AD & DA

[Kari Laine]

Hi,

I am wondering how AD- and DA-converters are implemented.
Any documents about the chips internal configuration?
Would it be in theory possible to implement them with discrete components?

Best Regards
Kari


PIC - ARM - DISPLAYS - RELAYS - MODULES - CONVERTERS - I2C - SPI -
KEYPADS - ACCESSORIES
http://www.byvac.com (I am just a satisfied customer)

 

[Bob Masta]

On Sun, 31 Jan 2010 12:55:50 +0200, Kari Laine
<klaine8@gmail.com> wrote:

Hi,

I am wondering how AD- and DA-converters are implemented.
Any documents about the chips internal configuration?
Would it be in theory possible to implement them with discrete components?

Indeed it would be possible, and of course that's
how it was done originally. Nowadays chips are
more practical for most applications, but simple
pulse averaging D/As are still used on the outputs
of microprocessors that don't have true D/As built
in.

On older systems with parallel ports, you can make
a simple 8-bit D/A with a handful of resistors.
See <http://www.daqarta.com/dw_ggdd.htm> for a
discussion of simple D/A circuits.

With a D/A in hand, you can turn it into an A/D
via "successive approximation", where the system
(computer, or internals of A/D chip) toggles the
D/A bits as it compares the D/A output to the
input signal. When they match close enough, that
bit pattern is the converted A/D value. See
<http://www.daqarta.com/dw_ggaa.htm> for more
info.

Best regards,




Bob Masta

DAQARTA v5.00
Data AcQuisition And Real-Time Analysis
www.daqarta.com
Scope, Spectrum, Spectrogram, Sound Level Meter
Frequency Counter, FREE Signal Generator
Pitch Track, Pitch-to-MIDI
DaqMusic - FREE MUSIC, Forever!
(Some assembly required)
Science (and fun!) with your sound card!

 

[Kari Laine]

Bob Masta wrote:

On Sun, 31 Jan 2010 12:55:50 +0200, Kari Laine
klaine8@gmail.com> wrote:
Bob Masta

DAQARTA v5.00
Data AcQuisition And Real-Time Analysis
www.daqarta.com
Scope, Spectrum, Spectrogram, Sound Level Meter
Frequency Counter, FREE Signal Generator
Pitch Track, Pitch-to-MIDI
DaqMusic - FREE MUSIC, Forever!
(Some assembly required)
Science (and fun!) with your sound card!

Thanks Bob!

Your product is interesting. How about Linux version?
My hobby at the moment is to write Linux software for the Velleman
PCSGU250 scope. It is not much yet. My math is poor so I have to study
little to be able to do the Fourier and sin(x)/x stuff.
I don't even yet to know what that sin(x)/x stuff is for...

Happy hacking to you !

Kari


--
PIC - ARM - DISPLAYS - RELAYS - MODULES - CONVERTERS - I2C - SPI -
KEYPADS - ACCESSORIES
http://www.byvac.com (I am just a satisfied customer)

 

[CC]

"Bob Masta" <N0Spam@daqarta.com> wrote in message
news:4b658f7c.1184126@news.eternal-september.org...

On Sun, 31 Jan 2010 12:55:50 +0200, Kari Laine
klaine8@gmail.com> wrote:

Hi,

I am wondering how AD- and DA-converters are implemented.
Any documents about the chips internal configuration?
Would it be in theory possible to implement them with discrete components?

Indeed it would be possible, and of course that's
how it was done originally. Nowadays chips are
more practical for most applications, but simple
pulse averaging D/As are still used on the outputs
of microprocessors that don't have true D/As built
in.

On older systems with parallel ports, you can make
a simple 8-bit D/A with a handful of resistors.
See <http://www.daqarta.com/dw_ggdd.htm> for a
discussion of simple D/A circuits.

With a D/A in hand, you can turn it into an A/D
via "successive approximation", where the system
(computer, or internals of A/D chip) toggles the
D/A bits as it compares the D/A output to the
input signal. When they match close enough, that
bit pattern is the converted A/D value. See
http://www.daqarta.com/dw_ggaa.htm> for more
info.

Best regards,




This post just got me thinking about the A/D concept. Just curious about the

basic concept (in 1 or 2 paragraphs). I think I understand how the process
works but just wondering if you sample a microphone and store its value,
that single sample represents the amplitude of the wave at that time and the
frequency is reconstructed by stringing multiple samples together.

I hope this is clear. I guess another way of saying this is how are both the
amplitude & freq derived from a single number (please keep it simple).

 

[Michael Black]

On Sun, 31 Jan 2010, Kari Laine wrote:

Hi,

I am wondering how AD- and DA-converters are implemented.
Any documents about the chips internal configuration?
Would it be in theory possible to implement them with discrete components?

There's very little that came about because of ICs. Most of them

are based on things that existed before ICs. The problem is that a lot of
those things were not practical in the days of tubes or even transistors,
they existed but either only in the laboratory or in very expensive
equipment, because it took up too many parts and too much space.

So synthesized tuning of radios was possible in the old days, but it
was only in very high end commercial equipment, but once ICs allowed
for high integration, they became common and indeed have certain
advantages in terms of cost and space (now they actually use up
less space than analog tuning did).

Even with things relatively recent, they often existed as discrete
components or at least using very common low integration ICs. So as
others have pointed out, D/A converters were in the form of a latch
on a databus and a set of resistors, and once you had that you could
add a comparator and use the computer to control the D/A converter and
make an A/D converter. Or even do it without a CPU, requiring more
hardware. Digital Signal Processing was being done before ICs to
handle the task came along.

A lot of the lowering of prices of consumer electronics comes because
the design goes through a few iterations. So a first VCR or computer
used a lot of common ICs that weren't very high density, and the cost
and size of the equipment would reflect that. As sales rose, the
manufacturer could afford to go to higher density components, which
also cut manufacturing costs, so the ICs become higher density and
less generic. That often happens a few times until there are virtually no
parts in a piece of equipment, and the ICs can't be used except for
that very specific use.

Michael

 

[Baron]

CC Inscribed thus:

"Bob Masta" <N0Spam@daqarta.com> wrote in message
news:4b658f7c.1184126@news.eternal-september.org...
On Sun, 31 Jan 2010 12:55:50 +0200, Kari Laine
klaine8@gmail.com> wrote:

Hi,

I am wondering how AD- and DA-converters are implemented.
Any documents about the chips internal configuration?
Would it be in theory possible to implement them with discrete
components?

Indeed it would be possible, and of course that's
how it was done originally. Nowadays chips are
more practical for most applications, but simple
pulse averaging D/As are still used on the outputs
of microprocessors that don't have true D/As built
in.

On older systems with parallel ports, you can make
a simple 8-bit D/A with a handful of resistors.
See <http://www.daqarta.com/dw_ggdd.htm> for a
discussion of simple D/A circuits.

With a D/A in hand, you can turn it into an A/D
via "successive approximation", where the system
(computer, or internals of A/D chip) toggles the
D/A bits as it compares the D/A output to the
input signal. When they match close enough, that
bit pattern is the converted A/D value. See
http://www.daqarta.com/dw_ggaa.htm> for more
info.

Best regards,




This post just got me thinking about the A/D concept. Just curious
about the basic concept (in 1 or 2 paragraphs). I think I understand
how the process works but just wondering if you sample a microphone
and store its value, that single sample represents the amplitude of
the wave at that time and the frequency is reconstructed by stringing
multiple samples together.

I hope this is clear. I guess another way of saying this is how are
both the amplitude & freq derived from a single number (please keep it
simple).

Google "Nyquist Criterion".

--
Best Regards:
Baron.

 

[John Larkin]

On Sun, 31 Jan 2010 12:28:09 -0500, "CC" <N@NE.nothing> wrote:

"Bob Masta" <N0Spam@daqarta.com> wrote in message
news:4b658f7c.1184126@news.eternal-september.org...
On Sun, 31 Jan 2010 12:55:50 +0200, Kari Laine
klaine8@gmail.com> wrote:

Hi,

I am wondering how AD- and DA-converters are implemented.
Any documents about the chips internal configuration?
Would it be in theory possible to implement them with discrete components?

Indeed it would be possible, and of course that's
how it was done originally. Nowadays chips are
more practical for most applications, but simple
pulse averaging D/As are still used on the outputs
of microprocessors that don't have true D/As built
in.

On older systems with parallel ports, you can make
a simple 8-bit D/A with a handful of resistors.
See <http://www.daqarta.com/dw_ggdd.htm> for a
discussion of simple D/A circuits.

With a D/A in hand, you can turn it into an A/D
via "successive approximation", where the system
(computer, or internals of A/D chip) toggles the
D/A bits as it compares the D/A output to the
input signal. When they match close enough, that
bit pattern is the converted A/D value. See
http://www.daqarta.com/dw_ggaa.htm> for more
info.

Best regards,




This post just got me thinking about the A/D concept. Just curious about the
basic concept (in 1 or 2 paragraphs). I think I understand how the process
works but just wondering if you sample a microphone and store its value,
that single sample represents the amplitude of the wave at that time and the
frequency is reconstructed by stringing multiple samples together.

Yes. In theory, to reconstruct the signal, you need to sample at a
rate at least twice the highest frequency in the signal. In practise,
3x or 4x works well. The resulting ADC samples may look ratty to the
eye, but if you later run them through a DAC and a lowpass filter, you
can almost-perfectly reconstruct the original smooth signal.

http://en.wikipedia.org/wiki/Sampling_theorem

People who don't understand this post elaborate web pages proving that
DVDs grossly distort music, which of course they don't.

John

 

[CC]

"John Larkin" <jjlarkin@highNOTlandTHIStechnologyPART.com> wrote in message
news:qfnbm5p34mi980l03fglbuf16s9i8hefl3@4ax.com...

On Sun, 31 Jan 2010 12:28:09 -0500, "CC" <N@NE.nothing> wrote:


"Bob Masta" <N0Spam@daqarta.com> wrote in message
news:4b658f7c.1184126@news.eternal-september.org...
On Sun, 31 Jan 2010 12:55:50 +0200, Kari Laine
klaine8@gmail.com> wrote:

Hi,

I am wondering how AD- and DA-converters are implemented.
Any documents about the chips internal configuration?
Would it be in theory possible to implement them with discrete
components?

Indeed it would be possible, and of course that's
how it was done originally. Nowadays chips are
more practical for most applications, but simple
pulse averaging D/As are still used on the outputs
of microprocessors that don't have true D/As built
in.

On older systems with parallel ports, you can make
a simple 8-bit D/A with a handful of resistors.
See <http://www.daqarta.com/dw_ggdd.htm> for a
discussion of simple D/A circuits.

With a D/A in hand, you can turn it into an A/D
via "successive approximation", where the system
(computer, or internals of A/D chip) toggles the
D/A bits as it compares the D/A output to the
input signal. When they match close enough, that
bit pattern is the converted A/D value. See
http://www.daqarta.com/dw_ggaa.htm> for more
info.

Best regards,




This post just got me thinking about the A/D concept. Just curious about
the
basic concept (in 1 or 2 paragraphs). I think I understand how the process
works but just wondering if you sample a microphone and store its value,
that single sample represents the amplitude of the wave at that time and
the
frequency is reconstructed by stringing multiple samples together.

Yes. In theory, to reconstruct the signal, you need to sample at a
rate at least twice the highest frequency in the signal. In practise,
3x or 4x works well. The resulting ADC samples may look ratty to the
eye, but if you later run them through a DAC and a lowpass filter, you
can almost-perfectly reconstruct the original smooth signal.

http://en.wikipedia.org/wiki/Sampling_theorem

People who don't understand this post elaborate web pages proving that
DVDs grossly distort music, which of course they don't.

John

got it. thanks

 

[whit3rd]

This post just got me thinking about the A/D concept. Just curious about the
basic concept (in 1 or 2 paragraphs).

Hah! Excuse for a math digression!
The vast majority of 'analog' signals are time-varying functions which
are smooth (continuous and with continuous and well-defined
finite time derivatives). That description excludes
some kinds of phenomena, like truly uncorrelated 'white' noise
(thus, we extremely pedantic types like to refer to 'pink' noise).

One measurement of a signal isn't enough to tell its development
in time, so a series of measurements is made; there are some kinds
of signals (periodic repeating ones) where that series can be
finite and still not 'miss' aspects of the underlying analog signal,
thus we speak of the series as 'oversampled' if it suffices to
determine the interesting part of the signal, and 'undersampled' if
it does not.

A nonperiodic signal can be packaged like a string of sausages
into a periodic one.

Then there are mathematical treatments (like Fourier analysis) that
can represent a signal in multiple ways, either a series of timed
voltage measurements or a spectrum of amplitude-and-phase
of frequencies. There's an inversion theorem that says
your 1000-measurements time series and the 500-frequencies-and-
phases spectrum are interchangeable representations (holding
the same information content).

Then it gets complicated; the picture on your TV/video screen is
a large array of changing hue/brightness/saturation triple quantities,
and it too can be re-represented in lots of ways so a digital
description
can be squirted through the RF airwaves and reconstructed.
The screen can be made into a mosaic of patches, each with
some information that changes (and is retransmitted often) as well
as other information that doesn't change (and is retransmitted
less often). Patches with high spatial frequency (lots of detail)
can be rendered with extra accuracy, patches with high time-variance
can be given faster updates (but color accuracy becomes less
important).
Background stationary objects get the most color refinement but the
slowest re-transmission of edge positions.

The important thing to remember, is that this is all manipulation of
measurements, of the kind of information that has an error estimate
attached to it: as long as your re-doing of the info doesn't amplify
the errors, it's 'legitimate' for whatever purpose drives you. The
use of a logarithm scale for plotting a function, or a frequency
scale for representing music, it's all just a mathematical
rethinking of the same information, in possibly more useful forms.

 

[Bob Masta]

On Sun, 31 Jan 2010 17:52:01 +0200, Kari Laine
<klaine8@gmail.com> wrote:

Bob Masta wrote:
On Sun, 31 Jan 2010 12:55:50 +0200, Kari Laine
klaine8@gmail.com> wrote:
Bob Masta

DAQARTA v5.00
Data AcQuisition And Real-Time Analysis
www.daqarta.com
Scope, Spectrum, Spectrogram, Sound Level Meter
Frequency Counter, FREE Signal Generator
Pitch Track, Pitch-to-MIDI
DaqMusic - FREE MUSIC, Forever!
(Some assembly required)
Science (and fun!) with your sound card!

Thanks Bob!

Your product is interesting. How about Linux version?

Sorry, it's one of those "maybe some day" kinds of
things. Most of the coding effort in a big
project goes into the user interface, which would
mean a major effort to convert from the Windows
API to something for a Linux GUI.

My hobby at the moment is to write Linux software for the Velleman
PCSGU250 scope. It is not much yet. My math is poor so I have to study
little to be able to do the Fourier and sin(x)/x stuff.
I don't even yet to know what that sin(x)/x stuff is for...

You might want to check out my "Gut-Level Fourier
Transforms" series at
<http://www.daqarta.com/author.htm>.

That should provide you with the basics, though
not the complete math to actually write a full
FFT. You *could* do a Discrete Fourier Transform
(DFT) with no more than this, but DFTs are pretty
slow by comparison.

I don't have a good Web link for basic FFT code,
but I haven't looked. I learned about FFT code
back in the days of paper, when dinosaurs walked
the Earth. A good source, if it's still
available, is Hal Chamberlin's "Musical
Applications of Microprocessors", which contains
explanations and complete working code for FFTs.
The code is in old-style BASIC, but you can easily
convert to your language of choice.

Best regards,



Bob Masta

DAQARTA v5.00
Data AcQuisition And Real-Time Analysis
www.daqarta.com
Scope, Spectrum, Spectrogram, Sound Level Meter
Frequency Counter, FREE Signal Generator
Pitch Track, Pitch-to-MIDI
DaqMusic - FREE MUSIC, Forever!
(Some assembly required)
Science (and fun!) with your sound card!