Chip with simple program for Toy

w_tom <w_tom1@usa.net> writes:
Home traditionally use two phase electricity.
Homes in the USA use single phase electricity, at two different
voltages. Two phase electricity was phased out (pun intended) about a
century ago when three phase power was found to be better, but can
still be found powering stepper motors.
 
Ditto with Robert. The filtering should be evaluated at the output
stage of each amp. You do not want to feed unwanted signals into the
next stage Unless they are so small you need to amplify them out. As
Robert indicated Low Z. no antennas and sometimes using larger wattage
resistors help with noise e.g. 1/2 watt instead of 1/4.
As far as amplifying radio stations, I think your speakers will only
go to 20k in frequency which rids that problem. Even if they get
through you will not hear them!
Thanks to you and Robert both for your responses. It looks like
adding a bandpass filter from stage to stage is the way to go, along
with lower resistances for controlling gain in the opamps. It sounds
like 10K on the input and a 100K pot on the feedback loop might be a
good choice. Any opinions?
 
Avoid multiple filtering ( low-pass or high-pass ) at the same frequency
If your project needs to pass official tests concerning EMC, also
apply RF filtering of outputs and supply. There is a lot of info
on the web about this.
Thanks for all of your recommendations. So what is the reason for not
filtering multiple times at the same frequency?
 
Brandon wrote:
Avoid multiple filtering ( low-pass or high-pass ) at the same frequency
If your project needs to pass official tests concerning EMC, also
apply RF filtering of outputs and supply. There is a lot of info
on the web about this.

Thanks for all of your recommendations. So what is the reason for not
filtering multiple times at the same frequency?
It will create uncontrolled filtering leading to to steep cut-off
and / or phase shift.

Better is:
Dominant filtering at first stage / input by using single pole
low-pass and high-pass filtering ( one C - one R )
All other stages get only frequency compensation by a little
C ( 22pF ) around the feedback.

Make sure that the in and outputs of a potmeter are AC-coupled
by using large enough capacitors.
These will most likely be electrolytics because of the value needed.
In a single supply circuit this will be no problem.
For a circuit with symmetrical supply there will be the discussion
of the ( polarized ) coupling capacitor being wrong polarized
every half cycle. But they are not because there is no voltage-
drop over these capacitors. Use high temp / high rel. types
and they will last.

The slightest DC into a potmeter will make it noisy within weeks.

Robert
 
stratus46@yahoo.com wrote:

Eeyore wrote:
w_tom wrote:
Home traditionally use two phase electricity.

In the USA.

In Europe and many (most?) other places it's single phase to homes.


Keep in mind that '2 phase' power in American homes is very similar to
what you use. We just have a center tap on the 240 to get the 120
times 2.

But, in all my Athlon PCs (and I would bet Intel does it too), it uses
a 3 phase converter to make the 50+ amp 1.65 Volt core power supply
Tell me about this '3 phase converter' will you ?

Graham
 
On Dec 6, 3:03 am, Eeyore <rabbitsfriendsandrelati...@hotmail.com>
wrote:
stratu...@yahoo.com wrote:
Eeyore wrote:
w_tom wrote:
Home traditionally use two phase electricity.

In the USA.

In Europe and many (most?) other places it's single phase to homes.

Keep in mind that '2 phase' power in American homes is very similar to
what you use. We just have a center tap on the 240 to get the 120
times 2.

But, in all my Athlon PCs (and I would bet Intel does it too), it uses
a 3 phase converter to make the 50+ amp 1.65 Volt core power supply

Tell me about this '3 phase converter' will you ?

Graham- Hide quoted text -

- Show quoted text -
Hi, Graham. You might want to take a look at this Linear Technology
article:

http://www.linear.com/pc/downloadDocument.do?id=4918

Cheers
Chris
 
On Thu, 06 Dec 2007 08:07:14 +0000, Bob Woodward <"Bob
Woodward"@no.org> wrote:

Brandon wrote:
Avoid multiple filtering ( low-pass or high-pass ) at the same frequency
If your project needs to pass official tests concerning EMC, also
apply RF filtering of outputs and supply. There is a lot of info
on the web about this.

Thanks for all of your recommendations. So what is the reason for not
filtering multiple times at the same frequency?

It will create uncontrolled filtering leading to to steep cut-off
and / or phase shift.
Amen to that. Steepness is not usually an issue, since
most of the time it's desireable. And in fact, cascading
simple single-pole stages won't have much effect on
steepness, but it might have a big effect on the cut-off
frequency.

Consider that the cut-off is the -3dB frequency... let's say
it's 10 kHz for a single stage. Cascade two stages together, and
at 10 kHz the response is now -6dB. The 3 dB point will be
effectively moved lower. For simple single-pole R-C stages,
it would be about 70% of the original, or 7 kHz in this example.

Multi-stage filters are normally designed with two poles per
gain stage. To get a specified overall cut-off frequency and
response shape, the stages have to be designed to work
together. Each has a slightly different cut-off and damping
(Q) which when cascaded give the target response.
However, note that the more stages you use, the tighter the
tolerance of each stage must be if you want to get the
desired response. 4 or 5 stages (8-10 poles) is about
the max I'd want to mess with, and even there you need to
hand-pick parts.

Best regards,



Best regards,


Bob Masta

DAQARTA v3.50
Data AcQuisition And Real-Time Analysis
www.daqarta.com
Scope, Spectrum, Spectrogram, FREE Signal Generator
Science with your sound card!
 
Bob Woodward wrote:

If your project needs to pass official tests concerning EMC, also
apply RF filtering of outputs and supply.
Usually completely unnecessary.


There is a lot of info
on the web about this.
A lot of BAD info it would seem if you came to that conclusion.

FYI, I have designed many audio mixing products that have been in volume
manufacture and not ONE of them needed explicit *filters* to pass the
required EMC tests for CE. Mostly needed just a few extra caps of small
value in a few judiciously chosen places. May of those were merely 'belt and
braces' too.

Graham
 
poogie wrote:

Ditto with Robert. The filtering should be evaluated at the output
stage of each amp. You do not want to feed unwanted signals into the
next stage Unless they are so small you need to amplify them out.
Errr... Hallo !

If it's audio, the bandwidth will be constrained to audio frequencies, so no
filtering will be required.


As Robert indicated Low Z. no antennas
What antennas ?


and sometimes using larger wattage
resistors help with noise e.g. 1/2 watt instead of 1/4.
Complete nonsense.

Graham
 
On Wed, 5 Dec 2007 07:47:47 -0800, "BobW"
<nimby_NEEDSPAM@roadrunner.com> wrote:

"Bob Masta" <NoSpam@daqarta.com> wrote in message
news:4756ab11.3122477@news.sysmatrix.net...

[snip]


For those who haven't encountered lock-in amps before, what
they do is essentially chop the input at the applied reference
frequency, then amplify, low-pass filter, and meter it. (The
"lock-in" part of the name is a red herring.) The reference frequency
is derived from the same source that excites the sensor (or whatever).
The chopping action essentially multiplies the input signal by the
reference frequency. (Modern units do an actual multiply.)
This produces a DC signal proportional to any input component at
the reference frequency, while all the product sum-and-difference
components are at higher frequencies that are easily removed via
the low-pass filter. Lock-ins can give stupendous noise rejection
if you set the filter bandwidth narrow enough, such that you can
extract signals that are more than 100 dB below the noise.

[snip]

Best regards,


Bob Masta


This multiplication to isolate an individual frequency is essentially how
the Fourier transform works. What advantage do these "lockin" amps have,
today, vs. doing the A->D conversion and then an FFT?
Not much for many applications, and some modern lock-ins
actually do use FFT methods.

But notice that the reference frequency in a lock-in is exactly the
frequency of interest. You can do that with an FFT, but it's
only really practical if the same system is generating the test
signal as well. Otherwise, you end up with response leakage
"skirts" that cause energy to appear at adjacent frequencies.
(Though window functions help this, they are never as good
as the synchronous reference approach.)

The lock-in typically uses a single-pole filter, the same as an FFT
(effectively). The FFT's filter gets narrower by using a bigger
sample set, which takes more computing power. (No big deal
these days.) But in the lock-in all that's needed is to increase
the R and/or C, so there is essentially no cost to cranking it
way up. (Other than the fact that you have to wait for it to
settle, just as the FFT requires that all N samples be acquired.)

Until fairly recently, it was hard to get A/Ds with the specs to rival
lock-in dynamic range. But with 24-bit converters now readily
available that's not an issue any more.

By the way, my earlier explanation was simplified in that it
assumed that the reference and response were exactly in
phase. That's rarely the case, so lock-ins have 2 channels
set 90 degrees apart, plus magnitude/phase readouts.
Otherwise, an input that was at the exact reference frequency
but 90 degrees out of phase would give 0 from a single
lock-in channel. There is exactly the same issue in FFTs,
which do all computations in real and imaginary (cosine and
sine) phases and combine the results to get magnitude and phase.

Interested parties can investigate the benefits of synchronous
reference frequencies using my Daqarta software. You don't need
to buy anything, because the signal generator portion is free.
(The signal inputs won't work after the trial period expires, but
the generator will keep working indefinitely.)

There is a built-in spectrum analyzer that can monitor the generator
output. If you set any arbitrary generator frequency, you will
clearly see the spectral "skirts" mentioned above, and as you
change the frequency you will see the response change from
a "tight skirt" when the frequency is nearly synchronous (ie same as
one of the intrinsic spectral lines of the FFT), to a billowing skirt
when it is halfway between lines. At that point the spectrum will
show nearly half the energy at each of these adjacent lines, plus
smaller amounts at higher and lower lines.

Now go to the Frequency Step dialog (in the Tone Freq dialog)
and set Step Lines to 1.000. When you change frequencies
after this, they will only fall exactly on FFT spectral lines.
The spectrum of a single sine will show a single peak at
that frequency, with no skirts at all.

Enjoy!

Best regards,







Best regards,


Bob Masta

DAQARTA v3.50
Data AcQuisition And Real-Time Analysis
www.daqarta.com
Scope, Spectrum, Spectrogram, FREE Signal Generator
Science with your sound card!
 
Brandon wrote:

Ditto with Robert. The filtering should be evaluated at the output
stage of each amp. You do not want to feed unwanted signals into the
next stage Unless they are so small you need to amplify them out. As
Robert indicated Low Z. no antennas and sometimes using larger wattage
resistors help with noise e.g. 1/2 watt instead of 1/4.
As far as amplifying radio stations, I think your speakers will only
go to 20k in frequency which rids that problem. Even if they get
through you will not hear them!

Thanks to you and Robert both for your responses. It looks like
adding a bandpass filter from stage to stage is the way to go
NO.

You've been given very bad, over-the-top and completey erroneous advice.

I'm a pro-audio designer by the way so I damn well ought to know.

Graham
 
Eeyore wrote:
Bob Woodward wrote:

If your project needs to pass official tests concerning EMC, also
apply RF filtering of outputs and supply.

Usually completely unnecessary.


There is a lot of info
on the web about this.

A lot of BAD info it would seem if you came to that conclusion.

FYI, I have designed many audio mixing products that have been in volume
manufacture and not ONE of them needed explicit *filters* to pass the
required EMC tests for CE. Mostly needed just a few extra caps of small
value in a few judiciously chosen places. May of those were merely 'belt and
braces' too.

Graham
Lets all kneel down and bow our heads for Eeyores eternal wisdom and
authocracy.

Now shut down this thread. It starts to smell .

Robert
 
On Thu, 06 Dec 2007 14:06:01 +0000, Eeyore
<rabbitsfriendsandrelations@hotmail.com> wrote:


poogie wrote:

and sometimes using larger wattage
resistors help with noise e.g. 1/2 watt instead of 1/4.

Complete nonsense.
---
Really???

What do you think the 'T' in

E = sqrt(4kTR df)

is about?

And which do you think will generate more noise: a 100 ohm 1/2 watt
resistor dissipating 1/2 a watt or a 100 ohm 1 watt resistor
dissipating 1/2 a watt?


--
JF
 
"Bob Masta" <NoSpam@daqarta.com> wrote in message
news:4757f9ad.2378055@news.sysmatrix.net...
On Wed, 5 Dec 2007 07:47:47 -0800, "BobW"
nimby_NEEDSPAM@roadrunner.com> wrote:


"Bob Masta" <NoSpam@daqarta.com> wrote in message
news:4756ab11.3122477@news.sysmatrix.net...

[snip]


For those who haven't encountered lock-in amps before, what
they do is essentially chop the input at the applied reference
frequency, then amplify, low-pass filter, and meter it. (The
"lock-in" part of the name is a red herring.) The reference frequency
is derived from the same source that excites the sensor (or whatever).
The chopping action essentially multiplies the input signal by the
reference frequency. (Modern units do an actual multiply.)
This produces a DC signal proportional to any input component at
the reference frequency, while all the product sum-and-difference
components are at higher frequencies that are easily removed via
the low-pass filter. Lock-ins can give stupendous noise rejection
if you set the filter bandwidth narrow enough, such that you can
extract signals that are more than 100 dB below the noise.

[snip]

Best regards,


Bob Masta


This multiplication to isolate an individual frequency is essentially how
the Fourier transform works. What advantage do these "lockin" amps have,
today, vs. doing the A->D conversion and then an FFT?


Not much for many applications, and some modern lock-ins
actually do use FFT methods.

But notice that the reference frequency in a lock-in is exactly the
frequency of interest. You can do that with an FFT, but it's
only really practical if the same system is generating the test
signal as well. Otherwise, you end up with response leakage
"skirts" that cause energy to appear at adjacent frequencies.
(Though window functions help this, they are never as good
as the synchronous reference approach.)

The lock-in typically uses a single-pole filter, the same as an FFT
(effectively). The FFT's filter gets narrower by using a bigger
sample set, which takes more computing power. (No big deal
these days.) But in the lock-in all that's needed is to increase
the R and/or C, so there is essentially no cost to cranking it
way up. (Other than the fact that you have to wait for it to
settle, just as the FFT requires that all N samples be acquired.)

Until fairly recently, it was hard to get A/Ds with the specs to rival
lock-in dynamic range. But with 24-bit converters now readily
available that's not an issue any more.

By the way, my earlier explanation was simplified in that it
assumed that the reference and response were exactly in
phase. That's rarely the case, so lock-ins have 2 channels
set 90 degrees apart, plus magnitude/phase readouts.
Otherwise, an input that was at the exact reference frequency
but 90 degrees out of phase would give 0 from a single
lock-in channel. There is exactly the same issue in FFTs,
which do all computations in real and imaginary (cosine and
sine) phases and combine the results to get magnitude and phase.

Interested parties can investigate the benefits of synchronous
reference frequencies using my Daqarta software. You don't need
to buy anything, because the signal generator portion is free.
(The signal inputs won't work after the trial period expires, but
the generator will keep working indefinitely.)

There is a built-in spectrum analyzer that can monitor the generator
output. If you set any arbitrary generator frequency, you will
clearly see the spectral "skirts" mentioned above, and as you
change the frequency you will see the response change from
a "tight skirt" when the frequency is nearly synchronous (ie same as
one of the intrinsic spectral lines of the FFT), to a billowing skirt
when it is halfway between lines. At that point the spectrum will
show nearly half the energy at each of these adjacent lines, plus
smaller amounts at higher and lower lines.

Now go to the Frequency Step dialog (in the Tone Freq dialog)
and set Step Lines to 1.000. When you change frequencies
after this, they will only fall exactly on FFT spectral lines.
The spectrum of a single sine will show a single peak at
that frequency, with no skirts at all.

Enjoy!

Best regards,

Bob Masta
Ahhh...yes. Knowing the target frequency ahead of time, and using
synchronous filtering, makes it much more accurate (as compared with a
general purpose FFT).

Thanks for the great information, Bob.
Bob
 
John Fields wrote:

Eeyore wrote:
poogie wrote:

and sometimes using larger wattage
resistors help with noise e.g. 1/2 watt instead of 1/4.

Complete nonsense.

---
Really???

What do you think the 'T' in

E = sqrt(4kTR df)

is about?

And which do you think will generate more noise: a 100 ohm 1/2 watt
resistor dissipating 1/2 a watt or a 100 ohm 1 watt resistor
dissipating 1/2 a watt?
Modern audio design doesn't run resistors hot in sensitive gain stages.

Graham
 
"Chris" <cfoley1064@yahoo.com> wrote in message
news:1ed84e43-8abe-4f1f-8b31-c0a2cdf8cd43@w40g2000hsb.googlegroups.com...
On Dec 6, 3:03 am, Eeyore <rabbitsfriendsandrelati...@hotmail.com
wrote:
stratu...@yahoo.com wrote:
Eeyore wrote:
w_tom wrote:
Home traditionally use two phase electricity.

In the USA.

In Europe and many (most?) other places it's single phase to
homes.

Keep in mind that '2 phase' power in American homes is very similar
to
what you use. We just have a center tap on the 240 to get the 120
times 2.

But, in all my Athlon PCs (and I would bet Intel does it too), it
uses
a 3 phase converter to make the 50+ amp 1.65 Volt core power supply

Tell me about this '3 phase converter' will you ?

Graham- Hide quoted text -

- Show quoted text -

Hi, Graham. You might want to take a look at this Linear Technology
article:

http://www.linear.com/pc/downloadDocument.do?id=4918

Cheers
Chris
Notice this 3PHASE powersupply has a DC input and some 3 phase
technology within the pwer supply.
When most people discuss American split phase or 50hz Three phase they
are talking about the mains distribution and not what some clever
engineer has built into the inner workings of a power supply.

John G.
 
DJ Delorie wrote:

w_tom <w_tom1@usa.net> writes:

Home traditionally use two phase electricity.


Homes in the USA use single phase electricity, at two different
voltages. Two phase electricity was phased out (pun intended) about a
century ago when three phase power was found to be better, but can
still be found powering stepper motors.
Really,



--
"I'd rather have a bottle in front of me than a frontal lobotomy"
http://webpages.charter.net/jamie_5
 
stratus46@yahoo.com wrote:

On Dec 5, 6:19 pm, Eeyore <rabbitsfriendsandrelati...@hotmail.com
wrote:

w_tom wrote:

Home traditionally use two phase electricity.

In the USA.

In Europe and many (most?) other places it's single phase to homes.

Graham


Keep in mind that '2 phase' power in American homes is very similar to
what you use. We just have a center tap on the 240 to get the 120
times 2.

But, in all my Athlon PCs (and I would bet Intel does it too), it uses
a 3 phase converter to make the 50+ amp 1.65 Volt core power supply

GG
I think you have your source count and phase count terminology mixed up.


--
"I'd rather have a bottle in front of me than a frontal lobotomy"
http://webpages.charter.net/jamie_5
 
On Thu, 06 Dec 2007 21:25:08 +0000, Eeyore
<rabbitsfriendsandrelations@hotmail.com> wrote:

John Fields wrote:

Eeyore wrote:
poogie wrote:

and sometimes using larger wattage
resistors help with noise e.g. 1/2 watt instead of 1/4.

Complete nonsense.

---
Really???

What do you think the 'T' in

E = sqrt(4kTR df)

is about?

And which do you think will generate more noise: a 100 ohm 1/2 watt
resistor dissipating 1/2 a watt or a 100 ohm 1 watt resistor
dissipating 1/2 a watt?

Modern audio design doesn't run resistors hot in sensitive gain stages.
---
Precisely because of the reason poogie stated, which you called
'complete nonsense', you utter hypocrite.


--
JF
 
John Fields wrote:

Eeyore wrote:
John Fields wrote:
Eeyore wrote:
poogie wrote:

and sometimes using larger wattage
resistors help with noise e.g. 1/2 watt instead of 1/4.

Complete nonsense.

---
Really???

What do you think the 'T' in

E = sqrt(4kTR df)

is about?

And which do you think will generate more noise: a 100 ohm 1/2 watt
resistor dissipating 1/2 a watt or a 100 ohm 1 watt resistor
dissipating 1/2 a watt?

Modern audio design doesn't run resistors hot in sensitive gain stages.

---
Precisely because of the reason poogie stated,
NO.


which you called 'complete nonsense', you utter hypocrite.
It is complete nonsense when you're talking about dissipating at maximum
maybe a few milliwatts in resisitors.

Audio designers don't throw away significant fractions of a watt as heat in
resistors. Stop pontificating about stuff you're not familiar with.


Graham
 

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