +/- 18V
supplies to get +4dBu cleanly!

TonyP.
Anonymous
August 4, 2004 10:56:22 PM

Archived from groups: rec.audio.pro (More info?)

In article <4110a4cc$0$30773$afc38c87@news.optusnet.com.au> TonyP@optus.net.com.au writes:

> > with enough headroom to drive his converter to distraction, he'll
> > definitely need a +/- 15V supply, or 18V if the chips will take it.
>
> This has me puzzled, why does he need to drive the converter into clipping?

Notice the word "headroom." You don't drive the converter into
clipping, you have enough headroom so that you can, and then you
don't. Since (unless he's going to build peak-reading meters into it)
there's no indication that the analog circuitry is clipping, you
design it so that it absolutely won't clip until it's gone beyone the
level that will clip the A/D converter - for which you almost always
have an indication.

> One nice thing about digital is that you know exactly how much level you
> need, and extra headroom is unnecessary. You certainly don't need +/- 18V
> supplies to get +4dBu cleanly!

No, but you need it to do +26 dBu cleanly. A "+4" converter that's
calibrated so that that level represents -20 dBFS requires +24 dBu in
order to reach full scale. Of course you might have a converter that
calibrates +4 dBu in to equaly as high a digital level as -10 dBFS and
for that, you'd only need a maximum analog output of +14 dBu to reach
full scale.

Since the calibration of an A/D converter is typically not adjustable
other than down by attenuating the input or up in a jump by setting it
to -10 dBV nominal if that's possible, you don't want to design a
level converter that's only adequate for your present converter
because you might get a different converter (with a different
calibration point) some day.

Or maybe some time the industry will establish a standard relationship
between analog and digital levels so we'll all know that we're talking
about the same thing when interfacing. In the meantime, users will
just have to pay us system engineers a lot of money to figure it out
for them, or blunder through it.

Any AES standards groups working on this?



--
I'm really Mike Rivers (mrivers@d-and-d.com)
However, until the spam goes away or Hell freezes over,
lots of IP addresses are blocked from this system. If
you e-mail me and it bounces, use your secret decoder ring
and reach me here: double-m-eleven-double-zero at yahoo
Anonymous
August 5, 2004 4:45:45 AM

Archived from groups: rec.audio.pro (More info?)

"Christian Serig" <cserig@artic.edu> wrote in message
news:2f4d4600.0408041117.6d3e7957@posting.google.com...
> [...]
> Also I probably will not build it. I will hook up all the wires to
> the jacks, but a buddy of mine who I do alot of work with just spends
> his days now learning electronics working up new modules for his
> modular synth. He would do the actual work. I just thought I would
> help him with a schematic.

Check out http://www.henryeng.com/Support_doc/MATCHBOX%20II.doc - includes
the schematic for the Henry Engineering matchbox, which is as good as any.
Of course you'll just want the unbalanced-to-balanced section, not the other
part.
Anonymous
August 6, 2004 8:52:17 PM

Archived from groups: rec.audio.pro (More info?)

"Mike Rivers" <mrivers@d-and-d.com> wrote in message
news:znr1091624925k@trad...
> In article <4110a4cc$0$30773$afc38c87@news.optusnet.com.au>
TonyP@optus.net.com.au writes:
> > > with enough headroom to drive his converter to distraction, he'll
> > > definitely need a +/- 15V supply, or 18V if the chips will take it.
> > This has me puzzled, why does he need to drive the converter into
clipping?

> Notice the word "headroom." You don't drive the converter into
> clipping, you have enough headroom so that you can, and then you
> don't. Since (unless he's going to build peak-reading meters into it)
> there's no indication that the analog circuitry is clipping, you
> design it so that it absolutely won't clip until it's gone beyone the
> level that will clip the A/D converter - for which you almost always
> have an indication.

On this point I don't agree. If the input level required for 0dB FS is
fixed, you don't need *ANY* headroom. You simply require a clean analog
signal up to the 0dB FS point. Any more is simply wasted. If it makes you
feel better, fine. Metering anywhere in a fixed gain system will tell you
exactly the same thing, once you have established the lowest overload point
and don't exceed it. I wouldn't want the analog stage clipping first, but
extra headroom is just a feel good factor with digital.

> > One nice thing about digital is that you know exactly how much level you
> > need, and extra headroom is unnecessary. You certainly don't need +/-
18V
> > supplies to get +4dBu cleanly!

> No, but you need it to do +26 dBu cleanly. A "+4" converter that's
> calibrated so that that level represents -20 dBFS requires +24 dBu in
> order to reach full scale. Of course you might have a converter that
> calibrates +4 dBu in to equaly as high a digital level as -10 dBFS and
> for that, you'd only need a maximum analog output of +14 dBu to reach
> full scale.

OK, now you have a point, you must first establish exactly where the 0dB FS
level is. If it's +26dBu, then you will indeed lose resolution by running
too far below that to avoid analog overload if your pre-amp can't manage
that level. However you also lose resolution by running analog stages too
far below their maximum, EIN being constant. Personally I would never design
such a device. +14dBu FS, with 12 dB pads on the input when necessary, is
much more useful IMO. If we were struggling to get 100dB SNR at +14 then it
would be a different matter, but most manufacturers converters are well past
that.

> Since the calibration of an A/D converter is typically not adjustable
> other than down by attenuating the input or up in a jump by setting it
> to -10 dBV nominal if that's possible, you don't want to design a
> level converter that's only adequate for your present converter
> because you might get a different converter (with a different
> calibration point) some day.

I thought you were the one who mentioned that you buy what you need today,
because tomorrows equipment will probably be better and cheaper by the time
you need it.
If not it must have been me :-)

TonyP.
Anonymous
August 6, 2004 8:52:18 PM

Archived from groups: rec.audio.pro (More info?)

In article <41132ab8$0$10612$afc38c87@news.optusnet.com.au> TonyP@optus.net.com.au writes:

> On this point I don't agree. If the input level required for 0dB FS is
> fixed, you don't need *ANY* headroom. You simply require a clean analog
> signal up to the 0dB FS point. Any more is simply wasted.

That would be valid if practice agreed with theory, but it doesn't.
Most analog circuits approach "not clean" slowly. In order to be
really clean at the level that produces 0 dBFS, you need to be able to
go above that level by some amount (depending on how much distortion
you're willing to accept at high levels) before reaching clipping.
Clipping isn't the only "distortion" you need to worry about.

Also, subjectively, audio that's clipped for just a couple of samples
is undetectable, so you want to be able to send clean level to the
converter even at its clipping level.

> If it makes you
> feel better, fine. Metering anywhere in a fixed gain system will tell you
> exactly the same thing, once you have established the lowest overload point
> and don't exceed it. I wouldn't want the analog stage clipping first, but
> extra headroom is just a feel good factor with digital.

The metering tells you what's happening right now (or what just
happened now that it's too late to do anything about it). Again,
you're arguing theory and I'm talking practice. If everyone sang level
sine waves, your system would be OK. But meters differ in how they
deal with complex waveforms and with dynamics, so the meters don't
always tell you everything. You want to allow for some ambiguity of
the metering.

> OK, now you have a point, you must first establish exactly where the 0dB FS
> level is.

Exactly - and since there's no industry standard, you have to
determine this for every system if you want to be able to push things
to the limit. And once you know that, you need to choose a source that
can produce that level. And if you're talking about a mic preamp, it
may need more GAIN in order to produce that level. A singer produces a
given sound pressure level. Let's say that with the preamp cranked all
the way open it has 60 dB of gain, and that produces a peak level (a
function of the singer's volume, the distance from the mic, and the
mic's sensitivity - also not standardized) of +16 dBu. Let's assume
that the distortion is acceptable at that level.

Let's say this makes the digital meters on the A/D converter hit
-8 dBFS on peaks. If you're paranoid about losing resolution (maybe
you're working at 8-bits?) what do you have to do in order to make the
meters hit FS on peaks?

Well, you could raise the input sensitivity of the A/D converter, but
we've established that this is fixed in the hardware. So you can't do
that. You could tell the singer to sing louder or move closer to the
mic, or switch to a mic with higher sensitivity, but that might make
changees that aren't acceptable otherwise. You could turn up the
preamp gain, but it's already at maximum. So you need more gain.

> If it's +26dBu, then you will indeed lose resolution by running
> too far below that to avoid analog overload

How much resolution do you think you'll lose, and how do you think
that will affect what you hear? I hate to sound like Phil here, but
you really should do a listening test. There is so much "audible"
headroom with decent 24-bit converters that you can easily work in the
eyeball average range of -20 to -10 dBFS and it will sound just fine.

Another problem that can occur if you're mixing multiple channels all
firing away at full scale output. Converter pairs (such as in a sound
card or a "workstation" converter that has A/D and D/A in a single
box) are generally designed to be "unity gain" so that if it takes
+24 dBU to push the input to full scale, playing back a full scale
recording will produce +24 dBu at the output of the D/A converter. Try
to mix too many of those in an analog mixer and you'll have to reduce
the gain at the mixer. Try to mix too many of those in a digital mixer
(hardware or software) and you may become one of the "mixing in the
box sounds bad" camp.

> However you also lose resolution by running analog stages too
> far below their maximum, EIN being constant.

The problem with that argument is that EIN isn't constant. It's a
function of gain, which is why it's nearly given (on the spec sheet
anyway - the engineering lab is another story) at maximum gain.

Noise floor is more nearly constant, and you can indeed get down into
that noise if things are quiet enough. But given a modern system with
greater than 100 dB of signal-to-noise ratio in the worst of
conditions, it's not really a concern unless you're doing something
really dumb (or are simply trying to prove the point of what happens
when you do).

> Personally I would never design
> such a device. +14dBu FS, with 12 dB pads on the input when necessary, is
> much more useful IMO.

I agree, but you're not the one designing what Marketing has to sell.
The numbers look a lot better when you leave it to "the other guy" to
give you the signal that will get all the performance your device is
capable of.


--
I'm really Mike Rivers (mrivers@d-and-d.com)
However, until the spam goes away or Hell freezes over,
lots of IP addresses are blocked from this system. If
you e-mail me and it bounces, use your secret decoder ring
and reach me here: double-m-eleven-double-zero at yahoo
Anonymous
August 8, 2004 2:08:26 PM

Archived from groups: rec.audio.pro (More info?)

"Mike Rivers" <mrivers@d-and-d.com> wrote in message
news:znr1091793775k@trad...
> That would be valid if practice agreed with theory, but it doesn't.
> Most analog circuits approach "not clean" slowly. In order to be
> really clean at the level that produces 0 dBFS, you need to be able to
> go above that level by some amount (depending on how much distortion
> you're willing to accept at high levels) before reaching clipping.
> Clipping isn't the only "distortion" you need to worry about.

Yep, it's up to you to decide how clean. Just because you run an analog
stage 10dB below clipping is no guarantee that it is "CLEAN". And the lower
you go, the more noise your likely to get as well. Know your equipment is
the real answer.

> Also, subjectively, audio that's clipped for just a couple of samples
> is undetectable, so you want to be able to send clean level to the
> converter even at its clipping level.

Yep, but no need for more.

> > If it's +26dBu, then you will indeed lose resolution by running
> > too far below that to avoid analog overload
>
> How much resolution do you think you'll lose, and how do you think
> that will affect what you hear? I hate to sound like Phil here, but
> you really should do a listening test. There is so much "audible"
> headroom with decent 24-bit converters that you can easily work in the
> eyeball average range of -20 to -10 dBFS and it will sound just fine.

I totally agree, I said you lose resolution, it's up to the user to know
whether it's significant.

> > However you also lose resolution by running analog stages too
> > far below their maximum, EIN being constant.
>
> The problem with that argument is that EIN isn't constant. It's a
> function of gain, which is why it's nearly given (on the spec sheet
> anyway - the engineering lab is another story) at maximum gain.

The EIN being constant was a qualifier. Which is why I also said "in a fixed
gain system".

> > Personally I would never design
> > such a device. +14dBu FS, with 12 dB pads on the input when necessary,
is
> > much more useful IMO.
>
> I agree, but you're not the one designing what Marketing has to sell.
> The numbers look a lot better when you leave it to "the other guy" to
> give you the signal that will get all the performance your device is
> capable of.

OK, I'm glad we can agree on that.

TonyP.
!