Market for an A/D converter with 150dB dynamic range?
DigitalSignal
the answer.
I wonder if there is a market for an A/D converter with very high
dynamic range, say 150dB. I thought it might be useful because it can
eliminate the needs for multiple-gain front end.
By 150dB dynamic range I mean an A/D converter that can practically
measure a large signal as high as a few volts, and within the same time
period it can detect a signal as small as a few nano volts. To clarify,
when I say "within the same time period", it does not necessarily mean
at the exact same sample points.
Say if we sample a frame of signal with 1024 points. The first a few
hundred points have magnitude as high as a few volts while the last a
few hundred points can go as low as a few nano-volts. This is how I
define the dynamic range.
We do not talk about theoretical dynamic range. For a 24bit A/D
converter, the theoretical dynamic range is 6.02 dB/bit * N bits =
144dB. This definition is not very useful. We all know a 24bit A/D
usually has only 110dB dynamic range due to analog limitation.
The question is: do you see a market or application for such kind of
A/D converter if somebody got the products?
Andy Peters
> I wonder if there is a market for an A/D converter with very high
> dynamic range, say 150dB. I thought it might be useful because it can
> eliminate the needs for multiple-gain front end.
(snip)
> We do not talk about theoretical dynamic range. For a 24bit A/D
> converter, the theoretical dynamic range is 6.02 dB/bit * N bits =
> 144dB. This definition is not very useful. We all know a 24bit A/D
> usually has only 110dB dynamic range due to analog limitation.
So, why do you think your 150 dB dynamic range converter will not have
these same "analog limitations?"
-a
BobG
difference between 44, 88, and 192KHz. Then they listen to 128kbit mpeg.
J.A. Legris
Do you mean like this?
>From http://en.wikipedia.org/wiki/Analog-to-digital_converter :
"Because of this, logarithmic ADCs are very common in voice
communication systems to increase the dynamic range of the
representable values while retaining fine-granular fidelity in the
low-amplitude region. An 8 bit a-law or the µ-law logarithmic ADC
covers the wide dynamic range and has a high resolution in the critical
low-amplitude region, that would otherwise require a 12-bit linear
ADC."
Or maybe this:
>From http://adsabs.harvard.edu/abs/1978ITCS...25..522K :
"An eight-bit exponential converter has been constructed with a dynamic
range of eight decades and a relative error of about three percent.
With the same technology, logarithmic A/D converters and floating-point
D/A and A/D converters can be built."
--
Joe Legris
Eeyore
Andy Peters wrote:
Because he read it in a book ?
Maybe he doesn't know they are 'physical' limitations ?
Graham
Eeyore
"J.A. Legris" wrote:
> DigitalSignal wrote:
>
> > The question is: do you see a market or application for such kind of
> > A/D converter if somebody got the products?
>
> Do you mean like this?
>
> >From http://en.wikipedia.org/wiki/Analog-to-digital_converter :
>
> "Because of this, logarithmic ADCs are very common in voice
> communication systems to increase the dynamic range of the
> representable values while retaining fine-granular fidelity in the
> low-amplitude region. An 8 bit a-law or the ต-law logarithmic ADC
> covers the wide dynamic range and has a high resolution in the critical
> low-amplitude region, that would otherwise require a 12-bit linear
> ADC."
>
> Or maybe this:
>
> >From http://adsabs.harvard.edu/abs/1978ITCS...25..522K :
>
> "An eight-bit exponential converter has been constructed with a dynamic
> range of eight decades and a relative error of about three percent.
> With the same technology, logarithmic A/D converters and floating-point
> D/A and A/D converters can be built."
No. Because non-linear conversion is well known to have unpleasant audible
artifacts.
Graham
martin griffith
<Bassm...@yahoo.com> wrote:
K2-W op-amp front end?
martin
Genome
"DigitalSignal" <digitals...@yahoo.com> wrote in message
news:1160589230.5...@c28g2000cwb.googlegroups.com...[Bollocks]
>
> The question is: do you see a market or application for such kind of
> A/D converter if somebody got the products?
>
You need to brush up on your marketing skills.
In case you missed it we all have mobile phones these days and ring tones
with a dynamic range of 150dB are not duh riguer.
DNA
Ancient_Hacker
Square that if you dare to get the power ratio.
So if the highest voltage out was 16 volts, the lowest would be one
microvolt.
Problem is, if you calculate the thermal noise across say a 10,000 ohm
input across a 200KHz bandwidth (for those golden ears), it's going to
be a whole lot more than a microvolt, like nearly 20 microvolts peak
to peak.
So even if you had a full 24-bit D/A, you can't read out the bottom
four bits due to thermal noise.
Also it takes a heck of an amplifier to play that kind of music. If
you're listening to really gentle music at the milliwatt level, say
exercising the bottom six bits, then you put on The Hole, you need like
2^36 times more power, which will burn out your local power plant.
martin griffith
"Ancient_Hacker" <gr...@comcast.net> wrote:
You missed out how much CO2 it would produce, and how the planet
would slow down each time a modern popular "tune" was played.
Nice toy in a BMW, but it would probably demolecularise the cocaine
the driver had on him, (and the driver hopefully)
martin
Eeyore
Ancient_Hacker wrote:
> A 24-bit A/D has a dynamic range of about 16 MILLION (in volts).
> Square that if you dare to get the power ratio.
>
> So if the highest voltage out was 16 volts, the lowest would be one
> microvolt.
>
> Problem is, if you calculate the thermal noise across say a 10,000 ohm
> input across a 200KHz bandwidth (for those golden ears), it's going to
> be a whole lot more than a microvolt, like nearly 20 microvolts peak
> to peak.
Think more like 200 ohms and 20kHz ok ?
It's around -132dBu from memory
Typical audio mixer clip level is +20dBu so you have 152 dB dynamic range.
> So even if you had a full 24-bit D/A, you can't read out the bottom
> four bits due to thermal noise.
>
> Also it takes a heck of an amplifier to play that kind of music. If
> you're listening to really gentle music at the milliwatt level, say
> exercising the bottom six bits, then you put on The Hole, you need like
> 2^36 times more power, which will burn out your local power plant.
It's not the reproduction chain that's the issue.
Graham
DigitalSignal
is a linear converter.
DigitalSignal
a wide range dynamic range A/D converters? I've never heard of it.
mi...@sushi.com
There is a market for large dynamic range converters in studio
applications. This is because most modern music is a mixture of mono
sources. Mix two source, you get 3db more noise, etc. etc. For the end
user, 16 bits is fine, but I wouldn't turn down 24/96k.
Eeyore
DigitalSignal wrote:
> Genome, could you elaborate a bit? are you saying cellphones are using
> a wide range dynamic range A/D converters? I've never heard of it.
He's pulling your plonker.
He's a nitwit troll.
Graham
Tom Bruhns
On Oct 11, 10:53 am, "DigitalSignal" <digitalsignal...@yahoo.com>
wrote:
> I posted a similar question in the rec.audio.pro group but did not get
> the answer.
>
> I wonder if there is a market for an A/D converter with very high
> dynamic range, say 150dB. I thought it might be useful because it can
> eliminate the needs for multiple-gain front end.
>
> By 150dB dynamic range I mean an A/D converter that can practically
> measure a large signal as high as a few volts, and within the same time
> period it can detect a signal as small as a few nano volts. To clarify,
> when I say "within the same time period", it does not necessarily mean
> at the exact same sample points.
>
I suppose it needs a lot more clarification. I'd like to know about
distortion versus signal amplitude, about spurious responses versus
signal amplitude, and about noise floor (which may also vary with
signal amplitude). It would be a pretty decent converter indeed if it
has a broadband noise floor that's 150dB below full scale, and the
bandwidth is at least audio (apparently what you're interested in).
Actually, I'd be happy with an 18 bit converter that will run at
150Ms/s and has distortion and spurious products worst-case 120dB below
the input signal level, and broadband noise 90dB below a +10dBm full
scale...
Cheers,
Tom
Ancient_Hacker
mi...@sushi.com wrote:
> There is a market for large dynamic range converters in studio
> applications. This is because most modern music is a mixture of mono
> sources. Mix two source, you get 3db more noise, etc. etc.
I could be wrong but I don't see the logic here. With either analog
or digital mixing. Please explain. As long as you don't do something
stupid, like attenuate the incoming signals so they're down in the
noise, where is the extra noise coming from? And what kind of signal
source has that kind of dynamic range? And what medium or amplifier or
listener can handle or listen to anything more than a 60db range?
One possible application would be a "digital radio". The signals
hitting an antenna can easily span 120db, if you're close to an AM
radio station yet want to listen to Radio Antipodal. Let me know when
you have that 24 bit 60 MHz converter ready. Preferably with 12 bits
of exponent, 12 bits of mantissa. For under $5.
Michael A. Terrell
>
> mi...@sushi.com wrote:
>
> > There is a market for large dynamic range converters in studio
> > applications. This is because most modern music is a mixture of mono
> > sources. Mix two source, you get 3db more noise, etc. etc.
>
> I could be wrong but I don't see the logic here. With either analog
> or digital mixing. Please explain. As long as you don't do something
> stupid, like attenuate the incoming signals so they're down in the
> noise, where is the extra noise coming from? And what kind of signal
> source has that kind of dynamic range? And what medium or amplifier or
> listener can handle or listen to anything more than a 60db range?
No analog signal is noiseless, so every time you double the number of
channels the noise goes up another 3 dB.
> One possible application would be a "digital radio". The signals
> hitting an antenna can easily span 120db, if you're close to an AM
> radio station yet want to listen to Radio Antipodal. Let me know when
> you have that 24 bit 60 MHz converter ready. Preferably with 12 bits
> of exponent, 12 bits of mantissa. For under $5.
RF / IF AGC is a simpler way to improve the dynamic range. reduce
the gain for local signals, and ramp it up to full gain for DX
reception.
--
Service to my country? Been there, Done that, and I've got my DD214 to
prove it.
Member of DAV #85.
Michael A. Terrell
Central Florida
mi...@sushi.com
Ancient_Hacker wrote:
> mi...@sushi.com wrote:
>
> > There is a market for large dynamic range converters in studio
> > applications. This is because most modern music is a mixture of mono
> > sources. Mix two source, you get 3db more noise, etc. etc.
>
> I could be wrong but I don't see the logic here. With either analog
> or digital mixing. Please explain.
See the post from Michael A. Terrell.
>As long as you don't do something
> stupid, like attenuate the incoming signals so they're down in the
> noise, where is the extra noise coming from? And what kind of signal
> source has that kind of dynamic range?
The individual signal is a mic'd instrument, electric instrument, or
voice. Granted, the individual signal won't have 150db dynamic range,
but you are constantly mixing audio signals,so you need to keep the
noise "build up" down. Pro-audio pushed for the 24 bit converters, at
least ADC, for their end of the business. They need to do a mix down
and deliver a signal that has a noise level suitable for the final 16
bit product.
And what medium or amplifier or
> listener can handle or listen to anything more than a 60db range?
>
> One possible application would be a "digital radio". The signals
> hitting an antenna can easily span 120db, if you're close to an AM
> radio station yet want to listen to Radio Antipodal. Let me know when
> you have that 24 bit 60 MHz converter ready. Preferably with 12 bits
> of exponent, 12 bits of mantissa. For under $5.
I want one too.
Ancient_Hacker
Michael A. Terrell wrote:
> No analog signal is noiseless, so every time you double the number of
> channels the noise goes up another 3 dB.
Still makes no sense. If you mix two analog signals and keep the total
signal level constant, how does the noise level go up? If that were
true, if you took a bazillion signal generators putting out a very
clean sine wave, you're telling me there's going to be a bazillion *
3db of noise?
> RF / IF AGC is a simpler way to improve the dynamic range. reduce
> the gain for local signals, and ramp it up to full gain for DX
> reception.
Indeed, but that requires analog front-end filtering, analog
multiplication for gain control, analog frequency conversion.
With a 150db A/D we could simply hook a wire to the input and digitally
extract any signal we wanted, no analog front end or tuning required.
Michael A. Terrell
>
> Michael A. Terrell wrote:
>
> > No analog signal is noiseless, so every time you double the number of
> > channels the noise goes up another 3 dB.
>
> Still makes no sense. If you mix two analog signals and keep the total
> signal level constant, how does the noise level go up? If that were
> true, if you took a bazillion signal generators putting out a very
> clean sine wave, you're telling me there's going to be a bazillion *
> 3db of noise?
Any noise will add when signals are mixed, so yes any residual noise
in those generators would go up +3 dB every time the number of
generators was doubled. Soon, the mixing losses would have the desired
signal so low its useless because they have to be impedance matched.
(Look up "Directional Coupler")
Turn on a fan, you get a little noise.
Turn on another identical fan (total = 2), the noise doubles. (+ 3 dB).
Turn on two more identical fans (total = 4), the noise doubles again (+6
dB).
Turn on four more identical fans (total = 8), and you're up to (+9 dB)
from a single fan. Its the same, whether the noise is mixed
electronically, or acoustically.
Ancient_Hacker
Michael A. Terrell wrote:
> Ancient_Hacker wrote:
> >
> > Michael A. Terrell wrote:
> >
> > > No analog signal is noiseless, so every time you double the number of
> > > channels the noise goes up another 3 dB.
> >
> > Still makes no sense. If you mix two analog signals and keep the total
> > signal level constant, how does the noise level go up? If that were
> > true, if you took a bazillion signal generators putting out a very
> > clean sine wave, you're telling me there's going to be a bazillion *
> > 3db of noise?
>
>
> Any noise will add when signals are mixed, so yes any residual noise
> in those generators would go up +3 dB every time the number of
> generators was doubled. Soon, the mixing losses would have the desired
> signal so low its useless because they have to be impedance matched.
> (Look up "Directional Coupler")
You are not getting the point. If you mix two audio sources each of
one volt of signal and one millivolt of noise, you get TWO volts of
signal and TWO millivolts of noise.
Of course you also turn down the overall gain, so as to not overload
the next stage. So you have ONE volt of signal and ONE millivolt of
noise. Not any more noise in proportion.
Ian Bell
>
> We do not talk about theoretical dynamic range. For a 24bit A/D
> converter, the theoretical dynamic range is 6.02 dB/bit * N bits =
> 144dB. This definition is not very useful. We all know a 24bit A/D
> usually has only 110dB dynamic range due to analog limitation.
>
And why do you think that is? - it has to do with the laws of physics and
how noise is related to resistance. The only ways to increase the actual
dynamic range to get somewhere near the theoretical is either to cool the
analog, reduce the source impedance or increase the maximum signal
handling. The last is the most practical but you would probably need a
bipolar process to get you 10V rms input.
Ian
Eeyore
"Michael A. Terrell" wrote:
> Ancient_Hacker wrote:
> >
> > Michael A. Terrell wrote:
> >
> > > No analog signal is noiseless, so every time you double the number of
> > > channels the noise goes up another 3 dB.
> >
> > Still makes no sense. If you mix two analog signals and keep the total
> > signal level constant, how does the noise level go up? If that were
> > true, if you took a bazillion signal generators putting out a very
> > clean sine wave, you're telling me there's going to be a bazillion *
> > 3db of noise?
>
> Any noise will add when signals are mixed, so yes any residual noise
> in those generators would go up +3 dB every time the number of
> generators was doubled. Soon, the mixing losses would have the desired
> signal so low its useless because they have to be impedance matched.
> (Look up "Directional Coupler")
The losses will attenuate the noise too.
Graham
Michael A. Terrell
Ok, whatever you say! That works when stacking receive antennas,
where they all receive the same signal.
OTOH, I'd never let you design a large audio mixer board. There is
thermal noise generated by the resistors in the summing network, and the
output voltage doubles only if everything is exactly in phase. been
there, done that. Why do you think there are switches for each channel
on a large audio board? To disconnect unused channels, to lower the
overall noise floor.
When you mix video signals, they have to be under one degree of phase
shift at the color burst frequency which is 3,579,545 Hz for NTSC.
Andy Peters
> You are not getting the point. If you mix two audio sources each of
> one volt of signal and one millivolt of noise, you get TWO volts of
> signal and TWO millivolts of noise.
Two comments:
a) by your math, you get 6 dB increases (voltage doubling is 6 dB
gain).
b) that only applies if you're mixing coherent (identical) signals.
Mixing a bunch of 1 kHz sine waves is not very interesting. When
dealing with incoherent signals, mixing two channels gives you a 3 dB
increase.
-a
Tom Bruhns
Michael A. Terrell wrote:
...
> RF / IF AGC is a simpler way to improve the dynamic range. reduce
> the gain for local signals, and ramp it up to full gain for DX
> reception.
:-) That's an overly simplistic view, to be sure. Consider strong
local signals on 710kHz and 770Khz, and I want to listen to a very weak
signal on 830kHz. Consider that the strong signals are each at 0dBm at
the antenna terminals of the receiver, and the 830kHz signal is at
-110dBm. Now figure out what sort of IIP3 you need to receive the
830kHz signal with interference from the distortion of the other two
down 20dB below the 830kHz signal.
Now go looking for a mixer that will give you that sort of
spurious-free dynamic range, even if you assume some decent but
practical tracking input filtering before the mixer. Or try it with
signals at 9.710MHz and 9.770MHz and 9.830MHz instead of the rather
easier MW band.
Good luck.
Cheers,
Tom
Michael A. Terrell
Don't you believe in a tuned input? Even the simple tuned loop
antenna in a cheap radio would reduce the level of the 710 and 780 KHz
signals. I listen to WSM (50 KW) in Nashville, Tn. on 650 AM, and there
is a local station in "The Villages" on 640 AM. As long as they are not
over modulating, I can pick up WSM with no trouble. Try it on some
crappy electronically tuned radio with no front end filtering, and you
won't hear anything. I have used, and built HF and MW receivers for
over 40 years. A crappy design is just that. You will still need AGC
on DX signals to reduce fading, unless you want to ride an RF gain
control, as well. I started with regen, then super regen receivers
before I moved on to superhet, multiple tuned RF stages, then dual and
triple conversion designs. The more work you do to filter the signal at
the front end, the easier it is to process at later stages.
9.710 MHz and 9.770 MHz and 9.830 MHz? Its easier to have proper
filtering at higher frequencies than on the BCB. Move on up to P band,
LL, UL and KU bands, where I worked at my last job with lots of custom
built tubular filters. One of our KU band receivers is aboard the ISS.
Ban
No, the noise will add up to 3dB louder, the signal will be 6dB, if it is
correlated. So to get more dynamic range, put all the inputs in parallel and
the noise improves by 3dB/doubling.
--
ciao Ban
Apricale, Italy
mi...@sushi.com
Andy Peters wrote:
> Ancient_Hacker wrote:
>
> > You are not getting the point. If you mix two audio sources each of
> > one volt of signal and one millivolt of noise, you get TWO volts of
> > signal and TWO millivolts of noise.
In digital land, the peak level is determined by spanning the maximum
number of bits. However, the music sources are not correlated. Singers
have to breathe, guitar notes hit a peak then decay, etc. So you mix
signals that are not at their peaks at all times, but the noise is
there as a constant signal. Hence the noise floor rises as you mix all
these signals.
Eeyore
Andy Peters wrote:
Exactly.
The noise also adds as the rms so that goes up by 3dB too.
Graham
whi...@gmail.com
Ancient_Hacker wrote:
> ... If you mix two audio sources each of
> one volt of signal and one millivolt of noise, you get TWO volts of
> signal and TWO millivolts of noise.
Others have pointed out that this is a fallacy; blending of
any two uncorrelated sources of signal and noise would
result in 1.4 volts of signal and 1.4 millivolts of noise.
Signals aren't just the voltage at one instant, they're averages of
variations
over long time periods; you have to do the addition of all
the variations and redo the average on the sum to get the
right answer.
David L. Jones
> I posted a similar question in the rec.audio.pro group but did not get
> the answer.
>
> I wonder if there is a market for an A/D converter with very high
> dynamic range, say 150dB. I thought it might be useful because it can
> eliminate the needs for multiple-gain front end.
>
> By 150dB dynamic range I mean an A/D converter that can practically
> measure a large signal as high as a few volts, and within the same time
> period it can detect a signal as small as a few nano volts. To clarify,
> when I say "within the same time period", it does not necessarily mean
> at the exact same sample points.
>
> hundred points have magnitude as high as a few volts while the last a
> few hundred points can go as low as a few nano-volts. This is how I
> define the dynamic range.
>
> converter, the theoretical dynamic range is 6.02 dB/bit * N bits =
> 144dB. This definition is not very useful. We all know a 24bit A/D
> usually has only 110dB dynamic range due to analog limitation.
>
> A/D converter if somebody got the products?
Seriously, yes. The Geophysical market is the main user of the highest
end ADCs available. 24bit ADCs used here have a usable dynamic range
approaching 130dB with very low power consumption.
The best ADC designers in the world can't get better than 130dB or so
for a few KHz bandwidth, but if you think you can do better than this
you'll be a very rich man, start work on it now and make sure you
invest all your time and money in it ;-)
Dave :)
DigitalSignal
Thanks for pointing out seismic measurement as a market. Let me ask a
question: while people are talking about the noise floor in this group
and it seems we have reached to a physical limit of heat noise, why
don't the seismic measurement devices have the same problem? How are
they taking the advantage of 130dB dynamic range?
D.S.
David L. Jones
Mainly because the dynamic range is specified over a lower bandwidth.
Dynamic range will increase with reduced bandwidth.
Have a look at the datasheet for this geophysical device for instance:
http://www.cirrus.com/en/products/pro/detail/P274.html
Typical dynamic range is 136dB for a 0 to 27Hz bandwidth.
BTW, the typical specs on this type of device are a bit conservative,
you can actually get better performance than that.
A few hundred Hz bandwidth might be useless for the audio industry, but
for the Geophysical seismic market, bandwidths of only a few dozen or a
few hundred Hz are all that is needed. Hence you geat greater dynamic
range, and lower power consumption. Low power consumption is important
because a typical seismic system will have thousands of these devices
spread over many kilometers of cabling, either floating behind a boat
in the ocean, on the sea bed, or on land.
Dave :)
Kevin White
whi...@gmail.com wrote:
> Ancient_Hacker wrote:
>
> > ... If you mix two audio sources each of
> > one volt of signal and one millivolt of noise, you get TWO volts of
> > signal and TWO millivolts of noise.
>
> Others have pointed out that this is a fallacy; blending of
> any two uncorrelated sources of signal and noise would
> result in 1.4 volts of signal and 1.4 millivolts of noise.
Yes, but if you don't wish to overload under any circumstances you have
to take to case where both inputs are at their maximum. In this case
you get 2V of signal and 1.4 millivolts of noise.
kevin
Eeyore
DigitalSignal wrote:
My guess would be very low impedance transducers.
Graham
Ian
"Eeyore" <rabbitsfriend...@hotmail.com> wrote in message
news:452F0ABC...@hotmail.com...
Regards
Ian
Ken Smith
Ian <ian_buc...@agilent.com> wrote:
>
>"Eeyore" <rabbitsfriend...@hotmail.com> wrote in message
>news:452F0ABC...@hotmail.com...
>> My guess would be very low impedance transducers.
>>
>> Graham
>>
>Not necessarily impedance, just low resistance.
No it is mostly low bandwidth that does it. Sampling at 4KHz is
considered "fast" in the seismic industry.
--
--
kens...@rahul.net forging knowledge
Ken Smith
DigitalSignal <digitals...@yahoo.com> wrote:
[...]
>By 150dB dynamic range I mean an A/D converter that can practically
>measure a large signal as high as a few volts, and within the same time
>period it can detect a signal as small as a few nano volts. To clarify,
>when I say "within the same time period", it does not necessarily mean
>at the exact same sample points.
>
>Say if we sample a frame of signal with 1024 points. The first a few
>hundred points have magnitude as high as a few volts while the last a
>few hundred points can go as low as a few nano-volts. This is how I
>define the dynamic range.
This sounds like they have reinvented the instantaneous floating point
converter. At low frequencies, 150dB would be no big problem to do with
an IFP converter.
Tom Bruhns
_Believe_ in a tuned input?? I'm not sure what that means. As it
turns out, a filter in front of any active devices is not an option for
me. That translates to needing high spurious free dynamic range. In
any event, if you have a look at my posting to which you replied,
you'll see that I DID explicitly mention filtering.
Interesting you think that filtering for signals spaced the same
absolute frequency differences is easier at 9.8MHz than at 0.75MHz.
The same attenuations will require loaded Qs 13 times as big.
Especially given that the filtering should track the LO, and given the
required loaded resonator Q (however many poles, and whatever shape you
want in the filter), getting any significant attenuation about 1%
offset from the center frequency should be an interesting design
challenge. In addition, mechanical tuning is simply not an option in a
great many applications. Maintaining frequency tracking among the LO
and one or more resonators (connected to a source of unknown and
varying impedance) to something noticably better than 1% over an octave
bandwidth should present an interesting challenge.
Though the inductances are a bit easier at 10MHz than at 0.7-0.8MHz,
I'd much rather deal with the far lower loaded Q that would be needed
for the MW filter. But in any event, as I noted above, filters aren't
an option for me anyway.
What in-band IIP3 will your receiver have? What degree of filtering
attenuation will actually be required to achieve the performance goal I
suggested, given that IIP3? Numbers will speak louder than
generalizations.
Cheers,
Tom
"I have used, designed and built high dynamic range wideband inputs for
spectral analysis applications for more than 40 hours."
Michael A. Terrell
>
> Michael A. Terrell wrote:
> >
> > Don't you believe in a tuned input? Even the simple tuned loop
> > antenna in a cheap radio would reduce the level of the 710 and 780 KHz
> > signals. I listen to WSM (50 KW) in Nashville, Tn. on 650 AM, and there
> > is a local station in "The Villages" on 640 AM. As long as they are not
> > over modulating, I can pick up WSM with no trouble. Try it on some
> > crappy electronically tuned radio with no front end filtering, and you
> > won't hear anything. I have used, and built HF and MW receivers for
> > over 40 years. A crappy design is just that. You will still need AGC
> > on DX signals to reduce fading, unless you want to ride an RF gain
> > control, as well. I started with regen, then super regen receivers
> > before I moved on to superhet, multiple tuned RF stages, then dual and
> > triple conversion designs. The more work you do to filter the signal at
> > the front end, the easier it is to process at later stages.
> >
> >
> > 9.710 MHz and 9.770 MHz and 9.830 MHz? Its easier to have proper
> > filtering at higher frequencies than on the BCB. Move on up to P band,
> > LL, UL and KU bands, where I worked at my last job with lots of custom
> > built tubular filters. One of our KU band receivers is aboard the ISS.
>
>
> turns out, a filter in front of any active devices is not an option for
> me. That translates to needing high spurious free dynamic range. In
> any event, if you have a look at my posting to which you replied,
> you'll see that I DID explicitly mention filtering.
You can't even use a bandpass filter to keep out of band crap from
causing desense problems?
> Interesting you think that filtering for signals spaced the same
> absolute frequency differences is easier at 9.8MHz than at 0.75MHz.
> The same attenuations will require loaded Qs 13 times as big.
> Especially given that the filtering should track the LO, and given the
> required loaded resonator Q (however many poles, and whatever shape you
> want in the filter), getting any significant attenuation about 1%
> offset from the center frequency should be an interesting design
> challenge. In addition, mechanical tuning is simply not an option in a
> great many applications. Maintaining frequency tracking among the LO
> and one or more resonators (connected to a source of unknown and
> varying impedance) to something noticably better than 1% over an octave
> bandwidth should present an interesting challenge.
Why is the input uncontrolled impedance? A 3 dB pad will make it
match better than a direct input, if you can stand the loss. What are
the actual frequencies you're interest in?
Electronic tracking is possible, with varactor tuned RF stages.
Either by creating a lookup table, or using an ADC to measure system
gain, and a DAC to set the tuning voltage per stage. It was even done
with vacuum tubes and servo motors turning butterfly capacitors in the
'60s for the VOA Bethany, Ohio transmitters built by National Radio.
> Though the inductances are a bit easier at 10MHz than at 0.7-0.8MHz,
> I'd much rather deal with the far lower loaded Q that would be needed
> for the MW filter. But in any event, as I noted above, filters aren't
> an option for me anyway.
>
> What in-band IIP3 will your receiver have? What degree of filtering
> attenuation will actually be required to achieve the performance goal I
> suggested, given that IIP3? Numbers will speak louder than
> generalizations.
The last reviver design I worked on is still under an NDA, and I no
longer have any of the paperwork. All I can say is that it was DSP based
telemetry equipment.
All of the test equipment in my home electronics shop was badly
damaged two years ago during the hurricane season when the roof sprung
hundreds of leaks, so I can't make any measurements. The project has
been on hold since then because I need $4000 for a new roof, and at
least that much more to replace equipment that can't be repaired. I
can't do the roof work myself due to my recent disabilities. I can't
afford to have done, so everything is at a standstill, maybe forever.
Tom Bruhns
It's common to have amplifiers with "noise figures" lower than 3dB. At
3dB, the amplifier (could just as well be an A/D input) contributes
noise power equal to that of an ideal resistor at room temperature,
about 4e-21 watts/Hz of available power delivered to a load, or
-174dBm/Hz. Amplifiers operating even at room temperature can have
noise figures well below 3dB. How useful that really is depends on how
noise-free the input signal is. For example, the signal from a
resistive strain gage operating at room temperature won't be better
than that -174dBm/Hz, but the noise signal from a dish antenna pointed
at certain places in space and looking at signals in the GHz region
could be very much lower. Practical amplifiers operating at room
temperature can have noise figures that are a small fraction of a dB.
If I'd like to detect small signals--signals with perhaps -150dBm power
in a 100Hz bandwidth--then I'd like to have instrumentation that has a
noise figure under 3dB to get a decent signal-to-noise ratio. If I
need to detect those signals in the presence of other large signals,
perhaps as big as -20dBm or even larger, that sets the full-scale that
the ADC must handle, because if the ADC is overloaded, all bets are off
on being able to detect the small signal. Requirements like this set
the allowable noise floor, of course, and also the allowable
distortion. The small signals are resolvable from the large ones
spectrally, but if I don't know ahead of time what frequencies either
will be on, I can't use filtering to remove the large signals--the ADC
just has to handle the whole range of amplitudes all at once. It's a
tough problem to push down the noise and the distortion at the same
time.
Such is often the life of people doing spectral analysis. There may
well be other "axes" along which the resolution could take place, but
in general, you're going to need digitization that captures a faithful
representation of the input signal, covering a wide (enough) dynamic
range to handle the largest inputs while keeping noise, distortion, and
spurious signals low.
Cheers,
Tom
jasen
>> Michael A. Terrell wrote:
>>> Any noise will add when signals are mixed, so yes any residual noise
>>> in those generators would go up +3 dB every time the number of
>>> generators was doubled. Soon, the mixing losses would have the desired
>>> signal so low its useless because they have to be impedance matched.
>>> (Look up "Directional Coupler")
> and the output voltage doubles only if everything is exactly in phase.
that would be a 6db increase in power, which would reduce the signal to
noise ration, as the noise only goes up 3db in power unless you have a
phase matched noise source :)
Bye.
Jasen
Ian Bell
>
> You are not getting the point. If you mix two audio sources each of
> one volt of signal and one millivolt of noise, you get TWO volts of
> signal and TWO millivolts of noise.
>
Not necessarily. If the two 1V audio sources are correlated (identical) then
they will make 2 volts at the output. The noise in the two channels will
not be correlated because it comes from two different sources so their
powers will add and their combined amplitude will only be root 2 times
their original values. The signal/noise ratio will now be 3dB greater.
That's why a magnetic tape head covering twice the tape width has a 3dB
better S/N ratio.
Ian
Ancient_Hacker
Ah, isnt the Internet wonderful. One guy says mixing sources increases
the noise.
I suggest common sense thought experiment suggests the noise doesnt get
worse.
Then some really smarty-pants says due to the non-correlation the noise
goes down.
All these options... wunerful...
Michael A. Terrell
Are you really that stupid? The audio signals I was talking about
are not in phase, or do you even want them all at the same level but the
noise added by each channel is still there, and the more channels on the
mixing buss, the higher the noise floor. If the signals were identical
and in phase you would only need one channel, no mixing buss, and you
would only have the noise from one channel.
Ancient_Hacker
Michael A. Terrell wrote:
> Ancient_Hacker wrote:
> >
> > Ah, isnt the Internet wonderful. One guy says mixing sources increases
> > the noise.
> > I suggest common sense thought experiment suggests the noise doesnt get
> > worse.
> > Then some really smarty-pants says due to the non-correlation the noise
> > goes down.
> >
> > All these options... wunerful...
>
>
> Are you really that stupid?
Sigh. One last try. Try this thought experiment: You have ONE
BILLION signal generators, all running off separate batteries, each
putting out one volt, each with a microvolt of noise. The noise is
120db below the signal. Got it?
Now you hire 1,000,000 <ethnic>s, give them all some wire, and tell
them to hook all the signal generators in series.
You take the two free ends and hook your "Harbor Freight $2.49
multimeter across the wires. Oops, meter blows out. There goes
$2.49!
You go to your junkbox and pull out the first two resistors you grab.
Lucky you, one is a 999,999,999 ohm 0.00000000000001% resistor. The
other one is a 1 ohm 0.00000000001% ohm resistor.
You hook up the 999... ohm one to the hot wire, the other end of that
one to one end of the 1 ohm resistor, and the other end of the 1 ohmer
to the grond wire from the generators.
So you have a ONE BILLION TO ONE voltage divider. Okay so far?
You go buy another harbor freight $2.49 meter. Put it across the one
ohm resistor and what do we measure?
By your reckoning, we should measure, lessee start with -120db, add
three db for each generator, that's, hmm, THREE BILLION DB. that's a
lot of noise! Divide that by the voltage divider of a billion 10^9 in
voltage, 10^18 in power, 180DB HMMM, we have 2,999,720DB OF NOISE!
By my reckoning, you'll measure one volt of signal, one microvolt of
noise.
Now please explain where I went wrong and why your answer is right.
Michael A. Terrell
Where do you get your mytical noiseless 999,999,999 resistors? ?
Ancient_Hacker
Michael A. Terrell wrote:
> Where do you get your mytical noiseless 999,999,999 resistors? ?
we don't need any resistors, we used just two, only because our $2.49
meter doesnt go up to a billion volts.
And they don't have to be noiseless, if anything that would add a few
microvolts to both our scenarios.
Michael A. Terrell
How are you going to hook all those imaginary generators in series
without radiation and ingression problems?